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This time could change but FYI ...


DISPATCH Session 1 (2 hours)
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From gao.yang2@zte.com.cn  Sun Oct 10 19:18:05 2010
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Subject: [dispatch] Comments on draft-avasarala-dispatch-comm-div-notification-05 //About forking
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In our implementation(the background system is IMS) of this draft, we find 
one problem about forking.

In section 6.9(Handling of Forked Requests) of this draft:

This specification only allows a single dialog to be constructed as a
   result of emitting an initial SUBSCRIBE request.  This guarantees
   that only a single notifier is generating notifications for a
   particular subscription to a particular user.

we could find that forked subscription is forbidden for this Event 
package.

While in real IMS network, there are cases the div-services are deployed 
in different ASs. Then, the subscription should be forked to the different 
ASs, if there are more than one AS do div-services for the specific 
service user.

We once considered one way to avoid the forking condition. The way is to 
let the user send different subscription for different div-service. But by 
the end user's viewpoint, he/she have no idea of the deployment of the 
div-servies(in ASs). So, he/she does't know when should he/she send 
separated subscription or not. 

So, I suggest to allow forking for this Event package.

Thanks,

Gao

===================================
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 e_mail : gao.yang2@zte.com.cn
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<br><font size=2 face="sans-serif">In our implementation(the background
system is IMS) of this draft, we find one problem about forking.</font>
<br>
<br><font size=2 face="sans-serif">In section 6.9(Handling of Forked Requests)
of this draft:</font>
<br>
<br><font size=2 face="sans-serif">This specification only allows a single
dialog to be constructed as a</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;result of emitting an initial
SUBSCRIBE request. &nbsp;This guarantees</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;that only a single notifier
is generating notifications for a</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;particular subscription
to a particular user.</font>
<br>
<br><font size=2 face="sans-serif">we could find that forked subscription
is forbidden for this Event package.</font>
<br>
<br><font size=2 face="sans-serif">While in real IMS network, there are
cases the div-services are deployed in different ASs. Then, the subscription
should be forked to the different ASs, if there are more than one AS do
div-services for the specific service user.</font>
<br>
<br><font size=2 face="sans-serif">We once considered one way to avoid
the forking condition. The way is to let the user send different subscription
for different div-service. But by the end user's viewpoint, he/she have
no idea of the deployment of the div-servies(in ASs). So, he/she does't
know when should he/she send separated subscription or not. &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">So, I suggest to allow forking for this
Event package.</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao</font>
<br>
<br><font size=2 face="sans-serif">===================================<br>
 Zip &nbsp; &nbsp;: 210012<br>
 Tel &nbsp; &nbsp;: 87211<br>
 Tel2 &nbsp; :(+86)-025-52877211<br>
 e_mail : gao.yang2@zte.com.cn<br>
===================================</font><br><pre>
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: <gao.yang2@zte.com.cn>, <dispatch@ietf.org>
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Cc: drsubirsaha@yahoo.com
Subject: Re: [dispatch] Comments on draft-avasarala-dispatch-comm-div-notification-05 //About forking
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Hi

=20

Yes we did not want to support forking for the CDIV package since the
CDIV notification is only intended for the subscriber who subscribed it
.=20

=20

Regards

Ranjit

=20

From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
Sent: Monday, October 11, 2010 7:47 AM
To: dispatch@ietf.org
Cc: Avasarala Ranjit-A20990; jbakker@rim.com; drsubirsaha@yahoo.com
Subject: Comments on draft-avasarala-dispatch-comm-div-notification-05
//About forking

=20


In our implementation(the background system is IMS) of this draft, we
find one problem about forking.=20

In section 6.9(Handling of Forked Requests) of this draft:=20

This specification only allows a single dialog to be constructed as a=20
   result of emitting an initial SUBSCRIBE request.  This guarantees=20
   that only a single notifier is generating notifications for a=20
   particular subscription to a particular user.=20

we could find that forked subscription is forbidden for this Event
package.=20

While in real IMS network, there are cases the div-services are deployