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Subject: [MMUSIC] Call for Papers: WiOpt'04
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Apologies if you receive multiple copies of this call!
Please note the deadline extension!

----------------------------------------------------------------------
  
                         DEADLINE EXTENSION

               The submission deadline has been extended
                     to the 12th of November, 2003.


        ***************************************************
        ********       2nd  CALL FOR PAPERS        ********
        ***************************************************
        *                                                 *
        *                   WiOpt '04                     *
        *                                                 *
        *     2nd Workshop on Modeling and Optimization   *
        *      in Mobile, Ad Hoc and Wireless Networks    *
        *                                                 *
        *        March 24 - 26, 2004, Cambridge, UK       *
        *                                                 *
        *                                                 *
        ***************************************************


                         Workshop Website:
           http://www.cl.cam.ac.uk/Research/SRG/wiopt04/


This workshop is intended to bring together leading researchers from
both academia and industry to present both foundations and practical
methods and tools for modeling and optimisation of mobile and ad hoc
networks.  The workshop will bring together researchers studying
optimisation issues of mobile networks from both the network
performance and the mobile networking perspectives:

*  Mobile Network Optimisation: Applications and Algorithms
*  Performance characterisation and optimisation of
   mobile/wireless networks and systems
*  Protocol design (MAC, Routing) for optimal mobile network
   utilisation
*  Tools for network optimisation
*  Energy efficiency in mobile networks
*  Security and co-operation models for mobile networks
*  Pricing and incentives in mobile and ad hoc networks.

Papers to be presented at WiOpt'04 will be selected based on extended
abstracts of up to 8 pages, double-spaced, 12 point font size. The
submission should not exceed 3000 words including tables and figures,
each of which count as 250 words.  Authors of accepted presentations
will have the opportunity to publish either the extended abstract or
the full paper in a proceedings volume. Full papers will further be
considered for publication in a special issue of the MONET journal. 

Plenary Speakers:

Bruce Hajek (University of Illinois at Urbana-Champaign, USA) 
Ian Akyildiz (Georgia Institute of Technology, USA) 


IMPORTANT DATES:
Submission of extended abstract: Nov. 12, 2003 (EXTENDED!).
Notification of acceptance:      Jan. 15, 2004.
Camera ready copy:               Feb. 15, 2004.

(Further details on the workshop website,
http://www.cl.cam.ac.uk/Research/SRG/wiopt04/)



General chairman: Imrich Chlamtac

Vice General Chair: Roger Whitaker 

Program Co-Chairs: Marco Conti and Tamer Basar

Program committee of the conference:

Eitan Altman, Venkat Anantharam, Konstantin Avrachenkov, Andrew
Campbell, X. Cao, A. Chockalingam, Jon Crowcroft, Anthony Ephremides,
Silvia Giordano, Stephen V. Hanly, Jean-Pierre Hubaux, Holger Karl,
Peter Key, P. R. Kumar, Peter Marbach, Derek Mcauley, Archan Misra,
Philippe Nain, Sven Oestring, Yoshikuni Onozato, Ariel Orda, Ramesh
Rao, Catherine Rosenberg, Ness Shroff, Violet Syrotiuk, Thierry
Turletti, Adam Wolisz, Jeff Wieselthier, Moshe Zukerman

Local Arrangements Co-Chair: Sven Oestring, Julian Chesterfield and
	Rajiv Chakravorty  

Treasurer: Meng How Lim

Publicity Chair: Holger Karl


The workshop is co-sponsored by
        * Univ. of Cambridge
        * IIT (Istituto di Informatica e Telematica)
        * IEEE IT Society (technical co-sponsorship)
(ACM sponsorship is pending.)

Additional financial support is provided by:
        * Intel
        * Microsoft Research



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From exim@www1.ietf.org  Sat Nov  1 05:49:42 2003
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From: "Sami Peltotalo" <sami.peltotalo@tut.fi>
To: <mmusic@ietf.org>
Subject: RE: [MMUSIC] Service/Session Announcement for Multicast Apps
Date: Fri, 31 Oct 2003 16:36:19 +0200
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Hi,

I am currently working with topic, which handles service and session =
announcements for multicast applications. SAP/SDP doesn't suit very well =
for describing services, which could be categorized to different groups, =
for example sports, news etc. So that the receiver could build some kind =
of service hierarchy from the announcements, and the user is able to =
browse the services from categories he or she is interested. At least I =
have chosen another protocols than SAP/SDP in my work. RMT working =
group's FLUTE protocol seems to be very good choise for the transmission =
of service descriptions, so I am using it. I can't use SAP because it =
has limited payload size and payload can't be segmented to different =
packets (my service and session descriptions are much larger than for =
example one SDP description). I am using XML in my service and session =
descriptions so it was nice to notice that SDPng has now again =
XML-Schema. Has anyone else considered to use FLUTE for transmission of =
session announcements?=20

Cheers, Sami

> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf Of
> Rod.Walsh@nokia.com
> Sent: 15. lokakuuta 2003 12:58
> To: mmusic@ietf.org; mboned@network-services.uoregon.edu
> Subject: [MMUSIC] Service/Session Announcement for Multicast Apps
>=20
>=20
> Does anyone have a problem with a new and better announcement=20
> protocol than SAP?
>=20
> We need a few things which SAP doesn't handle too well, which is=20
> one of the things covered by the Internet Media Guide effort in=20
> MMUSIC:=20
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-img-req-00.txt
>=20
> I guess anyone involved in applications/services over multicast=20
> could be interested, especially where point-to-point isn't desirable.
>=20
> I wanted to get an idea of any strong opinions before I stick out=20
> head and propose a real protocol task for MMUSIC next month.
>=20
> Cheers, Rod.
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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Date: Sun, 02 Nov 2003 15:58:19 +0100
From: Joerg Ott <jo@tzi.uni-bremen.de>
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Subject: [MMUSIC] Draft Agenda for MPLS
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Folks,

MMUSIC will be meeting once in MPLS (on Friday morning).
Please find attached our draft agenda for this meeting.
If anything is missing or incorrect, please let us know.

Thanks,
Colin and Joerg

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

Multiparty Multimedia Session Control (MMUSIC)

CHAIRS: 
Joerg Ott <jo@informatik.uni-bremen.de>
Colin Perkins <csp@csperkins.org>


Friday, 14 November 2003, 09:00 - 11:30
---------------------------------------

0900    Agenda Bashing and Status Update               (Chairs, 15)

         Recent RFCs
         draft-ietf-mmusic-sdp4nat-05.txt		-> RFC 3605

         With RFC editor
         draft-ietf-mmusic-sdpng-trans-04.txt

	With IESG
         draft-ietf-mmusic-sdp-new-15.txt
	draft-rajeshkumar-mmusic-gpmd-03.txt

	With AD
         draft-ietf-mmusic-sdp-comedia-05.txt
	draft-ietf-mmusic-sdp-bwparam-05.txt
	draft-ietf-mmusic-srcfilter-05.txt

         Misc Updates
	draft-camarillo-mmusic-alt-01.txt
	draft-ietf-mmusic-kmgmt-ext-09.txt
	draft-ietf-mmusic-sdp-implem-00.txt
         draft-ietf-mmusic-sdp-new-15.txt

0915 
Internet Media Guides			       (Manyfolks, 30)
	draft-ietf-mmusic-img-req-00.txt
	draft-nomura-mmusic-img-framework-02.txt
	(draft-luoma-mmusic-img-metadata-02.txt fyi only)
	(draft-luoma-mmusic-img-muppet-03.txt fyi only)

0945 
SDP for alternate IP versions		       (Camarillo, 10)
	draft-camarillo-mmusic-alt-02.txt

0955 
SDP Security Descriptions for Media Streams    (Andreasen, 15)
	draft-ietf-mmusic-sdescriptions-02.txt
	draft-andreasen-mmusic-securityprecondition-00.txt

1010 
SDP parameters to support H.263/H.261 options  (Even/Ismail, 15)
	draft-even-avt-h263-h261-options-00.txt
	draft-nismail-mmusic-sdp-video-00.txt

1025 
Interactive Connectivity Establishment (ICE)    (Rosenberg, 20)
	draft-ietf-mmusic-ice-00.txt
	draft-rosenberg-sipping-ice-01.txt

1045 
SDPng Update					(Kutscher, 15)
	draft-ietf-mmusic-sdpng-07.txt

1100 
RTSP Update				 (Westerlund/Zeng, 30)
	draft-ietf-mmusic-rfc2326bis-05.txt
	draft-ietf-mmusic-rtsp-nat-01.txt
	draft-zeng-mmusic-rtsp-end-of-stream-01.txt

1130 
Close




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From: "Peter Lewis" <peter.lewis@upperside.fr>
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Which proposal will be selected as the IEEE 802.15.3a standard? A =
consortium has just proposed to offer a RANDZ access to its patents. The =
industry is now waiting for the reply from the supporters of the other =
proposal.=20
This battle between technology heavyweights underlines the growing =
importance of a technology expected to generate $1.39 billion in revenue =
by 2007.
All questions and answers will be delivered during the UWB Summit to be =
held in Paris next 2-5 December.
All details at:
http://www.upperside.fr/newmobtech/uwb03cfp.htm

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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT color=3D#111111 size=3D2>Which proposal will be selected as =
the=20
<STRONG>IEEE 802.15.3a standard</STRONG>? A consortium has just proposed =
to=20
offer a RANDZ access to its patents. The industry is now waiting for the =
reply=20
from the supporters of the other proposal. </FONT></DIV>
<DIV><FONT color=3D#111111 size=3D2>This battle between technology =
heavyweights=20
underlines the growing importance of a technology expected to generate =
$1.39=20
billion in revenue by 2007.</FONT></DIV>
<DIV><FONT color=3D#111111 size=3D2>All questions and answers will be =
delivered=20
during the <STRONG>UWB Summit </STRONG>to be held in Paris next 2-5=20
December.</FONT></DIV>
<DIV><FONT color=3D#111111 size=3D2>All details at:</FONT></DIV>
<DIV><FONT color=3D#111111 size=3D2><A=20
href=3D"http://www.upperside.fr/newmobtech/uwb03cfp.htm">http://www.upper=
side.fr/newmobtech/uwb03cfp.htm</A></FONT></DIV></BODY></HTML>

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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
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Subject: Re: [MMUSIC] RTSP transport negotiation
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Hi,

In this matter, are there really a need to included a Range header to 
indicate when the SETUP request is to change parameters? What is the use 
case for this. You can't send the SETUP to move to the new transport 
prior to that the new transport interface exist. Therefore I don't see a 
problem with having it change parameters as soon as the request is 
received. Unless you have transport interfaces that remain available 
during some overlapping time you will of course have a result of losing 
packets during the switch.

Regarding the use of Range header in the SETUP. If appropriately 
motivated it can of course be changed. However the potential backwards 
compatibility problems needs to be considered. If this work becomes 
large in impact, we might have to consider splitting it into a separate 
document. However so far I don't think there are sufficient need.

I am sorry that I haven't had time to properly review your proposal.

Cheers

Magnus


-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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From: "Elwell, John" <john.elwell@siemens.com>
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Subject: [MMUSIC] Clarification on use of sendonly for hold
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In section 8.4 of RFC3264 it says:

"Typically, when a user "presses" hold, the agent will generate an offer
with all streams in the SDP indicating a direction of sendonly, and it will
also locally mute, so that no media is sent to the far  end, and no media is
played out."

This seems strange. If the UA is not sending any media to the far end, why
not use "inactive" instead of "sendonly"?

John Elwell (john.elwell@siemens.com)


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<P><FONT SIZE=2>In section 8.4 of RFC3264 it says:</FONT>
</P>

<P><FONT SIZE=2>&quot;Typically, when a user &quot;presses&quot; hold, the agent will generate an offer with all streams in the SDP indicating a direction of sendonly, and it will also locally mute, so that no media is sent to the far&nbsp; end, and no media is played out.&quot;</FONT></P>

<P><FONT SIZE=2>This seems strange. If the UA is not sending any media to the far end, why not use &quot;inactive&quot; instead of &quot;sendonly&quot;?</FONT>
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<P><FONT SIZE=2>John Elwell (john.elwell@siemens.com)</FONT>
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CC: "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Clarification on use of sendonly for hold
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That sort of makes sense, but it presents a problem. It doesn't provide 
a clear indication of which end *wants* the call to be on hold, and 
which one doesn't.

	Paul

Elwell, John wrote:
> In section 8.4 of RFC3264 it says:
> 
> "Typically, when a user "presses" hold, the agent will generate an offer 
> with all streams in the SDP indicating a direction of sendonly, and it 
> will also locally mute, so that no media is sent to the far  end, and no 
> media is played out."
> 
> This seems strange. If the UA is not sending any media to the far end, 
> why not use "inactive" instead of "sendonly"?
> 
> John Elwell (john.elwell@siemens.com)
> 


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Subject: Re: [MMUSIC] Clarification on use of sendonly for hold
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To allow for situations where there is muzak or other media placed 
while the call is on hold. That said, there is nothing wrong with 
using inactive if you want, if you have no plans to do that.

-Jonathan R.

Elwell, John wrote:

> In section 8.4 of RFC3264 it says:
> 
> "Typically, when a user "presses" hold, the agent will generate an offer 
> with all streams in the SDP indicating a direction of sendonly, and it 
> will also locally mute, so that no media is sent to the far  end, and no 
> media is played out."
> 
> This seems strange. If the UA is not sending any media to the far end, 
> why not use "inactive" instead of "sendonly"?
> 
> John Elwell (john.elwell@siemens.com)
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



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Subject: Re: [MMUSIC] Clarification on use of sendonly for hold
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Its not the feature "on-hold" that is being signaled. Its the fact 
that media is not desired, which can be signaled for any number of 
reasons.

-Jonathan R.

Paul Kyzivat wrote:

> That sort of makes sense, but it presents a problem. It doesn't provide 
> a clear indication of which end *wants* the call to be on hold, and 
> which one doesn't.
> 
>     Paul
> 
> Elwell, John wrote:
> 
>> In section 8.4 of RFC3264 it says:
>>
>> "Typically, when a user "presses" hold, the agent will generate an 
>> offer with all streams in the SDP indicating a direction of sendonly, 
>> and it will also locally mute, so that no media is sent to the far  
>> end, and no media is played out."
>>
>> This seems strange. If the UA is not sending any media to the far end, 
>> why not use "inactive" instead of "sendonly"?
>>
>> John Elwell (john.elwell@siemens.com)
>>
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Paul Kyzivat
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Cc: "Elwell, John" <john.elwell@siemens.com>,
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Subject: RE: [MMUSIC] Clarification on use of sendonly for hold
Date: Tue, 4 Nov 2003 08:06:57 +0100 
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Hi,

I think it's also to think about what is sent in the ANSWER (it's not very well described in RFC3264).

IF the UAS knows the UAS wishes to put the call on-hold (sending "sendonly" in the offer doesn't always have to indicate that, as Jonathain said) it may answer with "inactive". However, if the UAC simply, for whatever reason, doesn't want to receive, but still send, media (by sending "sendonly" in the offer) the UAS may answer with "recvonly".

Again, this should probably be a little more clarified in RFC3264.

Regards,

Christer Holmberg
Ericsson Finland







> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf Of
> Jonathan Rosenberg
> Sent: 4. marraskuuta 2003 6:49
> To: Paul Kyzivat
> Cc: Elwell, John; 'mmusic@ietf.org'
> Subject: Re: [MMUSIC] Clarification on use of sendonly for hold
> 
> 
> Its not the feature "on-hold" that is being signaled. Its the fact 
> that media is not desired, which can be signaled for any number of 
> reasons.
> 
> -Jonathan R.
> 
> Paul Kyzivat wrote:
> 
> > That sort of makes sense, but it presents a problem. It 
> doesn't provide 
> > a clear indication of which end *wants* the call to be on hold, and 
> > which one doesn't.
> > 
> >     Paul
> > 
> > Elwell, John wrote:
> > 
> >> In section 8.4 of RFC3264 it says:
> >>
> >> "Typically, when a user "presses" hold, the agent will generate an 
> >> offer with all streams in the SDP indicating a direction 
> of sendonly, 
> >> and it will also locally mute, so that no media is sent to 
> the far  
> >> end, and no media is played out."
> >>
> >> This seems strange. If the UA is not sending any media to 
> the far end, 
> >> why not use "inactive" instead of "sendonly"?
> >>
> >> John Elwell (john.elwell@siemens.com)
> >>
> > 
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

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Date: Tue, 04 Nov 2003 07:59:28 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [MMUSIC] Clarification on use of sendonly for hold
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'mmusic@ietf.org'" <mmusic@ietf.org>
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Jonathan,

Your reply below, and also the one you sent to Paul Kyzivat, are in line
with how I had interpreted things. However, we have encountered
implementations that offer sendonly on audio but do not supply music. The
problem we then have is whether a UAS should supply its own music. If it
receives inactive, clearly it is at liberty to do so. If it receives
sendonly, I believe it should not supply local music, because there may be
music or some other audio from the far end (although in many cases there is
not).

Some clarification in RFC3264 would be beneficial.

John


> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
> Sent: 04 November 2003 00:34
> To: Elwell, John
> Cc: 'mmusic@ietf.org'
> Subject: Re: [MMUSIC] Clarification on use of sendonly for hold
> 
> 
> To allow for situations where there is muzak or other media placed 
> while the call is on hold. That said, there is nothing wrong with 
> using inactive if you want, if you have no plans to do that.
> 
> -Jonathan R.
> 
> Elwell, John wrote:
> 
> > In section 8.4 of RFC3264 it says:
> > 
> > "Typically, when a user "presses" hold, the agent will 
> generate an offer 
> > with all streams in the SDP indicating a direction of 
> sendonly, and it 
> > will also locally mute, so that no media is sent to the far 
>  end, and no 
> > media is played out."
> > 
> > This seems strange. If the UA is not sending any media to 
> the far end, 
> > why not use "inactive" instead of "sendonly"?
> > 
> > John Elwell (john.elwell@siemens.com)
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> 

_______________________________________________
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Subject: RE: [MMUSIC] RTSP transport negotiation
Date: Tue, 4 Nov 2003 13:57:20 +0100
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Hi Geetha,

recently I've implemented the solution a) that you mentioned using well =
know
open source RTSP client and server implementation.
I noticed that sending the SETUP during the play state (to change the =
"destination" of RTP packets)
and then send the PLAY with the Range header doesn't work very well due =
to syncronization problems.
The only working solution I found, was to send the PAUSE method before =
the SETUP
and the PLAY. I think this behaviour is due to the RTSP client-server =
implementation that
I've used and maybe it doesn't happen with other implementation.

Anyway, I think that (in the case of client mobility) the best solution =
should be that the client sends only the SETUP request with the new =
"client_ip" without any other RTSP message (PAUSE and PLAY).

Regards,
Enrico



------------------------------------------------------------------
Ing Enrico M. Balatti
CEFRIEL - Politecnico di Milano
ICT Center of Excellence For Research, Innovation,
Education & industrial Labs partnership
via Fucini 2, 20133 Milano (ITALY)
voice: +39 0223954327 fax: +39 0223954527
e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   www.cefriel.it =
<http://www.cefriel.it>=20
------------------------------------------------------------------


-----Original Message-----
From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
Sent: venerd=EC 31 ottobre 2003 19.44
To: Enrico Balatti
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] RTSP transport negotiation


hi Enrico,

Thanks for the comments..

That's true - per the current spec we cannot have a Range header in the
SETUP request.
However, if there were no Range header in the SETUP request during play
state, the client would have to either=20
(a) send a PLAY request with a Range header and use the PLAY response=20
    for synchronization, or,
(b) live with the range selected by the server when it receives a SETUP=20
    during play state.

With (a) there is the issue of an extra PLAY request-response. It might
just be better for the client to issue a PAUSE from the old transport
setting, send a SETUP request to modify transport and PLAY to resume
playing from the Range of the client's choosing. Here the transport
negotiation happens in Ready state not Play state.

With (b) there is a high likelihood of synchronization problems when
transport parameters change - RTP packets would get delivered to the old
transport, and discarded; the server has no way of knowing the gap in
time between when it processes the SETUP during play and when the
packets it already sent didn't actually get used by the player. This
implies media glitches at the client are a real possibility.

My inclination is to go with either specifying a Range header in the
SETUP request during play state or send the SETUP request without Range,
follow it up with a PLAY request with Range header. In the latter case
we'd be violating another restriction in the spec about issuing PLAY
requests during Play state, I believe. So this implies the spec would
have to modify restrictions on either the SETUP request or the PLAY
request.

It would be good to get more comments on this, especially from the
authors.

thanks
geetha



On Wed, 2003-10-29 at 06:02, Enrico Balatti wrote:
> Dear Geetha,
>=20
> >... now change client IP address ....
> >
> >     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
> >           CSeq: 5
> >	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
> >	              client_port=3D8002-8003;server_port=3D9004-9005
> >	   Range: npt=3D5-
> >           Session: 12345678
>=20
>=20
> I'm not sure the Range field can be aplicated to the SETUP request.
> From the specification it seems the Range is applicable on the SETUP =
response
> but not in the SETUP request.
>=20
>    Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD =20
>=20
>    Range                 R           -   -    -    o     o   -
>    Range                 r           -   -    c    m*   m*   -
>=20
>=20
> Best Regards,
> Enrico
>=20
>=20
>=20
>=20
> ------------------------------------------------------------------
> Ing Enrico M. Balatti
> CEFRIEL - Politecnico di Milano
> ICT Center of Excellence For Research, Innovation,
> Education & industrial Labs partnership
> via Fucini 2, 20133 Milano (ITALY)
> voice: +39 0223954327 fax: +39 0223954527
> e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   =
www.cefriel.it <http://www.cefriel.it>=20
> ------------------------------------------------------------------
>=20
>=20
> -----Original Message-----
> From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
> Sent: mercoled=EC 22 ottobre 2003 7.26
> To: mmusic@ietf.org
> Subject: [MMUSIC] RTSP transport negotiation
>=20
>=20
> Hello everyone,
>=20
> Here is the promised write-up on RTSP transport negotiation..sorry for
> the delay. If there is agreement on the content of this note, I can
> write up text for the RTSP draft.
>=20
> thanks
> geetha
>=20
>=20
> Topic: RTSP Transport negotiation using SETUP during play state.
>=20
> Introduction
>=20
> The RTSP specification allows for negotiation of transport parameters
> via the SETUP method, while the RTSP session is in play state. The
> transport parameters that can be negotiated and the exact form of this
> negotiation is under-specified in the RTSP documents. This note is an
> attempt to clarify some of the issues around transport negotiation for
> inclusion in the ongoing revision of RTSP.
>=20
> The recent interest in mobility for RTSP implementations also impacts
> transport parameters used by the session. For instance, a wireless
> RTSP client could be assigned a new IP address, or may experience
> change of network characteristics, while it is in the midst of a RTSP
> session. These issues are also addressed in this note.
>=20
>=20
> Outline:
>=20
> 1. Objective
> 2. Enumeration of all Transport Parameters
> 3. Transport Parameters Excluded from Negotiation
> 4. Negotiable Transport Parameters
> 5. Impact of Transport Negotiation
> 6. Examples
> 	6.1 Client IP Address Changes
> 	6.2 UDP -> TCP
> 	6.3 TCP -> UDP
> 7. Other transport related changes to RTSP draft
> 8. Summary
>=20
>=20
>=20
> 1. Objective:=20
>=20
> The main goals of this document are to
> 	* Provide an enumeration of transport parameters.
> 	* Identify transport parameters that cannot be negotiated
> 	  with minimal impact on other sessions.
> 	* Propose transport parameters that can be negotiated during
> 	  play state.
> 	* Determine implications of changing these parameters via a
> 	  SETUP request while the session is in play state.
> 	* Provide examples demonstrating transport negotiation.
>=20
>=20
> 2. Enumeration of all Transport Parameters
>=20
>    The following is a list of all transport parameters in the current
>    RTSP draft.
>=20
> 	unicast/multicast
> 	destination
> 	source
> 	mode (=3DPLAY)
> 	interleaved
> 	layers (for multicast)
> 	ttl (for multicast
> 	port (for multicast)
> 	client_port (for RTP)
> 	server_port (for RTP)
> 	ssrc (for RTP)
> 	url
>=20
>=20
> 3. Transport Parameters Excluded from Negotiation
>=20
>    It is recommended that the following parameters be excluded from =
the
>    negotiation process.
>=20
> 	mode
> 	url
> 	unicast/multicast
> 	layers
> 	ttl
> 	ssrc
> 	destination
> 	source
>=20
>=20
>     As the "mode" parameter is restricted to a single value in the
> current draft, it can be excluded.
>=20
>     As per the recent proposal to not support addition of urls via =
SETUP
> during play state, the "url" parameter can be excluded.=20
>=20
>     Changing from multicast to unicast based on a single RTSP =
session's
> SETUP, would impact all the other clients accessing the multicast
> session.=20
> Several issues need to be considered in this case, including transport
> negotiation. Similarly, changing from unicast to multicast also has
> implications beyond a single RTSP session. Changes to other
> multicast-related parameters, "layers", "ssrc", "ttl" most likely
> requires specification beyond transport negotiation.
> The recommendation is to *not* specify changing the transport from
> "unicast" to "multicast" or vice versa, and also not specify changes
> to the "layers" and "ttl" parameters, in this draft. It is recommedned =

> that negotiation of multicast-related transport parameters should be
> deferred to an RTSP-extension document.=20
>=20
>     Once an RTSP session has been initiated and the "destination" and
> "source" have been established - i.e. a multicast "destination" has
> been identified, and the stream "source" has been identified, there
> should be no need to change these parameters.  It is recommended that
> the "destination" and "source" parameters not be negotiated when the
> session is in play state.  The case of the client IP address change is
> signaled in a different manner, outlined in the next section.
>=20
>=20
> 4. Negotiable Transport Parameters
>=20
>    Hence there are three main conditions that require transport
> negotiation for unicast:
>=20
> 	* client IP address changes
> 	* client wants to switch transport from UDP -> TCP=20
> 	* client wants to switch transport from TCP -> UDP
>=20
>    Changes in the client IP address can be indicated using a new
> "client_ip" parameter. Hence the parameters that are impacted by
> transport negotiation are:
>=20
> 	client_ip
> 	interleaved
> 	client_port
> 	server_port
>=20
> =20
> 5. Impact of Transport Negotiation
>=20
>    The first column describes the primary change, the second and third
> columns describe any secondary transport parameters that MUST or=20
> MAY change as a result of the primary change.
>=20
>=20
> Changed			MUST			MAY=09
> -------			----			---
> =09
> Client IP address	client_ip		client_port
> 						server_port
>=20
> UDP -> TCP		interleaved		Server may change
> 			Drop client_port	interleaved channel ids
> 			Drop server_port	in response.
>=20
> TCP -> UDP		client_port in request	client_ip
> 			server_port in response client_port in response=09
> 			Drop interleaved
>=20
>=20
>=20
> 6. Examples
>=20
> 6.1 Client IP Address Changes
>=20
>      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.1
> 	   CSeq: 1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   CSeq: 1
>            Content-Type: application/sdp
> 	   Content-Length: 164
>=20
> 	   v=3D0
>            o=3D- 2890844256 2890842807 IN IP4 172.16.2.93
>            s=3DRTSP Session
>            i=3DAn Example of RTSP Session Usage
>            a=3Dcontrol:rtsp://foo/twister
>            t=3D0 0
>            m=3Daudio 0 RTP/AVP 0
>            a=3Dcontrol:rtsp://foo/twister/audio
>            m=3Dvideo 0 RTP/AVP 26
>            a=3Dcontrol:rtsp://foo/twister/video
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
>            CSeq: 2
>            Transport: RTP/AVP;unicast;client_port=3D8000-8001
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 2
>            Transport: RTP/AVP;unicast;client_port=3D8000-8001;
>                       server_port=3D9000-9001
>            Session: 12345678
>=20
>      C->S: SETUP rtsp://foo/twister/video RTSP/1.0
>            CSeq: 3
>            Transport: RTP/AVP;unicast;client_port=3D8002-8003
>            Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 3
>            Transport: RTP/AVP;unicast;client_port=3D8002-8003;
>                       server_port=3D9004-9005
>            Session: 12345678
>=20
>      C->S: PLAY rtsp://foo/twister RTSP/1.0
>            CSeq: 4
>            Range: npt=3D0-
>            Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 4
>            Session: 12345678
>            RTP-Info: url=3Drtsp://foo/twister/video;
>              seq=3D9810092;rtptime=3D3450012,
> 	   url=3Drtsp://foo/twister/audio;seq=3D5923;rtptime=3D22234432
> 	   Range: npt=3D0-30.76
>=20
> ... now change client IP address ....
>=20
>      C->S: SETUP rtsp://foo/twister/video RTSP/1.0
>            CSeq: 5
> 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
> 	              client_port=3D8002-8003;server_port=3D9004-9005
> 	   Range: npt=3D5-
>            Session: 12345678
> 		=09
>      S->C: RTSP/1.0 200 OK
>            CSeq: 5
> 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
> 		      client_port=3D8002-8003;server_port=3D9004-9005
> 	   RTP-Info: url=3Drtsp://foo/twister/video;
> 		    seq=3D<new>;rtptime=3D<new>,
> 	   Range: npt=3D5-30.76
> 	   Session: 12345678
>=20
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
>            CSeq: 6
> 	   Transport:RTP/AVP;unicast;client_ip=3D<new IP address>;
> 		     client_port=3D8000-8001;server_port=3D9000-9001
> 	   Range: npt=3D5-
>            Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 6
> 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
> 		      client_port=3D8000-8001;server_port=3D9000-9001
> 	   RTP-Info:url=3Drtsp://foo/twister/audio;
> 		    seq=3D<new>;rtptime=3D<new>
> 	   Range: npt=3D5-30.76=09
> 	   Session: 12345678
>=20
>=20
>=20
>     =20
> 6.2 UDP -> TCP
>=20
>      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
> 	   CSeq: 1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   Content-Length: 230
> 	   Content-Type: application/sdp
> 	   Cseq: 1
>=20
> 	   v=3D0
> 	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
> 	   s=3DRTSP Session
> 	   c=3DIN IP4 0.0.0.0
> 	   t=3D0 0
> 	   a=3Dcontrol:*
> 	   a=3Drange:npt=3D0-10.566000
> 	   m=3Dvideo 0 RTP/AVP 26
> 	   a=3Dcontrol:rtsp://foo/twister/video
> 	   m=3Daudio 0 RTP/AVP 0
> 	   a=3Dcontrol:rtsp://foo/twister/audio
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 2
> 	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001
>=20
>      S->C: RTSP/1.0 200 OK
> 	   CSeq: 2
> 	   Session: 12345678
> 	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
> 		      server_port=3D9000-9001
>=20
>      C->S: PLAY rtsp://foo/twister RTSP/1.0
> 	   CSeq: 3
> 	   Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   CSeq: 3
> 	   RTP-Info: =
url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D1339949552
> 	   Range: npt=3D0-
> 	   Session: 12345678
>=20
> ... now request TCP ...
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 4
> 	   Transport: RTP/AVP/TCP;interleaved=3D0-1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Cseq: 4
> 	   Transport: RTP/AVP/TCP;unicast;
> 		      client_ip=3D<client IP>;interleaved=3D0-1
> 	   RTP-Info: =
url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<new>
> 	   Range: npt=3D0-10.56600
> 	   Session: 12345678
>=20
>=20
>=20
> 6.3 TCP -> UDP
>=20
>      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
> 	   CSeq: 1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   Content-Length: 230
> 	   Content-Type: application/sdp
> 	   Cseq: 1
>=20
> 	   v=3D0
> 	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
> 	   s=3DRTSP Session
> 	   c=3DIN IP4 0.0.0.0
> 	   t=3D0 0
> 	   a=3Dcontrol:*
> 	   a=3Drange:npt=3D0-10.566000
> 	   m=3Dvideo 0 RTP/AVP 26
> 	   a=3Dcontrol:rtsp://foo/twister/video
> 	   m=3Daudio 0 RTP/AVP 0
> 	   a=3Dcontrol:rtsp://foo/twister/audio
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 2
> 	   Transport: RTP/AVP/TCP;interleaved=3D0-1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Cseq: 2
> 	   Session: 12345678
> 	   Transport: RTP/AVP/TCP;unicast;client_ip=3D<client =
IP>;interleaved=3D0-1
>=20
>      C->S: PLAY rtsp://foo/twister RTSP/1.0
> 	   CSeq: 3
> 	   Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   Cseq: 3
> 	   RTP-Info: =
url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D1339949552
> 	   Range: npt=3D0-
> 	   Session: 12345678
>=20
> ... now request UDP ...
>=20
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 4
> 	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001
> 	   Range: npt=3D0-
>=20
>      S->C: RTSP/1.0 200 OK
> 	   CSeq: 4
> 	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
> 		      server_port=3D9000-9001
> 	   RTP-Info: =
url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<new>
> 	   Range: npt=3D0-10.56600
> 	   Session: 12345678
>=20
>=20
>=20
>=20
> 7. Other transport related changes to RTSP draft
>=20
> A few comments on draft-ietf-mmusic-rfc2326bis-04.txt relating to
> transport parameters.
>=20
> (a) Section 13.40:
>     Page 81: Are "src_addr" and "dest_addr" parameters
>     going away? If these are relevant for NAT, perhaps they belong in
>     the NAT document?
>=20
>     Page 83-83: BNF indicates "client_port" and
>     "server_port" will be indicated as 'port1-port2'. This doesn't =
seem
>     adequate for the case where the RTP and RTCP ports are =
non-adjacent
>     (per the new RTP spec rfc 3550).=20
>=20
>     Proposal:
> 	client_port: rtp=3D<port>, rtcp=3D<port>
> 	server_port: rtp=3D<port>, rtcp=3D<port>
>=20
> (b) Appendix B, B.1.1
>     Page 115: Mentions "src_addresses" and
>     "dest_addresses" - what are these?
>=20
>     Page 116: "non-continuous" ports - might want to rephrase to
>     "non-adjacent" (or "non-continguous") ports?
>=20
>=20
>=20
> 8. Summary
>=20
>=20
> This note recommends negotiation of a few transport parameters and
> specifies mechanisms to do so. One new transport parameter "client_ip" =

> is recommended for use in cases where the client's IP address changes
> during the session.=20
>=20
> It is also recommended that the client_port and server_port parameters
> be specified further with the use of tags "rtp_port" and "rtcp_port"
> to handle the case where these ports are non-contiguous.
--=20
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: Re: [MMUSIC] RTSP transport negotiation
In-reply-to: <3FA62E59.9040407@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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hi Magnus,

My assumption was that SETUP during play happens *after* the client has
switched to the new transport..synchronization is an issue in this case.
I agree that if we can assume that the client requests new transport
before switching to the new transport, there is probably no
synchronization issue. I have two follow-up questions on this:

a) How does a mobile client get notified about new transports - such as 
   a new IP address?  How far ahead of time does the client get
   notified?

b) What is the range of time over which the transport switch happens?
   Consider a scenario where the client requests an IP address change
   for example, via SETUP with no Range, and the server starts sending
   packets to the new IP. Is there a chance that the buffers at the new
   IP get overrun, before the client switches to the new IP?
   It would help if the network guarantees that the transport switch
   will not lose any packets due to buffer overflows,
   
thanks
geetha  



On Mon, 2003-11-03 at 02:30, Magnus Westerlund wrote:
> Hi,
> 
> In this matter, are there really a need to included a Range header to 
> indicate when the SETUP request is to change parameters? What is the use 
> case for this. You can't send the SETUP to move to the new transport 
> prior to that the new transport interface exist. Therefore I don't see a 
> problem with having it change parameters as soon as the request is 
> received. Unless you have transport interfaces that remain available 
> during some overlapping time you will of course have a result of losing 
> packets during the switch.
> 
> Regarding the use of Range header in the SETUP. If appropriately 
> motivated it can of course be changed. However the potential backwards 
> compatibility problems needs to be considered. If this work becomes 
> large in impact, we might have to consider splitting it into a separate 
> document. However so far I don't think there are sufficient need.
> 
> I am sorry that I haven't had time to properly review your proposal.
> 
> Cheers
> 
> Magnus
-- 
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: RE: [MMUSIC] RTSP transport negotiation
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hi Enrico,

Thanks for sharing your experience with implementing this.

I wouldn't mind dropping the Range header from SETUP, as long as
we also specify (please see my reply to Magnus on the topic):

(a) To support transport negotiation in play state, the client may issue
    SETUP to change transport *before* switching to the new transport,
    and,
(b) the client will switch to the new transport before buffers overrun
    at the new transport.=20

If transport negotiation is required in play state and (a) and (b)
cannot be assumed, the client should issue a PAUSE, SETUP(*) and PLAY
- this implies that transport negotiation occurs in ready state and not
in play state.


Does that sound reasonable?

geetha


On Tue, 2003-11-04 at 04:57, Enrico Balatti wrote:
> Hi Geetha,
>=20
> recently I've implemented the solution a) that you mentioned using well=
 know
> open source RTSP client and server implementation.
> I noticed that sending the SETUP during the play state (to change the "=
destination" of RTP packets)
> and then send the PLAY with the Range header doesn't work very well due=
 to syncronization problems.
> The only working solution I found, was to send the PAUSE method before =
the SETUP
> and the PLAY. I think this behaviour is due to the RTSP client-server i=
mplementation that
> I've used and maybe it doesn't happen with other implementation.
>=20
> Anyway, I think that (in the case of client mobility) the best solution=
 should be that the client sends only the SETUP request with the new "cli=
ent_ip" without any other RTSP message (PAUSE and PLAY).
>=20
> Regards,
> Enrico
>=20
>=20
>=20
> ------------------------------------------------------------------
> Ing Enrico M. Balatti
> CEFRIEL - Politecnico di Milano
> ICT Center of Excellence For Research, Innovation,
> Education & industrial Labs partnership
> via Fucini 2, 20133 Milano (ITALY)
> voice: +39 0223954327 fax: +39 0223954527
> e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   www.cefriel.it=
 <http://www.cefriel.it>=20
> ------------------------------------------------------------------
>=20
>=20
> -----Original Message-----
> From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
> Sent: venerd=EC 31 ottobre 2003 19.44
> To: Enrico Balatti
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] RTSP transport negotiation
>=20
>=20
> hi Enrico,
>=20
> Thanks for the comments..
>=20
> That's true - per the current spec we cannot have a Range header in the
> SETUP request.
> However, if there were no Range header in the SETUP request during play
> state, the client would have to either=20
> (a) send a PLAY request with a Range header and use the PLAY response=20
>     for synchronization, or,
> (b) live with the range selected by the server when it receives a SETUP=
=20
>     during play state.
>=20
> With (a) there is the issue of an extra PLAY request-response. It might
> just be better for the client to issue a PAUSE from the old transport
> setting, send a SETUP request to modify transport and PLAY to resume
> playing from the Range of the client's choosing. Here the transport
> negotiation happens in Ready state not Play state.
>=20
> With (b) there is a high likelihood of synchronization problems when
> transport parameters change - RTP packets would get delivered to the ol=
d
> transport, and discarded; the server has no way of knowing the gap in
> time between when it processes the SETUP during play and when the
> packets it already sent didn't actually get used by the player. This
> implies media glitches at the client are a real possibility.
>=20
> My inclination is to go with either specifying a Range header in the
> SETUP request during play state or send the SETUP request without Range=
,
> follow it up with a PLAY request with Range header. In the latter case
> we'd be violating another restriction in the spec about issuing PLAY
> requests during Play state, I believe. So this implies the spec would
> have to modify restrictions on either the SETUP request or the PLAY
> request.
>=20
> It would be good to get more comments on this, especially from the
> authors.
>=20
> thanks
> geetha
>=20
>=20
>=20
> On Wed, 2003-10-29 at 06:02, Enrico Balatti wrote:
> > Dear Geetha,
> >=20
> > >... now change client IP address ....
> > >
> > >     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
> > >           CSeq: 5
> > >	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
> > >	              client_port=3D8002-8003;server_port=3D9004-9005
> > >	   Range: npt=3D5-
> > >           Session: 12345678
> >=20
> >=20
> > I'm not sure the Range field can be aplicated to the SETUP request.
> > From the specification it seems the Range is applicable on the SETUP =
response
> > but not in the SETUP request.
> >=20
> >    Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD =20
> >=20
> >    Range                 R           -   -    -    o     o   -
> >    Range                 r           -   -    c    m*   m*   -
> >=20
> >=20
> > Best Regards,
> > Enrico
> >=20
> >=20
> >=20
> >=20
> > ------------------------------------------------------------------
> > Ing Enrico M. Balatti
> > CEFRIEL - Politecnico di Milano
> > ICT Center of Excellence For Research, Innovation,
> > Education & industrial Labs partnership
> > via Fucini 2, 20133 Milano (ITALY)
> > voice: +39 0223954327 fax: +39 0223954527
> > e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   www.cefriel.=
it <http://www.cefriel.it>=20
> > ------------------------------------------------------------------
> >=20
> >=20
> > -----Original Message-----
> > From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
> > Sent: mercoled=EC 22 ottobre 2003 7.26
> > To: mmusic@ietf.org
> > Subject: [MMUSIC] RTSP transport negotiation
> >=20
> >=20
> > Hello everyone,
> >=20
> > Here is the promised write-up on RTSP transport negotiation..sorry fo=
r
> > the delay. If there is agreement on the content of this note, I can
> > write up text for the RTSP draft.
> >=20
> > thanks
> > geetha
> >=20
> >=20
> > Topic: RTSP Transport negotiation using SETUP during play state.
> >=20
> > Introduction
> >=20
> > The RTSP specification allows for negotiation of transport parameters
> > via the SETUP method, while the RTSP session is in play state. The
> > transport parameters that can be negotiated and the exact form of thi=
s
> > negotiation is under-specified in the RTSP documents. This note is an
> > attempt to clarify some of the issues around transport negotiation fo=
r
> > inclusion in the ongoing revision of RTSP.
> >=20
> > The recent interest in mobility for RTSP implementations also impacts
> > transport parameters used by the session. For instance, a wireless
> > RTSP client could be assigned a new IP address, or may experience
> > change of network characteristics, while it is in the midst of a RTSP
> > session. These issues are also addressed in this note.
> >=20
> >=20
> > Outline:
> >=20
> > 1. Objective
> > 2. Enumeration of all Transport Parameters
> > 3. Transport Parameters Excluded from Negotiation
> > 4. Negotiable Transport Parameters
> > 5. Impact of Transport Negotiation
> > 6. Examples
> > 	6.1 Client IP Address Changes
> > 	6.2 UDP -> TCP
> > 	6.3 TCP -> UDP
> > 7. Other transport related changes to RTSP draft
> > 8. Summary
> >=20
> >=20
> >=20
> > 1. Objective:=20
> >=20
> > The main goals of this document are to
> > 	* Provide an enumeration of transport parameters.
> > 	* Identify transport parameters that cannot be negotiated
> > 	  with minimal impact on other sessions.
> > 	* Propose transport parameters that can be negotiated during
> > 	  play state.
> > 	* Determine implications of changing these parameters via a
> > 	  SETUP request while the session is in play state.
> > 	* Provide examples demonstrating transport negotiation.
> >=20
> >=20
> > 2. Enumeration of all Transport Parameters
> >=20
> >    The following is a list of all transport parameters in the current
> >    RTSP draft.
> >=20
> > 	unicast/multicast
> > 	destination
> > 	source
> > 	mode (=3DPLAY)
> > 	interleaved
> > 	layers (for multicast)
> > 	ttl (for multicast
> > 	port (for multicast)
> > 	client_port (for RTP)
> > 	server_port (for RTP)
> > 	ssrc (for RTP)
> > 	url
> >=20
> >=20
> > 3. Transport Parameters Excluded from Negotiation
> >=20
> >    It is recommended that the following parameters be excluded from t=
he
> >    negotiation process.
> >=20
> > 	mode
> > 	url
> > 	unicast/multicast
> > 	layers
> > 	ttl
> > 	ssrc
> > 	destination
> > 	source
> >=20
> >=20
> >     As the "mode" parameter is restricted to a single value in the
> > current draft, it can be excluded.
> >=20
> >     As per the recent proposal to not support addition of urls via SE=
TUP
> > during play state, the "url" parameter can be excluded.=20
> >=20
> >     Changing from multicast to unicast based on a single RTSP session=
's
> > SETUP, would impact all the other clients accessing the multicast
> > session.=20
> > Several issues need to be considered in this case, including transpor=
t
> > negotiation. Similarly, changing from unicast to multicast also has
> > implications beyond a single RTSP session. Changes to other
> > multicast-related parameters, "layers", "ssrc", "ttl" most likely
> > requires specification beyond transport negotiation.
> > The recommendation is to *not* specify changing the transport from
> > "unicast" to "multicast" or vice versa, and also not specify changes
> > to the "layers" and "ttl" parameters, in this draft. It is recommedne=
d=20
> > that negotiation of multicast-related transport parameters should be
> > deferred to an RTSP-extension document.=20
> >=20
> >     Once an RTSP session has been initiated and the "destination" and
> > "source" have been established - i.e. a multicast "destination" has
> > been identified, and the stream "source" has been identified, there
> > should be no need to change these parameters.  It is recommended that
> > the "destination" and "source" parameters not be negotiated when the
> > session is in play state.  The case of the client IP address change i=
s
> > signaled in a different manner, outlined in the next section.
> >=20
> >=20
> > 4. Negotiable Transport Parameters
> >=20
> >    Hence there are three main conditions that require transport
> > negotiation for unicast:
> >=20
> > 	* client IP address changes
> > 	* client wants to switch transport from UDP -> TCP=20
> > 	* client wants to switch transport from TCP -> UDP
> >=20
> >    Changes in the client IP address can be indicated using a new
> > "client_ip" parameter. Hence the parameters that are impacted by
> > transport negotiation are:
> >=20
> > 	client_ip
> > 	interleaved
> > 	client_port
> > 	server_port
> >=20
> > =20
> > 5. Impact of Transport Negotiation
> >=20
> >    The first column describes the primary change, the second and thir=
d
> > columns describe any secondary transport parameters that MUST or=20
> > MAY change as a result of the primary change.
> >=20
> >=20
> > Changed			MUST			MAY=09
> > -------			----			---
> > =09
> > Client IP address	client_ip		client_port
> > 						server_port
> >=20
> > UDP -> TCP		interleaved		Server may change
> > 			Drop client_port	interleaved channel ids
> > 			Drop server_port	in response.
> >=20
> > TCP -> UDP		client_port in request	client_ip
> > 			server_port in response client_port in response=09
> > 			Drop interleaved
> >=20
> >=20
> >=20
> > 6. Examples
> >=20
> > 6.1 Client IP Address Changes
> >=20
> >      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.1
> > 	   CSeq: 1
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   CSeq: 1
> >            Content-Type: application/sdp
> > 	   Content-Length: 164
> >=20
> > 	   v=3D0
> >            o=3D- 2890844256 2890842807 IN IP4 172.16.2.93
> >            s=3DRTSP Session
> >            i=3DAn Example of RTSP Session Usage
> >            a=3Dcontrol:rtsp://foo/twister
> >            t=3D0 0
> >            m=3Daudio 0 RTP/AVP 0
> >            a=3Dcontrol:rtsp://foo/twister/audio
> >            m=3Dvideo 0 RTP/AVP 26
> >            a=3Dcontrol:rtsp://foo/twister/video
> >=20
> >      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> >            CSeq: 2
> >            Transport: RTP/AVP;unicast;client_port=3D8000-8001
> >=20
> >      S->C: RTSP/1.0 200 OK
> >            CSeq: 2
> >            Transport: RTP/AVP;unicast;client_port=3D8000-8001;
> >                       server_port=3D9000-9001
> >            Session: 12345678
> >=20
> >      C->S: SETUP rtsp://foo/twister/video RTSP/1.0
> >            CSeq: 3
> >            Transport: RTP/AVP;unicast;client_port=3D8002-8003
> >            Session: 12345678
> >=20
> >      S->C: RTSP/1.0 200 OK
> >            CSeq: 3
> >            Transport: RTP/AVP;unicast;client_port=3D8002-8003;
> >                       server_port=3D9004-9005
> >            Session: 12345678
> >=20
> >      C->S: PLAY rtsp://foo/twister RTSP/1.0
> >            CSeq: 4
> >            Range: npt=3D0-
> >            Session: 12345678
> >=20
> >      S->C: RTSP/1.0 200 OK
> >            CSeq: 4
> >            Session: 12345678
> >            RTP-Info: url=3Drtsp://foo/twister/video;
> >              seq=3D9810092;rtptime=3D3450012,
> > 	   url=3Drtsp://foo/twister/audio;seq=3D5923;rtptime=3D22234432
> > 	   Range: npt=3D0-30.76
> >=20
> > ... now change client IP address ....
> >=20
> >      C->S: SETUP rtsp://foo/twister/video RTSP/1.0
> >            CSeq: 5
> > 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
> > 	              client_port=3D8002-8003;server_port=3D9004-9005
> > 	   Range: npt=3D5-
> >            Session: 12345678
> > 		=09
> >      S->C: RTSP/1.0 200 OK
> >            CSeq: 5
> > 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
> > 		      client_port=3D8002-8003;server_port=3D9004-9005
> > 	   RTP-Info: url=3Drtsp://foo/twister/video;
> > 		    seq=3D<new>;rtptime=3D<new>,
> > 	   Range: npt=3D5-30.76
> > 	   Session: 12345678
> >=20
> >=20
> >      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> >            CSeq: 6
> > 	   Transport:RTP/AVP;unicast;client_ip=3D<new IP address>;
> > 		     client_port=3D8000-8001;server_port=3D9000-9001
> > 	   Range: npt=3D5-
> >            Session: 12345678
> >=20
> >      S->C: RTSP/1.0 200 OK
> >            CSeq: 6
> > 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
> > 		      client_port=3D8000-8001;server_port=3D9000-9001
> > 	   RTP-Info:url=3Drtsp://foo/twister/audio;
> > 		    seq=3D<new>;rtptime=3D<new>
> > 	   Range: npt=3D5-30.76=09
> > 	   Session: 12345678
> >=20
> >=20
> >=20
> >     =20
> > 6.2 UDP -> TCP
> >=20
> >      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
> > 	   CSeq: 1
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   Content-Base: rtsp://foo/twister/
> > 	   Content-Length: 230
> > 	   Content-Type: application/sdp
> > 	   Cseq: 1
> >=20
> > 	   v=3D0
> > 	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
> > 	   s=3DRTSP Session
> > 	   c=3DIN IP4 0.0.0.0
> > 	   t=3D0 0
> > 	   a=3Dcontrol:*
> > 	   a=3Drange:npt=3D0-10.566000
> > 	   m=3Dvideo 0 RTP/AVP 26
> > 	   a=3Dcontrol:rtsp://foo/twister/video
> > 	   m=3Daudio 0 RTP/AVP 0
> > 	   a=3Dcontrol:rtsp://foo/twister/audio
> >=20
> >      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> > 	   CSeq: 2
> > 	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   CSeq: 2
> > 	   Session: 12345678
> > 	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
> > 		      server_port=3D9000-9001
> >=20
> >      C->S: PLAY rtsp://foo/twister RTSP/1.0
> > 	   CSeq: 3
> > 	   Session: 12345678
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   Content-Base: rtsp://foo/twister/
> > 	   CSeq: 3
> > 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D133=
9949552
> > 	   Range: npt=3D0-
> > 	   Session: 12345678
> >=20
> > ... now request TCP ...
> >=20
> >      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> > 	   CSeq: 4
> > 	   Transport: RTP/AVP/TCP;interleaved=3D0-1
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   Cseq: 4
> > 	   Transport: RTP/AVP/TCP;unicast;
> > 		      client_ip=3D<client IP>;interleaved=3D0-1
> > 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<n=
ew>
> > 	   Range: npt=3D0-10.56600
> > 	   Session: 12345678
> >=20
> >=20
> >=20
> > 6.3 TCP -> UDP
> >=20
> >      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
> > 	   CSeq: 1
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   Content-Base: rtsp://foo/twister/
> > 	   Content-Length: 230
> > 	   Content-Type: application/sdp
> > 	   Cseq: 1
> >=20
> > 	   v=3D0
> > 	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
> > 	   s=3DRTSP Session
> > 	   c=3DIN IP4 0.0.0.0
> > 	   t=3D0 0
> > 	   a=3Dcontrol:*
> > 	   a=3Drange:npt=3D0-10.566000
> > 	   m=3Dvideo 0 RTP/AVP 26
> > 	   a=3Dcontrol:rtsp://foo/twister/video
> > 	   m=3Daudio 0 RTP/AVP 0
> > 	   a=3Dcontrol:rtsp://foo/twister/audio
> >=20
> >      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> > 	   CSeq: 2
> > 	   Transport: RTP/AVP/TCP;interleaved=3D0-1
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   Cseq: 2
> > 	   Session: 12345678
> > 	   Transport: RTP/AVP/TCP;unicast;client_ip=3D<client IP>;interleave=
d=3D0-1
> >=20
> >      C->S: PLAY rtsp://foo/twister RTSP/1.0
> > 	   CSeq: 3
> > 	   Session: 12345678
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   Content-Base: rtsp://foo/twister/
> > 	   Cseq: 3
> > 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D133=
9949552
> > 	   Range: npt=3D0-
> > 	   Session: 12345678
> >=20
> > ... now request UDP ...
> >=20
> >=20
> >      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> > 	   CSeq: 4
> > 	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001
> > 	   Range: npt=3D0-
> >=20
> >      S->C: RTSP/1.0 200 OK
> > 	   CSeq: 4
> > 	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
> > 		      server_port=3D9000-9001
> > 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<n=
ew>
> > 	   Range: npt=3D0-10.56600
> > 	   Session: 12345678
> >=20
> >=20
> >=20
> >=20
> > 7. Other transport related changes to RTSP draft
> >=20
> > A few comments on draft-ietf-mmusic-rfc2326bis-04.txt relating to
> > transport parameters.
> >=20
> > (a) Section 13.40:
> >     Page 81: Are "src_addr" and "dest_addr" parameters
> >     going away? If these are relevant for NAT, perhaps they belong in
> >     the NAT document?
> >=20
> >     Page 83-83: BNF indicates "client_port" and
> >     "server_port" will be indicated as 'port1-port2'. This doesn't se=
em
> >     adequate for the case where the RTP and RTCP ports are non-adjace=
nt
> >     (per the new RTP spec rfc 3550).=20
> >=20
> >     Proposal:
> > 	client_port: rtp=3D<port>, rtcp=3D<port>
> > 	server_port: rtp=3D<port>, rtcp=3D<port>
> >=20
> > (b) Appendix B, B.1.1
> >     Page 115: Mentions "src_addresses" and
> >     "dest_addresses" - what are these?
> >=20
> >     Page 116: "non-continuous" ports - might want to rephrase to
> >     "non-adjacent" (or "non-continguous") ports?
> >=20
> >=20
> >=20
> > 8. Summary
> >=20
> >=20
> > This note recommends negotiation of a few transport parameters and
> > specifies mechanisms to do so. One new transport parameter "client_ip=
"=20
> > is recommended for use in cases where the client's IP address changes
> > during the session.=20
> >=20
> > It is also recommended that the client_port and server_port parameter=
s
> > be specified further with the use of tags "rtp_port" and "rtcp_port"
> > to handle the case where these ports are non-contiguous.
--=20
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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I agree that clarification would be good. The good and the bad of the 
current scheme is that the directionality of send/recv/sendrecv/inactive 
doesn't correlate directly to the general notion of "on hold".

But there is a general notion of "on hold" even if it isn't very well 
defined. The problem is that the relationship between going on/off hold 
to the direction attributes isn't entirely clear, as evidenced by John 
Elwell's experiences below.

	Paul

Elwell, John wrote:
> Jonathan,
> 
> Your reply below, and also the one you sent to Paul Kyzivat, are in line
> with how I had interpreted things. However, we have encountered
> implementations that offer sendonly on audio but do not supply music. The
> problem we then have is whether a UAS should supply its own music. If it
> receives inactive, clearly it is at liberty to do so. If it receives
> sendonly, I believe it should not supply local music, because there may be
> music or some other audio from the far end (although in many cases there is
> not).
> 
> Some clarification in RFC3264 would be beneficial.
> 
> John
> 
> 
> 
>>-----Original Message-----
>>From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
>>Sent: 04 November 2003 00:34
>>To: Elwell, John
>>Cc: 'mmusic@ietf.org'
>>Subject: Re: [MMUSIC] Clarification on use of sendonly for hold
>>
>>
>>To allow for situations where there is muzak or other media placed 
>>while the call is on hold. That said, there is nothing wrong with 
>>using inactive if you want, if you have no plans to do that.
>>
>>-Jonathan R.
>>
>>Elwell, John wrote:
>>
>>
>>>In section 8.4 of RFC3264 it says:
>>>
>>>"Typically, when a user "presses" hold, the agent will 
>>
>>generate an offer 
>>
>>>with all streams in the SDP indicating a direction of 
>>
>>sendonly, and it 
>>
>>>will also locally mute, so that no media is sent to the far 
>>
>> end, and no 
>>
>>>media is played out."
>>>
>>>This seems strange. If the UA is not sending any media to 
>>
>>the far end, 
>>
>>>why not use "inactive" instead of "sendonly"?
>>>
>>>John Elwell (john.elwell@siemens.com)
>>>
>>
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>http://www.jdrosen.net                      PHONE: (973) 952-5000
>>http://www.dynamicsoft.com
>>
>>
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


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Subject: RE: [MMUSIC] RTSP transport negotiation
Date: Wed, 5 Nov 2003 09:58:15 +0100
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Thread-Topic: [MMUSIC] RTSP transport negotiation
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From: "Enrico Balatti" <balatti@cefriel.it>
To: "Geetha Srikantan" <Geetha.Srikantan@Sun.COM>
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Hi,

comments inline:

>a) How does a mobile client get notified about new transports - such as =

>   a new IP address?  How far ahead of time does the client get
>   notified?

In my opinion the notification can be done in two different ways:

1) you have have a mobile client application(like MobileIP, CellularIP =
etc..)=20
   running on you mobile host. In this case this application can inform =
you when
   the handoff happens (before or after the switching to a new =
AccessPoint).
   and if you have changed your IP address or you are reachable from =
outside the new domain
   via a new Public IP address.
=20
2) you have not a mobile client application. You get a new IP address =
via DHCP.
   In this case I think the dhcp client should sends you a trigger to =
inform of the change.

While the 1) can be a very quickly method, I think that 2) is very =
slowly due to the dhcp protocol.

>b) What is the range of time over which the transport switch happens?
>   Consider a scenario where the client requests an IP address change
>   for example, via SETUP with no Range, and the server starts sending
>   packets to the new IP. Is there a chance that the buffers at the new
>   IP get overrun, before the client switches to the new IP?
>   It would help if the network guarantees that the transport switch
>   will not lose any packets due to buffer overflows,

I think that your assumption that "the SETUP during play state happens =
after
the client has switched to the new handoff" is right.=20
Its is very common that when a mobile host enters in a new IP domain, it =
gets
a private IP address or if it use a mobile protocols application, this =
protocol allows
it to maintain its "home address". In both these case the mobile host =
will be reachable
from the "internet" via a Public IP address (NAT). So, in the case of =
rtsp, the mobile client
in a play state has to know the new Public IP address an rtp ports (for =
example using the stun protocol),=20
create the binding on the NAT, and then send the SETUP with the new =
"client_ip".

Best Regards,

------------------------------------------------------------------
Ing Enrico M. Balatti
CEFRIEL - Politecnico di Milano
ICT Center of Excellence For Research, Innovation,
Education & industrial Labs partnership
via Fucini 2, 20133 Milano (ITALY)
voice: +39 0223954327 fax: +39 0223954527
e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   www.cefriel.it =
<http://www.cefriel.it>=20
------------------------------------------------------------------


-----Original Message-----
From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
Sent: marted=EC 4 novembre 2003 19.05
To: Magnus Westerlund
Cc: Enrico Balatti; mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP transport negotiation


hi Magnus,

My assumption was that SETUP during play happens *after* the client has
switched to the new transport..synchronization is an issue in this case.
I agree that if we can assume that the client requests new transport
before switching to the new transport, there is probably no
synchronization issue. I have two follow-up questions on this:

a) How does a mobile client get notified about new transports - such as=20
   a new IP address?  How far ahead of time does the client get
   notified?

b) What is the range of time over which the transport switch happens?
   Consider a scenario where the client requests an IP address change
   for example, via SETUP with no Range, and the server starts sending
   packets to the new IP. Is there a chance that the buffers at the new
   IP get overrun, before the client switches to the new IP?
   It would help if the network guarantees that the transport switch
   will not lose any packets due to buffer overflows,
  =20
thanks
geetha =20



On Mon, 2003-11-03 at 02:30, Magnus Westerlund wrote:
> Hi,
>=20
> In this matter, are there really a need to included a Range header to=20
> indicate when the SETUP request is to change parameters? What is the =
use=20
> case for this. You can't send the SETUP to move to the new transport=20
> prior to that the new transport interface exist. Therefore I don't see =
a=20
> problem with having it change parameters as soon as the request is=20
> received. Unless you have transport interfaces that remain available=20
> during some overlapping time you will of course have a result of =
losing=20
> packets during the switch.
>=20
> Regarding the use of Range header in the SETUP. If appropriately=20
> motivated it can of course be changed. However the potential backwards =

> compatibility problems needs to be considered. If this work becomes=20
> large in impact, we might have to consider splitting it into a =
separate=20
> document. However so far I don't think there are sufficient need.
>=20
> I am sorry that I haven't had time to properly review your proposal.
>=20
> Cheers
>=20
> Magnus
--=20
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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Hi Geetha,

Thanks for your views. Comments inline

Geetha Srikantan wrote:
> hi Magnus,
> 
> My assumption was that SETUP during play happens *after* the client has
> switched to the new transport..synchronization is an issue in this case.
> I agree that if we can assume that the client requests new transport
> before switching to the new transport, there is probably no
> synchronization issue. I have two follow-up questions on this:
> 

Yes, I would also think that in most cases the application will not know 
about the change of transport until it actually happens. This will 
probably result in the loss of some media packets. This loss can be 
dealt with in a few different way depending on the what the application 
can use: media level retransmission, RTSP go back through PAUSE and PLAY.

However in regards the question about what your assumption are and why 
you need Range in the request I think your answer clearly points out 
that there are no need for Range in the Request. There seem to be 
reasons for it to be allowed in the Response, to indicate the correct 
new synchronization after the change of address.

So are you still seeing any need for Range in the SETUP request? If that 
can be motivated I definitely think there will be possibility to change 
the specification to allow for this. However this would need further 
investigation.


> a) How does a mobile client get notified about new transports - such as 
>    a new IP address?  How far ahead of time does the client get
>    notified?

I would say that this is not a question we can answer. I think we simply 
need to ensure that the solution works with both sufficient notification 
and without. However without overlap in transport availability, there 
will be losses of media packets.


> 
> b) What is the range of time over which the transport switch happens?
>    Consider a scenario where the client requests an IP address change
>    for example, via SETUP with no Range, and the server starts sending
>    packets to the new IP. Is there a chance that the buffers at the new
>    IP get overrun, before the client switches to the new IP?
>    It would help if the network guarantees that the transport switch
>    will not lose any packets due to buffer overflows,
>    

As I have stated prior I think we must require the client to have access 
to an interface it sends traffic to. If there is overlap I would assume 
that it can receive data on both during this period. I don't find it 
reasonable to get any guarantees from the networks. I would assume that 
any switch of network depends on the change of access technology, for 
example by switching from cable to WLAN or to 3G mobile network. The 
probability that there will exist any mobility functionality between 
these networks seems very small.

I think that we should design our solution to allow the mobility, only 
note that it may loss some packets

Cheers


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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Folks,

attached is a slightly updated MMUSIC agenda.
The HTML version with all documents linked is available as usual
from

     http://www.dmn.tzi.org/ietf/mmusic/

Joerg

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

Multiparty Multimedia Session Control (MMUSIC)

CHAIRS: Joerg Ott <jo@informatik.uni-bremen.de>
Colin Perkins <csp@csperkins.org>


Friday, 14 November 2003, 09:00 - 11:30
---------------------------------------

0900    Agenda Bashing and Status Update               (Chairs, 15)

         Recent RFCs
         draft-ietf-mmusic-sdp4nat-05.txt                -> RFC 3605

         With RFC editor
         draft-ietf-mmusic-sdpng-trans-04.txt

         With IESG
         draft-ietf-mmusic-sdp-new-15.txt
         draft-rajeshkumar-mmusic-gpmd-03.txt

         With AD
         draft-ietf-mmusic-sdp-comedia-05.txt
         draft-ietf-mmusic-sdp-bwparam-05.txt
         draft-ietf-mmusic-srcfilter-05.txt

         Misc Updates
         draft-ietf-mmusic-kmgmt-ext-09.txt
         draft-ietf-mmusic-sdp-implem-00.txt
         draft-ietf-mmusic-sdp-new-15.txt

0915    Internet Media Guides                           (Manyfolks, 30)
         draft-ietf-mmusic-img-req-00.txt
         draft-nomura-mmusic-img-framework-02.txt
         (draft-luoma-mmusic-img-metadata-02.txt fyi only)
         (draft-luoma-mmusic-img-muppet-03.txt fyi only)

0945    SDP for alternate IP versions                   (Camarillo, 10)
         draft-camarillo-mmusic-alt-02.txt

0955    SDP Security Descriptions for Media Streams     (Andreasen, 15)
         draft-ietf-mmusic-sdescriptions-02.txt
         draft-andreasen-mmusic-securityprecondition-00.txt

0945    SDP for alternate IP versions                   (Camarillo, 10)
         draft-camarillo-mmusic-alt-02.txt

0955    SDP Security Descriptions for Media Streams     (Andreasen, 15)
         draft-ietf-mmusic-sdescriptions-02.txt
         draft-andreasen-mmusic-securityprecondition-00.txt

1010    SDP parameters to support H.263/H.261 options   (Even/Ismail, 15)
         draft-even-avt-h263-h261-options-00.txt
         draft-nismail-mmusic-sdp-video-00.txt

1025    Interactive Connectivity Establishment (ICE)    (Rosenberg, 20)
         draft-ietf-mmusic-ice-00.txt
         draft-rosenberg-sipping-ice-01.txt

1045    SDPng Update                                    (Kutscher, 15)
         draft-ietf-mmusic-sdpng-07.txt

1100    RTSP Update                              (Westerlund/Zeng, 30)
         draft-ietf-mmusic-rfc2326bis-05.txt
         draft-ietf-mmusic-rtsp-nat-01.txt
         draft-zeng-rtsp-end-of-stream-00.txt

1130    Close

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

                                 NOTE WELL

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Date: Wed, 05 Nov 2003 10:37:33 -0800
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: RE: [MMUSIC] RTSP transport negotiation
In-reply-to: <6F5FB8313FE82C4EA10DA4DA40B90949F74810@swing.cefriel.it>
To: Enrico Balatti <balatti@cefriel.it>
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Enrico,

Thanks for the background..this helps to understand the scenario better.

geetha


On Wed, 2003-11-05 at 00:58, Enrico Balatti wrote:
> Hi,
>=20
> comments inline:
>=20
> >a) How does a mobile client get notified about new transports - such a=
s=20
> >   a new IP address?  How far ahead of time does the client get
> >   notified?
>=20
> In my opinion the notification can be done in two different ways:
>=20
> 1) you have have a mobile client application(like MobileIP, CellularIP =
etc..)=20
>    running on you mobile host. In this case this application can inform=
 you when
>    the handoff happens (before or after the switching to a new AccessPo=
int).
>    and if you have changed your IP address or you are reachable from ou=
tside the new domain
>    via a new Public IP address.
> =20
> 2) you have not a mobile client application. You get a new IP address v=
ia DHCP.
>    In this case I think the dhcp client should sends you a trigger to i=
nform of the change.
>=20
> While the 1) can be a very quickly method, I think that 2) is very slow=
ly due to the dhcp protocol.
>=20
> >b) What is the range of time over which the transport switch happens?
> >   Consider a scenario where the client requests an IP address change
> >   for example, via SETUP with no Range, and the server starts sending
> >   packets to the new IP. Is there a chance that the buffers at the ne=
w
> >   IP get overrun, before the client switches to the new IP?
> >   It would help if the network guarantees that the transport switch
> >   will not lose any packets due to buffer overflows,
>=20
> I think that your assumption that "the SETUP during play state happens =
after
> the client has switched to the new handoff" is right.=20
> Its is very common that when a mobile host enters in a new IP domain, i=
t gets
> a private IP address or if it use a mobile protocols application, this =
protocol allows
> it to maintain its "home address". In both these case the mobile host w=
ill be reachable
> from the "internet" via a Public IP address (NAT). So, in the case of r=
tsp, the mobile client
> in a play state has to know the new Public IP address an rtp ports (for=
 example using the stun protocol),=20
> create the binding on the NAT, and then send the SETUP with the new "cl=
ient_ip".
>=20
> Best Regards,
>=20
> ------------------------------------------------------------------
> Ing Enrico M. Balatti
> CEFRIEL - Politecnico di Milano
> ICT Center of Excellence For Research, Innovation,
> Education & industrial Labs partnership
> via Fucini 2, 20133 Milano (ITALY)
> voice: +39 0223954327 fax: +39 0223954527
> e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   www.cefriel.it=
 <http://www.cefriel.it>=20
> ------------------------------------------------------------------
>=20
>=20
> -----Original Message-----
> From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
> Sent: marted=EC 4 novembre 2003 19.05
> To: Magnus Westerlund
> Cc: Enrico Balatti; mmusic@ietf.org
> Subject: Re: [MMUSIC] RTSP transport negotiation
>=20
>=20
> hi Magnus,
>=20
> My assumption was that SETUP during play happens *after* the client has
> switched to the new transport..synchronization is an issue in this case.
> I agree that if we can assume that the client requests new transport
> before switching to the new transport, there is probably no
> synchronization issue. I have two follow-up questions on this:
>=20
> a) How does a mobile client get notified about new transports - such as=
=20
>    a new IP address?  How far ahead of time does the client get
>    notified?
>=20
> b) What is the range of time over which the transport switch happens?
>    Consider a scenario where the client requests an IP address change
>    for example, via SETUP with no Range, and the server starts sending
>    packets to the new IP. Is there a chance that the buffers at the new
>    IP get overrun, before the client switches to the new IP?
>    It would help if the network guarantees that the transport switch
>    will not lose any packets due to buffer overflows,
>   =20
> thanks
> geetha =20
>=20
>=20
>=20
> On Mon, 2003-11-03 at 02:30, Magnus Westerlund wrote:
> > Hi,
> >=20
> > In this matter, are there really a need to included a Range header to=
=20
> > indicate when the SETUP request is to change parameters? What is the =
use=20
> > case for this. You can't send the SETUP to move to the new transport=20
> > prior to that the new transport interface exist. Therefore I don't se=
e a=20
> > problem with having it change parameters as soon as the request is=20
> > received. Unless you have transport interfaces that remain available=20
> > during some overlapping time you will of course have a result of losi=
ng=20
> > packets during the switch.
> >=20
> > Regarding the use of Range header in the SETUP. If appropriately=20
> > motivated it can of course be changed. However the potential backward=
s=20
> > compatibility problems needs to be considered. If this work becomes=20
> > large in impact, we might have to consider splitting it into a separa=
te=20
> > document. However so far I don't think there are sufficient need.
> >=20
> > I am sorry that I haven't had time to properly review your proposal.
> >=20
> > Cheers
> >=20
> > Magnus
--=20
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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Date: Wed, 05 Nov 2003 10:54:51 -0800
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: Re: [MMUSIC] RTSP transport negotiation
In-reply-to: <3FA8DDB4.5090700@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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hi Magnus,

Yes, it looks like we can drop the Range header from SETUP request.
We probably want the Range header + RTP-Info in the SETUP response
for VOD streams...with the caveat that packets can be lost during the
switch. 

I don't believe all components of the RTP-Info header are relevant for
live streams that are 'reflected' by the server - here the server acts
as a translator. The RTP stream url is known, however the exact sequence
number and timestamp for RTP packets is determined by the source of the
live stream, not the server reflecting the stream. Hence, I recommend
that the RTP-Info carry no sequence number or rtptime for live streams.

Also, I noticed that the RTP spec requires change of SSRC when the
transport changes. This needs to be added to the transport negotiation 
notes.

I will update the notes and send it to the alias.

geetha


On Wed, 2003-11-05 at 03:23, Magnus Westerlund wrote:
> Hi Geetha,
> 
> Thanks for your views. Comments inline
> 
> Geetha Srikantan wrote:
> > hi Magnus,
> > 
> > My assumption was that SETUP during play happens *after* the client has
> > switched to the new transport..synchronization is an issue in this case.
> > I agree that if we can assume that the client requests new transport
> > before switching to the new transport, there is probably no
> > synchronization issue. I have two follow-up questions on this:
> > 
> 
> Yes, I would also think that in most cases the application will not know 
> about the change of transport until it actually happens. This will 
> probably result in the loss of some media packets. This loss can be 
> dealt with in a few different way depending on the what the application 
> can use: media level retransmission, RTSP go back through PAUSE and PLAY.
> 
> However in regards the question about what your assumption are and why 
> you need Range in the request I think your answer clearly points out 
> that there are no need for Range in the Request. There seem to be 
> reasons for it to be allowed in the Response, to indicate the correct 
> new synchronization after the change of address.
> 
> So are you still seeing any need for Range in the SETUP request? If that 
> can be motivated I definitely think there will be possibility to change 
> the specification to allow for this. However this would need further 
> investigation.
> 
> 
> > a) How does a mobile client get notified about new transports - such as 
> >    a new IP address?  How far ahead of time does the client get
> >    notified?
> 
> I would say that this is not a question we can answer. I think we simply 
> need to ensure that the solution works with both sufficient notification 
> and without. However without overlap in transport availability, there 
> will be losses of media packets.
> 
> 
> > 
> > b) What is the range of time over which the transport switch happens?
> >    Consider a scenario where the client requests an IP address change
> >    for example, via SETUP with no Range, and the server starts sending
> >    packets to the new IP. Is there a chance that the buffers at the new
> >    IP get overrun, before the client switches to the new IP?
> >    It would help if the network guarantees that the transport switch
> >    will not lose any packets due to buffer overflows,
> >    
> 
> As I have stated prior I think we must require the client to have access 
> to an interface it sends traffic to. If there is overlap I would assume 
> that it can receive data on both during this period. I don't find it 
> reasonable to get any guarantees from the networks. I would assume that 
> any switch of network depends on the change of access technology, for 
> example by switching from cable to WLAN or to 3G mobile network. The 
> probability that there will exist any mobility functionality between 
> these networks seems very small.
> 
> I think that we should design our solution to allow the mobility, only 
> note that it may loss some packets
> 
> Cheers
> 
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
-- 
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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Thread-Topic: [MMUSIC] RTSP transport negotiation
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From: "Gamze Seckin" <gamze@vidiator.com>
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Subject: [MMUSIC] sdp alt, alt-default-id and alt-group
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Dear mmusic members,

my apologies if this is discussed here before:

I am looking for ietf pointers (if there are any) on the sdp =
descriptions provided in =
http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/ document =
s4-30680.
on sdp "alt" and "alt-default-id" attributes and the session level =
grouping attribute, "alt-group".

best regards
g



-----------------------------------------------------------
Gamze Seckin, Ph.D.
Director of Research
Vidiator Technology (US) Inc.
411 108th Avenue NE. Suite 688
Bellevue, WA 98004 USA.
www.vidiator.com

Phone: 1.425.688.8811x2108
Email: gamze@vidiator.com

This e-mail message (including any attachment) is intended only for the =
personal use of the recipient(s) named above. This message is =
confidential and may be legally privileged. If you are not an intended =
recipient, you may not review, copy or distribute this message. If you =
have received this communication in error, please notify us immediately =
by e-mail and delete the original message. Any views or opinions =
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this message including any attachment does not create any legally =
binding rights or obligations whatsoever, which may only be created by =
the exchange of hard copy documents signed by a duly authorized =
representative of Vidiator Technology (US) Inc.
-----------------------------------------------------------


-----Original Message-----
From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
Sent: Wednesday, November 05, 2003 10:55 AM
To: Magnus Westerlund
Cc: Enrico Balatti; mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP transport negotiation



hi Magnus,

Yes, it looks like we can drop the Range header from SETUP request.
We probably want the Range header + RTP-Info in the SETUP response
for VOD streams...with the caveat that packets can be lost during the
switch.=20

I don't believe all components of the RTP-Info header are relevant for
live streams that are 'reflected' by the server - here the server acts
as a translator. The RTP stream url is known, however the exact sequence
number and timestamp for RTP packets is determined by the source of the
live stream, not the server reflecting the stream. Hence, I recommend
that the RTP-Info carry no sequence number or rtptime for live streams.

Also, I noticed that the RTP spec requires change of SSRC when the
transport changes. This needs to be added to the transport negotiation=20
notes.

I will update the notes and send it to the alias.

geetha


On Wed, 2003-11-05 at 03:23, Magnus Westerlund wrote:
> Hi Geetha,
>=20
> Thanks for your views. Comments inline
>=20
> Geetha Srikantan wrote:
> > hi Magnus,
> >=20
> > My assumption was that SETUP during play happens *after* the client =
has
> > switched to the new transport..synchronization is an issue in this =
case.
> > I agree that if we can assume that the client requests new transport
> > before switching to the new transport, there is probably no
> > synchronization issue. I have two follow-up questions on this:
> >=20
>=20
> Yes, I would also think that in most cases the application will not =
know=20
> about the change of transport until it actually happens. This will=20
> probably result in the loss of some media packets. This loss can be=20
> dealt with in a few different way depending on the what the =
application=20
> can use: media level retransmission, RTSP go back through PAUSE and =
PLAY.
>=20
> However in regards the question about what your assumption are and why =

> you need Range in the request I think your answer clearly points out=20
> that there are no need for Range in the Request. There seem to be=20
> reasons for it to be allowed in the Response, to indicate the correct=20
> new synchronization after the change of address.
>=20
> So are you still seeing any need for Range in the SETUP request? If =
that=20
> can be motivated I definitely think there will be possibility to =
change=20
> the specification to allow for this. However this would need further=20
> investigation.
>=20
>=20
> > a) How does a mobile client get notified about new transports - such =
as=20
> >    a new IP address?  How far ahead of time does the client get
> >    notified?
>=20
> I would say that this is not a question we can answer. I think we =
simply=20
> need to ensure that the solution works with both sufficient =
notification=20
> and without. However without overlap in transport availability, there=20
> will be losses of media packets.
>=20
>=20
> >=20
> > b) What is the range of time over which the transport switch =
happens?
> >    Consider a scenario where the client requests an IP address =
change
> >    for example, via SETUP with no Range, and the server starts =
sending
> >    packets to the new IP. Is there a chance that the buffers at the =
new
> >    IP get overrun, before the client switches to the new IP?
> >    It would help if the network guarantees that the transport switch
> >    will not lose any packets due to buffer overflows,
> >   =20
>=20
> As I have stated prior I think we must require the client to have =
access=20
> to an interface it sends traffic to. If there is overlap I would =
assume=20
> that it can receive data on both during this period. I don't find it=20
> reasonable to get any guarantees from the networks. I would assume =
that=20
> any switch of network depends on the change of access technology, for=20
> example by switching from cable to WLAN or to 3G mobile network. The=20
> probability that there will exist any mobility functionality between=20
> these networks seems very small.
>=20
> I think that we should design our solution to allow the mobility, only =

> note that it may loss some packets
>=20
> Cheers
>=20
>=20
> Magnus Westerlund
>=20
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
--=20
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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Subject: RE: [MMUSIC] sdp alt, alt-default-id and alt-group
Date: Thu, 6 Nov 2003 11:11:07 -0800
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Thread-Topic: [MMUSIC] sdp alt, alt-default-id and alt-group
Thread-Index: AcOklVdCNoiLrB6kRSW2swUJNDQ8gAAA5R9g
From: "Gamze Seckin" <gamze@vidiator.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
Cc: <mmusic@ietf.org>
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Hi Magnus,

Thanks. I do not see bugs and I think this proposal could be useful =
actually.
But I also think it is important that IETF takes into account these =
sections somehow, since they may have an impact on several proposals, =
say such as transport independent bw modifiers?

br
g


-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
Sent: Thursday, November 06, 2003 10:34 AM
To: Gamze Seckin
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] sdp alt, alt-default-id and alt-group


Hi Gamze,

No there are no IETF pointers about this proposal. This work is totally=20
created within the 3GPP SA4. There should probably have been some review =

  by the IETF before the acceptance of the proposal, but that has not=20
happened. However as the Release 6 has not been approved yet, any bugs=20
in this proposal can still be fixed.

If you are interested in this proposal please review chapter 5.3.3.3.=20
and  5.3.3.4 in the 3GPP TS 26.234 version 0.3.0 of Release 6=20
specification, which can be downloaded here:

http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/S4-030680.zip

This is a 200 kb zipped word document.

Cheers

Magnus

Gamze Seckin wrote:

> Dear mmusic members,
>=20
> my apologies if this is discussed here before:
>=20
> I am looking for ietf pointers (if there are any) on the sdp =
descriptions provided in =
http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/ document =
s4-30680.
> on sdp "alt" and "alt-default-id" attributes and the session level =
grouping attribute, "alt-group".
>=20
> best regards
> g
>=20
>=20
>=20
> -----------------------------------------------------------
> Gamze Seckin, Ph.D.
> Director of Research
> Vidiator Technology (US) Inc.
> 411 108th Avenue NE. Suite 688
> Bellevue, WA 98004 USA.
> www.vidiator.com
>=20
> Phone: 1.425.688.8811x2108
> Email: gamze@vidiator.com
>=20
> This e-mail message (including any attachment) is intended only for =
the personal use of the recipient(s) named above. This message is =
confidential and may be legally privileged. If you are not an intended =
recipient, you may not review, copy or distribute this message. If you =
have received this communication in error, please notify us immediately =
by e-mail and delete the original message. Any views or opinions =
expressed in this message are those of the author only. Furthermore, =
this message including any attachment does not create any legally =
binding rights or obligations whatsoever, which may only be created by =
the exchange of hard copy documents signed by a duly authorized =
representative of Vidiator Technology (US) Inc.
> -----------------------------------------------------------
>=20
>=20
> -----Original Message-----
> From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
> Sent: Wednesday, November 05, 2003 10:55 AM
> To: Magnus Westerlund
> Cc: Enrico Balatti; mmusic@ietf.org
> Subject: Re: [MMUSIC] RTSP transport negotiation
>=20
>=20
>=20
> hi Magnus,
>=20
> Yes, it looks like we can drop the Range header from SETUP request.
> We probably want the Range header + RTP-Info in the SETUP response
> for VOD streams...with the caveat that packets can be lost during the
> switch.=20
>=20
> I don't believe all components of the RTP-Info header are relevant for
> live streams that are 'reflected' by the server - here the server acts
> as a translator. The RTP stream url is known, however the exact =
sequence
> number and timestamp for RTP packets is determined by the source of =
the
> live stream, not the server reflecting the stream. Hence, I recommend
> that the RTP-Info carry no sequence number or rtptime for live =
streams.
>=20
> Also, I noticed that the RTP spec requires change of SSRC when the
> transport changes. This needs to be added to the transport negotiation =

> notes.
>=20
> I will update the notes and send it to the alias.
>=20
> geetha
>=20
>=20
> On Wed, 2003-11-05 at 03:23, Magnus Westerlund wrote:
>=20
>>Hi Geetha,
>>
>>Thanks for your views. Comments inline
>>
>>Geetha Srikantan wrote:
>>
>>>hi Magnus,
>>>
>>>My assumption was that SETUP during play happens *after* the client =
has
>>>switched to the new transport..synchronization is an issue in this =
case.
>>>I agree that if we can assume that the client requests new transport
>>>before switching to the new transport, there is probably no
>>>synchronization issue. I have two follow-up questions on this:
>>>
>>
>>Yes, I would also think that in most cases the application will not =
know=20
>>about the change of transport until it actually happens. This will=20
>>probably result in the loss of some media packets. This loss can be=20
>>dealt with in a few different way depending on the what the =
application=20
>>can use: media level retransmission, RTSP go back through PAUSE and =
PLAY.
>>
>>However in regards the question about what your assumption are and why =

>>you need Range in the request I think your answer clearly points out=20
>>that there are no need for Range in the Request. There seem to be=20
>>reasons for it to be allowed in the Response, to indicate the correct=20
>>new synchronization after the change of address.
>>
>>So are you still seeing any need for Range in the SETUP request? If =
that=20
>>can be motivated I definitely think there will be possibility to =
change=20
>>the specification to allow for this. However this would need further=20
>>investigation.
>>
>>
>>
>>>a) How does a mobile client get notified about new transports - such =
as=20
>>>   a new IP address?  How far ahead of time does the client get
>>>   notified?
>>
>>I would say that this is not a question we can answer. I think we =
simply=20
>>need to ensure that the solution works with both sufficient =
notification=20
>>and without. However without overlap in transport availability, there=20
>>will be losses of media packets.
>>
>>
>>
>>>b) What is the range of time over which the transport switch happens?
>>>   Consider a scenario where the client requests an IP address change
>>>   for example, via SETUP with no Range, and the server starts =
sending
>>>   packets to the new IP. Is there a chance that the buffers at the =
new
>>>   IP get overrun, before the client switches to the new IP?
>>>   It would help if the network guarantees that the transport switch
>>>   will not lose any packets due to buffer overflows,
>>>  =20
>>
>>As I have stated prior I think we must require the client to have =
access=20
>>to an interface it sends traffic to. If there is overlap I would =
assume=20
>>that it can receive data on both during this period. I don't find it=20
>>reasonable to get any guarantees from the networks. I would assume =
that=20
>>any switch of network depends on the change of access technology, for=20
>>example by switching from cable to WLAN or to 3G mobile network. The=20
>>probability that there will exist any mobility functionality between=20
>>these networks seems very small.
>>
>>I think that we should design our solution to allow the mobility, only =

>>note that it may loss some packets
>>
>>Cheers
>>
>>
>>Magnus Westerlund
>>
>>Multimedia Technologies, Ericsson Research EAB/TVA/A
>>----------------------------------------------------------------------
>>Ericsson AB                | Phone +46 8 4048287
>>Torshamsgatan 23           | Fax   +46 8 7575550
>>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic

--=20

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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Date: Thu, 06 Nov 2003 14:55:39 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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CC: Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] sdp alt, alt-default-id and alt-group
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Well, I've not seen the 3gpp docs, but there is an alt SDP parameter 
defined in ICE:

http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-00.txt

Is this the same as what is being used by 3gpp? If not, we have a 
serious name collision...

-Jonathan R.

Gamze Seckin wrote:

> Hi Magnus,
> 
> Thanks. I do not see bugs and I think this proposal could be useful actually.
> But I also think it is important that IETF takes into account these sections somehow, since they may have an impact on several proposals, say such as transport independent bw modifiers?
> 
> br
> g
> 
> 
> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Thursday, November 06, 2003 10:34 AM
> To: Gamze Seckin
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] sdp alt, alt-default-id and alt-group
> 
> 
> Hi Gamze,
> 
> No there are no IETF pointers about this proposal. This work is totally 
> created within the 3GPP SA4. There should probably have been some review 
>   by the IETF before the acceptance of the proposal, but that has not 
> happened. However as the Release 6 has not been approved yet, any bugs 
> in this proposal can still be fixed.
> 
> If you are interested in this proposal please review chapter 5.3.3.3. 
> and  5.3.3.4 in the 3GPP TS 26.234 version 0.3.0 of Release 6 
> specification, which can be downloaded here:
> 
> http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/S4-030680.zip
> 
> This is a 200 kb zipped word document.
> 
> Cheers
> 
> Magnus
> 
> Gamze Seckin wrote:
> 
> 
>>Dear mmusic members,
>>
>>my apologies if this is discussed here before:
>>
>>I am looking for ietf pointers (if there are any) on the sdp descriptions provided in http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/ document s4-30680.
>>on sdp "alt" and "alt-default-id" attributes and the session level grouping attribute, "alt-group".
>>
>>best regards
>>g
>>
>>
>>
>>-----------------------------------------------------------
>>Gamze Seckin, Ph.D.
>>Director of Research
>>Vidiator Technology (US) Inc.
>>411 108th Avenue NE. Suite 688
>>Bellevue, WA 98004 USA.
>>www.vidiator.com
>>
>>Phone: 1.425.688.8811x2108
>>Email: gamze@vidiator.com
>>
>>This e-mail message (including any attachment) is intended only for the personal use of the recipient(s) named above. This message is confidential and may be legally privileged. If you are not an intended recipient, you may not review, copy or distribute this message. If you have received this communication in error, please notify us immediately by e-mail and delete the original message. Any views or opinions expressed in this message are those of the author only. Furthermore, this message including any attachment does not create any legally binding rights or obligations whatsoever, which may only be created by the exchange of hard copy documents signed by a duly authorized representative of Vidiator Technology (US) Inc.
>>-----------------------------------------------------------
>>
>>
>>-----Original Message-----
>>From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
>>Sent: Wednesday, November 05, 2003 10:55 AM
>>To: Magnus Westerlund
>>Cc: Enrico Balatti; mmusic@ietf.org
>>Subject: Re: [MMUSIC] RTSP transport negotiation
>>
>>
>>
>>hi Magnus,
>>
>>Yes, it looks like we can drop the Range header from SETUP request.
>>We probably want the Range header + RTP-Info in the SETUP response
>>for VOD streams...with the caveat that packets can be lost during the
>>switch. 
>>
>>I don't believe all components of the RTP-Info header are relevant for
>>live streams that are 'reflected' by the server - here the server acts
>>as a translator. The RTP stream url is known, however the exact sequence
>>number and timestamp for RTP packets is determined by the source of the
>>live stream, not the server reflecting the stream. Hence, I recommend
>>that the RTP-Info carry no sequence number or rtptime for live streams.
>>
>>Also, I noticed that the RTP spec requires change of SSRC when the
>>transport changes. This needs to be added to the transport negotiation 
>>notes.
>>
>>I will update the notes and send it to the alias.
>>
>>geetha
>>
>>
>>On Wed, 2003-11-05 at 03:23, Magnus Westerlund wrote:
>>
>>
>>>Hi Geetha,
>>>
>>>Thanks for your views. Comments inline
>>>
>>>Geetha Srikantan wrote:
>>>
>>>
>>>>hi Magnus,
>>>>
>>>>My assumption was that SETUP during play happens *after* the client has
>>>>switched to the new transport..synchronization is an issue in this case.
>>>>I agree that if we can assume that the client requests new transport
>>>>before switching to the new transport, there is probably no
>>>>synchronization issue. I have two follow-up questions on this:
>>>>
>>>
>>>Yes, I would also think that in most cases the application will not know 
>>>about the change of transport until it actually happens. This will 
>>>probably result in the loss of some media packets. This loss can be 
>>>dealt with in a few different way depending on the what the application 
>>>can use: media level retransmission, RTSP go back through PAUSE and PLAY.
>>>
>>>However in regards the question about what your assumption are and why 
>>>you need Range in the request I think your answer clearly points out 
>>>that there are no need for Range in the Request. There seem to be 
>>>reasons for it to be allowed in the Response, to indicate the correct 
>>>new synchronization after the change of address.
>>>
>>>So are you still seeing any need for Range in the SETUP request? If that 
>>>can be motivated I definitely think there will be possibility to change 
>>>the specification to allow for this. However this would need further 
>>>investigation.
>>>
>>>
>>>
>>>
>>>>a) How does a mobile client get notified about new transports - such as 
>>>>  a new IP address?  How far ahead of time does the client get
>>>>  notified?
>>>
>>>I would say that this is not a question we can answer. I think we simply 
>>>need to ensure that the solution works with both sufficient notification 
>>>and without. However without overlap in transport availability, there 
>>>will be losses of media packets.
>>>
>>>
>>>
>>>
>>>>b) What is the range of time over which the transport switch happens?
>>>>  Consider a scenario where the client requests an IP address change
>>>>  for example, via SETUP with no Range, and the server starts sending
>>>>  packets to the new IP. Is there a chance that the buffers at the new
>>>>  IP get overrun, before the client switches to the new IP?
>>>>  It would help if the network guarantees that the transport switch
>>>>  will not lose any packets due to buffer overflows,
>>>>  
>>>
>>>As I have stated prior I think we must require the client to have access 
>>>to an interface it sends traffic to. If there is overlap I would assume 
>>>that it can receive data on both during this period. I don't find it 
>>>reasonable to get any guarantees from the networks. I would assume that 
>>>any switch of network depends on the change of access technology, for 
>>>example by switching from cable to WLAN or to 3G mobile network. The 
>>>probability that there will exist any mobility functionality between 
>>>these networks seems very small.
>>>
>>>I think that we should design our solution to allow the mobility, only 
>>>note that it may loss some packets
>>>
>>>Cheers
>>>
>>>
>>>Magnus Westerlund
>>>
>>>Multimedia Technologies, Ericsson Research EAB/TVA/A
>>>----------------------------------------------------------------------
>>>Ericsson AB                | Phone +46 8 4048287
>>>Torshamsgatan 23           | Fax   +46 8 7575550
>>>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>>
>>>
>>>_______________________________________________
>>>mmusic mailing list
>>>mmusic@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



From exim@www1.ietf.org  Thu Nov  6 16:43:26 2003
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Subject: RE: [MMUSIC] sdp alt, alt-default-id and alt-group
Date: Thu, 6 Nov 2003 13:41:37 -0800
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From: "Gamze Seckin" <gamze@vidiator.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, <mmusic@ietf.org>
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Hi Jonathan

The "alt" in the ice draft is different than the "alt" in the 3gpp =
document.

The purpose of the alt, alt-default-id and alt-group definitions in =
26.234 for Rel6 is to define alternate groupings for several streaming =
tracks, based on bitrate or language.=20
And it is an efficient method. I believe it should be taken into account =
by IETF for further work on sdp related efforts.=20

However I am not sure how to solve the naming overlap.

g

Gamze Seckin, Ph.D.
Vidiator Technology US
www.vidiator.com



-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Thursday, November 06, 2003 11:56 AM
To: Gamze Seckin
Cc: Magnus Westerlund; mmusic@ietf.org
Subject: Re: [MMUSIC] sdp alt, alt-default-id and alt-group


Well, I've not seen the 3gpp docs, but there is an alt SDP parameter=20
defined in ICE:

http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-00.txt

Is this the same as what is being used by 3gpp? If not, we have a=20
serious name collision...

-Jonathan R.

Gamze Seckin wrote:

> Hi Magnus,
>=20
> Thanks. I do not see bugs and I think this proposal could be useful =
actually.
> But I also think it is important that IETF takes into account these =
sections somehow, since they may have an impact on several proposals, =
say such as transport independent bw modifiers?
>=20
> br
> g
>=20
>=20
> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Thursday, November 06, 2003 10:34 AM
> To: Gamze Seckin
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] sdp alt, alt-default-id and alt-group
>=20
>=20
> Hi Gamze,
>=20
> No there are no IETF pointers about this proposal. This work is =
totally=20
> created within the 3GPP SA4. There should probably have been some =
review=20
>   by the IETF before the acceptance of the proposal, but that has not=20
> happened. However as the Release 6 has not been approved yet, any bugs =

> in this proposal can still be fixed.
>=20
> If you are interested in this proposal please review chapter 5.3.3.3.=20
> and  5.3.3.4 in the 3GPP TS 26.234 version 0.3.0 of Release 6=20
> specification, which can be downloaded here:
>=20
> http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/S4-030680.zip
>=20
> This is a 200 kb zipped word document.
>=20
> Cheers
>=20
> Magnus
>=20
> Gamze Seckin wrote:
>=20
>=20
>>Dear mmusic members,
>>
>>my apologies if this is discussed here before:
>>
>>I am looking for ietf pointers (if there are any) on the sdp =
descriptions provided in =
http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/ document =
s4-30680.
>>on sdp "alt" and "alt-default-id" attributes and the session level =
grouping attribute, "alt-group".
>>
>>best regards
>>g
>>
>>
>>
>>-----------------------------------------------------------
>>Gamze Seckin, Ph.D.
>>Director of Research
>>Vidiator Technology (US) Inc.
>>411 108th Avenue NE. Suite 688
>>Bellevue, WA 98004 USA.
>>www.vidiator.com
>>
>>Phone: 1.425.688.8811x2108
>>Email: gamze@vidiator.com
>>
>>This e-mail message (including any attachment) is intended only for =
the personal use of the recipient(s) named above. This message is =
confidential and may be legally privileged. If you are not an intended =
recipient, you may not review, copy or distribute this message. If you =
have received this communication in error, please notify us immediately =
by e-mail and delete the original message. Any views or opinions =
expressed in this message are those of the author only. Furthermore, =
this message including any attachment does not create any legally =
binding rights or obligations whatsoever, which may only be created by =
the exchange of hard copy documents signed by a duly authorized =
representative of Vidiator Technology (US) Inc.
>>-----------------------------------------------------------
>>
>>
>>-----Original Message-----
>>From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
>>Sent: Wednesday, November 05, 2003 10:55 AM
>>To: Magnus Westerlund
>>Cc: Enrico Balatti; mmusic@ietf.org
>>Subject: Re: [MMUSIC] RTSP transport negotiation
>>
>>
>>
>>hi Magnus,
>>
>>Yes, it looks like we can drop the Range header from SETUP request.
>>We probably want the Range header + RTP-Info in the SETUP response
>>for VOD streams...with the caveat that packets can be lost during the
>>switch.=20
>>
>>I don't believe all components of the RTP-Info header are relevant for
>>live streams that are 'reflected' by the server - here the server acts
>>as a translator. The RTP stream url is known, however the exact =
sequence
>>number and timestamp for RTP packets is determined by the source of =
the
>>live stream, not the server reflecting the stream. Hence, I recommend
>>that the RTP-Info carry no sequence number or rtptime for live =
streams.
>>
>>Also, I noticed that the RTP spec requires change of SSRC when the
>>transport changes. This needs to be added to the transport negotiation =

>>notes.
>>
>>I will update the notes and send it to the alias.
>>
>>geetha
>>
>>
>>On Wed, 2003-11-05 at 03:23, Magnus Westerlund wrote:
>>
>>
>>>Hi Geetha,
>>>
>>>Thanks for your views. Comments inline
>>>
>>>Geetha Srikantan wrote:
>>>
>>>
>>>>hi Magnus,
>>>>
>>>>My assumption was that SETUP during play happens *after* the client =
has
>>>>switched to the new transport..synchronization is an issue in this =
case.
>>>>I agree that if we can assume that the client requests new transport
>>>>before switching to the new transport, there is probably no
>>>>synchronization issue. I have two follow-up questions on this:
>>>>
>>>
>>>Yes, I would also think that in most cases the application will not =
know=20
>>>about the change of transport until it actually happens. This will=20
>>>probably result in the loss of some media packets. This loss can be=20
>>>dealt with in a few different way depending on the what the =
application=20
>>>can use: media level retransmission, RTSP go back through PAUSE and =
PLAY.
>>>
>>>However in regards the question about what your assumption are and =
why=20
>>>you need Range in the request I think your answer clearly points out=20
>>>that there are no need for Range in the Request. There seem to be=20
>>>reasons for it to be allowed in the Response, to indicate the correct =

>>>new synchronization after the change of address.
>>>
>>>So are you still seeing any need for Range in the SETUP request? If =
that=20
>>>can be motivated I definitely think there will be possibility to =
change=20
>>>the specification to allow for this. However this would need further=20
>>>investigation.
>>>
>>>
>>>
>>>
>>>>a) How does a mobile client get notified about new transports - such =
as=20
>>>>  a new IP address?  How far ahead of time does the client get
>>>>  notified?
>>>
>>>I would say that this is not a question we can answer. I think we =
simply=20
>>>need to ensure that the solution works with both sufficient =
notification=20
>>>and without. However without overlap in transport availability, there =

>>>will be losses of media packets.
>>>
>>>
>>>
>>>
>>>>b) What is the range of time over which the transport switch =
happens?
>>>>  Consider a scenario where the client requests an IP address change
>>>>  for example, via SETUP with no Range, and the server starts =
sending
>>>>  packets to the new IP. Is there a chance that the buffers at the =
new
>>>>  IP get overrun, before the client switches to the new IP?
>>>>  It would help if the network guarantees that the transport switch
>>>>  will not lose any packets due to buffer overflows,
>>>> =20
>>>
>>>As I have stated prior I think we must require the client to have =
access=20
>>>to an interface it sends traffic to. If there is overlap I would =
assume=20
>>>that it can receive data on both during this period. I don't find it=20
>>>reasonable to get any guarantees from the networks. I would assume =
that=20
>>>any switch of network depends on the change of access technology, for =

>>>example by switching from cable to WLAN or to 3G mobile network. The=20
>>>probability that there will exist any mobility functionality between=20
>>>these networks seems very small.
>>>
>>>I think that we should design our solution to allow the mobility, =
only=20
>>>note that it may loss some packets
>>>
>>>Cheers
>>>
>>>
>>>Magnus Westerlund
>>>
>>>Multimedia Technologies, Ericsson Research EAB/TVA/A
>>>----------------------------------------------------------------------=

>>>Ericsson AB                | Phone +46 8 4048287
>>>Torshamsgatan 23           | Fax   +46 8 7575550
>>>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>>
>>>
>>>_______________________________________________
>>>mmusic mailing list
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>=20
>=20

--=20
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Hi Gamze,

No there are no IETF pointers about this proposal. This work is totally 
created within the 3GPP SA4. There should probably have been some review 
  by the IETF before the acceptance of the proposal, but that has not 
happened. However as the Release 6 has not been approved yet, any bugs 
in this proposal can still be fixed.

If you are interested in this proposal please review chapter 5.3.3.3. 
and  5.3.3.4 in the 3GPP TS 26.234 version 0.3.0 of Release 6 
specification, which can be downloaded here:

http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/S4-030680.zip

This is a 200 kb zipped word document.

Cheers

Magnus

Gamze Seckin wrote:

> Dear mmusic members,
> 
> my apologies if this is discussed here before:
> 
> I am looking for ietf pointers (if there are any) on the sdp descriptions provided in http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/ document s4-30680.
> on sdp "alt" and "alt-default-id" attributes and the session level grouping attribute, "alt-group".
> 
> best regards
> g
> 
> 
> 
> -----------------------------------------------------------
> Gamze Seckin, Ph.D.
> Director of Research
> Vidiator Technology (US) Inc.
> 411 108th Avenue NE. Suite 688
> Bellevue, WA 98004 USA.
> www.vidiator.com
> 
> Phone: 1.425.688.8811x2108
> Email: gamze@vidiator.com
> 
> This e-mail message (including any attachment) is intended only for the personal use of the recipient(s) named above. This message is confidential and may be legally privileged. If you are not an intended recipient, you may not review, copy or distribute this message. If you have received this communication in error, please notify us immediately by e-mail and delete the original message. Any views or opinions expressed in this message are those of the author only. Furthermore, this message including any attachment does not create any legally binding rights or obligations whatsoever, which may only be created by the exchange of hard copy documents signed by a duly authorized representative of Vidiator Technology (US) Inc.
> -----------------------------------------------------------
> 
> 
> -----Original Message-----
> From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
> Sent: Wednesday, November 05, 2003 10:55 AM
> To: Magnus Westerlund
> Cc: Enrico Balatti; mmusic@ietf.org
> Subject: Re: [MMUSIC] RTSP transport negotiation
> 
> 
> 
> hi Magnus,
> 
> Yes, it looks like we can drop the Range header from SETUP request.
> We probably want the Range header + RTP-Info in the SETUP response
> for VOD streams...with the caveat that packets can be lost during the
> switch. 
> 
> I don't believe all components of the RTP-Info header are relevant for
> live streams that are 'reflected' by the server - here the server acts
> as a translator. The RTP stream url is known, however the exact sequence
> number and timestamp for RTP packets is determined by the source of the
> live stream, not the server reflecting the stream. Hence, I recommend
> that the RTP-Info carry no sequence number or rtptime for live streams.
> 
> Also, I noticed that the RTP spec requires change of SSRC when the
> transport changes. This needs to be added to the transport negotiation 
> notes.
> 
> I will update the notes and send it to the alias.
> 
> geetha
> 
> 
> On Wed, 2003-11-05 at 03:23, Magnus Westerlund wrote:
> 
>>Hi Geetha,
>>
>>Thanks for your views. Comments inline
>>
>>Geetha Srikantan wrote:
>>
>>>hi Magnus,
>>>
>>>My assumption was that SETUP during play happens *after* the client has
>>>switched to the new transport..synchronization is an issue in this case.
>>>I agree that if we can assume that the client requests new transport
>>>before switching to the new transport, there is probably no
>>>synchronization issue. I have two follow-up questions on this:
>>>
>>
>>Yes, I would also think that in most cases the application will not know 
>>about the change of transport until it actually happens. This will 
>>probably result in the loss of some media packets. This loss can be 
>>dealt with in a few different way depending on the what the application 
>>can use: media level retransmission, RTSP go back through PAUSE and PLAY.
>>
>>However in regards the question about what your assumption are and why 
>>you need Range in the request I think your answer clearly points out 
>>that there are no need for Range in the Request. There seem to be 
>>reasons for it to be allowed in the Response, to indicate the correct 
>>new synchronization after the change of address.
>>
>>So are you still seeing any need for Range in the SETUP request? If that 
>>can be motivated I definitely think there will be possibility to change 
>>the specification to allow for this. However this would need further 
>>investigation.
>>
>>
>>
>>>a) How does a mobile client get notified about new transports - such as 
>>>   a new IP address?  How far ahead of time does the client get
>>>   notified?
>>
>>I would say that this is not a question we can answer. I think we simply 
>>need to ensure that the solution works with both sufficient notification 
>>and without. However without overlap in transport availability, there 
>>will be losses of media packets.
>>
>>
>>
>>>b) What is the range of time over which the transport switch happens?
>>>   Consider a scenario where the client requests an IP address change
>>>   for example, via SETUP with no Range, and the server starts sending
>>>   packets to the new IP. Is there a chance that the buffers at the new
>>>   IP get overrun, before the client switches to the new IP?
>>>   It would help if the network guarantees that the transport switch
>>>   will not lose any packets due to buffer overflows,
>>>   
>>
>>As I have stated prior I think we must require the client to have access 
>>to an interface it sends traffic to. If there is overlap I would assume 
>>that it can receive data on both during this period. I don't find it 
>>reasonable to get any guarantees from the networks. I would assume that 
>>any switch of network depends on the change of access technology, for 
>>example by switching from cable to WLAN or to 3G mobile network. The 
>>probability that there will exist any mobility functionality between 
>>these networks seems very small.
>>
>>I think that we should design our solution to allow the mobility, only 
>>note that it may loss some packets
>>
>>Cheers
>>
>>
>>Magnus Westerlund
>>
>>Multimedia Technologies, Ericsson Research EAB/TVA/A
>>----------------------------------------------------------------------
>>Ericsson AB                | Phone +46 8 4048287
>>Torshamsgatan 23           | Fax   +46 8 7575550
>>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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Are WiMAX and WiFi complementary (local vs. metropolitan area =
networking)?=20
When and how will WiMAX evolve towards mobility?=20
Can WiMAX be considered as a migration path to 4G?=20
How is addressed the interoperability challenge?=20

These questions, among others, will be addressed during the WiMAX =
Summit, to be held in Paris on May 25 to 28, 2004.=20

A call for papers has been launched on-line at:
http://www.upperside.fr/wimax04/wimax04intro.htm

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<DIV><FONT size=3D2>Are WiMAX and WiFi complementary (local vs. =
metropolitan=20
area&nbsp;networking)? <BR>When and how will WiMAX evolve towards =
mobility?=20
<BR>Can WiMAX be considered as a migration path to 4G? <BR>How is =
addressed the=20
interoperability challenge? <BR><BR>These questions, among others, will =
be=20
addressed during the <FONT class=3Dtextebold color=3D#2d795f>WiMAX =
Summit</FONT>, to=20
be held in <SPAN class=3Dunderline>Paris on May 25 to 28, </SPAN>2004. =
<BR><BR>A=20
call for papers has been launched on-line at:</FONT></DIV>
<DIV><FONT size=3D2><A=20
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Hi Jonathan,

Yes, we have a collision in the name space. I detected it yesterday when 
I read ICE. However as I understand your proposal for "alt" is brand 
new. The 3GPP "alt" has been there for a while as being accepted 
solution and included in the specification. However as it isn't approved 
yet, it can be changed. However if your parameter is as newly defined as 
I think, it might be easier for you to rename it.

To avoid future problem with this, the 3GPP SDP parameters in the PSS 
specification should be registered. The question here is what level is 
needed for registering the parameters. The previous SDP specification is 
not clear and the new specification requires a specification, which will 
exist when Release 6 is finished. So any advice from the chair on 
appropriate level to register them? Is a registering request pointing to 
26.234 enough or should a draft be written for this? I would also like 
to point out that there are a few more attributes also defined:

- "a=X-predecbufsize:<size of the hypothetical pre-decoder buffer>"
This gives the suggested size of the Annex G hypothetical pre-decoder 
buffer in bytes.

- "a=X-initpredecbufperiod:<initial pre-decoder buffering period>"
This gives the required initial pre-decoder buffering period specified 
according to Annex G. Values are interpreted as clock ticks of a 90-kHz 
clock. That is, the value is incremented by one for each 1/90 000 
seconds. For example, value 180 000 corresponds to a two second initial 
pre-decoder buffering.

- "a=X-initpostdecbufperiod:<initial post-decoder buffering period>"
This gives the required initial post-decoder buffering period specified 
according to Annex G. Values are interpreted as clock ticks of a 90-kHz 
clock.

- "a=X-decbyterate:<peak decoding byte rate>"
This gives the peak decoding byte rate that was used to verify the 
compatibility of the stream with Annex G. Values are given in bytes per 
second.

- "a=framesize:<payload type number> <width>-<height>"
This gives the largest video frame size of H.263 streams.

- "a=3GPP-Adaptation-Support"

Which also should be registered.


Cheers

Magnus

Jonathan Rosenberg wrote:

> Well, I've not seen the 3gpp docs, but there is an alt SDP parameter 
> defined in ICE:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-00.txt
> 
> Is this the same as what is being used by 3gpp? If not, we have a 
> serious name collision...
> 
> -Jonathan R.
> 
> Gamze Seckin wrote:
> 
>> Hi Magnus,
>>
>> Thanks. I do not see bugs and I think this proposal could be useful 
>> actually.
>> But I also think it is important that IETF takes into account these 
>> sections somehow, since they may have an impact on several proposals, 
>> say such as transport independent bw modifiers?
>>
>> br
>> g
>>
>>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Thursday, November 06, 2003 10:34 AM
>> To: Gamze Seckin
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] sdp alt, alt-default-id and alt-group
>>
>>
>> Hi Gamze,
>>
>> No there are no IETF pointers about this proposal. This work is 
>> totally created within the 3GPP SA4. There should probably have been 
>> some review   by the IETF before the acceptance of the proposal, but 
>> that has not happened. However as the Release 6 has not been approved 
>> yet, any bugs in this proposal can still be fixed.
>>
>> If you are interested in this proposal please review chapter 5.3.3.3. 
>> and  5.3.3.4 in the 3GPP TS 26.234 version 0.3.0 of Release 6 
>> specification, which can be downloaded here:
>>
>> http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/S4-030680.zip
>>
>> This is a 200 kb zipped word document.
>>
>> Cheers
>>
>> Magnus
>>
>> Gamze Seckin wrote:
>>
>>
>>> Dear mmusic members,
>>>
>>> my apologies if this is discussed here before:
>>>
>>> I am looking for ietf pointers (if there are any) on the sdp 
>>> descriptions provided in 
>>> http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_28/Docs/ document 
>>> s4-30680.
>>> on sdp "alt" and "alt-default-id" attributes and the session level 
>>> grouping attribute, "alt-group".
>>>
>>> best regards
>>> g
>>>
>>>
>>>
>>> -----------------------------------------------------------
>>> Gamze Seckin, Ph.D.
>>> Director of Research
>>> Vidiator Technology (US) Inc.
>>> 411 108th Avenue NE. Suite 688
>>> Bellevue, WA 98004 USA.
>>> www.vidiator.com
>>>
>>> Phone: 1.425.688.8811x2108
>>> Email: gamze@vidiator.com
>>>
>>> This e-mail message (including any attachment) is intended only for 
>>> the personal use of the recipient(s) named above. This message is 
>>> confidential and may be legally privileged. If you are not an 
>>> intended recipient, you may not review, copy or distribute this 
>>> message. If you have received this communication in error, please 
>>> notify us immediately by e-mail and delete the original message. Any 
>>> views or opinions expressed in this message are those of the author 
>>> only. Furthermore, this message including any attachment does not 
>>> create any legally binding rights or obligations whatsoever, which 
>>> may only be created by the exchange of hard copy documents signed by 
>>> a duly authorized representative of Vidiator Technology (US) Inc.
>>> -----------------------------------------------------------
>>>
>>>
>>> -----Original Message-----
>>> From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
>>> Sent: Wednesday, November 05, 2003 10:55 AM
>>> To: Magnus Westerlund
>>> Cc: Enrico Balatti; mmusic@ietf.org
>>> Subject: Re: [MMUSIC] RTSP transport negotiation
>>>
>>>
>>>
>>> hi Magnus,
>>>
>>> Yes, it looks like we can drop the Range header from SETUP request.
>>> We probably want the Range header + RTP-Info in the SETUP response
>>> for VOD streams...with the caveat that packets can be lost during the
>>> switch.
>>> I don't believe all components of the RTP-Info header are relevant for
>>> live streams that are 'reflected' by the server - here the server acts
>>> as a translator. The RTP stream url is known, however the exact sequence
>>> number and timestamp for RTP packets is determined by the source of the
>>> live stream, not the server reflecting the stream. Hence, I recommend
>>> that the RTP-Info carry no sequence number or rtptime for live streams.
>>>
>>> Also, I noticed that the RTP spec requires change of SSRC when the
>>> transport changes. This needs to be added to the transport 
>>> negotiation notes.
>>>
>>> I will update the notes and send it to the alias.
>>>
>>> geetha
>>>
>>>
>>> On Wed, 2003-11-05 at 03:23, Magnus Westerlund wrote:
>>>
>>>
>>>> Hi Geetha,
>>>>
>>>> Thanks for your views. Comments inline
>>>>
>>>> Geetha Srikantan wrote:
>>>>
>>>>
>>>>> hi Magnus,
>>>>>
>>>>> My assumption was that SETUP during play happens *after* the client 
>>>>> has
>>>>> switched to the new transport..synchronization is an issue in this 
>>>>> case.
>>>>> I agree that if we can assume that the client requests new transport
>>>>> before switching to the new transport, there is probably no
>>>>> synchronization issue. I have two follow-up questions on this:
>>>>>
>>>>
>>>> Yes, I would also think that in most cases the application will not 
>>>> know about the change of transport until it actually happens. This 
>>>> will probably result in the loss of some media packets. This loss 
>>>> can be dealt with in a few different way depending on the what the 
>>>> application can use: media level retransmission, RTSP go back 
>>>> through PAUSE and PLAY.
>>>>
>>>> However in regards the question about what your assumption are and 
>>>> why you need Range in the request I think your answer clearly points 
>>>> out that there are no need for Range in the Request. There seem to 
>>>> be reasons for it to be allowed in the Response, to indicate the 
>>>> correct new synchronization after the change of address.
>>>>
>>>> So are you still seeing any need for Range in the SETUP request? If 
>>>> that can be motivated I definitely think there will be possibility 
>>>> to change the specification to allow for this. However this would 
>>>> need further investigation.
>>>>
>>>>
>>>>
>>>>
>>>>> a) How does a mobile client get notified about new transports - 
>>>>> such as  a new IP address?  How far ahead of time does the client get
>>>>>  notified?
>>>>
>>>>
>>>> I would say that this is not a question we can answer. I think we 
>>>> simply need to ensure that the solution works with both sufficient 
>>>> notification and without. However without overlap in transport 
>>>> availability, there will be losses of media packets.
>>>>
>>>>
>>>>
>>>>
>>>>> b) What is the range of time over which the transport switch happens?
>>>>>  Consider a scenario where the client requests an IP address change
>>>>>  for example, via SETUP with no Range, and the server starts sending
>>>>>  packets to the new IP. Is there a chance that the buffers at the new
>>>>>  IP get overrun, before the client switches to the new IP?
>>>>>  It would help if the network guarantees that the transport switch
>>>>>  will not lose any packets due to buffer overflows,
>>>>>  
>>>>
>>>>
>>>> As I have stated prior I think we must require the client to have 
>>>> access to an interface it sends traffic to. If there is overlap I 
>>>> would assume that it can receive data on both during this period. I 
>>>> don't find it reasonable to get any guarantees from the networks. I 
>>>> would assume that any switch of network depends on the change of 
>>>> access technology, for example by switching from cable to WLAN or to 
>>>> 3G mobile network. The probability that there will exist any 
>>>> mobility functionality between these networks seems very small.
>>>>
>>>> I think that we should design our solution to allow the mobility, 
>>>> only note that it may loss some packets
>>>>
>>>> Cheers
>>>>
>>>>
>>>> Magnus Westerlund
>>>>
>>>> Multimedia Technologies, Ericsson Research EAB/TVA/A
>>>> ----------------------------------------------------------------------
>>>> Ericsson AB                | Phone +46 8 4048287
>>>> Torshamsgatan 23           | Fax   +46 8 7575550
>>>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>>
>>
> 

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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Hi Jonathan,

I had a late night reading session yesterday that produce some comments.

1. I think there are still maybe a few places where ICE is to situation 
specific towards SIP. I will not go so long to require that mapping of 
transport addresses are not IP + transport protocol ports as this is 
what our NAT problem causes. However the constant reference to RTP 
should probably be removed, there are at least other proprietary media 
protocols. So I think one should talk about media protocol instead of 
RTP + RTCP.

2. I think some better motivation why the openly available password in 
the STUN request is acceptable. And then why the integrity protection 
gives advantages or not.

3. When the responder performs his connectivity checks and succeed, why 
does the initiator need to perform a counter check also? Some more 
should probably be spent on this. The initiator knows the source address 
as it sees it to send any packets to. The only piece of information 
missing is what responder connection interface it goes to. This could be 
identified using STUN supplements and the response message connection IDs.

Then the only reasons for doing a counter check is to possibly enable 
connectivity over a more prioritized connection. OR have I missed 
something about this.

4. Of course not doing it both way will not provide full protection 
against the RTP DOS attack. However you have not at all discussed the 
applicability of ICE for RTP DOS prevention.

5. I also are not finding anything about the retrying in the 
transmission of STUN request. Should these be done in any special way 
for ICE? Also there is very little discussion about timer consideration 
for performing the connectivity checks.

6. In example 6.2 i think it can be easily expanded to show one 
important functionality of ICE, the possibility to open up restricted 
NAT through the usage of multiple ICE cycles. If A is behind a 
restricted NAT it will be able to determine its outermost address and 
port. However in cycle 1, B will not be able to reach A. When then A 
tries B over the cycle this may succeed. Then B can verify this by 
STUNNING a second time. These possibilities are rather tuned down.

7. I think the XML description should be more generic and include 1..N 
transport addresses for each media session.

8. If anyone missed the other message, the usage of "alt" has a conflict 
with 3GPP TS 26.234, Rel 6 preliminary specification.

9. I think there should be some example of the complete mesh ICE 
actually tries. I know it will be a hard thing to draw using ASCII art.

10. If one uses ICE for other purposes then UNSAF mechanism it suddenly 
will not have any exit strategy for that purpose. However it should 
still be true that if no relays, etc. are need when people has migrated 
to IPv6, etc., then it will simply reduce the complexity of doing ICE.

11. When one performs ICE, are there any congestion control 
considerations to make? If one digs up a number of address locally the 
local interface might need to send a lot of packets in a short time as 
it is desired to perform this in parallel.


Best Regards


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



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Magnus Westerlund wrote:

> Hi Jonathan,
> 
> Yes, we have a collision in the name space. I detected it yesterday when 
> I read ICE. However as I understand your proposal for "alt" is brand 
> new. The 3GPP "alt" has been there for a while as being accepted 
> solution and included in the specification.

I had no idea about it.

> However as it isn't approved 
> yet, it can be changed. However if your parameter is as newly defined as 
> I think, it might be easier for you to rename it.

Yes, its no big deal to change. I am worried more about the potential 
for other namespace collisions.

> 
> To avoid future problem with this, the 3GPP SDP parameters in the PSS 
> specification should be registered. 

Ideally, they should be shared before registration too...

> The question here is what level is 
> needed for registering the parameters. The previous SDP specification is 
> not clear and the new specification requires a specification, which will 
> exist when Release 6 is finished. So any advice from the chair on 
> appropriate level to register them? 

Well, the tact we took with the SIP documents was this. 3gpp-specific 
headers were created through IETF RFCs that created "P-headers" that 
were not for general use. The ensured that the ietf retained ownership 
and change control for sip and its extensions, while giving 3gpp the 
ability to have things they needed. Given that the new sdp spec 
requires an rfc to register parameters, it would seem to imply a 
similar kind of thing - there should be a doc that goes through the 
ietf process that defines these attributes and registers them. The 
3gpp specs could then reference this doc.

I think we ultimately need AD input here on the direction this should 
go. The "alt" collision is a clear indicator that there is a REAL 
problem here.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Thread-Topic: bitrate in gpmd vs. a:bw (from draft-nismail-mmusic-sdp-video-00)
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Section 4.1 Open Issue on counting RTP headers in bitrate:
Bitrate is not a network bandwidth parameter, it is a decoder buffer =
depth parameter.  Thus bitrate needs to reflect the bitrate of the video =
stream, not the RTP stream.

I can't remember: did we decide that a:bw did count RTP headers?  If so, =
and you need to limit the RTP stream, there is an existing mechanism to =
deal with it.  If not, then bitrate is not needed -- just use a:bw.


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Section 4:
In the example "CIF=3D4 QCIF=3D3 SQCIF=3D2 XMAX=3D360 YMAX=3D240 =
MPI=3D2", the text says that the MPI=3D2 parameter goes with SQCIF=3D2.  =
How does it know?


Nits - Section 2
ABNF
Size =3D "QCIF" "=3D" "mpi | "CIF" "=3D" mpi
Size =3D "QCIF" "=3D" mpi | "CIF" "=3D" mpi
                  ^
Section 3 ABNF
"CIF4" "=3D" mpi I "CIF16" "=3D" mpi
"CIF4" "=3D" mpi | "CIF16" "=3D" mpi
               ^

Here #term means comma separated list of term.
Here #term means comma separated list of terms.
                                             ^


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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Eric Burger'" <eburger@snowshore.com>, avt@ietf.org, mmusic@ietf.org
Subject: RE: [MMUSIC] Comments on draft-even-avt-h263-h261-options-00
Date: Sat, 8 Nov 2003 13:16:16 +0200 
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Eric,
The 2 in SQCIF=2 is the MPI value. The MPI but it self relates to the custom
format x=360, y=240, MPI=2.
Roni

*************************************
Roni Even
VP Product Planning
Polycom Israel

Tel: +972-3-9251200
Cell: +972-55-481099
email:roni.even@polycom.co.il
*******************************************


-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com]
Sent: Friday, November 07, 2003 11:20 PM
To: avt@ietf.org; mmusic@ietf.org
Subject: [MMUSIC] Comments on draft-even-avt-h263-h261-options-00


Section 4:
In the example "CIF=4 QCIF=3 SQCIF=2 XMAX=360 YMAX=240 MPI=2", the text says
that the MPI=2 parameter goes with SQCIF=2.  How does it know?


Nits - Section 2
ABNF
Size = "QCIF" "=" "mpi | "CIF" "=" mpi
Size = "QCIF" "=" mpi | "CIF" "=" mpi
                  ^
Section 3 ABNF
"CIF4" "=" mpi I "CIF16" "=" mpi
"CIF4" "=" mpi | "CIF16" "=" mpi
               ^

Here #term means comma separated list of term.
Here #term means comma separated list of terms.
                                             ^


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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Eric Burger'" <eburger@snowshore.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] bitrate in gpmd vs. a:bw (from draft-nismail-mmusic-
	sdp-video-00)
Date: Sat, 8 Nov 2003 13:18:30 +0200 
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Eric,
I agree that here it should be the decoder bit rate and not the RTP. We
intend to merge our drafts and that was one of the issues I told Nermeen we
have to fix. It is only for H.263
Roni

*************************************
Roni Even
VP Product Planning
Polycom Israel

Tel: +972-3-9251200
Cell: +972-55-481099
email:roni.even@polycom.co.il
*******************************************


-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com]
Sent: Friday, November 07, 2003 11:20 PM
To: mmusic@ietf.org
Subject: [MMUSIC] bitrate in gpmd vs. a:bw (from
draft-nismail-mmusic-sdp-video-00)


Section 4.1 Open Issue on counting RTP headers in bitrate:
Bitrate is not a network bandwidth parameter, it is a decoder buffer depth
parameter.  Thus bitrate needs to reflect the bitrate of the video stream,
not the RTP stream.

I can't remember: did we decide that a:bw did count RTP headers?  If so, and
you need to limit the RTP stream, there is an existing mechanism to deal
with it.  If not, then bitrate is not needed -- just use a:bw.


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Subject: RE: [MMUSIC] bitrate in gpmd vs. a:bw (from draft-nismail-mmusic-sdp-video-00)
Date: Sat, 8 Nov 2003 19:48:17 -0800
Message-ID: <6677B3346233B94EBB11C060935101207749C0@vtg-um-e2k1.sj21ad.cisco.com>
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Thread-Index: AcOl6hx+AL1N94EjRHasEbmHfsi8NAAiRqtw
From: "John Restrick" <restrick@cisco.com>
To: "Even, Roni" <roni.even@polycom.co.il>,
        "Eric Burger" <eburger@snowshore.com>, <mmusic@ietf.org>
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Agree, it should not include RTP overhead.

As for using, bw as long as there is a way to associate that bandwidth
with other video information.  For example, I may be able to do H.263 up
to 768 kbps but H.263 with annex foo might only make sense up to 128kbps
- beyond that it might be counterproductive (increase computation time
for no quality benefit perhaps).=20


> -----Original Message-----
> From: Even, Roni [mailto:roni.even@polycom.co.il]=20
> Sent: Saturday, November 08, 2003 3:19 AM
> To: 'Eric Burger'; mmusic@ietf.org
> Subject: RE: [MMUSIC] bitrate in gpmd vs. a:bw (from=20
> draft-nismail-mmusic-sdp-video-00)
>=20
>=20
> Eric,
> I agree that here it should be the decoder bit rate and not=20
> the RTP. We intend to merge our drafts and that was one of=20
> the issues I told Nermeen we have to fix. It is only for H.263 Roni
>=20
> *************************************
> Roni Even
> VP Product Planning
> Polycom Israel
>=20
> Tel: +972-3-9251200
> Cell: +972-55-481099
> email:roni.even@polycom.co.il
> *******************************************
>=20
>=20
> -----Original Message-----
> From: Eric Burger [mailto:eburger@snowshore.com]
> Sent: Friday, November 07, 2003 11:20 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] bitrate in gpmd vs. a:bw (from
> draft-nismail-mmusic-sdp-video-00)
>=20
>=20
> Section 4.1 Open Issue on counting RTP headers in bitrate:=20
> Bitrate is not a network bandwidth parameter, it is a decoder=20
> buffer depth parameter.  Thus bitrate needs to reflect the=20
> bitrate of the video stream, not the RTP stream.
>=20
> I can't remember: did we decide that a:bw did count RTP=20
> headers?  If so, and you need to limit the RTP stream, there=20
> is an existing mechanism to deal with it.  If not, then=20
> bitrate is not needed -- just use a:bw.
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

_______________________________________________
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From exim@www1.ietf.org  Sat Nov  8 23:54:22 2003
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: John Restrick <restrick@cisco.com>
CC: "Even, Roni" <roni.even@polycom.co.il>,
        Eric Burger <eburger@snowshore.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] bitrate in gpmd vs. a:bw (from draft-nismail-mmusic-sdp-video-00)
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Hi,

Are you talking about the b=AS: or is a:bw something else? The meaning 
fo b=AS is that it includes all lower layer overhead including IP. But 
the video bitrate parameter probably makes sense to talk about at a 
bitstream level.

Cheers

Magnus

John Restrick wrote:

>Agree, it should not include RTP overhead.
>
>As for using, bw as long as there is a way to associate that bandwidth
>with other video information.  For example, I may be able to do H.263 up
>to 768 kbps but H.263 with annex foo might only make sense up to 128kbps
>- beyond that it might be counterproductive (increase computation time
>for no quality benefit perhaps). 
>
>
>  
>
>>-----Original Message-----
>>From: Even, Roni [mailto:roni.even@polycom.co.il] 
>>Sent: Saturday, November 08, 2003 3:19 AM
>>To: 'Eric Burger'; mmusic@ietf.org
>>Subject: RE: [MMUSIC] bitrate in gpmd vs. a:bw (from 
>>draft-nismail-mmusic-sdp-video-00)
>>
>>
>>Eric,
>>I agree that here it should be the decoder bit rate and not 
>>the RTP. We intend to merge our drafts and that was one of 
>>the issues I told Nermeen we have to fix. It is only for H.263 Roni
>>
>>*************************************
>>Roni Even
>>VP Product Planning
>>Polycom Israel
>>
>>Tel: +972-3-9251200
>>Cell: +972-55-481099
>>email:roni.even@polycom.co.il
>>*******************************************
>>
>>
>>-----Original Message-----
>>From: Eric Burger [mailto:eburger@snowshore.com]
>>Sent: Friday, November 07, 2003 11:20 PM
>>To: mmusic@ietf.org
>>Subject: [MMUSIC] bitrate in gpmd vs. a:bw (from
>>draft-nismail-mmusic-sdp-video-00)
>>
>>
>>Section 4.1 Open Issue on counting RTP headers in bitrate: 
>>Bitrate is not a network bandwidth parameter, it is a decoder 
>>buffer depth parameter.  Thus bitrate needs to reflect the 
>>bitrate of the video stream, not the RTP stream.
>>
>>I can't remember: did we decide that a:bw did count RTP 
>>headers?  If so, and you need to limit the RTP stream, there 
>>is an existing mechanism to deal with it.  If not, then 
>>bitrate is not needed -- just use a:bw.
>>
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic
>>
>>    
>>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic
>  
>


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Subject: [MMUSIC] Free beer in Mpls
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To announce the availability of free beer over the Internet, I'd need a =
pretty good announcement protocol. Something that can handle the good =
news regardless of the language I chose to describe it in, that scales =
up to several thousand potential users, that gives decent reliability =
and doesn't congest the bars or the back account too heavily.

If you've got any interest in media & content discovery, there's a =
pretty good chance you'll need to announce it in this way for multicast =
networks. And even more so for scaling up to many thousands (or =
millions) for mass media. We've got some well developed ideas on how to =
do this, and now we want your opinions. Take a look at =
draft-ietf-mmusic-img-framework-00.txt and tell us what you think!

Cheers, Rod.

PS Sorry about the gratuitous use of temptation. But it's the jet lagand =
lack of sleep - if you read this far you most definately deserve free =
beer and all good reviews end up in beer, even if it's not free :)
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-img-framework-00.tx=
t

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Hi RTSP fans!

We will have a offline meeting on Tuseday from lunch until the 
coffebreak. We are gathering at 11.40 in the main lobby on gound floor 
and will continue until 15.15. People interested in participating should 
RSVP. Discussion topics will be sent to the list later.

Cheers

Magnus


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Thread-Topic: RTSP SET_PARAMETER with no parameters.
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Dear MMUSIC members and RTSP experts,

I have a question related to the RTSP SET_PARAMETER method.
Is it possible to issue an RTSP SET_PARAMETER method without a parameter =
?
Here is an example:

	SET_PARAMETER rtsp://server/content.3gp RTSP/1.0
	CSeq: 123
	Session: 12345678
	Bandwidth: 64000
	Standard_Defined_Header: some_parameter=3Dnew_value_of_some_parameter
	Content-length: 0=20
	Content-type: text/parameters


Basically the aim is to update the "Standard_Defined_Header" header =
field. I was wondering if this update could be done with such a =
SET_PARAMETER request. And if so, how should be the response behaviour =
of the server to that?

Another alternative may be the usage of OPTIONS method with a Session =
ID. Could you experts comment on which way is better?

Cheers,
Emre


_________________________________
Emre Baris Aksu
Nokia Mobile Software
P.O. Box 88, FIN-33100 Tampere
Tel: +358 7180 08000
e-mail: emre.aksu@nokia.com
=20


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From: Tom Marshall <tmarshall@real.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP offline meeting
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Could you push that back a bit?  I'm not arriving until about 7:00 Tuesday
evening.

On Mon, Nov 10, 2003 at 04:40:49AM +0100, Magnus Westerlund wrote:
> Hi RTSP fans!
>=20
> We will have a offline meeting on Tuseday from lunch until the=20
> coffebreak. We are gathering at 11.40 in the main lobby on gound floor=20
> and will continue until 15.15. People interested in participating should=
=20
> RSVP. Discussion topics will be sent to the list later.

--=20
It is by the fortune of God that, in this country, we have three benefits:
freedom of speech, freedom of thought, and the wisdom never to use either.
        -- Mark Twain

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Date: Mon, 10 Nov 2003 08:43:17 -0800
To: "Even, Roni" <roni.even@polycom.co.il>
From: Nermeen Ismail <nismail@cisco.com>
Subject: RE: [MMUSIC] bitrate in gpmd vs. a:bw (from
  draft-nismail-mmusic- sdp-video-00)
Cc: "'Eric Burger'" <eburger@snowshore.com>, mmusic@ietf.org
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I also agree that bitrate should represent that of the video stream rather 
than the RTP stream. However the main
difference between it and the BW parameter is that it is defined for each 
pair of video encoding and resolution. The a:bw
is defined per media line.

nermeen



>Eric,
>I agree that here it should be the decoder bit rate and not the RTP. We
>intend to merge our drafts and that was one of the issues I told Nermeen we
>have to fix. It is only for H.263
>Roni
>
>*************************************
>Roni Even
>VP Product Planning
>Polycom Israel
>
>Tel: +972-3-9251200
>Cell: +972-55-481099
>email:roni.even@polycom.co.il
>*******************************************
>
>
>-----Original Message-----
>From: Eric Burger [mailto:eburger@snowshore.com]
>Sent: Friday, November 07, 2003 11:20 PM
>To: mmusic@ietf.org
>Subject: [MMUSIC] bitrate in gpmd vs. a:bw (from
>draft-nismail-mmusic-sdp-video-00)
>
>
>Section 4.1 Open Issue on counting RTP headers in bitrate:
>Bitrate is not a network bandwidth parameter, it is a decoder buffer depth
>parameter.  Thus bitrate needs to reflect the bitrate of the video stream,
>not the RTP stream.
>
>I can't remember: did we decide that a:bw did count RTP headers?  If so, and
>you need to limit the RTP stream, there is an existing mechanism to deal
>with it.  If not, then bitrate is not needed -- just use a:bw.
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic
>
>_______________________________________________
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Folks,

please find attached a slightly revised MMUSIC agenda.

Multiparty Multimedia Session Control (MMUSIC)
	
CHAIRS: 
Joerg Ott <jo@informatik.uni-bremen.de>
Colin Perkins <csp@csperkins.org>


Friday, 14 November 2003, 09:00 - 11:30, Salon D
------------------------------------------------

0900    Agenda Bashing and Status Update               (Chairs, 15)

         Recent RFCs
         draft-ietf-mmusic-sdp4nat-05.txt		-> RFC 3605

         With RFC editor
         draft-ietf-mmusic-sdpng-trans-04.txt

	With IESG
         draft-ietf-mmusic-sdp-new-15.txt
	draft-rajeshkumar-mmusic-gpmd-03.txt

	With AD
         draft-ietf-mmusic-sdp-comedia-05.txt
	draft-ietf-mmusic-sdp-bwparam-05.txt
	draft-ietf-mmusic-srcfilter-05.txt

         Misc Updates
	draft-camarillo-mmusic-alt-01.txt
	draft-ietf-mmusic-kmgmt-ext-09.txt
	draft-ietf-mmusic-sdp-implem-00.txt
         draft-ietf-mmusic-sdp-new-15.txt

0915 
Internet Media Guides				(Manyfolks, 30)
	draft-ietf-mmusic-img-req-00.txt
	draft-nomura-mmusic-img-framework-02.txt
	(draft-luoma-mmusic-img-metadata-02.txt fyi only)
	(draft-luoma-mmusic-img-muppet-03.txt fyi only)

0945 
SDP for alternate IP versions			(Camarillo, 10)
	draft-camarillo-mmusic-alt-02.txt

0955 
SDP Security Descriptions for Media Streams	(Andreasen, 15)
	draft-ietf-mmusic-sdescriptions-02.txt
	draft-andreasen-mmusic-securityprecondition-00.txt

1010 
SDP parameters to support H.263/H.261 options	(Even/Ismail, 30)
	draft-even-avt-h263-h261-options-00.txt
	draft-nismail-mmusic-sdp-video-00.txt

1040 
Interactive Connectivity Establishment (ICE)	(Rosenberg, 20)
	draft-ietf-mmusic-ice-00.txt
	draft-rosenberg-sipping-ice-01.txt

1100 
RTSP Update				    (Westerlund/Zeng, 30)
	draft-ietf-mmusic-rfc2326bis-05.txt
	draft-ietf-mmusic-rtsp-nat-01.txt
	draft-zeng-mmusic-rtsp-end-of-stream-01.txt

1130 
Close




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

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Hi Tom,

No, I don't think we can do that. At least I have no other hole in my 
agenda sufficiently long to have time for real discussion. It will be 
unfortunate if you can't participate.

Magnus

Tom Marshall wrote:

>Could you push that back a bit?  I'm not arriving until about 7:00
>Tuesday
>evening.
>
>On Mon, Nov 10, 2003 at 04:40:49AM +0100, Magnus Westerlund wrote:
>  
>
>>Hi RTSP fans!
>>
>>We will have a offline meeting on Tuseday from lunch until the 
>>coffebreak. We are gathering at 11.40 in the main lobby on gound floor
>>    
>>
>
>  
>
>>and will continue until 15.15. People interested in participating
>>    
>>
>should 
>  
>
>>RSVP. Discussion topics will be sent to the list later.
>>    
>>
>
>  
>


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To: emre.aksu@nokia.com
CC: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP SET_PARAMETER with no parameters.
References: <450DCD80CA58884D95FF1840ED26A7A02E46A3@trebe007.europe.nokia.com>
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Hi Emre,

Please see my comments inline.

emre.aksu@nokia.com wrote:

>Dear MMUSIC members and RTSP experts,
>
>I have a question related to the RTSP SET_PARAMETER method.
>Is it possible to issue an RTSP SET_PARAMETER method without a parameter ?
>

Yes, but is rather a SET_PARAMETER without a body.

>Here is an example:
>
>	SET_PARAMETER rtsp://server/content.3gp RTSP/1.0
>	CSeq: 123
>	Session: 12345678
>	Bandwidth: 64000
>	Standard_Defined_Header: some_parameter=new_value_of_some_parameter
>	Content-length: 0 
>	Content-type: text/parameters
>
>  
>
Therfore it would not contain any of the Content-x headers.

>Basically the aim is to update the "Standard_Defined_Header" header field. I was wondering if this update could be done with such a SET_PARAMETER request. And if so, how should be the response behaviour of the server to that?
>

The response to such a request is a 200 OK. However that will not 
necessary mean that the recipient has understood all headers, only the 
mandatory. Therefore if one has important functionality and do not know 
appriori that it will behave in this way one should use a Require header 
with a feature tag for the functionality.

>
>Another alternative may be the usage of OPTIONS method with a Session ID. Could you experts comment on which way is better?
>

Options is not better. Options has another semantics and induce other 
overhead. So the usage of SET_PARAMETER is the right one.

Cheers

Magnus


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] RTSP off line meeting
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Hi RTSP fans,

First I would like to announce a new meeting place. We now meet at 11.40 
at the Message board at floor three, in front of the registration desk. 
We will go directly to find somewhere to sit down and talk. Therefore I 
will ask that people either bring something to eat or abstain from lunch.

I have also managed to make an agenda. If people has additions, you can 
mail them to the list or simply remember them until the meeting.

Cheers

Magnus

RTSP issues agenda for discussions in MPLS

1. Administrative

2. Since last meeting (1st of October).
a. http://rtsp.org/bug507882 - RTP timestamp and pause
    - Aravind to write up text

b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
end of a media clip
    - Await review from the WG.

c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
    - Needs text

d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
   - Await review from the WG.

e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
   - Await review from the WG.

f. http://rtsp.org/bug531189 - How does REDIRECT method work
   Everything except the following seem to be OK:
     - REDIRECT and the usage of Close:Connection.

g. http://rtsp.org/bug810678 - How to indicate end of media stream
   - We need to document in the RTSP specification regarding RTP what
     happens when a source times out a SSRC or receives a RTCP BYE for a
     source.

   - Using BYE is not suitable way to indicate the end of the stream.
     The motivation for this can be seen in Thomas proposal.

   - Look into SIP to see how they have resolved the discrepancies
     between media and signalling channel.
    
   - See draft-ietf-zeng-rtsp-end-of-stream-00.txt
     * RTCP
     * RTSP
     * Both  

h. http://rtsp.org/bug770522 - Request to response time is not limited.
   - Sean to write up text.

i. http://rtsp.org/bug810681 -     The updated specification version number
   - Awaiting investigation results regarding changes by Aravind.
   - Question of intentions.

j. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
   - Has been removed from spec.

k. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
   - Awaiting review from the WG.

l. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
   - Text sent to mailing list, comments?

m. http://rtsp.org/bug810666 - Removal of SETUP in play state
   - Removed from draft

n. http://rtsp.org/bug811815 - Use of c= in SDP to know possible address 
classes

o. http://rtsp.org/bug795314 - Caching of OPTIONS request


3. New issues

a. http://rtsp.org/bug - Should all ABNF syntax be gathered in one chapter?

b. http://rtsp.org/bug829528 - Usage of Grouping of media lines with RTSP.

4. Other RTSP related things

- RTSP NAT, what are our intentions with this work.



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Subject: RE: [MMUSIC] RTSP SET_PARAMETER with no parameters.
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HI Magnus,
Thanks for the clarification.


> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf Of
> ext Magnus Westerlund
> Sent: 10 November, 2003 20:30
> To: Aksu Emre (NMP-MSW/Tampere)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] RTSP SET_PARAMETER with no parameters.
>=20
>=20
> Hi Emre,
>=20
> Please see my comments inline.
>=20
> emre.aksu@nokia.com wrote:
>=20
> >Dear MMUSIC members and RTSP experts,
> >
> >I have a question related to the RTSP SET_PARAMETER method.
> >Is it possible to issue an RTSP SET_PARAMETER method without=20
> a parameter ?
> >
>=20
> Yes, but is rather a SET_PARAMETER without a body.
>=20
> >Here is an example:
> >
> >	SET_PARAMETER rtsp://server/content.3gp RTSP/1.0
> >	CSeq: 123
> >	Session: 12345678
> >	Bandwidth: 64000
> >	Standard_Defined_Header:=20
> some_parameter=3Dnew_value_of_some_parameter
> >	Content-length: 0=20
> >	Content-type: text/parameters
> >
> > =20
> >
> Therfore it would not contain any of the Content-x headers.
>=20
> >Basically the aim is to update the "Standard_Defined_Header"=20
> header field. I was wondering if this update could be done=20
> with such a SET_PARAMETER request. And if so, how should be=20
> the response behaviour of the server to that?
> >
>=20
> The response to such a request is a 200 OK. However that will not=20
> necessary mean that the recipient has understood all headers,=20
> only the=20
> mandatory. Therefore if one has important functionality and=20
> do not know=20
> appriori that it will behave in this way one should use a=20
> Require header=20
> with a feature tag for the functionality.
>=20
> >
> >Another alternative may be the usage of OPTIONS method with=20
> a Session ID. Could you experts comment on which way is better?
> >
>=20
> Options is not better. Options has another semantics and induce other=20
> overhead. So the usage of SET_PARAMETER is the right one.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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From: Sean Sheedy <seans@ncube.com>
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Subject: [MMUSIC] RTSP: Proposed text for client timeout
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Here's a first cut at some proposed text describing client request
timeouts in RTSP.  Please seen http://rtsp.org/bug770522 for more
details.

Sean

*****

Servers SHOULD respond to requests in a timely manner even when a
reliable transport such as TCP is used.  Similarly, clients SHOULD wait
for a sufficient time for a response before concluding that the server
will not be acting upon its request.

Appropriate server and client time values are based on T1, which is a
bounded estimate of the round-trip time (RTT).  The RTT is estimated as
in TCP (RFC 1323), with an initial value of 500 ms.  The client MAY
derive a more refined estimate of the RTT using the Timestamp header as
described in Section 13.39.  A client implementation MAY cache the last
RTT measurement as the initial value for future connections.

T1 is equal to the RTT, with the restriction that T1 is never less than
500 ms.  This restriction is necessary because, while the client may
adjust its RTT estimate, the server does not know the client's value.

A client SHOULD wait at least 20*T1 before concluding that the server
will not be responding to its request.

A server SHOULD respond to all requests within 10*(minimum value of T1);
that is, within 5 seconds.  If the server is not ready to send its final
response within this time, it SHOULD send a 100 response.  After
receiving a 100 response, the client SHOULD wait for further responses. 
The server MAY send additional 100 responses.  Eventually, the server
MUST send a final response indicating the success or failure of the
request.


-- 
Sean Sheedy                                         seans@ncube.com
(503) 531-6427




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Hi,

I checked the rtsp nat traversal draft =
(draft-ietf-mmusic-rtsp-nat-01.txt) but could not find a comprehensive =
comparison between STUN vs Symmetric RTP.

Is anybody actually using STUN?
How about symmetric RTP?

Which one would you recommend if you had to actually deploy it ?
Any pointers appreciated.


Br

g

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From: Tom Marshall <tmarshall@real.com>
To: Sean Sheedy <seans@ncube.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP: Proposed text for client timeout
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> Servers SHOULD respond to requests in a timely manner even when a
> reliable transport such as TCP is used.  Similarly, clients SHOULD wait
> for a sufficient time for a response before concluding that the server
> will not be acting upon its request.

I agree with these statements.  Unfortunately, it is very difficult to
define "sufficient" above, as this is affected by both server load and
network conditions.

> Appropriate server and client time values are based on T1, which is a
> bounded estimate of the round-trip time (RTT).  The RTT is estimated as
> in TCP (RFC 1323), with an initial value of 500 ms.  The client MAY
> derive a more refined estimate of the RTT using the Timestamp header as
> described in Section 13.39.  A client implementation MAY cache the last
> RTT measurement as the initial value for future connections.

The TCP has access to control packets (ACKs) that the application typically
does not.  How can an application distinguish variations in RTT due to
packet loss vs. congestion vs. physical propagation delay?  Is it even
possible to have a meaningful RTT measurement in networks that have packet
loss on the physical layer (eg. wireless)?

As I am sitting here typing this using the wireless signal from the IETF in
a hotel around the block from the conference, I'm getting sporadic packet
loss.  If my SSH connection behaved according to this RTSP proposal, I would
not be able to finish writing this email.

> T1 is equal to the RTT, with the restriction that T1 is never less than
> 500 ms.  This restriction is necessary because, while the client may
> adjust its RTT estimate, the server does not know the client's value.
>=20
> A client SHOULD wait at least 20*T1 before concluding that the server
> will not be responding to its request.
>=20
> A server SHOULD respond to all requests within 10*(minimum value of T1);
> that is, within 5 seconds.  If the server is not ready to send its final
> response within this time, it SHOULD send a 100 response.  After
> receiving a 100 response, the client SHOULD wait for further responses.=
=20
> The server MAY send additional 100 responses.  Eventually, the server
> MUST send a final response indicating the success or failure of the
> request.

How were the 500ms and the T1 multiples arrived at?

Discussion in RFC 1323 characterizes the RTT measurement as a signal
processing problem.  I am admittedly not a signal processing expert, but I
think that if we are to speak of RTT, it must be done with this
characteristic in mind.  In particular, we cannot assume that the reader
understands the implications of the wide variety of network technologies,
TCP introduced delays, etc.  "RTT" needs to be better defined in terms of
how it is measured, how often it is measured, and what unexpected factors
can influence it.  I'm very concerned that someone will write an RTSP client
that does an RTT measurement with the first RTSP message and gets some small
result like 3ms, then drops the connection when a response is delayed by
60ms because the server happens to be a bit busy and/or a couple packets get
dropped due to congestion.  This is why I am opposed to acting on negative
feedback.  I think that the only sensible option for a protocol that runs on
top of TCP is to use positive feedback exclusively, and let the user decide
when too much time has passed without response (possibly via a configurable
setting with sensible defaults).

--=20
Beware of Programmers who carry screwdrivers.
        -- Leonard Brandwein

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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-840304 ] CSEQ is direction dependent in numberspaces. (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Tue, 11 Nov 2003 13:54:14 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-840304 ] CSEQ is direction dependent in numberspaces.

Bugs item #840304, was opened at 2003-11-11 22:54
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=840304&group_id=23194

Category: Editorial
Group: Resolved - Need write up
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: CSEQ is direction dependent in numberspaces.

Initial Comment:
IETF 58 off-line discussion resulted in one point to 
ensure is edited in:It needs to be clarified that CSeq 
sequence number spaces are direction dependent. The 
C->S Requests uses a different space then S->C 
Requests.



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

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From: Tom Marshall <tmarshall@real.com>
To: Sean Sheedy <seans@ncube.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP: Proposed text for client timeout
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> Since what constitutes acceptable delays in RTSP is determined by human
> perception, the source of the delay (packet loss, network congestion,
> server load, etc.) is irrelevant.  The only question that needs to be
> answered from the client's perspective is:  Is the server working on the
> request, or not?  I think it's important that a user interface provides
> this information to the human waiting for the media stream.

Absolutely.  So why are we proposing to measure RTT at all?  Why don't we
just say this:

  The server SHOULD respond to all requests as soon as practicable.  If a
  request takes a significant amount of time to process, or there is an
  expectation that it will, the server SHOULD send 100 responses at regular
  intervals so that the client knows its request is still being processed.

  The client SHOULD give the server a reasonable amount of time to process =
its
  requests.  If an unreasonable amount of time elapses without receiving any
  response, the client MAY assume the server is unresponsive and abort the
  connection.  If a 100 response is received, the client MUST reset its
  response timer.

The only remaining question would be what constitutes a reasonable amount of
time and a regular interval.

--=20
I put instant coffee in a microwave and almost went back in time.
        -- Steven Wright

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Date: Wed, 12 Nov 2003 14:49:46 -0600
From: Aravind Narasimhan <Aravind.Narasimhan@Sun.COM>
To: mmusic@ietf.org
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Subject: [MMUSIC] RTSP offline meeting Notes - 11/11
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  The meeting followed the agenda that Magnus sent out on 11/10

Attendees:

Thomas Zeng,
Henning Schulzerine
Geetha Srikantan
Victor Varsa
Sean Sheedy
Adrea Basso
Magnus Westerlund
Colin Perkins
Aravind Narasimhan


Issues:

2A. RTP Timestamps under pause clarification. 
- Aravind to write up text  - Not Done.  Hopefully by the end of week.

2B. PLAY request at the end of a media clip
- Awaiting review

2C. Implicit Redirect
- Philippe/Aravind to write up text?

2D. Aggregate URL deriving
- Awaiting review. 
- HS:  Is there motivation for deriving the URL?  It should be opaque?  
I can't use for something else?
- TZ:  It might conflict with an actual URL?
- HS:  Seems like a recipie for strange cases.
- TZ:  SDP from two different sources that you want to mix.
- HS:  The server should give you the URL.  If the SDP doesn't have an 
aggregate URL than the media can't be played in aggregate fashion.
- HS:  Concerned about specifying generating URL.  How do you know what 
you are getting?  What if it is not there?
- TZ:  Have SETUP two separate streams.  Need to control together. 
Session ID should be enough, but URL is needed for PLAY request.
- AN:  URL is used for proxies and accounting.
- CP:  State explicitly what needs to be done.
- HS:  Server needs to accept the need to do it.
- CP:  Imposing rules on URL is hard.  Nontraditional URIs.  What do you do.
- HS:  Toss this feature and verbiage from the spec. because it is hard 
to specify.
- MW:  Leave it as open issue.  See if Tom has issue.  But will remove 
from spec if no objections.
Conclusion:  Hard feature to specify.  We should not try to do it 
because we will do a poor job of it.  Will remove from spec if no 
objections.

2E:  A=range in SDP indicates live media
- HS:  An undefined range means there is no defined a range forward or back.
- GS:  How about RTP-Info.  Does it go away?
- MW:  You need RTP-Info
- AN:  Server does not know NPT.  Can't generate it in many cases.
- HS:  How do you do Tivo like model?
- HS:  From whereever you started recording to now.
- CP:  It's not quite that simple.  You can play past the pause point 
because the Tivo is still recording.
- HS:  Dont' want to break the spec.  Don't want to label things. 
Because you may have part live and part not live.
- MW:  Two aspects.  We need something new, but we can't break range.
- HS:  If range abs time doesnt support this model.
- HS:  If you use UTC, then you know if you are asking for a time 
tomorrow then it is live.
- CP:  How about clock skew.
- HS:  If we can do time till now, it works with the Tivo Case.
- TZ:  If you say UTC -, then the slider size is going to be hard to 
determine.
- HS:  a user interface issue.
Conclusion: 
  + In the SDP range if there is an open range, then it is live. We need 
to check the Tivo case is supported
  + Need Live media defintion: media with out a defined end point during 
playback.
  + SDP is referencing RTSP for range format.  Need to register with IANA.

2.F  REDIRECT with Connetion close:
- MW:  Normally connection is closed by the client.  The server may 
close the connection after 200k is recieved.
- TZ:  a bit rude bit its fine.
Conclusion:  Server MAY drop the connection after it has recieved 200 OK 
response on a REDIRECT.

2G.  End of Stream
- TZ:  Need a persistant connection
- MW:  Is there a usecase in SIP for End of playout notification?
- HS:  There Is media level, application level and , session level
- HS:  It is hard to dectect the last packet stream.  Technically hard.
- HS:  What is the use case for needing end of stream in the RTP/RTCP?  
End of segment/ End of stream is it a more generic problem?
- TZ:  This is more like an event.
- HS:  DTMF-AD  was trying to signal time synchornized notification.  
Not universally popular.
- TZ:  Not event registration frame work.  Is something like SIP event 
framework overkill?
- AN:  The client in RTSP doesnt know the media has ended if you play 
Range -
- TZ:  Also for live.
- AN:  but the live server doesnt know when the range ends
- HS:  How about send an announce with the new SDP that stream's end 
range has changed?
- HS:  How about BYE
- TZ:  Don't wnat to do BYE because it gets rid of the SSRC and you want 
to keep RTCP reporting.
- TZ:  Also want to report error
- HS:  Make that generic.  Is the consumer a machine or is it for human 
user?
- AN:  Once persistence is established you can add all kinds of server 
reporting features.
- SS:  Almost always it will be a machine that will consume the error 
information.
- TZ:  A more generic framework for header that can be in end of stream 
and in Announce.
- HS:  Use Announce to send the SDP with the new range header
- TZ:  You dont need the whole range header.  You alredy have have SDP.
- HS:  You have a trade of with efficiently and future supoprt.  Try to 
overly optimize is not a good idea.
- AN:  No reason why RTCP can't say end of stream as a separate thing.
- HS:  Yes.  But keep it separarte and have AVT work on it.
- MW:  Use ANNOUNCE.  to do this.  Revivie Sean's old spec.
Conclusion:  Keep the end of stream messages in the control channel 
rather than the data chanel. TZ will take ownership.
- Keep the idea of server trying to start a coneection with client as a 
separate issue.

2H.  How long to wait for response?
- SS:  5 Seconds?
- HS:  10 times longer than SIP.
- SS:  This is RTSP over TCP.  UDP is lot longer
- HS:  100 is a indication that it will succeed.  You want to send it 
early rather than later.  So 5 seconds may be too long.
- TS:  The first 100 response is easy.  Do we need to send it periodically.
- HS:  There is a reasons to send useful feedback, but do we need to 
send more than that.
- HS:  Should not be mandated
- AN:  Server MAY / SHOULD send 100, but client MUST deal with it.
- CP:  If the server knows its going to take appreciable amount of time, 
send 100 immediately (less than 2 seconds).
- TZ:  Sean's proposal is reasonable.
- CP:  If the server should send in a "couple" of seconds.
- CP:  It's hard to place a time limit.
- TZ:  Implementor's guide.
- HS:  Not a good idea because it's hard to do, and may contradict the spec.
- TZ:  Provide some motivation example.
Conclusion:  Sean's writeup seems reasonable.  Change the timelimit to 2 
seconds and make sure that wording is there to indicate server should do 
this immediately if it know it's going to take a long time.  Need 
motivation example.

2I.  Version number of new spec
- MW: will try to make it 1.0
- HS: Protocols don't generally get a second chance
- HS: If you want to go with a new protocol version than get some better 
features.
- TZ: There are servers out there that wont interop.
- CP: Document where the potential interop problems will be.
- AN: We will be hard pressed to keep the current version number.  But 
let's see after the change spec is done.
 
2J.  Teardown * and Ready-nm
- Removed from spec. Awaiting WG review

2K.  How to handle sync in re-SETUP
- Awaiting review from the WG.
- Need to look at the level of mandate for RTP-Info in live case.

2L.  Re-Setup of Transport parameters
- GS:  Not to unicast -> multicast. 
- HS:  May want to switch from multicast to unicast. . .
- MW:  Take to list.

2M.  Removal of SETUP in play state
- Removed from spec.
- CP:  Keep it unspecified rather then saying that it cannot be done.  
That breaks being able to do that in the future.
- AN:  Explicitly state that it is unspecified instead of cant be done.
Conclusion:  Remove but state that it is unspecified and a new draft 
needs to be writted to make it work.

2N.  Use of c= in SDP to know possible address classes
- CP: It is reasonable to connet to the server with one protocol and use 
a different protocol for the media.
- TZ: Is it reasonable to have two, (one for IPv4, and one for IPv6).
- CP: You can specify alternatives.
- MW: 3338+gonzolo. Use the alt group thing.

Conclusion:
Should look at address classes in the SDP to figure what protocol to use 
(need to add SDP section of RTSP)
Look at c= line syntax.  Multiple c lines can be use in media field?  
SIP resolution was that multiple c lines should not be used to express 
multiple connection address.

2O.  How should proxies cache?  OPTIONS. 
Conclusion:
Caching dropped.  make it explicitly unspecified in the spec.

3A.  ABNF should be moved to end. 
- HS:  Difficult problem with SIP spec is that the when the semantic  
description is given, it should reference ABNF name.
- CP:  IETF has tools as well has henning.

3B.  . . .

4.  How to do RTSP over NAT?

NAT Requirements? 
Priorities:
 1. Server is deployed in open network.
 2. Server behind firewall next.
 3. If a client is not dual hosted, it should not have to deal with 
additional protocols.
 4. Must have defense against DOS
 5. Security.
 6. MUST deal with symmetric nat.
 7. ALG.  The firewall traversal should not interfere with ALGs. 
 8. Advise to ALG writers not to screw it up.  

Aravind


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To: Tom Marshall <tmarshall@real.com>
CC: Sean Sheedy <seans@ncube.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP: Proposed text for client timeout
References: <20031112204019.GA28461@real.com>
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Hi,

I think Tom's text is basically what the conclusions was of yesterday's 
RTSP meeting.

I think there was talk about being a little bit more strict on some 
cases. I think there was some talk about saying: A server MUST send a 
1xx response if the request takes or is expected to take more than 2 
seconds to process. The server SHOULD send further 1xx messages if 
further information about the processing of the request becomes available.

On the client side we talked about writing some general cautioning that 
the client needs to take transport delays into consideration.

Cheers

Magnus

Tom Marshall wrote:

>>Since what constitutes acceptable delays in RTSP is determined by
>>    
>>
>human
>  
>
>>perception, the source of the delay (packet loss, network congestion,
>>server load, etc.) is irrelevant.  The only question that needs to be
>>answered from the client's perspective is:  Is the server working on
>>    
>>
>the
>  
>
>>request, or not?  I think it's important that a user interface
>>    
>>
>provides
>  
>
>>this information to the human waiting for the media stream.
>>    
>>
>
>Absolutely.  So why are we proposing to measure RTT at all?  Why don't
>we
>just say this:
>
>  The server SHOULD respond to all requests as soon as practicable.  If
>a
>  request takes a significant amount of time to process, or there is an
>  expectation that it will, the server SHOULD send 100 responses at
>regular
>  intervals so that the client knows its request is still being
>processed.
>
>  The client SHOULD give the server a reasonable amount of time to
>process its
>  requests.  If an unreasonable amount of time elapses without receiving
>any
>  response, the client MAY assume the server is unresponsive and abort
>the
>  connection.  If a 100 response is received, the client MUST reset its
>  response timer.
>
>The only remaining question would be what constitutes a reasonable
>amount of
>time and a regular interval.
>
>  
>


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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Aravind Narasimhan'" <Aravind.Narasimhan@Sun.COM>, mmusic@ietf.org
Subject: RE: [MMUSIC] RTSP offline meeting Notes - 11/11
Date: Wed, 12 Nov 2003 13:05:26 -0800
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Thanks for the comments. Two  important corrections, see inline.

> -----Original Message-----
> From: Aravind Narasimhan [mailto:Aravind.Narasimhan@Sun.COM]
> Sent: Wednesday, November 12, 2003 12:50 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] RTSP offline meeting Notes - 11/11
> 
> 
>   The meeting followed the agenda that Magnus sent out on 11/10
> 
> Attendees:
> 
> Thomas Zeng,
> Henning Schulzerine
> Geetha Srikantan
> Victor Varsa
> Sean Sheedy
> Adrea Basso
> Magnus Westerlund
> Colin Perkins
> Aravind Narasimhan
> 
> 
> Issues:
> 
> 2A. RTP Timestamps under pause clarification. 
> - Aravind to write up text  - Not Done.  Hopefully by the end of week.
> 
> 2B. PLAY request at the end of a media clip
> - Awaiting review
> 
> 2C. Implicit Redirect
> - Philippe/Aravind to write up text?
> 
> 2D. Aggregate URL deriving
> - Awaiting review. 
> - HS:  Is there motivation for deriving the URL?  It should 
> be opaque?  
> I can't use for something else?
> - TZ:  It might conflict with an actual URL?
> - HS:  Seems like a recipie for strange cases.
> - TZ:  SDP from two different sources that you want to mix.
> - HS:  The server should give you the URL.  If the SDP 
> doesn't have an 
> aggregate URL than the media can't be played in aggregate fashion.
> - HS:  Concerned about specifying generating URL.  How do you 
> know what 
> you are getting?  What if it is not there?
> - TZ:  Have SETUP two separate streams.  Need to control together. 
> Session ID should be enough, but URL is needed for PLAY request.
> - AN:  URL is used for proxies and accounting.
> - CP:  State explicitly what needs to be done.
> - HS:  Server needs to accept the need to do it.
> - CP:  Imposing rules on URL is hard.  Nontraditional URIs.  
> What do you do.
> - HS:  Toss this feature and verbiage from the spec. because 
> it is hard 
> to specify.
> - MW:  Leave it as open issue.  See if Tom has issue.  But 
> will remove 
> from spec if no objections.
> Conclusion:  Hard feature to specify.  We should not try to do it 
> because we will do a poor job of it.  Will remove from spec if no 
> objections.
> 
> 2E:  A=range in SDP indicates live media
> - HS:  An undefined range means there is no defined a range 
> forward or back.
> - GS:  How about RTP-Info.  Does it go away?
> - MW:  You need RTP-Info
> - AN:  Server does not know NPT.  Can't generate it in many cases.
> - HS:  How do you do Tivo like model?
> - HS:  From whereever you started recording to now.
> - CP:  It's not quite that simple.  You can play past the pause point 
> because the Tivo is still recording.
> - HS:  Dont' want to break the spec.  Don't want to label things. 
> Because you may have part live and part not live.
> - MW:  Two aspects.  We need something new, but we can't break range.
> - HS:  If range abs time doesnt support this model.
> - HS:  If you use UTC, then you know if you are asking for a time 
> tomorrow then it is live.
> - CP:  How about clock skew.
> - HS:  If we can do time till now, it works with the Tivo Case.
> - TZ:  If you say UTC -, then the slider size is going to be hard to 
> determine.
> - HS:  a user interface issue.
> Conclusion: 
>   + In the SDP range if there is an open range, then it is 
> live. We need 
> to check the Tivo case is supported
>   + Need Live media defintion: media with out a defined end 
> point during 
> playback.
>   + SDP is referencing RTSP for range format.  Need to 
> register with IANA.
> 
> 2.F  REDIRECT with Connetion close:
> - MW:  Normally connection is closed by the client.  The server may 
> close the connection after 200k is recieved.
> - TZ:  a bit rude bit its fine.
> Conclusion:  Server MAY drop the connection after it has 
> recieved 200 OK 
> response on a REDIRECT.
> 
> 2G.  End of Stream
> - TZ:  Need a persistant connection
> - MW:  Is there a usecase in SIP for End of playout notification?
> - HS:  There Is media level, application level and , session level
> - HS:  It is hard to dectect the last packet stream.  
> Technically hard.
> - HS:  What is the use case for needing end of stream in the 
> RTP/RTCP?  
> End of segment/ End of stream is it a more generic problem?
> - TZ:  This is more like an event.
> - HS:  DTMF-AD  was trying to signal time synchornized notification.  
> Not universally popular.
> - TZ:  Not event registration frame work.  Is something like 
> SIP event 
> framework overkill?
> - AN:  The client in RTSP doesnt know the media has ended if you play 
> Range -
> - TZ:  Also for live.
> - AN:  but the live server doesnt know when the range ends
> - HS:  How about send an announce with the new SDP that stream's end 
> range has changed?
> - HS:  How about BYE
> - TZ:  Don't wnat to do BYE because it gets rid of the SSRC 
> and you want 
> to keep RTCP reporting.
> - TZ:  Also want to report error
> - HS:  Make that generic.  Is the consumer a machine or is it 
> for human 
> user?
> - AN:  Once persistence is established you can add all kinds 
> of server 
> reporting features.
> - SS:  Almost always it will be a machine that will consume the error 
> information.
> - TZ:  A more generic framework for header that can be in end 
> of stream 
> and in Announce.
> - HS:  Use Announce to send the SDP with the new range header
> - TZ:  You dont need the whole range header.  
What I said was:   You don't need the whole SDP here, just an updated Range
plus Reason header.
> You alredy have 
> have SDP.
> - HS:  You have a trade of with efficiently and future 
> supoprt.  Try to 
> overly optimize is not a good idea.
> - AN:  No reason why RTCP can't say end of stream as a separate thing.
> - HS:  Yes.  But keep it separarte and have AVT work on it.
> - MW:  Use ANNOUNCE.  to do this.  Revivie Sean's old spec.
> Conclusion:  Keep the end of stream messages in the control channel 
> rather than the data chanel. TZ will take ownership.
> - Keep the idea of server trying to start a coneection with 
> client as a 
> separate issue.
> 
> 2H.  How long to wait for response?
> - SS:  5 Seconds?
> - HS:  10 times longer than SIP.
> - SS:  This is RTSP over TCP.  UDP is lot longer
> - HS:  100 is a indication that it will succeed.  You want to send it 
> early rather than later.  So 5 seconds may be too long.
> - TS:  The first 100 response is easy.  Do we need to send it 
> periodically.
> - HS:  There is a reasons to send useful feedback, but do we need to 
> send more than that.
> - HS:  Should not be mandated
> - AN:  Server MAY / SHOULD send 100, but client MUST deal with it.
> - CP:  If the server knows its going to take appreciable 
> amount of time, 
> send 100 immediately (less than 2 seconds).
> - TZ:  Sean's proposal is reasonable.
> - CP:  If the server should send in a "couple" of seconds.
> - CP:  It's hard to place a time limit.
> - TZ:  Implementor's guide.
> - HS:  Not a good idea because it's hard to do, and may 
> contradict the spec.
> - TZ:  Provide some motivation example.
> Conclusion:  Sean's writeup seems reasonable.  Change the 
> timelimit to 2 
> seconds and make sure that wording is there to indicate 
> server should do 
> this immediately if it know it's going to take a long time.  Need 
> motivation example.
> 
> 2I.  Version number of new spec
> - MW: will try to make it 1.0
> - HS: Protocols don't generally get a second chance
> - HS: If you want to go with a new protocol version than get 
> some better 
> features.
> - TZ: There are servers out there that wont interop.
> - CP: Document where the potential interop problems will be.
> - AN: We will be hard pressed to keep the current version 
> number.  But 
> let's see after the change spec is done.
>  
> 2J.  Teardown * and Ready-nm
> - Removed from spec. Awaiting WG review
> 
> 2K.  How to handle sync in re-SETUP
> - Awaiting review from the WG.
> - Need to look at the level of mandate for RTP-Info in live case.
> 
> 2L.  Re-Setup of Transport parameters
> - GS:  Not to unicast -> multicast. 
> - HS:  May want to switch from multicast to unicast. . .
> - MW:  Take to list.
> 
> 2M.  Removal of SETUP in play state
> - Removed from spec.
> - CP:  Keep it unspecified rather then saying that it cannot 
> be done.  
> That breaks being able to do that in the future.
> - AN:  Explicitly state that it is unspecified instead of 
> cant be done.
> Conclusion:  Remove but state that it is unspecified and a new draft 
> needs to be writted to make it work.
> 
> 2N.  Use of c= in SDP to know possible address classes
> - CP: It is reasonable to connet to the server with one 
> protocol and use 
> a different protocol for the media.
> - TZ: Is it reasonable to have two, (one for IPv4, and one for IPv6).
> - CP: You can specify alternatives.
> - MW: 3338+gonzolo. Use the alt group thing.
> 
> Conclusion:
> Should look at address classes in the SDP to figure what 
> protocol to use 
> (need to add SDP section of RTSP)
> Look at c= line syntax.  Multiple c lines can be use in media field?  
> SIP resolution was that multiple c lines should not be used 
> to express 
> multiple connection address.
> 
> 2O.  How should proxies cache?  OPTIONS. 
> Conclusion:
> Caching dropped.  make it explicitly unspecified in the spec.
> 
> 3A.  ABNF should be moved to end. 
> - HS:  Difficult problem with SIP spec is that the when the semantic  
> description is given, it should reference ABNF name.
> - CP:  IETF has tools as well has henning.
> 
> 3B.  . . .
> 
> 4.  How to do RTSP over NAT?
> 
> NAT Requirements? 
> Priorities:
>  1. Server is deployed in open network.
>  2. Server behind firewall next.
>  3. If a client is not dual hosted,
If a client is neither behind NAT/FW, nor dual-hosted, it should not have to
pay additional price (in terms of delay, at least).
> it should not have to deal with 
> additional protocols.
>  4. Must have defense against DOS
>  5. Security.
>  6. MUST deal with symmetric nat.
>  7. ALG.  The firewall traversal should not interfere with ALGs. 
>  8. Advise to ALG writers not to screw it up.  
> 
> Aravind
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

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From: Tom Marshall <tmarshall@real.com>
To: Aravind Narasimhan <Aravind.Narasimhan@Sun.COM>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP offline meeting Notes - 11/11
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> 2D. Aggregate URL deriving
> - Awaiting review.=20
> - HS:  Is there motivation for deriving the URL?  It should be opaque? =
=20
> I can't use for something else?
> - TZ:  It might conflict with an actual URL?
> - HS:  Seems like a recipie for strange cases.
> - TZ:  SDP from two different sources that you want to mix.
> - HS:  The server should give you the URL.  If the SDP doesn't have an=20
> aggregate URL than the media can't be played in aggregate fashion.
> - HS:  Concerned about specifying generating URL.  How do you know what=
=20
> you are getting?  What if it is not there?
> - TZ:  Have SETUP two separate streams.  Need to control together.=20
> Session ID should be enough, but URL is needed for PLAY request.
> - AN:  URL is used for proxies and accounting.
> - CP:  State explicitly what needs to be done.
> - HS:  Server needs to accept the need to do it.
> - CP:  Imposing rules on URL is hard.  Nontraditional URIs.  What do you =
do.
> - HS:  Toss this feature and verbiage from the spec. because it is hard=
=20
> to specify.
> - MW:  Leave it as open issue.  See if Tom has issue.  But will remove=20
> from spec if no objections.
> Conclusion:  Hard feature to specify.  We should not try to do it=20
> because we will do a poor job of it.  Will remove from spec if no=20
> objections.

I don't have any objection either way here.  It would be nice if we could
provide a recommendation to make this work but if not, it's not a big deal.=
=20
There is no real-world use case as far as I can tell.  My theoretical use
case (playing audio from one file and video from another) is not supported
by our server.  If a use case pops up, the implementor can deal with it.

> 2G.  End of Stream
> - TZ:  Need a persistant connection
> - MW:  Is there a usecase in SIP for End of playout notification?
> - HS:  There Is media level, application level and , session level
> - HS:  It is hard to dectect the last packet stream.  Technically hard.
> - HS:  What is the use case for needing end of stream in the RTP/RTCP? =
=20
> End of segment/ End of stream is it a more generic problem?
> - TZ:  This is more like an event.
> - HS:  DTMF-AD  was trying to signal time synchornized notification. =20
> Not universally popular.
> - TZ:  Not event registration frame work.  Is something like SIP event=20
> framework overkill?
> - AN:  The client in RTSP doesnt know the media has ended if you play=20
> Range -
> - TZ:  Also for live.
> - AN:  but the live server doesnt know when the range ends
> - HS:  How about send an announce with the new SDP that stream's end=20
> range has changed?
> - HS:  How about BYE
> - TZ:  Don't wnat to do BYE because it gets rid of the SSRC and you want=
=20
> to keep RTCP reporting.
> - TZ:  Also want to report error
> - HS:  Make that generic.  Is the consumer a machine or is it for human=
=20
> user?
> - AN:  Once persistence is established you can add all kinds of server=20
> reporting features.
> - SS:  Almost always it will be a machine that will consume the error=20
> information.
> - TZ:  A more generic framework for header that can be in end of stream=
=20
> and in Announce.
> - HS:  Use Announce to send the SDP with the new range header
> - TZ:  You dont need the whole range header.  You alredy have have SDP.
> - HS:  You have a trade of with efficiently and future supoprt.  Try to=
=20
> overly optimize is not a good idea.
> - AN:  No reason why RTCP can't say end of stream as a separate thing.
> - HS:  Yes.  But keep it separarte and have AVT work on it.
> - MW:  Use ANNOUNCE.  to do this.  Revivie Sean's old spec.
> Conclusion:  Keep the end of stream messages in the control channel=20
> rather than the data chanel. TZ will take ownership.
> - Keep the idea of server trying to start a coneection with client as a=
=20
> separate issue.

If we take an END_OF_STREAM proposal, I think it should be an extension
signalled by a feature tag.  I believe this is a deficiency in the RTP
transport and we should bring the issue up in AVT.  We have heard from
someone that has implemented a transport level stream end notification so we
know that it's both possible and deployed.  We also know that END_OF_STREAM
has issues with connectionless control that a transport solution does not.

> 2H.  How long to wait for response?
> - SS:  5 Seconds?
> - HS:  10 times longer than SIP.
> - SS:  This is RTSP over TCP.  UDP is lot longer
> - HS:  100 is a indication that it will succeed.  You want to send it=20
> early rather than later.  So 5 seconds may be too long.
> - TS:  The first 100 response is easy.  Do we need to send it periodicall=
y.
> - HS:  There is a reasons to send useful feedback, but do we need to=20
> send more than that.
> - HS:  Should not be mandated
> - AN:  Server MAY / SHOULD send 100, but client MUST deal with it.
> - CP:  If the server knows its going to take appreciable amount of time,=
=20
> send 100 immediately (less than 2 seconds).
> - TZ:  Sean's proposal is reasonable.
> - CP:  If the server should send in a "couple" of seconds.
> - CP:  It's hard to place a time limit.
> - TZ:  Implementor's guide.
> - HS:  Not a good idea because it's hard to do, and may contradict the sp=
ec.
> - TZ:  Provide some motivation example.
> Conclusion:  Sean's writeup seems reasonable.  Change the timelimit to 2=
=20
> seconds and make sure that wording is there to indicate server should do=
=20
> this immediately if it know it's going to take a long time.  Need=20
> motivation example.

Please see my message from a few minutes ago.  I don't think RTT
measurements are meaningful.  In addition to being unaware of the physical
layer and TCP retransmission delays, this proposal makes it clear that
servers have real and occasionally significant processing delays.  This
makes any RTT measurement extremely unreliable.  I think that we should drop
the RTT measurements and focus on the actual issue, which is total delay
seen by the client.

I have implementation concerns with using 100 responses on the server side.=
=20
As an example, our server is highly asynchronous.  It does nonblocking reads
and writes on all I/O, including files.  Further, complex datatypes (such as
mpeg4) require quite a few read/seek operations to formulate an SDP for the
DESCRIBE response.  Now suppose the server is busy.  Really busy.  It's
taking, say, 100ms or so to process all of its events each time through its
main event loop.  The server doesn't know how long it is going to take to
formulate the SDP because it doesn't know exactly how many reads it will
take.  And to make matters worse, sending 100 responses both generates more
network traffic and more event traffic (you need to set a timer so that you
know when the response is taking too long).  It may even make the difference
between handling the requests and failing.  Is sending one or more 100
responses really the best thing to do in this situation?

--=20
Most people are too busy to have time for anything important.

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Hi,

some comments, removing any sections without comments:

Aravind Narasimhan wrote:

>
> 2E:  A=range in SDP indicates live media
> - HS:  An undefined range means there is no defined a range forward or 
> back.
> - GS:  How about RTP-Info.  Does it go away?
> - MW:  You need RTP-Info
> - AN:  Server does not know NPT.  Can't generate it in many cases.
> - HS:  How do you do Tivo like model?
> - HS:  From whereever you started recording to now.
> - CP:  It's not quite that simple.  You can play past the pause point 
> because the Tivo is still recording.
> - HS:  Dont' want to break the spec.  Don't want to label things. 
> Because you may have part live and part not live.
> - MW:  Two aspects.  We need something new, but we can't break range.
> - HS:  If range abs time doesnt support this model.
> - HS:  If you use UTC, then you know if you are asking for a time 
> tomorrow then it is live.
> - CP:  How about clock skew.
> - HS:  If we can do time till now, it works with the Tivo Case.
> - TZ:  If you say UTC -, then the slider size is going to be hard to 
> determine.
> - HS:  a user interface issue.
> Conclusion:  + In the SDP range if there is an open range, then it is 
> live. We need to check the Tivo case is supported
>  + Need Live media defintion: media with out a defined end point 
> during playback.
>  + SDP is referencing RTSP for range format.  Need to register with IANA.


I think that we had a conclusion to write a informative note about TIVO 
scenario signalling using large future values. The question is how this 
relates to the range limitations and what happens when a client asks for 
a futuer point. As I can see it the server can answer with what it 
actually will play out.

>
> 2G.  End of Stream
> - TZ:  Need a persistant connection
> - MW:  Is there a usecase in SIP for End of playout notification?
> - HS:  There Is media level, application level and , session level
> - HS:  It is hard to dectect the last packet stream.  Technically hard.
> - HS:  What is the use case for needing end of stream in the 
> RTP/RTCP?  End of segment/ End of stream is it a more generic problem?
> - TZ:  This is more like an event.
> - HS:  DTMF-AD  was trying to signal time synchornized notification.  
> Not universally popular.
> - TZ:  Not event registration frame work.  Is something like SIP event 
> framework overkill?
> - AN:  The client in RTSP doesnt know the media has ended if you play 
> Range -
> - TZ:  Also for live.
> - AN:  but the live server doesnt know when the range ends
> - HS:  How about send an announce with the new SDP that stream's end 
> range has changed?
> - HS:  How about BYE
> - TZ:  Don't wnat to do BYE because it gets rid of the SSRC and you 
> want to keep RTCP reporting.
> - TZ:  Also want to report error
> - HS:  Make that generic.  Is the consumer a machine or is it for 
> human user?
> - AN:  Once persistence is established you can add all kinds of server 
> reporting features.
> - SS:  Almost always it will be a machine that will consume the error 
> information.
> - TZ:  A more generic framework for header that can be in end of 
> stream and in Announce.
> - HS:  Use Announce to send the SDP with the new range header
> - TZ:  You dont need the whole range header.  You alredy have have SDP.
> - HS:  You have a trade of with efficiently and future supoprt.  Try 
> to overly optimize is not a good idea.
> - AN:  No reason why RTCP can't say end of stream as a separate thing.
> - HS:  Yes.  But keep it separarte and have AVT work on it.
> - MW:  Use ANNOUNCE.  to do this.  Revivie Sean's old spec.
> Conclusion:  Keep the end of stream messages in the control channel 
> rather than the data chanel. TZ will take ownership.
> - Keep the idea of server trying to start a coneection with client as 
> a separate issue.
>
> 2H.  How long to wait for response?
> - SS:  5 Seconds?
> - HS:  10 times longer than SIP.
> - SS:  This is RTSP over TCP.  UDP is lot longer
> - HS:  100 is a indication that it will succeed.  You want to send it 
> early rather than later.  So 5 seconds may be too long.
> - TS:  The first 100 response is easy.  Do we need to send it 
> periodically.
> - HS:  There is a reasons to send useful feedback, but do we need to 
> send more than that.
> - HS:  Should not be mandated
> - AN:  Server MAY / SHOULD send 100, but client MUST deal with it.
> - CP:  If the server knows its going to take appreciable amount of 
> time, send 100 immediately (less than 2 seconds).
> - TZ:  Sean's proposal is reasonable.
> - CP:  If the server should send in a "couple" of seconds.
> - CP:  It's hard to place a time limit.
> - TZ:  Implementor's guide.
> - HS:  Not a good idea because it's hard to do, and may contradict the 
> spec.
> - TZ:  Provide some motivation example.
> Conclusion:  Sean's writeup seems reasonable.  Change the timelimit to 
> 2 seconds and make sure that wording is there to indicate server 
> should do this immediately if it know it's going to take a long time.  
> Need motivation example.

I think we ad more discussion in the direction that we don't need exact 
rules for clients on how long they should wait, rather clarify why the 
client should wait sufficently long.

>
> 2I.  Version number of new spec
> - MW: will try to make it 1.0
> - HS: Protocols don't generally get a second chance
> - HS: If you want to go with a new protocol version than get some 
> better features.
> - TZ: There are servers out there that wont interop.
> - CP: Document where the potential interop problems will be.
> - AN: We will be hard pressed to keep the current version number.  But 
> let's see after the change spec is done. 


I think the hennings important comments, that a solution if one creates 
non-interoperable changes to document them and explain how to support 
them, but clearly show that they are deprecated and can be removed at a 
later stage when old implementation has been upgraded.

> 3B.  . . . 


This will be included due to 2N.


Cheers

Magnus



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Subject: [MMUSIC] SDP security descriptions - updates and open issues in -02
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Greetings

We submitted an update to the SDP Security Descriptions draft which you
can find at:

 http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions-02.t=
xt

The draft was reworked to more clearly divide it into two:
- A core security description framework
- An SRTP profile of the security description core.

For each of these, details on high-level operation, parameter
definitions, use with and without offer/answer, and a grammar was
provided. Also, the following more detailed changes were introduced:
- Offer/Answer highlights differences between unicast/multicast, and
initial/subsequent offer.
- Rekeying addressed via offer/answer; everything can be changed (not
just keys)
- Additional rules and considerations for communicating the SRTP SRC
parameter (SSRC, SEQ, ROC).
- Removed SRTP =93use=94 parameter (decryption, encryption or both) - now
inferred from context
- Added Window-Size Hint and <From,To> to SRTP profile
- Extension rules were defined for key methods, SRTP crypto-suites, and
SRTP session parameters
- IANA considerations and registration procedures were added
- Grammar was updated

Finally, in Section 9 (and in a couple of places throughout the
document), we identified the following open issues:

1) The use of security descriptions, and in particular SRTP security
descriptions, with multicast streams where offer/answer is being used is
not well understood and results in several complications.

2) Another aspect related to multicast is the SRC parameter which is
used to instantiate crypto contexts. The question is how this parameter
should be used with multicast streams, and in particular if it should be
allowed to instantiate multiple crypto context. There is also an issue
here in how to deal with "SSRC collisions" which is primarily a concern
for multicast.

3) The security descriptions do not deal with hierarchically encoded
streams (or at least they have not been considered). Note that this is
only an issue for multicast streams.

4) The current mechanism does not allow for a key to be specified as
being an encryption or decryption key or both; instead this is inferred
from the context (e.g. unicast offer).  Should there be a mechanism to
allow a key to be tagged as an encryption, decryption
or both key ? Again, this is mostly seen as an issue for multicast
streams.

Support for multicast streams is at the heart of these issues, and
solving them in a way that is satisfactory to multicast in general is
something the authors believe will require considerable work and result
in a significant delay in progressing the draft. We therefore propose to
restrict the scope of the draft to deal with unicast streams only and
leave support for multicast streams for further study. If there is
sufficient interest in solving all the multicast issues, these can then
be worked on in a seperate document that can still use the general
mechanism defined in the current draft.

Please let us know if you disagree with this proposal.

Thanks and regards

            Flemming, Mark, and Dan



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Hi Jonathan:

I finally got around to go through your draft. As always, I enjoy your
writing style very much, plus that this version is much improved than the
previous one.

Some comments (more or less from RTSP perspective):

1. Section 5.1, last paragraph:

There is a statement:  "However, in
   order to communicate with the peer, at least one of the offered
   addresses has to be guaranteed to work with any peer that might be
   called. This generally requires that at least one of the derived
   addresses be obtained from a relay service (such as TURN) that exist
   within the public Internet."

I don't agree this statement. Consider the case of RTSP streaming, if the
server is in the open, and client behind NAT, 
I fail to see the need to require "relay service (such as TURN)".  

2. Section 5.5, page 13, also on page 14, 3rd paragraph

There is a statement:  "This BindingRequest MUST contain the USERNAME
   attribute, and the value of the USERNAME MUST equal the username from
   that alternate element. Similarly, the BindingRequest MUST contain a
   PASSWORD attribute, and the value of the PASSWORD MUST equal the
   password from the alternate element."
Why can't a USERNAME be globally unqiue for each RTP and RTCP port? That way
USERNAME alone is enough to figure out the alternate.
Inclusion of password in the BindingRequest is a security compromise
(especially that such message is  sent in the clear), and it contradicts
rfc3489,
section 9.3, where the password is not included in the BindingRequest.

It is important not to send password in the open.

3. Also page 13: on the lock down problem,  the solution I see would be to
have multiple TURN servers available, so as to  make it possible to try
alternates in parrellel.
TURN servers in general are not very scalable any way, so it is not
unthinkable to deploy multiple STUN servers ...

4. section 6.2 second paragraph:

The scenario described is not going to work for symetric NAT, am I correct?
If so, it probably is useful to state it explicitly.

5. Section 7: xml schema for ICE Messages:

There is one attribute ("extension base?") called "rtp-info". Does this have
anything to do with "rtp-info" in RFC2326? It doesn't match the parameters
in rfc2326's Rtp-info header.
 I suspect  this to be a name space conflict?


6. It would nice to have a mapping from ICE to RTSP.

thanks,
thomas




> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, November 07, 2003 7:49 AM
> To: Jonathan Rosenberg; IETF MMUSIC WG
> Subject: [MMUSIC] Comment on draft-ietf-mmusic-ice-00.txt
> 
> 
> Hi Jonathan,
> 
> I had a late night reading session yesterday that produce 
> some comments.
> 
> 1. I think there are still maybe a few places where ICE is to 
> situation 
> specific towards SIP. I will not go so long to require that 
> mapping of 
> transport addresses are not IP + transport protocol ports as this is 
> what our NAT problem causes. However the constant reference to RTP 
> should probably be removed, there are at least other 
> proprietary media 
> protocols. So I think one should talk about media protocol instead of 
> RTP + RTCP.
> 
> 2. I think some better motivation why the openly available 
> password in 
> the STUN request is acceptable. And then why the integrity protection 
> gives advantages or not.
> 
> 3. When the responder performs his connectivity checks and 
> succeed, why 
> does the initiator need to perform a counter check also? Some more 
> should probably be spent on this. The initiator knows the 
> source address 
> as it sees it to send any packets to. The only piece of information 
> missing is what responder connection interface it goes to. 
> This could be 
> identified using STUN supplements and the response message 
> connection IDs.
> 
> Then the only reasons for doing a counter check is to possibly enable 
> connectivity over a more prioritized connection. OR have I missed 
> something about this.
> 
> 4. Of course not doing it both way will not provide full protection 
> against the RTP DOS attack. However you have not at all discussed the 
> applicability of ICE for RTP DOS prevention.
> 
> 5. I also are not finding anything about the retrying in the 
> transmission of STUN request. Should these be done in any special way 
> for ICE? Also there is very little discussion about timer 
> consideration 
> for performing the connectivity checks.
> 
> 6. In example 6.2 i think it can be easily expanded to show one 
> important functionality of ICE, the possibility to open up restricted 
> NAT through the usage of multiple ICE cycles. If A is behind a 
> restricted NAT it will be able to determine its outermost address and 
> port. However in cycle 1, B will not be able to reach A. When then A 
> tries B over the cycle this may succeed. Then B can verify this by 
> STUNNING a second time. These possibilities are rather tuned down.
> 
> 7. I think the XML description should be more generic and 
> include 1..N 
> transport addresses for each media session.
> 
> 8. If anyone missed the other message, the usage of "alt" has 
> a conflict 
> with 3GPP TS 26.234, Rel 6 preliminary specification.
> 
> 9. I think there should be some example of the complete mesh ICE 
> actually tries. I know it will be a hard thing to draw using 
> ASCII art.
> 
> 10. If one uses ICE for other purposes then UNSAF mechanism 
> it suddenly 
> will not have any exit strategy for that purpose. However it should 
> still be true that if no relays, etc. are need when people 
> has migrated 
> to IPv6, etc., then it will simply reduce the complexity of doing ICE.
> 
> 11. When one performs ICE, are there any congestion control 
> considerations to make? If one digs up a number of address 
> locally the 
> local interface might need to send a lot of packets in a 
> short time as 
> it is desired to perform this in parallel.
> 
> 
> Best Regards
> 
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

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All presenters in tomorrow's MMUSIC session:

please send us a copy of your slides in PS, PDF or PPT
(PPT preferred) until 8am tomorrow morning.

Thanks,
Joerg


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Hi Everyone

We had an IMG "design team bar bof" with Colin & Joerg on IMG in the =
Brit's Pub and collected multiple review comments face-to-face. We're =
updating the IMG requirements and framework and will release updated IDs =
ASAP.

Please review and comment the documents by email (and also tomorrow =
morning if you're with us in Minneapolis).

Here's a summary of the comments and some responses over the good old =
Internet...=20

Cheers, Rod.


---

*** IMG Document Review, IETF-58 / 10.November.2003 / Brit's Pub ***


=3D=3D Requirements Draft =3D=3D

RQ1. requirements should be numbered to make it easy to reperence (and =
identify) them

RQ2. "Agnostic" can have some negative conertainions so swap "agnostic" =
for "supports many different ones".

RQ3. Make it explicit that we assume many more receivers than senders as =
it's important to make the primary use clear.

RQ4. The receiver can be intermittently connected but it is unclear =
whether the document is saying the sender could be too. Show the =
assumption that, for the duration of interest, the sender is continually =
connected.

RQ5. Customizing and Scalability are somewhat exclusive. For general =
scalability we are talking of possibly millions of users, for =
customisation involving receiver -> sender/network messaging we can =
expect scalability to considerably smaller numbers.

RQ6. There are a few requirements draft terminology problems...
The term IMG -> changed to IMG metadata where appropriate, and IMG and =
IMG metadata definitions updated/added:
IMG networks -> networks
IMG transport -> IMG transport protocol
IMG system -> clarified in places
complete IMG knowledge -> complete IMG metadata knowledge
IMG originators -> IMG senders
IMG components -> IMG entities
IMG content -> IMG metadata


=3D=3D IMG Framework =3D=3D

FW1. 3.2 is conent use cases. The use of the term "file" was verified in =
3.2.1 - it (correctly) refers to a content type which maybe described =
(and have it's transport media described) in IMG metadata. Minor chanes =
to emphasis this will be made.

FW2. Where to put use cases - it was felt that they're ok where they are

FW3. A "reference/pointer" to an IMG delivery session would be useful - =
it gives for descriptors to define an IMG delivery session. (an SDP =
descriptor?) Clarification for this is added in 4.1.3.

FW4. Framework should not mix the description of IMGs with what is in =
the IETF scope and what is not. Thus the two are now seperated.

FW5. [Note - several other standards bodies could be good places for =
application specific metadata]

FW6. We condisered and change the "SHOULD" to "MUST" for the requirement =
on authenticating fragments of IMG metadata study the feasibility and =
imply options -> Relates, especially, to transceiver/proxy issues of =
middlebox fragmentation, reassembly, aggregation, and augmentation. I.e. =
can a middlebox authenticate original parts of the metadata with the =
original source 'signature' and new/modified parts with it's own =
'signature'. Generally it's not going to be needed. If it is, the =
middlebox either gets the source to re-sign the fragment, or the =
original metadata must be reasonably fragmented already (allowing the =
middlebox to replace a small fragment without changing the authenticity =
of the remainder). Easy fixes in the req doc.

FW7. It's "IMG in IETF" scope to look at IMG Delivery (link traversing), =
whereas it is not "IMG in IETF" scope to standardise =
application-specific metadata including data models. This practically =
means multicast transport (=A76.3.1), Usage of Unicast Transport =
Protocols (=A76.3.2) and The Metadata Envelope (=A76.3.3) are =
recommended for IETF work items. Whereas for the Baseline (Meta)Data =
Model Specification (=A76.3.4) we acknowledge the need to exchange and =
agree on a baseline metadata model and application specific metadata for =
the purposes of interoperability testing between different =
implementations of IMG related IETF protocols - but the IETF standards =
process is not required for this.

FW8. Each connectity/network use case should show that unicast or =
multicast is needed, which they do

FW9. identifier - URL in HTTP, what about delta? how to describe the =
correlations? Identifier is likely to be a URI in all cases. Same ID for =
deltas, just that the payload type (in metadata envelope) would be =
different. We also discussed that SDP-delta would need to be a different =
type from SDP. Solution belongs to the metadata envelope spec.

FW10. Framework draft terminology problems
proxy -> deleted
IMG information -> IMG metadata
IMG protocol -> IMG transport protocol (in 2/3 places)
announcement protocol -> added a note that SAP is an announement =
protocol, but don't really need to define a fixed term

---


Comments are welcome!

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Hello,

A few comments on the latest draft..

1. Page 18, feature tags: It would be good to mention which ones are
   defined currently and refer to section 18 where they are described
   in detail. The text refers to several instances of feature-tags
   well before they are described..it's confusing for the reader. 

2. Page 26: How does table 8.1 relate to entity tags? Am not sure if
   the table needs to be moved?

3. Page 60, last part of 13.4: "The headers value is valid....media
   supports." It's not clear what this means..might want to reword..

4. RTP-Info for Live sessions - specification to be revisited (as 
   discussed in the offline meeting on 11//11).

5. Hope that we can include some text specifying transport negotiation  
   when we've decided what it is.


geetha

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Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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Hi Geetha,

Geetha Srikantan wrote:

>Hello,
>
>A few comments on the latest draft..
>
>1. Page 18, feature tags: It would be good to mention which ones are
>   defined currently and refer to section 18 where they are described
>   in detail. The text refers to several instances of feature-tags
>   well before they are described..it's confusing for the reader. 
>

I can understand this, there are probably a lot of these things that has 
to do with inconsistency of the whole draft.

>
>2. Page 26: How does table 8.1 relate to entity tags? Am not sure if
>   the table needs to be moved?
>

Table 1, are response codes. They do not relate to entity tags, however 
the table is floating to where it fits, accroding to Latex.

>
>3. Page 60, last part of 13.4: "The headers value is valid....media
>   supports." It's not clear what this means..might want to reword..
>

Agree, it need some rewording. Proposal: "The server SHOULD indicate 
what range units it supports, for the given media resource in the SETUP 
response."
Is this clearer?

>
>4. RTP-Info for Live sessions - specification to be revisited (as 
>   discussed in the offline meeting on 11//11).
>

Yes.

>
>5. Hope that we can include some text specifying transport negotiation  
>   when we've decided what it is.
>  
>

Yes, please review the proposed text, send comments to the mailing list.

Cheers

Magnus


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From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: Re: [MMUSIC] comments on  draft-ietf-mmusic-rfc2326bis-05.txt
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hi Magnus,

I'm sure most of the editorial nits will get addressed in a couple of
revs.

More inline...

 
On Thu, 2003-11-13 at 15:49, Magnus Westerlund wrote:
> Hi Geetha,
> >
> >3. Page 60, last part of 13.4: "The headers value is valid....media
> >   supports." It's not clear what this means..might want to reword..
> >
> 
> Agree, it need some rewording. Proposal: "The server SHOULD indicate 
> what range units it supports, for the given media resource in the SETUP 
> response."
> Is this clearer?
> 
sounds good..

> >
> >4. RTP-Info for Live sessions - specification to be revisited (as 
> >   discussed in the offline meeting on 11//11).
> >
> 
> Yes.
> 
great..

> >
> >5. Hope that we can include some text specifying transport negotiation  
> >   when we've decided what it is.
> >  
> >
> 
> Yes, please review the proposed text, send comments to the mailing list.
> 

I did take a look at the text in the draft..it looks good. I was hoping
there would be some text to specify which transport parameters can be
negotiated. It would be ok to leave out negotiation of some parameters
to an extension document (such as multicast->unicast), however we should
say so explicitly in the text?
If people are in agreement wrt on which parameters to negotiate (see my
note earlier), I can write up some text for this.

> Cheers
> 
> Magnus
> 
> 
geetha

> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
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-- 
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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From: Peltotalo Sami <peltotas@cs.tut.fi>
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Subject: [MMUSIC] IMG metadata
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Hi,

Has someone read Luoma's metadata dratf
(http://www.ietf.org/internet-drafts/draft-luoma-mmusic-img-metadata-02.txt).
My question concerns about the relational part
of the IMG metadata. For example, we might have a use case where content
providers (IP Service Portal in the metadata model) announce their own
services (many IMG roots) and then IMG transceiver
(see IMG framework draft) merges those to a single IMG root. I think this
database might grow to quite big, and probably it is too large to be forwarded
ahead for example in a single set of metadata.

Has anyone ideas how the relational part of the
metadata might be splitted into smaller pieces, which can then be gathered
and rendered, in the receiver, as a service hierarchy?

How does it sound, if the IMG transceiver sends a set of metadata,
which informs the content providers, that are included in its IMG
(database). After that, it sends the content providers' IMGs. This
first set of metadata helps receivers to notice when they have reveived
the complete IMG from the IMG transceiver.

Metadata spec. says that there is root ID for IMG database. So for
example in the above mentioned use case, the first content provider had ID one, and the
second had ID two. Now the IMG transceiver's IMG database has ID three.
So the IMG tranceiver sends the content providers' IMG with its own IMG
root ID (three). Is this correct?

Cheers, Sami


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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <mmusic@ietf.org>
Subject: [mmusic] Inclusion of media type = "text" in SDP
Date: Fri, 14 Nov 2003 12:31:40 -0600
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This is a request for a minor addition to draft-ietf-mmusic-sdp-new-15.tx=
t
to acknowledge use of the media type "text" in sessions.

An RTP transport for real time character by character transmission of tex=
t
was defined a couple of years ago in RFC2793.
It is now successfully used in conversational applications, usually combi=
ned
with voice and/or video to enable the participants in the session to
communicate in any preferred medium. One important user group is deaf and
hard-of-hearing people, but the feature is of course of value for all.

In the past, most multimedia implementors have been eager to create
interoperability in video and audio, while the text part has been left
behind and iimplemented as manufacturer specific message-oriented
"chat"-additions. It is time for a change, to raise all the three media o=
f
video, text and audio to the same status of interoperability and deployme=
nt.
That will give SIP based conversational services a real benefit.

This is also needed for the transcoding work.

By mistake no registration of "text" as an allowed media in the m=3D
descriptor of SDP has yet taken place. This was discussed in the mmusic
meeting today, and the conclusion was that since it is already a MIME top
level medium, it is just a matter of including it in the sdp document to
make it allowed.

Therefore I suggest the addition of the word "text" in the following thre=
e
places in draft-ietf-mmusic-sdp-new-15.txt .
In discussion with Colin, we came to the conclusion that this action woul=
d
be sufficient for acknowledging the text medium.


--------in section 5.14 ----------------------------------------------
5.14 Media Announcements ("m=3D")

      m=3D<media> <port> <transport> <fmt list>

   A session description may contain a number of media descriptions.
   Each media description starts with an "m=3D" field, and is terminated
   by either the next "m=3D" field or by the end of the session
   description. A media field has several sub-fields.

   The first sub-field is the media type.  Currently defined media are
   "audio", "video", "text", "application", "data" and "control", though
this
   list may be extended in future.

-----further down, same section -----------

   The fourth and subsequent sub-fields are media formats.  For audio
   text and video, these SHOULD reference a MIME sub-type describing the
   format under the "audio", "text" and "video" top-level MIME types.

------In Appendix A------------------------------

; sub-rules of 'm=3D'
      media =3D               token
                            ;typically "audio", "video", "text",
"application"
                            ;or "data"
------End of
proposal-----------------------------------------------------------------=
---
--------

Regards

Gunnar
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------




_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "'Sipping@Ietf. Org'" <sipping@ietf.org>, <mmusic@ietf.org>
Cc: "Toip list" <toip@snowshore.com>
Subject: [mmusic] Inclusion of media type = "text" in SDP
Date: Fri, 14 Nov 2003 12:24:12 -0600
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This is a request for a minor addition to draft-ietf-mmusic-sdp-new-15.tx=
t
to acknowledge use of the media type "text" in sessions.

An RTP transport for real time character by character transmission of tex=
t
was defined a couple of years ago in RFC2793.
It is now successfully used in conversational applications, usually combi=
ned
with voice and/or video to enable the participants in the session to
communicate in any preferred medium. One important user group is deaf and
hard-of-hearing people, but the feature is of course of value for all.

In the past, most multimedia implementors have been eager to create
interoperability in video and audio, while the text part has been left
behind and iimplemented as manufacturer specific message-oriented
"chat"-additions. It is time for a change, to raise all the three media o=
f
video, text and audio to the same status of interoperability and deployme=
nt.
That will give SIP based conversational services a real benefit.

This is also needed for the transcoding work.

By mistake no registration of "text" as an allowed media in the m=3D
descriptor of SDP has yet taken place. This was discussed in the mmusic
meeting today, and the conclusion was that since it is already a MIME top
level medium, it is just a matter of including it in the sdp document to
make it allowed.

Therefore I suggest the addition of the word "text" in the following thre=
e
places in draft-ietf-mmusic-sdp-new-15.txt .
In discussion with Colin, we came to the conclusion that this action woul=
d
be sufficient for acknowledging the text medium.


--------in section 5.14 ----------------------------------------------
5.14 Media Announcements ("m=3D")

      m=3D<media> <port> <transport> <fmt list>

   A session description may contain a number of media descriptions.
   Each media description starts with an "m=3D" field, and is terminated
   by either the next "m=3D" field or by the end of the session
   description. A media field has several sub-fields.

   The first sub-field is the media type.  Currently defined media are
   "audio", "video", "text", "application", "data" and "control", though
this
   list may be extended in future.

-----further down, same section -----------

   The fourth and subsequent sub-fields are media formats.  For audio
   text and video, these SHOULD reference a MIME sub-type describing the
   format under the "audio", "text" and "video" top-level MIME types.

------In Appendix A------------------------------

; sub-rules of 'm=3D'
      media =3D               token
                            ;typically "audio", "video", "text",
"application"
                            ;or "data"
------End of
proposal-----------------------------------------------------------------=
---
--------

Regards

Gunnar
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------




_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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References: <BHEHLFPKIPMLPFNFAHJKEEMFDBAA.gunnar.hellstrom@omnitor.se>
Date: Sun, 16 Nov 2003 12:32:04 -0800
To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>,
        "'Sipping@Ietf. Org'" <sipping@ietf.org>, mmusic@ietf.org
From: Dave Singer <singer@apple.com>
Subject: Re: [mmusic] Inclusion of media type = "text" in SDP
Cc: Toip list <toip@snowshore.com>
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There is also the 3G timed text draft, of course,=20
and the upcoming MPEG and W3C work in timed text.


At 12:24 PM -0600 11/14/03, Gunnar Hellstrom wrote:
>This is a request for a minor addition to draft-ietf-mmusic-sdp-new-15.txt
>to acknowledge use of the media type "text" in sessions.
>
>An RTP transport for real time character by character transmission of text
>was defined a couple of years ago in RFC2793.
>It is now successfully used in conversational applications, usually combine=
d
>with voice and/or video to enable the participants in the session to
>communicate in any preferred medium. One important user group is deaf and
>hard-of-hearing people, but the feature is of course of value for all.
>
>In the past, most multimedia implementors have been eager to create
>interoperability in video and audio, while the text part has been left
>behind and iimplemented as manufacturer specific message-oriented
>"chat"-additions. It is time for a change, to raise all the three media of
>video, text and audio to the same status of interoperability and deployment=
=2E
>That will give SIP based conversational services a real benefit.
>
>This is also needed for the transcoding work.
>
>By mistake no registration of "text" as an allowed media in the m=3D
>descriptor of SDP has yet taken place. This was discussed in the mmusic
>meeting today, and the conclusion was that since it is already a MIME top
>level medium, it is just a matter of including it in the sdp document to
>make it allowed.
>
>Therefore I suggest the addition of the word "text" in the following three
>places in draft-ietf-mmusic-sdp-new-15.txt .
>In discussion with Colin, we came to the conclusion that this action would
>be sufficient for acknowledging the text medium.
>
>
>--------in section 5.14 ----------------------------------------------
>5.14 Media Announcements ("m=3D")
>
>       m=3D<media> <port> <transport> <fmt list>
>
>    A session description may contain a number of media descriptions.
>    Each media description starts with an "m=3D" field, and is terminated
>    by either the next "m=3D" field or by the end of the session
>    description. A media field has several sub-fields.
>
>    The first sub-field is the media type.  Currently defined media are
>    "audio", "video", "text", "application", "data" and "control", though
>this
>    list may be extended in future.
>
>-----further down, same section -----------
>
>    The fourth and subsequent sub-fields are media formats.  For audio
>    text and video, these SHOULD reference a MIME sub-type describing the
>    format under the "audio", "text" and "video" top-level MIME types.
>
>------In Appendix A------------------------------
>
>; sub-rules of 'm=3D'
>       media =3D               token
>                             ;typically "audio", "video", "text",
>"application"
>                             ;or "data"
>------End of
>proposal-------------------------------------------------------------------=
-
>--------
>
>Regards
>
>Gunnar
>-------------------------------------------
>Gunnar Hellstr=F6m
>Omnitor AB
>Renathv=E4gen 2
>SE 121 37 Johanneshov
>SWEDEN
>+46 8 556 002 03
>Mob: +46 708 204 288
>www.omnitor.se
>Gunnar.Hellstrom@Omnitor.se
>--------------------------------------------
>
>
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic


-- 
David Singer
Apple Computer/QuickTime

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/mmusic



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Subject: Re: [mmusic] Inclusion of media type = "text" in SDP
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Gunnar,

On Friday, Nov 14, 2003, at 18:24 Europe/London, Gunnar Hellstrom wrote:
> This is a request for a minor addition to 
> draft-ietf-mmusic-sdp-new-15.txt
> to acknowledge use of the media type "text" in sessions.
...
> Therefore I suggest the addition of the word "text" in the following 
> three
> places in draft-ietf-mmusic-sdp-new-15.txt .
> In discussion with Colin, we came to the conclusion that this action 
> would
> be sufficient for acknowledging the text medium.
>
>
> --------in section 5.14 ----------------------------------------------
> 5.14 Media Announcements ("m=")
>
>       m=<media> <port> <transport> <fmt list>
>
>    A session description may contain a number of media descriptions.
>    Each media description starts with an "m=" field, and is terminated
>    by either the next "m=" field or by the end of the session
>    description. A media field has several sub-fields.
>
>    The first sub-field is the media type.  Currently defined media are
>    "audio", "video", "text", "application", "data" and "control", 
> though
> this
>    list may be extended in future.
>
> -----further down, same section -----------
>
>    The fourth and subsequent sub-fields are media formats.  For audio
>    text and video, these SHOULD reference a MIME sub-type describing 
> the
>    format under the "audio", "text" and "video" top-level MIME types.
>
> ------In Appendix A------------------------------
>
> ; sub-rules of 'm='
>       media =               token
>                             ;typically "audio", "video", "text",
> "application"
>                             ;or "data"
> ------End of proposal--

These will be in draft-ietf-mmusic-sdp-new-16.

-- 
Colin Perkins
http://csperkins.org/


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Subject: Re: [MMUSIC] STUN vs Symmetric RTP
From: Cullen Jennings <fluffy@cisco.com>
To: Gamze Seckin <gamze@vidiator.com>, <mmusic@ietf.org>
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STUN is shipping in several vendors SIP phones. STUN like things that are
not exactly STUN but work about the same way are are also done by many
vendors.

If you could only do one, I would do STUN before commedia but of course
doing all the stuff in ICE (draft-ietf-mmusic-ice) would probably work out
best. 

Cullen


On 11/11/03 10:01 AM, "Gamze Seckin" <gamze@vidiator.com> wrote:

> 
> Hi,
> 
> I checked the rtsp nat traversal draft (draft-ietf-mmusic-rtsp-nat-01.txt) but
> could not find a comprehensive comparison between STUN vs Symmetric RTP.
> 
> Is anybody actually using STUN?
> How about symmetric RTP?
> 
> Which one would you recommend if you had to actually deploy it ?
> Any pointers appreciated.
> 
> 
> Br
> 
> g
> 
> -----------------------------------------------------------
> Gamze Seckin, Ph.D.
> Director of Research
> Vidiator Technology (US) Inc.
> 411 108th Avenue NE. Suite 688
> Bellevue, WA 98004 USA.
> www.vidiator.com
> 
> Phone: 1.425.688.8811x2108
> Email: gamze@vidiator.com
> 
> This e-mail message (including any attachment) is intended only for the
> personal use of the recipient(s) named above. This message is confidential and
> may be legally privileged. If you are not an intended recipient, you may not
> review, copy or distribute this message. If you have received this
> communication in error, please notify us immediately by e-mail and delete the
> original message. Any views or opinions expressed in this message are those of
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> create any legally binding rights or obligations whatsoever, which may only be
> created by the exchange of hard copy documents signed by a duly authorized
> representative of Vidiator Technology (US) Inc.
> -----------------------------------------------------------
> 
> _______________________________________________
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> https://www1.ietf.org/mailman/listinfo/mmusic
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Cullen Jennings wrote:

> STUN is shipping in several vendors SIP phones. STUN like things that are
> not exactly STUN but work about the same way are are also done by many
> vendors.
> 
> If you could only do one, I would do STUN before commedia but of course
> doing all the stuff in ICE (draft-ietf-mmusic-ice) would probably work out
> best. 

To add to that, if you do ICE, the client to client tests that use 
STUN would get you equivalent functionality to symmetric RTP. The 
benefit is that ICE will allow your client to work in lots of 
different cases, not just the limited ones where symmetric RTP works. 
In particular, if both clients are behidn NAT, symmetric RTP doesnt help.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Subject: Re: [midcom] Re: [MMUSIC] Status of TURN
From: Cullen Jennings <fluffy@cisco.com>
To: Pete Cordell <pete@tech-know-ware.com>, Mmusic <mmusic@ietf.org>,
        Midcom <midcom@ietf.org>
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If people have specific technical issues about the security of TURN that are
not documented in the current draft - it would be really nice to know about
them. I always hear there are some but I never seem to hear what exactly
they are. I do understand Melinda's general concerns on if media relays
should be done at all no matter how they are done.

Thanks Cullen


On 10/21/03 10:14 PM, "Pete Cordell" <pete@tech-know-ware.com> wrote:

> What you say Christian is technically absolutely correct.
> 
> However this communities appetite for implementing and labelling their
> products with even -00 draft material suggests to me that many implementers
> (and probably their customers) are not bothered by these subtleties of IETF
> process.
> 
> Certainly it would help to know whether informational or proposed standard
> is
> being sought for this draft.
> 
> Pete.
> 
> =============================================
> Pete Cordell
> +44 1473 635863
> =============================================
> 
> ----- Original Message -----
> From: "Christian Huitema" <huitema@windows.microsoft.com>
> To: "Pete Cordell" <pete@tech-know-ware.com>; "Melinda Shore"
> <mshore@cisco.com>
> Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>; "Peterson, Jon"
> <jon.peterson@neustar.biz>; "Mmusic" <mmusic@ietf.org>; <midcom@ietf.org>
> Sent: 20 October 2003 20:27
> Subject: RE: [midcom] Re: [MMUSIC] Status of TURN
> 
> 
> Individual submissions are a well established part of the process, and
> should be considered a fine way to make progress when a working group
> would not. There is a catch, however. Individual contributions are very
> seldom published as proposed standards -- they normally end up as either
> experimental or informational. The criteria for publishing something as
> informational ought to be, demonstration of at least some significant
> interest in the community -- but not necessarily demonstration of
> consensus. The bar for refusing publication in the presence of enough
> interest has to be somewhat high -- basically, the group has to be
> convinced that implementing the proposal as is would damage the
> Internet.
> 
> Obviously, publication as a proposed standard would require the
> equivalent of WG consensus. I don't believe there is such consensus on
> TURN.
> 
> -- Christian Huitema
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 
> _______________________________________________
> midcom mailing list
> midcom@ietf.org
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From: "Peter Lewis" <peter.lewis@upperside.fr>
To: <mmusic@ietf.org>
Date: Tue, 18 Nov 2003 10:27:13 +0100
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Subject: [MMUSIC] 2004 WiFi Voice Conference
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The 2004 WiFi Voice Conference will stand in Paris on May 25 to 28, =
2004.=20
The call for papers dead line has been extended until December 15, 2003. =

Get more details at:
http://www.upperside.fr/wifivoice04/wifivoice04intro.htm

We also remind you that registration is still open to the New Mobile =
Technologies Forum (UWB, IPCN and IPv6 Mobile sessions) at:
http://www.upperside.fr/newmobtech/newmobtech03intro.htm



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2919.6307" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV>
<DIV><FONT size=3D2>
<DIV><FONT size=3D2>The <SPAN class=3Dtextebold><FONT =
color=3D#2a9085>2004 WiFi Voice=20
Conference</FONT></SPAN> will stand in Paris on <SPAN =
class=3Dtextebold>May 25 to=20
28, 2004</SPAN>.&nbsp;<BR>The call for papers dead line has been =
extended until=20
<SPAN class=3Dunderline>December 15, 2003.</SPAN> </FONT></DIV>
<DIV><FONT size=3D2>Get more details at:</FONT></DIV>
<DIV><FONT size=3D2><A=20
href=3D"http://www.upperside.fr/wifivoice04/wifivoice04intro.htm">http://=
www.upperside.fr/wifivoice04/wifivoice04intro.htm</A></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>We also remind you that registration is still open to the <FONT=20
color=3D#008080>New Mobile Technologies Forum </FONT>(UWB, IPCN and IPv6 =
Mobile=20
sessions) at:</DIV>
<DIV><A=20
href=3D"http://www.upperside.fr/newmobtech/newmobtech03intro.htm">http://=
www.upperside.fr/newmobtech/newmobtech03intro.htm</A></DIV></FONT></DIV><=
/DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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From exim@www1.ietf.org  Wed Nov 19 08:00:25 2003
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From: Stig Venaas <Stig.Venaas@uninett.no>
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Subject: [MMUSIC] some comments on IPV and ICE
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First of all, I haven't studied this topic much, so bare with me for
any misunderstandings.

I have some doubts regarding IPV (draft-camarillo-mmusic-alt-02.txt).
As I understand it, IPV would be useful to ipv4-only or ipv6-only
hosts, but has significant drawbacks for dual-stack hosts not
supporting it. I believe that most IPv6 hosts today, are dual-stack
and will continue to be so for some time. An exceptiom might be some
new environments like 3G networks. I think it's better to add special
requirements for the special cases, that is, require IPv6-only hosts
to support a new solution.

I think it's important to be able to express two different things in
SDP. One is a number of addresses (possibly addresses of different
address families) where the client may pick any one, the other is a
number of addresses (again possibly different families) where the
client is supposed to use all.

Unless I've misunderstood something, the latter is fine, and the
former can be done with ICE, except that there is a preferred
address. That there's a preferred address isn't a problem I think.

If indeed the alt attribute is useful in general, isn't it better
to define this separately from ICE, and have ICE refer to that?

Stig

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [MMUSIC] some comments on IPV and ICE
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Hi,

> If indeed the alt attribute is useful in general, isn't it better
> to define this separately from ICE, and have ICE refer to that?

This is exactly what the draft does right now. You may be referring to 
an old version of that draft.

Regarding your comment about dual-stack UAs, I do not see any problem 
related to the use of IPV.

Thanks,

Gonzalo


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Hi,

I've sent "Strategies for Media Applications Using DCCP" to the
Internet-Drafts archive.  While we're waiting for it to be added, it's also
available at http://www.phelan-4.com/dccp/draft-phelan-dccp-media-00.txt.
The plan is that this will soon evolve into a general DCCP User Guide.

Abstract:
   This document discusses strategies for using DCCP as the transport
   protocol for streaming media applications.  Of particular interest is
   how media streams can be adapted to the smoothly varying transmit
   rate requirements of CCID3, or TCP-Friendly Rate Control (TFRC).
   Also explored is the resulting network behavior of streams using
   these strategies and the fairness between these streams and TCP-based
   streams.

Comments will be appreciated.  In particular, I make the suggestion that
media apps should transmit at up to twice their nominal maximum rate, when
DCCP allows it, in order to provide protection from competing TCP
applications.  I know that the first time I heard someone make a similar
suggestion I was horrified, but as I developed the issues it seemed to be
necessary.  Reactions to this will be appreciated.

Tom P.

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/mmusic



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Date: Thu, 20 Nov 2003 02:31:31 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Tom,

I don't think this solution will be acceptable to many of the network 
providers of interactive voip services. Many of them are insistent on 
using low rate codecs, like g.723, in order to reduce the costs of 
capacity in their networks. Doubling the rate would defeat the purpose 
of using these low rate codecs in the first place.

-Jonathan R.

Phelan, Tom wrote:


> Comments will be appreciated.  In particular, I make the suggestion that
> media apps should transmit at up to twice their nominal maximum rate, when
> DCCP allows it, in order to provide protection from competing TCP
> applications.  I know that the first time I heard someone make a similar
> suggestion I was horrified, but as I developed the issues it seemed to be
> necessary.  Reactions to this will be appreciated.
> 
> Tom P.
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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Subject: RE: [MMUSIC] DCCP Media Guide
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Hi Jonathan,

Thanks for the comments.  The actual proposal is to send up to double the
nominal transmit rate when there's available bandwidth, and to reduce the
rate as necessary if there isn't enough bandwidth.  This doubling is to
allow fair sharing of the chokepoint bandwidth (between DCCP media apps and
TCP apps), and to avoid what appears to me to be the potential for bullying
from TCP applications that want to suck up all the bandwidth they can.  It
doesn't preclude using low-rate codecs, or mean that the access link must
support the double rate for the connection to function well.

This recommendation was meant for best-effort environments, where the voice
traffic receives no special treatment, and must compete with TCP apps for
network resources.  I can see how this would require a network that gave
special treatment to voice packets to provision double the capacity, though.

This fairness problem is difficult.  All of the research I'm familiar with
approaches it from the direction of other apps being fair to TCP.  I'm not
aware of any other methods for ensuring that non-TCP apps get their fair
share when competing with TCP.  If you (or anyone else) knows of something
I'd appreciate the pointer.

Tom P.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, November 20, 2003 2:32 AM
> To: Phelan, Tom
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: Re: [MMUSIC] DCCP Media Guide
> 
> 
> Tom,
> 
> I don't think this solution will be acceptable to many of the network 
> providers of interactive voip services. Many of them are insistent on 
> using low rate codecs, like g.723, in order to reduce the costs of 
> capacity in their networks. Doubling the rate would defeat 
> the purpose 
> of using these low rate codecs in the first place.
> 
> -Jonathan R.
> 
> Phelan, Tom wrote:
> 
> 
> > Comments will be appreciated.  In particular, I make the 
> suggestion that
> > media apps should transmit at up to twice their nominal 
> maximum rate, when
> > DCCP allows it, in order to provide protection from competing TCP
> > applications.  I know that the first time I heard someone 
> make a similar
> > suggestion I was horrified, but as I developed the issues 
> it seemed to be
> > necessary.  Reactions to this will be appreciated.
> > 
> > Tom P.
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>,
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Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 11:04:49 -0500
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To me, this is just nonsense.
I can't imagine anyone deploying such a solution.

This is forcing the problem to fit the solution, and it's not
the right way to go.  If you want it to work, the protocol
has to take the characteristics of the flows as they are, and not
as you would like them to be.  Real time media, streaming or realtime,
is either constant bit rate, or pretty close to it.  That's how it 
works.  Our challenge is to make the protocols work with this data,
not change the data, especially silly ideas like doubling the bandwidth.

We make video telephones.  They nominally send 4 megabits.  Would
you like us to double that, under any circumstances?  

I thought not. 

Brian

> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Thursday, November 20, 2003 10:36 AM
> To: 'Jonathan Rosenberg'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Hi Jonathan,
> 
> Thanks for the comments.  The actual proposal is to send up 
> to double the
> nominal transmit rate when there's available bandwidth, and 
> to reduce the
> rate as necessary if there isn't enough bandwidth.  This 
> doubling is to
> allow fair sharing of the chokepoint bandwidth (between DCCP 
> media apps and
> TCP apps), and to avoid what appears to me to be the 
> potential for bullying
> from TCP applications that want to suck up all the bandwidth 
> they can.  It
> doesn't preclude using low-rate codecs, or mean that the 
> access link must
> support the double rate for the connection to function well.
> 
> This recommendation was meant for best-effort environments, 
> where the voice
> traffic receives no special treatment, and must compete with 
> TCP apps for
> network resources.  I can see how this would require a 
> network that gave
> special treatment to voice packets to provision double the 
> capacity, though.
> 
> This fairness problem is difficult.  All of the research I'm 
> familiar with
> approaches it from the direction of other apps being fair to 
> TCP.  I'm not
> aware of any other methods for ensuring that non-TCP apps get 
> their fair
> share when competing with TCP.  If you (or anyone else) knows 
> of something
> I'd appreciate the pointer.
> 
> Tom P.
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Thursday, November 20, 2003 2:32 AM
> > To: Phelan, Tom
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: Re: [MMUSIC] DCCP Media Guide
> > 
> > 
> > Tom,
> > 
> > I don't think this solution will be acceptable to many of 
> the network 
> > providers of interactive voip services. Many of them are 
> insistent on 
> > using low rate codecs, like g.723, in order to reduce the costs of 
> > capacity in their networks. Doubling the rate would defeat 
> > the purpose 
> > of using these low rate codecs in the first place.
> > 
> > -Jonathan R.
> > 
> > Phelan, Tom wrote:
> > 
> > 
> > > Comments will be appreciated.  In particular, I make the 
> > suggestion that
> > > media apps should transmit at up to twice their nominal 
> > maximum rate, when
> > > DCCP allows it, in order to provide protection from competing TCP
> > > applications.  I know that the first time I heard someone 
> > make a similar
> > > suggestion I was horrified, but as I developed the issues 
> > it seemed to be
> > > necessary.  Reactions to this will be appreciated.
> > > 
> > > Tom P.
> > > 
> > > _______________________________________________
> > > mmusic mailing list
> > > mmusic@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > 
> > 
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > Chief Technology Officer                    Parsippany, NJ 
> 07054-2711
> > dynamicsoft
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 11:47:47 -0500
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Hi Brian,

Thanks for the comments.  See inline ...

Tom P.

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Thursday, November 20, 2003 11:05 AM
> To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> To me, this is just nonsense.
> I can't imagine anyone deploying such a solution.
> 
> This is forcing the problem to fit the solution, and it's not
> the right way to go.  If you want it to work, the protocol
> has to take the characteristics of the flows as they are, and not
> as you would like them to be.  Real time media, streaming or realtime,
> is either constant bit rate, or pretty close to it.  That's how it 
> works.  Our challenge is to make the protocols work with this data,
> not change the data, especially silly ideas like doubling the 
> bandwidth.
> 
> We make video telephones.  They nominally send 4 megabits.  Would
> you like us to double that, under any circumstances?  
> 
> I thought not. 

Would you prefer to be forced unnecessarily into packet loss and a lower
encoding rate by greedy TCP applications?  Or do you have another solution?

I don't think I'm forcing the problem to fit the solution, just perhaps that
the problem I'm trying to solve is different than you perceive.  The problem
that I'm trying to solve is how can an application that is content with a
maximum bandwidth compete with TCP applications that are insatiably greedy,
in a way that satisfies some concept of fairness in all directions?

The solution that I see is to provide a buffer against TCP probing.  This
would involve sending more data than is nominally necessary.  If the network
has capacity for more than 4M bps, then why shouldn't a media app make use
of it?  A TCP app would.

I must admit that my initial reactions to suggestions like this were very
similar to yours.  I encourage you to look at the draft for more detail on
the issues that convinced me otherwise.

> 
> Brian
> 
> > -----Original Message-----
> > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > Sent: Thursday, November 20, 2003 10:36 AM
> > To: 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > 
> > 
> > Hi Jonathan,
> > 
> > Thanks for the comments.  The actual proposal is to send up 
> > to double the
> > nominal transmit rate when there's available bandwidth, and 
> > to reduce the
> > rate as necessary if there isn't enough bandwidth.  This 
> > doubling is to
> > allow fair sharing of the chokepoint bandwidth (between DCCP 
> > media apps and
> > TCP apps), and to avoid what appears to me to be the 
> > potential for bullying
> > from TCP applications that want to suck up all the bandwidth 
> > they can.  It
> > doesn't preclude using low-rate codecs, or mean that the 
> > access link must
> > support the double rate for the connection to function well.
> > 
> > This recommendation was meant for best-effort environments, 
> > where the voice
> > traffic receives no special treatment, and must compete with 
> > TCP apps for
> > network resources.  I can see how this would require a 
> > network that gave
> > special treatment to voice packets to provision double the 
> > capacity, though.
> > 
> > This fairness problem is difficult.  All of the research I'm 
> > familiar with
> > approaches it from the direction of other apps being fair to 
> > TCP.  I'm not
> > aware of any other methods for ensuring that non-TCP apps get 
> > their fair
> > share when competing with TCP.  If you (or anyone else) knows 
> > of something
> > I'd appreciate the pointer.
> > 
> > Tom P.
> > 
> > > -----Original Message-----
> > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > Sent: Thursday, November 20, 2003 2:32 AM
> > > To: Phelan, Tom
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > 
> > > 
> > > Tom,
> > > 
> > > I don't think this solution will be acceptable to many of 
> > the network 
> > > providers of interactive voip services. Many of them are 
> > insistent on 
> > > using low rate codecs, like g.723, in order to reduce the 
> costs of 
> > > capacity in their networks. Doubling the rate would defeat 
> > > the purpose 
> > > of using these low rate codecs in the first place.
> > > 
> > > -Jonathan R.
> > > 
> > > Phelan, Tom wrote:
> > > 
> > > 
> > > > Comments will be appreciated.  In particular, I make the 
> > > suggestion that
> > > > media apps should transmit at up to twice their nominal 
> > > maximum rate, when
> > > > DCCP allows it, in order to provide protection from 
> competing TCP
> > > > applications.  I know that the first time I heard someone 
> > > make a similar
> > > > suggestion I was horrified, but as I developed the issues 
> > > it seemed to be
> > > > necessary.  Reactions to this will be appreciated.
> > > > 
> > > > Tom P.
> > > > 
> > > > _______________________________________________
> > > > mmusic mailing list
> > > > mmusic@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > 
> > > 
> > > -- 
> > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > Chief Technology Officer                    Parsippany, NJ 
> > 07054-2711
> > > dynamicsoft
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> > > 
> > 
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> > 
> 

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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From: philippe.gentric@philips.com
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Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
To: "Phelan, Tom" <tphelan@sonusnet.com>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>
Cc: "'avt@ietf.org'" <avt@ietf.org>, "'dccp@ietf.org'" <dccp@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
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Brian, all,

It may provide some help in understanding the situation
to describe your exemple as follows:

* you send video at 4 Mb/s, good.

* it happens that you are competing with (a lot of ) TCP

AND that you want to be fair to it:

* so when that happens you decrease your rate to 2 Mb/s for "a while"
because otherwise you are not fair.

* but at some moment you will want to come back up to 4 Mb/s right ?

=> the point is that the **only way**  to know if you "can" do that, is to try it....

regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   "'Rosen, Brian'" <Brian.Rosen@marconi.com>                                    
                                                   cc:   "'dccp@ietf.org'" <dccp@ietf.org>                                             
                                                    "'avt@ietf.org'" <avt@ietf.org>                                                    
                                                    "'mmusic@ietf.org'" <mmusic@ietf.org>                                              
               "Phelan, Tom"                        (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
               <tphelan@sonusnet.com>              Subject:    RE: [AVT] RE: [MMUSIC] DCCP Media Guide                                 
                                                                                                                                       
               Sent by:                            Classification:                                                                     
               mmusic-admin@ietf.org                                                                                                   
                                                                                                                                       
               20/11/2003 17:47                                                                                                        
                                                                                                                                       
                                                                                                                                       




Hi Brian,

Thanks for the comments.  See inline ...

Tom P.

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Thursday, November 20, 2003 11:05 AM
> To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
>
>
> To me, this is just nonsense.
> I can't imagine anyone deploying such a solution.
>
> This is forcing the problem to fit the solution, and it's not
> the right way to go.  If you want it to work, the protocol
> has to take the characteristics of the flows as they are, and not
> as you would like them to be.  Real time media, streaming or realtime,
> is either constant bit rate, or pretty close to it.  That's how it
> works.  Our challenge is to make the protocols work with this data,
> not change the data, especially silly ideas like doubling the
> bandwidth.
>
> We make video telephones.  They nominally send 4 megabits.  Would
> you like us to double that, under any circumstances?
>
> I thought not.

Would you prefer to be forced unnecessarily into packet loss and a lower
encoding rate by greedy TCP applications?  Or do you have another solution?

I don't think I'm forcing the problem to fit the solution, just perhaps that
the problem I'm trying to solve is different than you perceive.  The problem
that I'm trying to solve is how can an application that is content with a
maximum bandwidth compete with TCP applications that are insatiably greedy,
in a way that satisfies some concept of fairness in all directions?

The solution that I see is to provide a buffer against TCP probing.  This
would involve sending more data than is nominally necessary.  If the network
has capacity for more than 4M bps, then why shouldn't a media app make use
of it?  A TCP app would.

I must admit that my initial reactions to suggestions like this were very
similar to yours.  I encourage you to look at the draft for more detail on
the issues that convinced me otherwise.

>
> Brian
>
> > -----Original Message-----
> > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > Sent: Thursday, November 20, 2003 10:36 AM
> > To: 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > Hi Jonathan,
> >
> > Thanks for the comments.  The actual proposal is to send up
> > to double the
> > nominal transmit rate when there's available bandwidth, and
> > to reduce the
> > rate as necessary if there isn't enough bandwidth.  This
> > doubling is to
> > allow fair sharing of the chokepoint bandwidth (between DCCP
> > media apps and
> > TCP apps), and to avoid what appears to me to be the
> > potential for bullying
> > from TCP applications that want to suck up all the bandwidth
> > they can.  It
> > doesn't preclude using low-rate codecs, or mean that the
> > access link must
> > support the double rate for the connection to function well.
> >
> > This recommendation was meant for best-effort environments,
> > where the voice
> > traffic receives no special treatment, and must compete with
> > TCP apps for
> > network resources.  I can see how this would require a
> > network that gave
> > special treatment to voice packets to provision double the
> > capacity, though.
> >
> > This fairness problem is difficult.  All of the research I'm
> > familiar with
> > approaches it from the direction of other apps being fair to
> > TCP.  I'm not
> > aware of any other methods for ensuring that non-TCP apps get
> > their fair
> > share when competing with TCP.  If you (or anyone else) knows
> > of something
> > I'd appreciate the pointer.
> >
> > Tom P.
> >
> > > -----Original Message-----
> > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > Sent: Thursday, November 20, 2003 2:32 AM
> > > To: Phelan, Tom
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: Re: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > Tom,
> > >
> > > I don't think this solution will be acceptable to many of
> > the network
> > > providers of interactive voip services. Many of them are
> > insistent on
> > > using low rate codecs, like g.723, in order to reduce the
> costs of
> > > capacity in their networks. Doubling the rate would defeat
> > > the purpose
> > > of using these low rate codecs in the first place.
> > >
> > > -Jonathan R.
> > >
> > > Phelan, Tom wrote:
> > >
> > >
> > > > Comments will be appreciated.  In particular, I make the
> > > suggestion that
> > > > media apps should transmit at up to twice their nominal
> > > maximum rate, when
> > > > DCCP allows it, in order to provide protection from
> competing TCP
> > > > applications.  I know that the first time I heard someone
> > > make a similar
> > > > suggestion I was horrified, but as I developed the issues
> > > it seemed to be
> > > > necessary.  Reactions to this will be appreciated.
> > > >
> > > > Tom P.
> > > >
> > > > _______________________________________________
> > > > mmusic mailing list
> > > > mmusic@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > >
> > >
> > > --
> > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > Chief Technology Officer                    Parsippany, NJ
> > 07054-2711
> > > dynamicsoft
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> > >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
>

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Date: Thu, 20 Nov 2003 12:34:57 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [MMUSIC] DCCP Media Guide
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Phelan, Tom wrote:

> Hi Jonathan,
> 
> Thanks for the comments.  The actual proposal is to send up to double the
> nominal transmit rate when there's available bandwidth, and to reduce the
> rate as necessary if there isn't enough bandwidth.  This doubling is to
> allow fair sharing of the chokepoint bandwidth (between DCCP media apps and
> TCP apps), and to avoid what appears to me to be the potential for bullying
> from TCP applications that want to suck up all the bandwidth they can.  It
> doesn't preclude using low-rate codecs, or mean that the access link must
> support the double rate for the connection to function well.
> 
> This recommendation was meant for best-effort environments, where the voice
> traffic receives no special treatment, and must compete with TCP apps for
> network resources.  I can see how this would require a network that gave
> special treatment to voice packets to provision double the capacity, though.

What about networks where there is also a lot of UDP traffic? For 
example, these same networks providing voip traffic today. If I 
incrementally add dccp support, it seems like I will need to increase 
the provisioned capacity so I dont push out the UDP, no?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>,
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Subject: RE: [MMUSIC] DCCP Media Guide
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Hi Jonathan,

See inline ...

Tom P.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, November 20, 2003 12:35 PM
> To: Phelan, Tom
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: Re: [MMUSIC] DCCP Media Guide
> 
> 
> 
> 
> Phelan, Tom wrote:
> 
> > Hi Jonathan,
> > 
> > Thanks for the comments.  The actual proposal is to send up 
> to double the
> > nominal transmit rate when there's available bandwidth, and 
> to reduce the
> > rate as necessary if there isn't enough bandwidth.  This 
> doubling is to
> > allow fair sharing of the chokepoint bandwidth (between 
> DCCP media apps and
> > TCP apps), and to avoid what appears to me to be the 
> potential for bullying
> > from TCP applications that want to suck up all the 
> bandwidth they can.  It
> > doesn't preclude using low-rate codecs, or mean that the 
> access link must
> > support the double rate for the connection to function well.
> > 
> > This recommendation was meant for best-effort environments, 
> where the voice
> > traffic receives no special treatment, and must compete 
> with TCP apps for
> > network resources.  I can see how this would require a 
> network that gave
> > special treatment to voice packets to provision double the 
> capacity, though.
> 
> What about networks where there is also a lot of UDP traffic? For 
> example, these same networks providing voip traffic today. If I 
> incrementally add dccp support, it seems like I will need to increase 
> the provisioned capacity so I dont push out the UDP, no?

Yes, DCCP will behave towards UDP (and vice versa) pretty much the same as
TCP behaves toward UDP.  For streaming media apps following the guidelines
in the DCCP Media Guide, it actually can be a little better than TCP versus
UDP, because there is a maximum transmit rate.

> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 

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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>,
        "Phelan, Tom" <tphelan@sonusnet.com>,
        "'Rosen, Brian'"
	 <Brian.Rosen@marconi.com>
Cc: "'avt@ietf.org'" <avt@ietf.org>, "'dccp@ietf.org'" <dccp@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 13:04:29 -0500
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Hi Phillipe,

The situation I'm worried about is a little different than you describe.
Say you have a 10M bps link, over which there's a 4M bps video stream and a
long-lived TCP app.  The TCP app will take the remaining 6M bps, and then
push for more, which might cause the video stream to lose packets.  The
video stream then drops to 2M bps, and the TCP app expands to 8M bps.  This
hardly seems fair.

The video app can protect itself from this by transmitting enough more than
4M bps to buffer itself against the TCP probes.  In this case we'd come to a
relatively steady state where both apps are sending roughly 5M bps.  This
seems reasonably fair to me, since the TCP app gets what it would get if it
were competing with other TCP apps, and the video app gets to send at its
full rate.

Tom P.

> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, November 20, 2003 12:23 PM
> To: Phelan, Tom; 'Rosen, Brian'
> Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Brian, all,
> 
> It may provide some help in understanding the situation
> to describe your exemple as follows:
> 
> * you send video at 4 Mb/s, good.
> 
> * it happens that you are competing with (a lot of ) TCP
> 
> AND that you want to be fair to it:
> 
> * so when that happens you decrease your rate to 2 Mb/s for "a while"
> because otherwise you are not fair.
> 
> * but at some moment you will want to come back up to 4 Mb/s right ?
> 
> => the point is that the **only way**  to know if you "can" 
> do that, is to try it....
> 
> regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                    To:   
> "'Rosen, Brian'" <Brian.Rosen@marconi.com>                    
>                 
>                                                    cc:   
> "'dccp@ietf.org'" <dccp@ietf.org>                             
>                 
>                                                     
> "'avt@ietf.org'" <avt@ietf.org>                               
>                      
>                                                     
> "'mmusic@ietf.org'" <mmusic@ietf.org>                         
>                      
>                "Phelan, Tom"                        (bcc: 
> Philippe Gentric/MP4-SUR/CE/PHILIPS)                          
>                
>                <tphelan@sonusnet.com>              Subject:   
>  RE: [AVT] RE: [MMUSIC] DCCP Media Guide                      
>            
>                                                               
>                                                               
>            
>                Sent by:                            
> Classification:                                               
>                       
>                mmusic-admin@ietf.org                          
>                                                               
>            
>                                                               
>                                                               
>            
>                20/11/2003 17:47                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
> 
> 
> 
> 
> Hi Brian,
> 
> Thanks for the comments.  See inline ...
> 
> Tom P.
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: Thursday, November 20, 2003 11:05 AM
> > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > To me, this is just nonsense.
> > I can't imagine anyone deploying such a solution.
> >
> > This is forcing the problem to fit the solution, and it's not
> > the right way to go.  If you want it to work, the protocol
> > has to take the characteristics of the flows as they are, and not
> > as you would like them to be.  Real time media, streaming 
> or realtime,
> > is either constant bit rate, or pretty close to it.  That's how it
> > works.  Our challenge is to make the protocols work with this data,
> > not change the data, especially silly ideas like doubling the
> > bandwidth.
> >
> > We make video telephones.  They nominally send 4 megabits.  Would
> > you like us to double that, under any circumstances?
> >
> > I thought not.
> 
> Would you prefer to be forced unnecessarily into packet loss 
> and a lower
> encoding rate by greedy TCP applications?  Or do you have 
> another solution?
> 
> I don't think I'm forcing the problem to fit the solution, 
> just perhaps that
> the problem I'm trying to solve is different than you 
> perceive.  The problem
> that I'm trying to solve is how can an application that is 
> content with a
> maximum bandwidth compete with TCP applications that are 
> insatiably greedy,
> in a way that satisfies some concept of fairness in all directions?
> 
> The solution that I see is to provide a buffer against TCP 
> probing.  This
> would involve sending more data than is nominally necessary.  
> If the network
> has capacity for more than 4M bps, then why shouldn't a media 
> app make use
> of it?  A TCP app would.
> 
> I must admit that my initial reactions to suggestions like 
> this were very
> similar to yours.  I encourage you to look at the draft for 
> more detail on
> the issues that convinced me otherwise.
> 
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > Sent: Thursday, November 20, 2003 10:36 AM
> > > To: 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > Hi Jonathan,
> > >
> > > Thanks for the comments.  The actual proposal is to send up
> > > to double the
> > > nominal transmit rate when there's available bandwidth, and
> > > to reduce the
> > > rate as necessary if there isn't enough bandwidth.  This
> > > doubling is to
> > > allow fair sharing of the chokepoint bandwidth (between DCCP
> > > media apps and
> > > TCP apps), and to avoid what appears to me to be the
> > > potential for bullying
> > > from TCP applications that want to suck up all the bandwidth
> > > they can.  It
> > > doesn't preclude using low-rate codecs, or mean that the
> > > access link must
> > > support the double rate for the connection to function well.
> > >
> > > This recommendation was meant for best-effort environments,
> > > where the voice
> > > traffic receives no special treatment, and must compete with
> > > TCP apps for
> > > network resources.  I can see how this would require a
> > > network that gave
> > > special treatment to voice packets to provision double the
> > > capacity, though.
> > >
> > > This fairness problem is difficult.  All of the research I'm
> > > familiar with
> > > approaches it from the direction of other apps being fair to
> > > TCP.  I'm not
> > > aware of any other methods for ensuring that non-TCP apps get
> > > their fair
> > > share when competing with TCP.  If you (or anyone else) knows
> > > of something
> > > I'd appreciate the pointer.
> > >
> > > Tom P.
> > >
> > > > -----Original Message-----
> > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > To: Phelan, Tom
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > >
> > > >
> > > > Tom,
> > > >
> > > > I don't think this solution will be acceptable to many of
> > > the network
> > > > providers of interactive voip services. Many of them are
> > > insistent on
> > > > using low rate codecs, like g.723, in order to reduce the
> > costs of
> > > > capacity in their networks. Doubling the rate would defeat
> > > > the purpose
> > > > of using these low rate codecs in the first place.
> > > >
> > > > -Jonathan R.
> > > >
> > > > Phelan, Tom wrote:
> > > >
> > > >
> > > > > Comments will be appreciated.  In particular, I make the
> > > > suggestion that
> > > > > media apps should transmit at up to twice their nominal
> > > > maximum rate, when
> > > > > DCCP allows it, in order to provide protection from
> > competing TCP
> > > > > applications.  I know that the first time I heard someone
> > > > make a similar
> > > > > suggestion I was horrified, but as I developed the issues
> > > > it seemed to be
> > > > > necessary.  Reactions to this will be appreciated.
> > > > >
> > > > > Tom P.
> > > > >
> > > > > _______________________________________________
> > > > > mmusic mailing list
> > > > > mmusic@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > >
> > > >
> > > > --
> > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > Chief Technology Officer                    Parsippany, NJ
> > > 07054-2711
> > > > dynamicsoft
> > > > jdrosen@dynamicsoft.com                     FAX:   
> (973) 952-5050
> > > > http://www.jdrosen.net                      PHONE: 
> (973) 952-5000
> > > > http://www.dynamicsoft.com
> > > >
> > >
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> > >
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 
> 
> 
> 

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Hi,
=20
I tried to get the following file from internet and using
mailserv@etf.org.
=20
draft-ietf-mmusic-confarch-03.txt
=20
Can any one send me this file?
 =20
Thanks 'n Rgds
Vandana
=20

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I tried to get the following file from internet and =
using <a
href=3D"mailto:mailserv@etf.org">mailserv@etf.org</a>.<o:p></o:p></span><=
/font></p>

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3.txt</span></font></span><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p=
>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Can any one send me this =
file?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;mso-no-proof:yes'>&nbsp; <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;mso-no-proof:yes'>Thanks 'n =
Rgds<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;mso-no-proof:yes'>Vandana<o:p></o:p></span></font></p>

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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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        avt@ietf.org, dccp@ietf.org, mmusic@ietf.org,
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Subject: Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
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Hi, Tom,

It seems to me that you are basically: 1) questioning the fairness of TCP itself
and, 2) you want to do something in the app level to outmaneuver TCP so as to
achieve your term of fairness.

I am not convinced this is the right thing to do with DCCP.

I remember similar issue was discussed on tsvwg list (I am cc'ing it). Maybe
people in tsvwg can share their thoughts on this as well.

regards,
-Qiaobing

"Phelan, Tom" wrote:

> Hi Phillipe,
>
> The situation I'm worried about is a little different than you describe.
> Say you have a 10M bps link, over which there's a 4M bps video stream and a
> long-lived TCP app.  The TCP app will take the remaining 6M bps, and then
> push for more, which might cause the video stream to lose packets.  The
> video stream then drops to 2M bps, and the TCP app expands to 8M bps.  This
> hardly seems fair.
>
> The video app can protect itself from this by transmitting enough more than
> 4M bps to buffer itself against the TCP probes.  In this case we'd come to a
> relatively steady state where both apps are sending roughly 5M bps.  This
> seems reasonably fair to me, since the TCP app gets what it would get if it
> were competing with other TCP apps, and the video app gets to send at its
> full rate.
>
> Tom P.
>
> > -----Original Message-----
> > From: philippe.gentric@philips.com
> > [mailto:philippe.gentric@philips.com]
> > Sent: Thursday, November 20, 2003 12:23 PM
> > To: Phelan, Tom; 'Rosen, Brian'
> > Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > Brian, all,
> >
> > It may provide some help in understanding the situation
> > to describe your exemple as follows:
> >
> > * you send video at 4 Mb/s, good.
> >
> > * it happens that you are competing with (a lot of ) TCP
> >
> > AND that you want to be fair to it:
> >
> > * so when that happens you decrease your rate to 2 Mb/s for "a while"
> > because otherwise you are not fair.
> >
> > * but at some moment you will want to come back up to 4 Mb/s right ?
> >
> > => the point is that the **only way**  to know if you "can"
> > do that, is to try it....
> >
> > regards,
> >
> >
> > Philippe Gentric
> > Chief Architect
> > Philips Software
> > philippe.gentric@philips.com
> > http://www.software.philips.com
> >
> >
> >
> >
> >
> >
> >
> >
> >                                                    To:
> > "'Rosen, Brian'" <Brian.Rosen@marconi.com>
> >
> >                                                    cc:
> > "'dccp@ietf.org'" <dccp@ietf.org>
> >
> >
> > "'avt@ietf.org'" <avt@ietf.org>
> >
> >
> > "'mmusic@ietf.org'" <mmusic@ietf.org>
> >
> >                "Phelan, Tom"                        (bcc:
> > Philippe Gentric/MP4-SUR/CE/PHILIPS)
> >
> >                <tphelan@sonusnet.com>              Subject:
> >  RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> >
> >
> >                Sent by:
> > Classification:
> >
> >                mmusic-admin@ietf.org
> >
> >
> >
> >
> >
> >                20/11/2003 17:47
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Hi Brian,
> >
> > Thanks for the comments.  See inline ...
> >
> > Tom P.
> >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > Sent: Thursday, November 20, 2003 11:05 AM
> > > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > To me, this is just nonsense.
> > > I can't imagine anyone deploying such a solution.
> > >
> > > This is forcing the problem to fit the solution, and it's not
> > > the right way to go.  If you want it to work, the protocol
> > > has to take the characteristics of the flows as they are, and not
> > > as you would like them to be.  Real time media, streaming
> > or realtime,
> > > is either constant bit rate, or pretty close to it.  That's how it
> > > works.  Our challenge is to make the protocols work with this data,
> > > not change the data, especially silly ideas like doubling the
> > > bandwidth.
> > >
> > > We make video telephones.  They nominally send 4 megabits.  Would
> > > you like us to double that, under any circumstances?
> > >
> > > I thought not.
> >
> > Would you prefer to be forced unnecessarily into packet loss
> > and a lower
> > encoding rate by greedy TCP applications?  Or do you have
> > another solution?
> >
> > I don't think I'm forcing the problem to fit the solution,
> > just perhaps that
> > the problem I'm trying to solve is different than you
> > perceive.  The problem
> > that I'm trying to solve is how can an application that is
> > content with a
> > maximum bandwidth compete with TCP applications that are
> > insatiably greedy,
> > in a way that satisfies some concept of fairness in all directions?
> >
> > The solution that I see is to provide a buffer against TCP
> > probing.  This
> > would involve sending more data than is nominally necessary.
> > If the network
> > has capacity for more than 4M bps, then why shouldn't a media
> > app make use
> > of it?  A TCP app would.
> >
> > I must admit that my initial reactions to suggestions like
> > this were very
> > similar to yours.  I encourage you to look at the draft for
> > more detail on
> > the issues that convinced me otherwise.
> >
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > > Sent: Thursday, November 20, 2003 10:36 AM
> > > > To: 'Jonathan Rosenberg'
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > >
> > > >
> > > > Hi Jonathan,
> > > >
> > > > Thanks for the comments.  The actual proposal is to send up
> > > > to double the
> > > > nominal transmit rate when there's available bandwidth, and
> > > > to reduce the
> > > > rate as necessary if there isn't enough bandwidth.  This
> > > > doubling is to
> > > > allow fair sharing of the chokepoint bandwidth (between DCCP
> > > > media apps and
> > > > TCP apps), and to avoid what appears to me to be the
> > > > potential for bullying
> > > > from TCP applications that want to suck up all the bandwidth
> > > > they can.  It
> > > > doesn't preclude using low-rate codecs, or mean that the
> > > > access link must
> > > > support the double rate for the connection to function well.
> > > >
> > > > This recommendation was meant for best-effort environments,
> > > > where the voice
> > > > traffic receives no special treatment, and must compete with
> > > > TCP apps for
> > > > network resources.  I can see how this would require a
> > > > network that gave
> > > > special treatment to voice packets to provision double the
> > > > capacity, though.
> > > >
> > > > This fairness problem is difficult.  All of the research I'm
> > > > familiar with
> > > > approaches it from the direction of other apps being fair to
> > > > TCP.  I'm not
> > > > aware of any other methods for ensuring that non-TCP apps get
> > > > their fair
> > > > share when competing with TCP.  If you (or anyone else) knows
> > > > of something
> > > > I'd appreciate the pointer.
> > > >
> > > > Tom P.
> > > >
> > > > > -----Original Message-----
> > > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > > To: Phelan, Tom
> > > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > > >
> > > > >
> > > > > Tom,
> > > > >
> > > > > I don't think this solution will be acceptable to many of
> > > > the network
> > > > > providers of interactive voip services. Many of them are
> > > > insistent on
> > > > > using low rate codecs, like g.723, in order to reduce the
> > > costs of
> > > > > capacity in their networks. Doubling the rate would defeat
> > > > > the purpose
> > > > > of using these low rate codecs in the first place.
> > > > >
> > > > > -Jonathan R.
> > > > >
> > > > > Phelan, Tom wrote:
> > > > >
> > > > >
> > > > > > Comments will be appreciated.  In particular, I make the
> > > > > suggestion that
> > > > > > media apps should transmit at up to twice their nominal
> > > > > maximum rate, when
> > > > > > DCCP allows it, in order to provide protection from
> > > competing TCP
> > > > > > applications.  I know that the first time I heard someone
> > > > > make a similar
> > > > > > suggestion I was horrified, but as I developed the issues
> > > > > it seemed to be
> > > > > > necessary.  Reactions to this will be appreciated.
> > > > > >
> > > > > > Tom P.
> > > > > >
> > > > > > _______________________________________________
> > > > > > mmusic mailing list
> > > > > > mmusic@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > > >
> > > > >
> > > > > --
> > > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > > Chief Technology Officer                    Parsippany, NJ
> > > > 07054-2711
> > > > > dynamicsoft
> > > > > jdrosen@dynamicsoft.com                     FAX:
> > (973) 952-5050
> > > > > http://www.jdrosen.net                      PHONE:
> > (973) 952-5000
> > > > > http://www.dynamicsoft.com
> > > > >
> > > >
> > > > _______________________________________________
> > > > Audio/Video Transport Working Group
> > > > avt@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/avt
> > > >
> > >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
> >
> >
> >
> >
>
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html


_______________________________________________
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Caitlin Bestler'" <cait@asomi.com>,
        Qiaobing Xie
	 <Qiaobing.Xie@motorola.com>
Cc: mmusic@ietf.org, TSV WG list <tsvwg@ietf.org>,
        "Phelan, Tom"
	 <tphelan@sonusnet.com>, avt@ietf.org, dccp@ietf.org,
        philippe.gentric@philips.com
Subject: RE: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 17:14:23 -0500
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The reason to "avoid" TCP friendly rate control is, at least so far,
because there is no way yet proposed to do it without destroying the 
usability of the RTP flow.

Part of the problem is that the only "rate control" that works is to
negotiate an acceptable rate in the signalling before the stream starts.
There really isn't any other way to change the rate, at least for
interactive streams.

For non-interactive streams, you can always buffer more, which allows
you some short term variation.  I don't think that any of the proposed
mechanisms would actually work even in that case, but maybe there is
a way to do something.  Non interactive streams need CBR averaged over
the buffer time.  That can be up to seconds in some cases, hundreds
of milliseconds in other cases, but over that interval, your goodput
must equate to the transmit offer rate times the interval (some small
loss rate acceptable).

It would be nice if there was a way.  I haven't seen one yet.

Brian

> -----Original Message-----
> From: Caitlin Bestler [mailto:cait@asomi.com]
> Sent: Thursday, November 20, 2003 5:02 PM
> To: Qiaobing Xie
> Cc: mmusic@ietf.org; 'Rosen, Brian'; TSV WG list; Phelan, Tom;
> avt@ietf.org; dccp@ietf.org; philippe.gentric@philips.com
> Subject: Re: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP 
> Media Guide
> 
> 
> 
> On Nov 20, 2003, at 1:26 PM, Qiaobing Xie wrote:
> 
> > Hi, Tom,
> >
> > It seems to me that you are basically: 1) questioning the 
> fairness of 
> > TCP itself
> > and, 2) you want to do something in the app level to 
> outmaneuver TCP 
> > so as to
> > achieve your term of fairness.
> >
> > I am not convinced this is the right thing to do with DCCP.
> >
> > I remember similar issue was discussed on tsvwg list (I am 
> cc'ing it). 
> > Maybe
> > people in tsvwg can share their thoughts on this as well.
> 
> 
> The fairness of TCP itself could use some questioning -- but I do not
> believe that is a reason to evade TCP friendly rate control.
> 
> Eventually, there should be a solution for proper support of 
> predictably
> variable rate flows that have fixed delivery requirements. But that 
> solution
> should be developed considering TCP, SCTP and DCCP as a whole.
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 

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Cc: mmusic@ietf.org, "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
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On Nov 20, 2003, at 1:26 PM, Qiaobing Xie wrote:

> Hi, Tom,
>
> It seems to me that you are basically: 1) questioning the fairness of 
> TCP itself
> and, 2) you want to do something in the app level to outmaneuver TCP 
> so as to
> achieve your term of fairness.
>
> I am not convinced this is the right thing to do with DCCP.
>
> I remember similar issue was discussed on tsvwg list (I am cc'ing it). 
> Maybe
> people in tsvwg can share their thoughts on this as well.


The fairness of TCP itself could use some questioning -- but I do not
believe that is a reason to evade TCP friendly rate control.

Eventually, there should be a solution for proper support of predictably
variable rate flows that have fixed delivery requirements. But that 
solution
should be developed considering TCP, SCTP and DCCP as a whole.


_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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From: Colin Perkins <csp@csperkins.org>
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On Wednesday, Nov 19, 2003, at 09:18 Europe/London,=20
vandana.yesoda@wipro.com wrote:
> I tried to get the following file from internet and using=20
> mailserv@etf.org.=A0
> draft-ietf-mmusic-confarch-03.txt
> Can any one send me this file?

That draft expired 3 years ago, and is likely only of historical=20
interest now. You can find a copy at=20
http://www.ietf.org/proceedings/00jul/I-D/mmusic-confarch-03.txt

--=20
Colin Perkins
http://csperkins.org/


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From: Stig Venaas <Stig.Venaas@uninett.no>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] some comments on IPV and ICE
Message-ID: <20031121100230.GC13937@sverresborg.uninett.no>
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Thanks Gonzalo, but I still think there are problems, see below.

On Wed, Nov 19, 2003 at 03:17:39PM +0200, Gonzalo Camarillo wrote:
[...]
> Regarding your comment about dual-stack UAs, I do not see any problem 
> related to the use of IPV.

Wouldn't dual-stack UAs that don't support IPV try to use both
media descriptions and try to establish two connections/streams?

Isn't that also why it in section 5 of IPV says:

   It is STRONGLY RECOMMENDED that dual-stack IPv6/IPv4 hosts implement
   the IPV semantics. Dual-stack hosts that failed to implement IPV
   would need more RTTs to establish a session with a single-stack host.
   When acting as answerers, they would establish more media streams
   than needed. This could increase the session bandwidth in the first
   instants of the session, until the remote end could issue a new offer
   with only one m= line.

Since there's alreay a large number of systems that are dual-stack,
and there probably will be even more in the future, I think one
should try to avoid such a requirement.

I understand that the IPV solution is useful to single-stack systems,
but I'm afraid it will create too many problems for existing dual-stack
systems.

Stig

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Hi,

I got the same comment during the MMUSIC presentation. I have added a 
section that defines a SIP option-tag and that explains what can happen 
in you do not use it (basically it explains what you also point out in 
your mail.)

I intend to release a new version of the draft pretty soon. I hope it 
will address all the comments that I have received from the group.

Thanks for your comments,

Gonzalo

Stig Venaas wrote:

> Thanks Gonzalo, but I still think there are problems, see below.
> 
> On Wed, Nov 19, 2003 at 03:17:39PM +0200, Gonzalo Camarillo wrote:
> [...]
> 
>>Regarding your comment about dual-stack UAs, I do not see any problem 
>>related to the use of IPV.
> 
> 
> Wouldn't dual-stack UAs that don't support IPV try to use both
> media descriptions and try to establish two connections/streams?
> 
> Isn't that also why it in section 5 of IPV says:
> 
>    It is STRONGLY RECOMMENDED that dual-stack IPv6/IPv4 hosts implement
>    the IPV semantics. Dual-stack hosts that failed to implement IPV
>    would need more RTTs to establish a session with a single-stack host.
>    When acting as answerers, they would establish more media streams
>    than needed. This could increase the session bandwidth in the first
>    instants of the session, until the remote end could issue a new offer
>    with only one m= line.
> 
> Since there's alreay a large number of systems that are dual-stack,
> and there probably will be even more in the future, I think one
> should try to avoid such a requirement.
> 
> I understand that the IPV solution is useful to single-stack systems,
> but I'm afraid it will create too many problems for existing dual-stack
> systems.
> 
> Stig

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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Brian, you wrote:

>The reason to "avoid" TCP friendly rate control is, at least so far,
>because there is no way yet proposed to do it without destroying the
>usability of the RTP flow.

I aggree this is a chalenge and especially I think that the availability
of codecs for which the rate can be rapidly changed over wide ranges is key.

>Part of the problem is that the only "rate control" that works is to
>negotiate an acceptable rate in the signalling before the stream starts.
>There really isn't any other way to change the rate, at least for
>interactive streams.

This is where progress must be done: there is a need for signaling
that the rate must change, during the streaming.

>For non-interactive streams, you can always buffer more, which allows
>you some short term variation.  I don't think that any of the proposed
>mechanisms would actually work even in that case, but maybe there is
>a way to do something.

Stream switching is a solution.

regards,

Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: mmusic@ietf.org, TSV WG list <tsvwg@ietf.org>, avt@ietf.org, dccp@ietf.org
CC: "Phelan, Tom" <tphelan@sonusnet.com>
References: <CAC8ED3E4E27EE4BBCF7030CC74A267D011E45A6@sonusmail01.sonusnet.com> <3FBD1558.C10047DF@motorola.com> <22EDA8FA-1BA5-11D8-8F72-000393814604@asomi.com>
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Subject: [MMUSIC] DCCP Media Guide - STOP crossposting
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Hi,

Please don't crosspost to all these lists. It creates problems for the 
list managers and will fragment the discussion.

As long as we are discussing issues related to media and DCCP please 
keep these topics on the AVT mailing list.

Other DCCP related issues goes on the DCCP list.

Best Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'vandana.yesoda@wipro.com'" <vandana.yesoda@wipro.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] draft-ietf-mmusic-confarch-03.txt
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Hi,
If you are interested in conferencing architecture look at
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framework
-01.txt
<http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framewor
k-01.txt> 
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framework
-01.txt
<http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framewor
k-01.txt> 
 
Roni
*************************************
Roni Even
Polycom Israel
Tel: +972-3-9251200
Cell: +972-55-481099
email:roni.even@polycom.co.il
*******************************************
-----Original Message-----
From: vandana.yesoda@wipro.com [mailto:vandana.yesoda@wipro.com]
Sent: Wednesday, November 19, 2003 11:18 AM
To: mmusic@ietf.org
Subject: [MMUSIC] draft-ietf-mmusic-confarch-03.txt


Hi,
 
I tried to get the following file from internet and using mailserv@etf.org
<mailto:mailserv@etf.org> .
 
draft-ietf-mmusic-confarch-03.txt
 
Can any one send me this file?
  
Thanks 'n Rgds
Vandana
 

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</xml><![endif]-->
<STYLE>@page Section1 {size: 8.5in 11.0in; margin: 1.0in 1.25in 1.0in =
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	{mso-style-name:"Table Normal";
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	font-family:"Times New Roman";}
</style>
<![endif]--></HEAD>
<BODY lang=3DEN-US style=3D"tab-interval: .5in" vLink=3Dpurple =
link=3Dblue>
<DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2>If you=20
are interested in conferencing architecture look at</FONT></SPAN></DIV>
<DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2><A=20
href=3D"http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferenci=
ng-framework-01.txt">http://www.softarmor.com/wgdb/docs/draft-ietf-sippi=
ng-conferencing-framework-01.txt</A></FONT></SPAN></DIV>
<DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2><A=20
href=3D"http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferenci=
ng-framework-01.txt">http://www.softarmor.com/wgdb/docs/draft-ietf-sippi=
ng-conferencing-framework-01.txt</A></FONT></SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Roni</FONT></SPAN></DIV>
<P dir=3Dltr><FONT face=3DArial=20
size=3D2>*************************************</FONT></P>
<P dir=3Dltr><FONT face=3DArial size=3D2>Roni Even</FONT></P>
<P dir=3Dltr><FONT face=3DArial size=3D2>Polycom Israel</FONT></P>
<P dir=3Dltr><FONT face=3DArial size=3D2>Tel: +972-3-9251200</FONT></P>
<P dir=3Dltr><FONT face=3DArial size=3D2>Cell: =
+972-55-481099</FONT></P>
<P dir=3Dltr><FONT face=3DArial =
size=3D2>email:roni.even@polycom.co.il</FONT></P>
<P dir=3Dltr><FONT face=3DArial=20
size=3D2>*******************************************</FONT></P>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> =
vandana.yesoda@wipro.com=20
  [mailto:vandana.yesoda@wipro.com]<BR><B>Sent:</B> Wednesday, November =
19, 2003=20
  11:18 AM<BR><B>To:</B> mmusic@ietf.org<BR><B>Subject:</B> [MMUSIC]=20
  draft-ietf-mmusic-confarch-03.txt<BR><BR></FONT></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Hi,<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">I tried to get the =
following file=20
  from internet and using <A=20
  =
href=3D"mailto:mailserv@etf.org">mailserv@etf.org</A>.<o:p></o:p></SPAN>=
</FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><SPAN class=3DGramE><FONT face=3DArial =
size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">draft-ietf-mmusic-confarch-03.txt</SPAN></FONT></SPAN><FONT=20
  face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Can any one send me =
this=20
  file?<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">&nbsp;=20
  <o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">Thanks 'n=20
  Rgds<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">Vandana<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: =
12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P></DIV></BLOCKQUOTE></BODY></HTM=
L>

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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <mmusic@ietf.org>, <vandana.yesoda@wipro.com>,
        "Even, Roni" <roni.even@polycom.co.il>
Cc: "Toip list" <toip@snowshore.com>
Subject: RE: [MMUSIC] draft-ietf-mmusic-confarch-03.txt
Date: Sun, 23 Nov 2003 11:11:59 +0100
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A good draft!

Just in order to keep the awareness alive that we have three common real
time streaming media to handle in conferences, ( video, text and audio ) =
I
have added a couple of proposed examples of media policies regarding the
text medium that was missing.
Otherwise the draft is general and valid for these three media and more.



4.6 Conference Policy
-------snip------
---------- note proposed additions about the text medium in the examples
below----assuming that the basic display of text from each participant
displayed under video from that participant is the common understanding
already established in other specifications-----------------

   Some examples of media policies include:

   o  The video output is the picture of the loudest speaker (video
      follows audio).

   o  The audio from each participant will be mixed with equal weight,
      and distributed to all other participants.

   o  The video from a participant can be associated with a text generati=
ng
participant,
      so that when this video is output, the text from the text source is
also displayed.
      (useful for real time subtitling)
=2E

   o  The audio, text and video that is distributed is the one selected b=
y
the
      floor control server.

   o  The text from all participants is displayed in one common area in
traditional IRC style.
-----------------------------------------end of
proposal-----------------------------------------------------------------=
---
------------------------------------

Regards
Gunnar Hellstr=F6m
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


  -----Original Message-----
  From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf Of
Even, Roni
  Sent: Sunday, November 23, 2003 9:25 AM
  To: 'vandana.yesoda@wipro.com'; mmusic@ietf.org
  Subject: RE: [MMUSIC] draft-ietf-mmusic-confarch-03.txt


  Hi,
  If you are interested in conferencing architecture look at

http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framew=
ork
-01.txt

http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framew=
ork
-01.txt

  Roni
  *************************************

  Roni Even

  Polycom Israel

  Tel: +972-3-9251200

  Cell: +972-55-481099

  email:roni.even@polycom.co.il

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

    -----Original Message-----
    From: vandana.yesoda@wipro.com [mailto:vandana.yesoda@wipro.com]
    Sent: Wednesday, November 19, 2003 11:18 AM
    To: mmusic@ietf.org
    Subject: [MMUSIC] draft-ietf-mmusic-confarch-03.txt


    Hi,



    I tried to get the following file from internet and using
mailserv@etf.org.



    draft-ietf-mmusic-confarch-03.txt



    Can any one send me this file?



    Thanks 'n Rgds

    Vandana



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<BODY lang=3DEN-US style=3D"tab-interval: .5in" vLink=3Dpurple =
link=3Dblue>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D624035409-23112003>A good=20
draft!</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D624035409-23112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D624035409-23112003>Just=20
in order to keep the awareness alive that we have three common real time =

streaming media to handle in conferences, ( video, text and audio ) I =
have added=20
a couple of proposed examples of media policies regarding the text =
medium that=20
was missing. </SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D624035409-23112003>Otherwise the draft is general and valid for =
these=20
three media and more.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV><FONT face=3DArial=20
color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><BR>4.6 =
Conference=20
Policy<BR><SPAN class=3D624035409-23112003></SPAN>-<SPAN=20
class=3D624035409-23112003>------snip------</SPAN><BR><SPAN=20
class=3D624035409-23112003></SPAN><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
size=3D2>-<SPAN class=3D624035409-23112003>--------- note proposed =
additions about=20
the text medium in the examples below----assuming that the basic display =
of text=20
from each participant displayed under video from that participant is the =
common=20
understanding already established in other=20
specifications-----------------</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D624035409-23112003></SPAN></FONT></FONT></FONT><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT><BR>&nbsp;&nbsp; Some examples of media policies=20
include:<BR><BR>&nbsp;&nbsp; o&nbsp; The video output is the picture of =
the=20
loudest speaker (video<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; follows=20
audio).<BR><BR>&nbsp;&nbsp; o&nbsp; The audio from each participant will =
be=20
mixed with equal weight,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and =
distributed to=20
all other participants.</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV>&nbsp;&nbsp; o&nbsp;&nbsp;<SPAN class=3D624035409-23112003>The =
video from=20
a</SPAN><SPAN class=3D624035409-23112003>&nbsp;participant can&nbsp;be =
associated=20
with a text generating participant,</SPAN></DIV>
<DIV><SPAN class=3D624035409-23112003>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; so =
that when=20
this video is output, the text from the text source is&nbsp;also=20
displayed.<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (useful for&nbsp;real time=20
subtitling)&nbsp;</SPAN></DIV>
<DIV><SPAN class=3D624035409-23112003></SPAN>.<BR><BR>&nbsp;&nbsp; =
o&nbsp; The=20
audio<SPAN class=3D624035409-23112003>, text</SPAN>&nbsp;and video that =
is=20
distributed is the one selected by the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
floor=20
control server.<BR><BR>&nbsp;&nbsp; o&nbsp; The <SPAN=20
class=3D624035409-23112003>text</SPAN>&nbsp;<SPAN =
class=3D624035409-23112003>from=20
all participants is displayed in one common area in traditional IRC=20
style.</SPAN><BR><SPAN class=3D624035409-23112003></SPAN><FONT =
face=3DArial><FONT=20
color=3D#0000ff><FONT size=3D2>-<SPAN=20
class=3D624035409-23112003>----------------------------------------end =
of=20
proposal-----------------------------------------------------------------=
---------------------------------------</SPAN></FONT></FONT></FONT><BR></=
DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D624035409-23112003>Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D624035409-23112003>Gunnar=20
Hellstr=F6m</SPAN></FONT></DIV>
<P><FONT size=3D2>-------------------------------------------<BR>Gunnar=20
Hellstr=F6m<BR>Omnitor AB<BR>Renathv=E4gen 2<BR>SE 121 37=20
Johanneshov<BR>SWEDEN<BR>+46 8 556 002 03<BR>Mob: +46 708 204=20
288<BR>www.omnitor.se<BR>Gunnar.Hellstrom@Omnitor.se<BR>-----------------=
---------------------------<BR>&nbsp;</FONT>=20
</P>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> =
mmusic-admin@ietf.org=20
  [mailto:mmusic-admin@ietf.org]<B>On Behalf Of </B>Even, =
Roni<BR><B>Sent:</B>=20
  Sunday, November 23, 2003 9:25 AM<BR><B>To:</B> =
'vandana.yesoda@wipro.com';=20
  mmusic@ietf.org<BR><B>Subject:</B> RE: [MMUSIC]=20
  draft-ietf-mmusic-confarch-03.txt<BR><BR></FONT></DIV>
  <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Hi,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2>If=20
  you are interested in conferencing architecture look =
at</FONT></SPAN></DIV>
  <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2><A=20
  =
href=3D"http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencin=
g-framework-01.txt">http://www.softarmor.com/wgdb/docs/draft-ietf-sipping=
-conferencing-framework-01.txt</A></FONT></SPAN></DIV>
  <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2><A=20
  =
href=3D"http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencin=
g-framework-01.txt">http://www.softarmor.com/wgdb/docs/draft-ietf-sipping=
-conferencing-framework-01.txt</A></FONT></SPAN></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Roni</FONT></SPAN></DIV>
  <P dir=3Dltr><FONT face=3DArial=20
  size=3D2>*************************************</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>Roni Even</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>Polycom Israel</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>Tel: =
+972-3-9251200</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>Cell: =
+972-55-481099</FONT></P>
  <P dir=3Dltr><FONT face=3DArial =
size=3D2>email:roni.even@polycom.co.il</FONT></P>
  <P dir=3Dltr><FONT face=3DArial=20
  size=3D2>*******************************************</FONT></P>
  <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> =
vandana.yesoda@wipro.com=20
    [mailto:vandana.yesoda@wipro.com]<BR><B>Sent:</B> Wednesday, =
November 19,=20
    2003 11:18 AM<BR><B>To:</B> mmusic@ietf.org<BR><B>Subject:</B> =
[MMUSIC]=20
    draft-ietf-mmusic-confarch-03.txt<BR><BR></FONT></DIV>
    <DIV class=3DSection1>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Hi,<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">I tried to get the =
following=20
    file from internet and using <A=20
    =
href=3D"mailto:mailserv@etf.org">mailserv@etf.org</A>.<o:p></o:p></SPAN><=
/FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><SPAN class=3DGramE><FONT face=3DArial =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">draft-ietf-mmusic-confarch-03.txt</SPAN></FONT></SPAN><FONT=20
    face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Can any one send me =
this=20
    file?<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">&nbsp;=20
    <o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">Thanks 'n=20
    Rgds<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">Vandana<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: =
12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P></DIV></BLOCKQUOTE></BLOCKQUOTE>=
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Date: Fri, 21 Nov 2003 10:10:02 -0500
To: "Phelan, Tom" <tphelan@sonusnet.com>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Cc: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
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At 11:47 AM 11/20/2003 -0500, Phelan, Tom wrote:

<snip>

>I don't think I'm forcing the problem to fit the solution, just perhaps that
>the problem I'm trying to solve is different than you perceive.  The problem
>that I'm trying to solve is how can an application that is content with a
>maximum bandwidth compete with TCP applications that are insatiably greedy,
>in a way that satisfies some concept of fairness in all directions?
>
>The solution that I see is to provide a buffer against TCP probing.  This
>would involve sending more data than is nominally necessary.  If the network
>has capacity for more than 4M bps, then why shouldn't a media app make use
>of it?  A TCP app would.

<snip>

I think this snippet captures the essence of the argument.

An "insatiably greedy" TCP application has an infinite amount of data to
send. And "fairness" in "TCP-land" is ONLY FAIRNESS IN INSTANTANEOUS
BANDWIDTH ALLOCATION (across TCP/DCCP/SCTP connections/associations).

This is historically how TCP has been viewed. But wait ... if you delay
the "infinite amount of data" in the TCP socket .... what bad happens?
Nothing, it just takes longer (from a "work conservation viewpoint") to send
the data. Did TCP suffer? Well ... it got less work per unit time ... but
nothing really bad occurred for the "insatiable TCP" case.

Most real-time data is DELAY SENSITIVE. To real-time data ... it would be
"fair" to delay some of the TCP data during it's bursts (like some video codecs
have when a high-entropy scene change occurs). That is because TCP
can afford to "wait a little longer" to get it's data across. The bursts 
are usually
bounded in time (a talkspurt, a scene change, etc.) ... and TCP can usually
afford to wait through these bursts. From the real-time application's 
perspective
they are being "good citizens" by not using bandwidth when they don't
need to (during VAD/silence or no video scene changes). Can't the TCP apps
step aside for a moment and acknowledge their good citizenship?

To be fair ... we really need to re-evaluate the issue of "fairness" 
capturing more
than just an INSTANTANEOUS BANDWIDTH ALLOCATION dimension. Is it "fair"
to slow down TCP more than present mechanisms allow? For a "insatiably greedy"
TCP connection the answer may be yes. For a TCP session with a reasonably
bounded amount of data the answer maybe no - because there is a linear
relationship with how long the data will take to get to the other side and
the slowdown rate.

This is a really tough problem.

Why this post? Simply because I'd desire each camp to widen their vision
of what is "fair/right/best" from their own myopic point of view. TCP'ers 
should
try to capture the spirit of Brian's comment ... that "forcing the problem
to fit [TCP's view of] a solution" is not appropriate. And that is, in 
fact, what
I perceive all the well-intentioned "TCP rate control" crowd to be doing here.

Please widen your perspective of the problem. Then comment. Don't expect
people like Brian to be happy with the "you must fit in to TCPs view of the
world" when most of their real-time applications are already being good
citizens (by not sending "all they can" when they don't have to). Do you
like being further punished because you weren't greedy when you could
have been?

Michael Ramalho


_______________________________________________
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From: "Roy, Radhika R, ALABS" <rrroy@att.com>
To: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>, <mmusic@ietf.org>,
        <vandana.yesoda@wipro.com>, "Even, Roni" <roni.even@polycom.co.il>
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I agree with Gunnar.
=20
I am adding more. Kindly see inline [RRR].
=20
Best regards,
Radhika
-----Original Message-----
From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
Sent: Sunday, November 23, 2003 5:12 AM
To: mmusic@ietf.org; vandana.yesoda@wipro.com; Even, Roni
Cc: Toip list
Subject: RE: [MMUSIC] draft-ietf-mmusic-confarch-03.txt


A good draft!
=20
Just in order to keep the awareness alive that we have three common real =
time streaming media to handle in conferences, ( video, text and audio ) =
I have added a couple of proposed examples of media policies regarding =
the text medium that was missing.=20
Otherwise the draft is general and valid for these three media and more.
=20
=20


4.6 Conference Policy
-------snip------
---------- note proposed additions about the text medium in the examples =
below----assuming that the basic display of text from each participant =
displayed under video from that participant is the common understanding =
already established in other specifications-----------------

   Some examples of media policies include:

   o  The video output is the picture of the loudest speaker (video
      follows audio).

   o  The audio from each participant will be mixed with equal weight,
      and distributed to all other participants.
[Roy, Radhika R, ALSO] If the real-time text is used as an alternative =
of audio from any participant, the mixing and distribution should also =
be taken into account appropriately.
=20
   o  The video from a participant can be associated with a text =
generating participant,
      so that when this video is output, the text from the text source =
is also displayed.
      (useful for real time subtitling)=20
.

   o  The audio, text and video that is distributed is the one selected =
by the
      floor control server.
[Roy, Radhika R, ALSO]  If real-time text is used as an alternative of =
audio from any participant, the selection and distribution of the media =
should also be taken into account appropriately by the floor control =
server.

   o  The text from all participants is displayed in one common area in =
traditional IRC style.
-----------------------------------------end of =
proposal-----------------------------------------------------------------=
---------------------------------------

Regards
Gunnar Hellstr=F6m
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------
 =20
-----Original Message-----
From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf Of =
Even, Roni
Sent: Sunday, November 23, 2003 9:25 AM
To: 'vandana.yesoda@wipro.com'; mmusic@ietf.org
Subject: RE: [MMUSIC] draft-ietf-mmusic-confarch-03.txt


Hi,
If you are interested in conferencing architecture look at
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framew=
ork-01.txt
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencing-framew=
ork-01.txt
=20
Roni
*************************************
Roni Even
Polycom Israel
Tel: +972-3-9251200
Cell: +972-55-481099
email:roni.even@polycom.co.il
*******************************************
-----Original Message-----
From: vandana.yesoda@wipro.com [mailto:vandana.yesoda@wipro.com]
Sent: Wednesday, November 19, 2003 11:18 AM
To: mmusic@ietf.org
Subject: [MMUSIC] draft-ietf-mmusic-confarch-03.txt


Hi,
=20
I tried to get the following file from internet and using =
mailserv@etf.org.
=20
draft-ietf-mmusic-confarch-03.txt
=20
Can any one send me this file?
 =20
Thanks 'n Rgds
Vandana
=20

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<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D055414501-24112003>I=20
agree with Gunnar.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D055414501-24112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D055414501-24112003>I am=20
adding more. Kindly see&nbsp;inline [RRR].</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D055414501-24112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D055414501-24112003>Best=20
regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D055414501-24112003>Radhika</SPAN></FONT></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Gunnar Hellstrom=20
  [mailto:gunnar.hellstrom@omnitor.se]<BR><B>Sent:</B> Sunday, November =
23, 2003=20
  5:12 AM<BR><B>To:</B> mmusic@ietf.org; vandana.yesoda@wipro.com; Even, =

  Roni<BR><B>Cc:</B> Toip list<BR><B>Subject:</B> RE: [MMUSIC]=20
  draft-ietf-mmusic-confarch-03.txt<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D624035409-23112003>A=20
  good draft!</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D624035409-23112003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D624035409-23112003>Just=20
  in order to keep the awareness alive that we have three common real =
time=20
  streaming media to handle in conferences, ( video, text and audio ) I =
have=20
  added a couple of proposed examples of media policies regarding the =
text=20
  medium that was missing. </SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D624035409-23112003>Otherwise the draft is general and valid =
for these=20
  three media and more.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>&nbsp;</DIV><FONT face=3DArial=20
  color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><BR>4.6 =
Conference=20
  Policy<BR><SPAN class=3D624035409-23112003></SPAN>-<SPAN=20
  class=3D624035409-23112003>------snip------</SPAN><BR><SPAN=20
  class=3D624035409-23112003></SPAN><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2>-<SPAN class=3D624035409-23112003>--------- note proposed =
additions about=20
  the text medium in the examples below----assuming that the basic =
display of=20
  text from each participant displayed under video from that participant =
is the=20
  common understanding already established in other=20
  specifications-----------------</SPAN></FONT></FONT></FONT></DIV>
  <DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
  class=3D624035409-23112003></SPAN></FONT></FONT></FONT><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff=20
  size=3D2></FONT><BR>&nbsp;&nbsp; Some examples of media policies=20
  include:<BR><BR>&nbsp;&nbsp; o&nbsp; The video output is the picture =
of the=20
  loudest speaker (video<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; follows=20
  audio).<BR><BR>&nbsp;&nbsp; o&nbsp; The audio from each participant =
will be=20
  mixed with equal weight,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and =
distributed to=20
  all other participants.<BR><SPAN class=3D055414501-24112003><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>[Roy, Radhika R, ALSO]&nbsp;If the real-time =
text is used=20
  as an alternative of audio&nbsp;from any participant, the mixing and=20
  distribution should also be taken into account=20
  appropriately.</FONT></SPAN></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV>&nbsp;&nbsp; o&nbsp;&nbsp;<SPAN class=3D624035409-23112003>The =
video from=20
  a</SPAN><SPAN class=3D624035409-23112003>&nbsp;participant can&nbsp;be =

  associated with a text generating participant,</SPAN></DIV>
  <DIV><SPAN class=3D624035409-23112003>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
so that=20
  when this video is output, the text from the text source is&nbsp;also=20
  displayed.<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (useful for&nbsp;real =
time=20
  subtitling)&nbsp;</SPAN></DIV>
  <DIV><SPAN class=3D624035409-23112003></SPAN>.<BR><BR>&nbsp;&nbsp; =
o&nbsp; The=20
  audio<SPAN class=3D624035409-23112003>, text</SPAN>&nbsp;and video =
that is=20
  distributed is the one selected by =
the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; floor=20
  control server.<BR><SPAN class=3D055414501-24112003><FONT face=3DArial =

  color=3D#0000ff size=3D2>[Roy, Radhika R, ALSO]&nbsp;&nbsp;If =
real-time text is=20
  used as an alternative of audio&nbsp;from any participant, =
the&nbsp;selection=20
  and distribution of the media should also be taken into account =
appropriately=20
  by the floor control server.</FONT></SPAN><BR><BR>&nbsp;&nbsp; o&nbsp; =
The=20
  <SPAN class=3D624035409-23112003>text</SPAN>&nbsp;<SPAN=20
  class=3D624035409-23112003>from all participants is displayed in one =
common area=20
  in traditional IRC style.</SPAN><BR><SPAN=20
  class=3D624035409-23112003></SPAN><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
  size=3D2>-<SPAN=20
  class=3D624035409-23112003>----------------------------------------end =
of=20
  =
proposal-----------------------------------------------------------------=
---------------------------------------</SPAN></FONT></FONT></FONT><BR></=
DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D624035409-23112003>Regards</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D624035409-23112003>Gunnar Hellstr=F6m</SPAN></FONT></DIV>
  <P><FONT =
size=3D2>-------------------------------------------<BR>Gunnar=20
  Hellstr=F6m<BR>Omnitor AB<BR>Renathv=E4gen 2<BR>SE 121 37=20
  Johanneshov<BR>SWEDEN<BR>+46 8 556 002 03<BR>Mob: +46 708 204=20
  =
288<BR>www.omnitor.se<BR>Gunnar.Hellstrom@Omnitor.se<BR>-----------------=
---------------------------<BR>&nbsp;</FONT>=20
  </P>
  <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> =
mmusic-admin@ietf.org=20
    [mailto:mmusic-admin@ietf.org]<B>On Behalf Of </B>Even, =
Roni<BR><B>Sent:</B>=20
    Sunday, November 23, 2003 9:25 AM<BR><B>To:</B> =
'vandana.yesoda@wipro.com';=20
    mmusic@ietf.org<BR><B>Subject:</B> RE: [MMUSIC]=20
    draft-ietf-mmusic-confarch-03.txt<BR><BR></FONT></DIV>
    <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>Hi,</FONT></SPAN></DIV>
    <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2>If=20
    you are interested in conferencing architecture look =
at</FONT></SPAN></DIV>
    <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2><A=20
    =
href=3D"http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencin=
g-framework-01.txt">http://www.softarmor.com/wgdb/docs/draft-ietf-sipping=
-conferencing-framework-01.txt</A></FONT></SPAN></DIV>
    <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff size=3D2><A=20
    =
href=3D"http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-conferencin=
g-framework-01.txt">http://www.softarmor.com/wgdb/docs/draft-ietf-sipping=
-conferencing-framework-01.txt</A></FONT></SPAN></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
    <DIV><SPAN class=3D708372308-23112003><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>Roni</FONT></SPAN></DIV>
    <P dir=3Dltr><FONT face=3DArial=20
    size=3D2>*************************************</FONT></P>
    <P dir=3Dltr><FONT face=3DArial size=3D2>Roni Even</FONT></P>
    <P dir=3Dltr><FONT face=3DArial size=3D2>Polycom Israel</FONT></P>
    <P dir=3Dltr><FONT face=3DArial size=3D2>Tel: =
+972-3-9251200</FONT></P>
    <P dir=3Dltr><FONT face=3DArial size=3D2>Cell: =
+972-55-481099</FONT></P>
    <P dir=3Dltr><FONT face=3DArial =
size=3D2>email:roni.even@polycom.co.il</FONT></P>
    <P dir=3Dltr><FONT face=3DArial=20
    size=3D2>*******************************************</FONT></P>
    <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
      <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
      size=3D2>-----Original Message-----<BR><B>From:</B> =
vandana.yesoda@wipro.com=20
      [mailto:vandana.yesoda@wipro.com]<BR><B>Sent:</B> Wednesday, =
November 19,=20
      2003 11:18 AM<BR><B>To:</B> mmusic@ietf.org<BR><B>Subject:</B> =
[MMUSIC]=20
      draft-ietf-mmusic-confarch-03.txt<BR><BR></FONT></DIV>
      <DIV class=3DSection1>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Hi,<o:p></o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">I tried to get the =
following=20
      file from internet and using <A=20
      =
href=3D"mailto:mailserv@etf.org">mailserv@etf.org</A>.<o:p></o:p></SPAN><=
/FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><SPAN class=3DGramE><FONT face=3DArial =
size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">draft-ietf-mmusic-confarch-03.txt</SPAN></FONT></SPAN><FONT=20
      face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p></o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Can any one send me =
this=20
      file?<o:p></o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">&nbsp;=20
      <o:p></o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">Thanks 'n=20
      Rgds<o:p></o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
      style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-no-proof: =
yes">Vandana<o:p></o:p></SPAN></FONT></P>
      <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN =

      style=3D"FONT-SIZE: =
12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P></DIV></BLOCKQUOTE></BLOCKQUOTE>=
</BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C3B22E.8A51B228--

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From: Mark Baugher <mbaugher@cisco.com>
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Subject: [MMUSIC] e2m, m2m security drafts
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hi
   I have read the three drafts dealing with end-to-middle and 
middle-to-middle security, both requirements and mechanisms.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-e2m-sec-reqs-00.txt
http://www.ietf.org/internet-drafts/draft-barnes-sipping-sec-inserted-info-01.txt
http://www.ietf.org/internet-drafts/draft-ono-sipping-end2middle-security-00.txt

As you may know, we developed SDP security descriptions to describe a key 
and security parameters for a media session.  We need to have end-to-end 
authentication and encryption of this information and secure multiparts 
looks like a promising approach to our security-description needs.  So, 
I'll have a lot of comments and questions on these drafts.

In this note, I'll limit myself to one question and one comment.

1. Section 5.1 of 
http://www.ietf.org/internet-drafts/draft-ono-sipping-end2middle-security-00.txt 
shows an example of an encrypted SDP in an INVITE but the key to this 
encrypted SDP is selectively shared with some proxies so they can view the 
encrypted data.  There is undoubtedly some information in the SDP that a 
proxy needs to view and some that it does not need to view, such as the 
end-to-end keying material.  Can we accommodate this requirement in the 
current proposals?  That is, could we keep the keying material encrypted 
end-to-end while allowing a proxy to view other parts of the SDP message?

2. Have you considered writing a core specification like RFC 1847 and allow 
for one or more profiles such as S/MIME (today) and PGP MIME (in the 
future)?  I understand that S/MIME is the designated SIP solution, but it 
might be wise to allow for future extensions.

Thanks for doing this work.

Mark


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Date: Wed, 26 Nov 2003 18:44:40 +0900
From: Kumiko Ono <ono.kumiko@lab.ntt.co.jp>
To: Mark Baugher <mbaugher@cisco.com>, mary.barnes@nortelnetworks.com,
        tachimoto.shinya@lab.ntt.co.jp
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Hi Mark,

Thanks for your comments. Inline-

<mbaugher@cisco.com> wrote:
>1. Section 5.1 of 
>http://www.ietf.org/internet-drafts/draft-ono-sipping-end2middle-security-
>00.txt 
>shows an example of an encrypted SDP in an INVITE but the key to this 
>encrypted SDP is selectively shared with some proxies so they can view the 
>encrypted data.  There is undoubtedly some information in the SDP that a 
>proxy needs to view and some that it does not need to view, such as the 
>end-to-end keying material.  Can we accommodate this requirement in the 
>current proposals?  That is, could we keep the keying material encrypted 
>end-to-end while allowing a proxy to view other parts of the SDP message?

Yes, definitely. 
Section 3.1 of the requirement draft says that 
"The solution MUST work even with  SIP end-to-end encryption for 
confidentiality service enabled".

To clarify this requirement, I would say:
- It MUST work even with end-to-end encryption of some data.
  That is, encrypted data can be shared with the end user and selected 
proxies.
- It MUST NOT violate end-to-end encryption when the encrypted data does 
not need to be shared with any proxies.
  
>2. Have you considered writing a core specification like RFC 1847 and allow 
>for one or more profiles such as S/MIME (today) and PGP MIME (in the 
>future)?  I understand that S/MIME is the designated SIP solution, but it 
>might be wise to allow for future extensions.

Hmm... in the future, such extensions might be needed.
However, I have not considered that yet.
I think there are still more work to use S/MIME practically in SIP.

Thanks a lot,

Kumiko

--
Kumiko Ono
NTT Network Service Systems Labs.
Tel +81 422 59 4508/Fax +81 422 60 4012

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From: "Drage, Keith (Keith)" <drage@lucent.com>
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Subject: [MMUSIC] Comments on draft-ietf-mmusic-sdp-new-15
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At IETF58 requests were made for a final review of this document before resubmitting to the IESG.

These comments are made as a result of that request.

If mmusic participants wish to enter into discussion on any of these, you snip out the remainder of the comments so that mail threads do not become burdened with text that is not under discussion.

regards

Keith

Keith Drage
Lucent Technologies
drage@lucent.com
tel: +44 1793 776249

Comments on draft-ietf-mmusic-sdp-new-15
----------------------------------------

1)	Introduction: section 1. 2nd paragraph. In:

   When initiating multimedia teleconferences, voice-over-IP calls,
   streaming video, or other real-time sessions, there is a requirement
   to convey media details, transport addresses, and other session
   description metadata to the participants.

the words "or other real-time sessions" imply that SDP is limited to describing real-time  sessions. SDP is proposed to be used to describe message sessions, which are immediate  delivery, but which are not traditionally known as real-time. It is proposed to delete the  words "real-time", leaving "or other sessions" with the word "session" being the defined  term.

2)	Terminology: section 2. The term "session advertisment" is never used in the  document. The word "advertisment" is used in the document, but in descriptive terms that  does not really justify the use of a defined term. It is suggested we delete the definition  "session advertisment" from section 2.

3)	Terminology. I section 2, the term "session announcement" is defined, but this term  is not consistently used through the document. There are a number of places where the text  within the document should be enhanced to use this term as below. However in other cases,  rather than using the term "session announcement", it would appear that the other term  "session description" would be more appropriate to use, see also below.

a)	section 3.1, title: "Multicast announcement" --> "Multicast session announcement".

b)	section 3.1, 2nd paragraph: 

   One protocol commonly used to implement such a distributed directory
   is the Session Announcement Protocol, SAP [8]. SDP provides the
   recommended session description format for such announcements.

should become:

   One protocol commonly used to implement such a distributed directory
   is the Session Announcement Protocol, SAP [8]. SDP provides the
   recommended session description format for such _session_ announcements.

c)	section 3.4, 2nd paragraph:

   SAP announcements do not suffer from this mismatch.

should become:

   Session announcements made using SAP do not suffer from this mismatch.

d)	section 4.5, 1st paragraph:

   When many session descriptions are being distributed by SAP, or any
   other advertisement mechanism, it may be desirable to filter
   announcements that are of interest from those that are not.  SDP
   supports a categorisation mechanism for sessions that is capable of
   being automated.

should become:

   When many session descriptions are being distributed by SAP, or any
   other advertisement mechanism, it may be desirable to filter
   _session_ announcements that are of interest from those that are not.  SDP
   supports a categorisation mechanism for sessions that is capable of
   being automated.

e)	Section 5. 1st paragraph. Text:

   ...Also, since announcements may be transported via very
   unreliable means or damaged by an intermediate caching server, the
   encoding was designed with strict order and formatting rules so that
   most errors would result in malformed announcements which could be
   detected easily and discarded.  This also allows rapid discarding of
   encrypted announcements for which a receiver does not have the
   correct key.

should become:

   ...Also, since announcements may be transported via very
   unreliable means or damaged by an intermediate caching server, the
   encoding was designed with strict order and formatting rules so that
   most errors would result in malformed _session_ announcements which could be
   detected easily and discarded.  This also allows rapid discarding of
   encrypted _session_ announcements for which a receiver does not have the
   correct key.

f)	Section 5. 6th paragraph. Text:

   The set of type letters is deliberately small and not intended to be
   extensible -- an SDP parser MUST completely ignore any announcement
   that contains a type letter that it does not understand...

should become:

   The set of type letters is deliberately small and not intended to be
   extensible -- an SDP parser MUST completely ignore any _session description_
   that contains a type letter that it does not understand...


g)	Section 5.2. <version> text:

   <version> is a version number for this announcement. It is needed for
      proxy announcements to detect which of several announcements for
      the same session is the most recent...

should become:

   <version> is a version number for this _session description_. It is needed for
      proxy announcements to detect which of several announcements for
      the same session is the most recent...

Incidentally I am not sure what "proxy announcements" are, so have not proposed a revision  here. This sentence could do with revision to more clearly explain what is meant.

h)	Section 5.4, text:

   The "i=" field is information about the session.  There may be at
   most one session-level "i=" field per session description, and at
   most one "i=" field per media. Although it may be omitted, this is
   NOT RECOMMENDED for session announcements, and user interfaces for
   composing sessions should require text to be entered.  If it is
   present it must contain ISO 10646 characters (but see also the
   "a=charset" attribute below).

Assume that what is meant here is SAP rather than announcements. The text should therefore  be modified to: 

   The "i=" field is information about the session.  There may be at
   most one session-level "i=" field per session description, and at
   most one "i=" field per media. Although it may be omitted, this is
   NOT RECOMMENDED for SAP, and user interfaces for
   composing sessions should require text to be entered.  If it is
   present it must contain ISO 10646 characters (but see also the
   "a=charset" attribute below).

However, surely if this is a SAP requirement, it should be in RFC 2974 and therefore this  sentence should be deleted from this document as outside the scope.

i)	Section 5.7. text:

   A session announcement MUST contain either at least one "c=" field in
   each media description (see below) or a single "c=" field at the
   session-level.  It MAY contain a single session-level "c=" field and
   additional "c=" field(s) per media description, in which case the
   per-media values override the session-level settings for the
   respective media.

should become:

   A session description MUST contain either at least one "c=" field in
   each media description (see below) or a single "c=" field at the
   session-level.  It MAY contain a single session-level "c=" field and
   additional "c=" field(s) per media description, in which case the
   per-media values override the session-level settings for the
   respective media.

j)	Section 5.10. text:

   Thus, the above announcement could also have been written:

should become:

   Thus, the above _session_ announcement could also have been written:

k)	Section 5.11. text:

   If a session is likely to last several years, it is expected that the
   session announcement will be modified periodically rather than
   transmit several years worth of adjustments in one announcement.

should become:

   If a session is likely to last several years, it is expected that the
   session announcement will be modified periodically rather than
   transmit several years worth of adjustments in one _session_ announcement.

l)	Section 5.13. text:

   Attribute interpretation depends on the media tool being invoked.
   Thus receivers of session descriptions should be configurable in
   their interpretation of announcements in general and of attributes in
   particular.

should become:

   Attribute interpretation depends on the media tool being invoked.
   Thus receivers of session descriptions should be configurable in
   their interpretation of session descriptions in general and of attributes in
   particular.


4)	Section 4, 2nd paragraph. Current text reads:

   The purpose of SDP is to convey information about media streams in
   multimedia sessions to allow the recipients of a session description
   to participate in the session.  SDP is primarily intended for use in
   an internetwork, although it is sufficiently general that it can
   describe conferences in other network environments.

   A multimedia session, for these purposes, is defined as a set of
   media streams that exist for some duration of time.  Media streams
   can be many-to-many.  The times during which the session is active
   need not be continuous.

We already have a definition of "multimedia" session in section 2, and this purported  definition here uses different words. It is proposed to delete the duplicated definition as  follows:

   The purpose of SDP is to convey information about media streams in
   multimedia sessions to allow the recipients of a session description
   to participate in the session.  SDP is primarily intended for use in
   an internetwork, although it is sufficiently general that it can
   describe conferences in other network environments. Media streams
   can be many-to-many.  The times during which the session is active
   need not be continuous.


5)	Section 4.6. The text:

   The SDP specification recommends the use of the ISO 10646 character
   sets in the UTF-8 encoding [3] to allow many different languages to
   be represented.  However, to assist in compact representations, SDP
   also allows other character sets such as ISO 8859-1 to be used when
   desired.  Internationalization only applies to free-text fields
   (session name and background information), and not to SDP as a whole.

allows users of SIP to not use UTF-8. SIP (see RFC 3261) is defined to use UTF-8, but  apparently does not specify how included SDP in message bodies should be encoded. It would  appear to me that all SDP uses in SIP message bodies should also be encoded in UTF-8. 

Is this lack of specification a fault of the SIP specification, or is it appropriate to add  words to sdp-new stating that:

   Where an application using SDP makes no further specification of the 
   character set to be used, the ISO 10646 character sets in the UTF-8 
   encoding [3] MUST be used.

6)	Section 5. 1st paragraph. In the text:

   ... However, since SDP may be used in environments where the maximum
   permissable size of a session description is limited (e.g. SAP
   announcements; SIP transported in UDP), the encoding is deliberately
   compact...

As SIP transport using SDP is currently considered to be less than ideal, it is probably not  a good idea to identify it as an example. Therefore delete the words "SIP transported in  UDP".

7)	Section 5.6. Phone numbers. Firstly the correct terminology is not:

   Phone numbers SHOULD be given in the conventional international
   format: preceded by a "+" and the international country code. 

but rather:

   Phone numbers SHOULD be given in the form of an international public
   telecommunication number (see ITU-T Recommendation E.164) preceded 
   by a "+".

Secondly, I do not see why this use of international number is a SHOULD. Is this to allow  the provision of numbers in an entirely different numbering plan, or the use of rfc2806bis,  or what. If there is some reason for only a recommendation, rather than mandatory use, then  I would have expected a sentence explaining it.

Thirdly, for the requirement:

   There must be a space or a hyphen ("-") between the country code and the
   rest of the phone number. 

I do not see a need for this requirement. It is not needed to parse the number, and it means  that numbers in the specified international format for text representation, ITU-T  Recommendation E.123 cannot be used without modification. I suggest that E.123 representation of international public telecommunication numbers should be valid for the phone number string.

This also requires the deletion of the associated comment in the BNF in appendix A, i.e.

                            ;there must be a space or hyphen between
                            ;the international code and the rest of
                            ;the number.

8)	Section 5.6. E-mail addresses. Text:

   The alternative RFC822 name quoting convention is also allowed for
   both email addresses and phone numbers.  For example:

RFC 822 has been made obsolete by RFC 2822. Is there some reason for quoting RFC 822 rather  than RFC 2822.

9)	Section 5.6, text:

   Both email addresses and phone numbers can have an optional free text
   string associated with them, normally giving the name of the person
   who may be contacted.  This should be enclosed in parenthesis if it
   is present.  For example:

      e=j.doe@example.com (Jane Doe)

From the BNF, the parenthesis are mandatory for the enclosure of the free text field.  Therefore this should be revised to:

   Both email addresses and phone numbers can have an optional free text
   string associated with them, normally giving the name of the person
   who may be contacted.  This MUST be enclosed in parenthesis if it
   is present.  For example:

      e=j.doe@example.com (Jane Doe)


10)	Section 5.7, 6th paragraph:

   For IP4 and IP6 addresses, the connection address is defined as
   follows:

these are values within the field, rather than the names of IP numbering schemes, thus I  proposes:

   Where the <address type> is IP4 or IP6, the connection address is defined as
   follows:

11)	Section 5.8. I am not convinced that the wording relating to X- naming is adequate.  I understood this was somewhat deprecated and this could be more emphasised. Currently use  of X- for anything would be conformant to this specification and I am not sure that was  intended. I think we should certainly get rid of the MAY quoted below.

   Tool writers MAY define experimental bandwidth modifiers by prefixing
   their modifier with "X-".  For example:

Maybe this would be better rephrased as

   A prefix "X-" is defined for bandwidth modifier names. This is intended 
   for experimental purposes only. For example:

12)	Section 5.14 defines a number of media types. Are these already top-level MIME  content types, as defined in 9.2.1, or are these to be registered by IANA as new media  types? Either section 5.14 or section 9.2.1 needs enhancement to reflect what IANA are  requested to do with these.

13)	Section 6. This section lists a large number of attribute definitions, most of which  do not appear to be defined elsewhere. Section 9.2.4 says new attributes should be  registered with IANA. Are IANA expected to pick attribute registrations from clause 6, and  all these are therefore intended to be registered, or not. Text needs to be added to either  section 6, or section 9.2.4 giving more detail of this process. 

Note that if any of these attributes are defined in other documents rather than here, I  would suggest an informative reference against the attribute giving the details of where it  is defined.

14)	Section 5.14 and section 9.2.2. In section 9.2.2, the references to "proto" field  seem to be references to the "protocol" field and or a <transport> element, and or  "transport protocol" defined in section 5.14. Certainly "proto" is not used as a field name  anywhere else in the document ecept in the BNF. Field names should be aligned one way or the  other.

15)	Section 9.2.5. Does this document register the bandwidth modifiers CT and AS defined  in section 5.8. If so this should be clearly stated in this section.

16)	Section 9.2.6. Does this document register the network type IN as defined in section  5.2 and 5.7. If so, this should be clearly stated in this section.

17)	Appendix A. Would the top level construct not be better defined as  "session-description" rather than "announcement". Not all SDP data constructs are  announcements.

18)	Appendix A. There is considerable misalignment between the BNF field names and those  in the main document. For example "v=" is "proto-version" in the BNF, and "protocol version"  in the main document (section 5.1). I would suggest changing this to "protocol-version" in  the BNF. "o=" is "origin-field" in the BNF and "origin" in the main document (section 5.2).  I would suggest we use "Origin Field" in the title of 5.2. Section 5.2 gives the expansion  of "origin" as 

   o=<username> <session id> <version> <network type> <address type>
        <address>

The BNF gives:

      origin-field =        "o=" username SP sess-id SP sess-version SP
                            nettype SP addrtype SP unicast-address CRLF

Further unnecessary misalignment of names. In this case I would change the BNF to 

      origin-field =        "o=" username SP session-id SP session-version SP
                            network-type SP address-type SP unicast-address CRLF

with appropriate modification of dependent expansions, and the 5.2 text to 

   o=<username> <session id> <version> <network type> <address type>
        <unicast address>

and so on for alignment through the rest of the BNF.

19)	Appendix A. In the expansion for "proto" the following comment is added

                   ;typically "RTP/AVP" or "udp" for IP4

This document is meant to cover IPv4 and IPv6. Why is this comment IPv4 specific, and why is  TCP not mentioned as one of the typical values.

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The IESG has received a request from the Multiparty Multimedia Session Control 
WG to consider the following document:

- 'A Transport Independent Bandwidth Modifier for the Session Description 
   Protocol (SDP) '
   <draft-ietf-mmusic-sdp-bwparam-05.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-12-10.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-bwparam-05.txt


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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] RTSP: Telecon 18.00 the 3rd of December with agenda
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Hi,

Next week on Wednesday (3rd of December) it is time for a RTSP telecon 
again. Please send a me a mail if you intend to participate. If you have 
comments on the agenda, please drop me a mail.

 Telecon details:
 Phone number +46 8 71 99500
 Participant PIN   67700 
 Date   12/03/2003 (MM/DD/YYYY) 
 Start Time   18:00 (HH:MM)  (CET)
 Duration 90 minutes

Cheers

Magnus Westerlund



Agenda

1. Opening of meeting and Administrative
   - Scribe

2. RTSP Core

2.1 Core specification bugs under developement.

a. http://rtsp.org/bug507882 - RTP timestamp and pause
   - Aravind to write up text

b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
end of a media clip
   - Await review from the WG.

c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
   - Needs text

d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
  - The explicit rules will be removed, possibility will be left open. 
Needs to be edited in.

e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
  - Needs a clarifying definition of Live
  - Need to check to possibility for TiVo cases.

f. http://rtsp.org/bug531189 - How does REDIRECT method work
  - Conclusion from Minneapolis needs to be edited in:  Server MAY drop the
    connection after it has recieved 200 OK response on a REDIRECT.

g. http://rtsp.org/bug770522 - Request to response time is not limited.
  - Sean has written up text.
  - Conclusion from Minneapolis:  Sean's writeup seems reasonable. 
    Change the timelimit to 2 seconds and make sure that wording is there
    to indicate server should do this immediately if it know it's going to
    take a long time.  Need motivation example.



h. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
  - Has been removed from spec, awaiting review.

i. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
  - Awaiting review from the WG.

j. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
  - Text sent to mailing list, we will dicuss this, please read
    
https://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg02030.html

k. http://rtsp.org/bug810666 - Removal of SETUP in play state
   - Clarify in specification that it is unspecified.

l. http://rtsp.org/bug811815 - Use of c= in SDP to know possible address 
classes
   - Conclusion from Minneapolis:
     Should look at address classes in the SDP to figure what protocol
     to use (need to add SDP section of RTSP). Look at c= line syntax. 
     Multiple c lines can be use in media field?  SIP resolution was
     that multiple c lines should not be used to express multiple
     connection address.

m. http://rtsp.org/bug795314 - Caching of OPTIONS request
   - conclusion from Minneapolis: Will be declared unspecified.
  
n. http://rtsp.org/bug850256 - When must RTP-Info be included?
  
o. http://rtsp.org/bug850265 - Removed features are unspecified, not 
forbidden   

p. http://rtsp.org/bug840304 - CSEQ is direction dependent in numberspaces.
   - Awaiting write up.

q. http://rtsp.org/bug850267 - Should all ABNF syntax be gathered in one 
chapter?

r. http://rtsp.org/bug829528 - Usage of Grouping of media lines with RTSP.


2.2 Tracking of items not require actions currently.

These items are for future processing when appropriate.

A. http://rtsp.org/bug810681 -     The updated specification version number
  - Awaiting investigation results regarding changes by Aravind.
  - Question of intentions.


3. RTSP Specifiction Extensions

3.1 End of Media
  http://rtsp.org/bug810678 - How to indicate end of media stream
  - We need to document in the RTSP specification regarding RTP what
    happens when a source times out a SSRC or receives a RTCP BYE for a
    source.

  - Using BYE is not suitable way to indicate the end of the stream.
    The motivation for this can be seen in Thomas proposal.

  - The creation of a new proposal.
 
 
3.2 RTSP and NAT issues
 
  - Requirements
    Priorities:
    1. Server is deployed in open network.
    2. Server behind firewall next.
    3. If a client is not dual hosted, it should not have to deal with 
additional protocols.
    4. Must have defense against DOS
    5. Security.
    6. MUST deal with symmetric nat.
    7. ALG.  The firewall traversal should not interfere with ALGs.
    8. Advise to ALG writers not to screw it up. 
 
    Review and clarify.
 
  - Protocol extension proposals
    What proposals do we have, and which people are sheeparding.
   
  - Update to RTSP and NAT WG draft
    * ALG recommendation
 
   
4. Next meeting

5. Meeting closed.


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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-850256 ] When must RTP-Info be included? (SourceForge.net)
   2. [ rtspspec-Bugs-850265 ] Removed features are unspecified, not forbidden (SourceForge.net)
   3. [ rtspspec-Bugs-850267 ] Gather all ABNF in syntax chapter (SourceForge.net)
   4. [ rtspspec-Bugs-850267 ] Gather all ABNF in syntax chapter (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 27 Nov 2003 06:06:29 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-850256 ] When must RTP-Info be included?

Bugs item #850256, was opened at 2003-11-27 15:06
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850256&group_id=23194

Category: General
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Rob Lanphier (robla)
Summary: When must RTP-Info be included?

Initial Comment:
At the RTSP discussion at IETF 58 there where some
discussion in regards if RTP-Info really can be
presented in all Play Responses when the media streams
are RTP. The argument was Live media. 

The current specification (05) requires this. 

Are there really a reason for changing this?

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850256&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 27 Nov 2003 06:25:59 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-850265 ] Removed features are unspecified, not forbidden

Bugs item #850265, was opened at 2003-11-27 15:25
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850265&group_id=23194

Category: Editorial
Group: Fix DS - Needs clarificiation
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Removed features are unspecified, not forbidden

Initial Comment:
In cases where feature has existed or could be defined,
but the core specification does not define them, this
should be clearly indicated in the text. 

A convention definition of unspecified can be added.
Then used in the interesting places in the text. 


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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850265&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 27 Nov 2003 06:38:10 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-850267 ] Gather all ABNF in syntax chapter

Bugs item #850267, was opened at 2003-11-27 15:38
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850267&group_id=23194

Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Gather all ABNF in syntax chapter

Initial Comment:
- All ABNF needs to be gathered in the syntax chapter.
- Ensure that it is in 2234 form
- When necessary put references into text when the ABNF
gets moved. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850267&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 27 Nov 2003 06:40:40 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-850267 ] Gather all ABNF in syntax chapter

Bugs item #850267, was opened at 2003-11-27 15:38
Message generated for change (Settings changed) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850267&group_id=23194

>Category: Editorial
>Group: Resolved - Need write up
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Gather all ABNF in syntax chapter

Initial Comment:
- All ABNF needs to be gathered in the syntax chapter.
- Ensure that it is in 2234 form
- When necessary put references into text when the ABNF
gets moved. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850267&group_id=23194




End of Rtspspec-bugs Digest


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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-850256 ] When must RTP-Info be included? (SourceForge.net)
   2. [ rtspspec-Bugs-850256 ] When must RTP-Info be included? (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 27 Nov 2003 22:18:31 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-850256 ] When must RTP-Info be included?

Bugs item #850256, was opened at 2003-11-27 23:06
Message generated for change (Comment added) made by kjh1905m
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850256&group_id=23194

Category: General
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Rob Lanphier (robla)
Summary: When must RTP-Info be included?

Initial Comment:
At the RTSP discussion at IETF 58 there where some
discussion in regards if RTP-Info really can be
presented in all Play Responses when the media streams
are RTP. The argument was Live media. 

The current specification (05) requires this. 

Are there really a reason for changing this?

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

Comment By: Jaehwan, Kim (kjh1905m)
Date: 2003-11-28 15:18

Message:
Logged In: YES 
user_id=365760

IMHO, it is enough that we focus on 2 cases: 1) the 
beginning of the live service 2) live PAUSE/Resume because 
we don't believe live streaming session supports random 
access.

1) The beginning of the live service
At this moment, there is no way for a player to see if the 
session is live or not. But if there is no RTP-Info, the player 
can start its session by the following way:
When a player gets the first rtp packet, it should store the 
rtptimestamp and has to subtract it from the remains.

2) Live PAUSE/Resume
Currently, we use monotonic rtptimestamp increament for 
random access. However, if a player requests PAUSE and 
Resume after a couple of minute, then rtptimestamp has to 
have the time gap. Without RTP-Info, how can the player 
remove the rtptimestamp gap? How can it continue the live 
session without any timegap? 

If RTSP Range header field also doesn't exist, a player can do 
that. From the begginning, the player should remember the 
first rtptimestamp and after the resume, we can just subtract 
the first from the lastest rtptimestamp. And then, we can 
calculate the real presentation time from it, comparing with 
the real PAUSE time gap.

But this behavior of a player is different from general on-
demand service having RTP-Info. Therefore, we need to chek 
our if the player can do that the both cases, RTP-Info and no 
for live. It means a general VOD player might not continue to 
work after live PAUSE.

Feel free to give me an advise regarding this matter if I am 
wrong.

Thank you!
Jae-Hwan



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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850256&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 27 Nov 2003 23:59:12 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-850256 ] When must RTP-Info be included?

Bugs item #850256, was opened at 2003-11-27 15:06
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850256&group_id=23194

Category: General
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Nobody/Anonymous (nobody)
Summary: When must RTP-Info be included?

Initial Comment:
At the RTSP discussion at IETF 58 there where some
discussion in regards if RTP-Info really can be
presented in all Play Responses when the media streams
are RTP. The argument was Live media. 

The current specification (05) requires this. 

Are there really a reason for changing this?

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-11-28 08:59

Message:
Logged In: YES 
user_id=302620

The argument why a server could not present a RTP-Info
header in a PLAY response was due to that the distribution
server might not be aware of inter media synchronization of
the media streams it forwards. If that is the case there
exist the possibility to perform intermedia synchronization
with RTCP. 

In regards to Jae-Whan's comments: 

Regarding "1) The beginning of the live service" the first
comment is that there is no way of knowing that a session is
live. This we have addressed through the definition that any
open range in the SDP a=range attribute shall be considered
Live. 

"When a player gets the first rtp packet, it should store the 
rtptimestamp and has to subtract it from the remains." I
don't see how this relates to the intermedia synchronization
problem. A client can't rely on that all media streams first
packet will represent the same point in time and would
therefore result in bad synchronization between the media. 

In regards to 2: In a live session if you are continuing
after a couple of minutes there is no reason for removing
the timestamp gap. In a live media session the media will
have advanced during the paused time, therefore there does
not seem to be any reason to synchronize the media prior to
the pause to the one after the pause. Therefore the player
should in a live case flush the buffers when it resumes play
after a pause. Thus the lack of RTP-Info in this case would
have no effect. As the RTP timestamp continues correctly
during the paused time the intermedia synchronization
established prior to pausing will still be valid

I think the main problem of not always requiring RTP-Info is
the intermedia synchronization. RTCP can be used, however
the time until RTCP has been received for all media may be
on order of seconds, thus possibly resulting in that
playback has to be commeneced without synchronization. 



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

Comment By: Jaehwan, Kim (kjh1905m)
Date: 2003-11-28 07:18

Message:
Logged In: YES 
user_id=365760

IMHO, it is enough that we focus on 2 cases: 1) the 
beginning of the live service 2) live PAUSE/Resume because 
we don't believe live streaming session supports random 
access.

1) The beginning of the live service
At this moment, there is no way for a player to see if the 
session is live or not. But if there is no RTP-Info, the player 
can start its session by the following way:
When a player gets the first rtp packet, it should store the 
rtptimestamp and has to subtract it from the remains.

2) Live PAUSE/Resume
Currently, we use monotonic rtptimestamp increament for 
random access. However, if a player requests PAUSE and 
Resume after a couple of minute, then rtptimestamp has to 
have the time gap. Without RTP-Info, how can the player 
remove the rtptimestamp gap? How can it continue the live 
session without any timegap? 

If RTSP Range header field also doesn't exist, a player can do 
that. From the begginning, the player should remember the 
first rtptimestamp and after the resume, we can just subtract 
the first from the lastest rtptimestamp. And then, we can 
calculate the real presentation time from it, comparing with 
the real PAUSE time gap.

But this behavior of a player is different from general on-
demand service having RTP-Info. Therefore, we need to chek 
our if the player can do that the both cases, RTP-Info and no 
for live. It means a general VOD player might not continue to 
work after live PAUSE.

Feel free to give me an advise regarding this matter if I am 
wrong.

Thank you!
Jae-Hwan



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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=850256&group_id=23194




End of Rtspspec-bugs Digest


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Date: Sat, 29 Nov 2003 10:11:45 -0800
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
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A few questions and suggestions about b= lines:

Section 5.8 of this draft mentions that alternative units can be used
to specify the bandwidth value. Is there a mechanism to communicate the
units within SDP? It would help to clarify this in the text.

When using the CT modifier for RTP, is there any recommendation on
whether to use peak bandwidth or average bandwidth for each session in
the conference? A guideline would help.

Can the AS modifier be used at the session level? From the text here
it sounds like it applies to the media-level, however in previous RTSP
discussions I gathered that AS can refer to session level also - I'm
still a little confused about this.

thanks
geetha

-- 
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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Date: Sat, 29 Nov 2003 10:16:01 -0800
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
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On page 24, the example shown:

	m=audio 49230 RTP/AVP 96 97 98
	a=rtpmap:96 L8/8000
	a=rtpmap:97 L16/8000
	a=rtpmap:98 L16/11025/2

Does this mean all three payloads can appear simultaneously on the same
port 49230?

thanks
geetha
-- 
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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From: Colin Perkins <csp@csperkins.org>
To: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] a=rtpmap in sdp-new-15
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--> Geetha Srikantan <Geetha.Srikantan@Sun.COM> writes:
> On page 24, the example shown:
> 
> 	m=audio 49230 RTP/AVP 96 97 98
> 	a=rtpmap:96 L8/8000
> 	a=rtpmap:97 L16/8000
> 	a=rtpmap:98 L16/11025/2
> 
> Does this mean all three payloads can appear simultaneously on the same
> port 49230?

It means that all three payloads can be used in the RTP session on port
49230, switching between them as appropriate.

-- 
Colin Perkins
csp@csperkins.org

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From: Colin Perkins <csp@csperkins.org>
To: "Drage, Keith (Keith)" <drage@lucent.com>
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Keith,

--> "Drage, Keith (Keith)" <drage@lucent.com> writes:
> At IETF58 requests were made for a final review of this document before
> resubmitting to the IESG.
> 
> These comments are made as a result of that request.
> 
> If mmusic participants wish to enter into discussion on any of these, you
> snip out the remainder of the comments so that mail threads do not become
> burdened with text that is not under discussion.

Thanks, your review is greatly appreciated. I'll work on addressing these
for the next version.

-- 
Colin Perkins
csp@csperkins.org

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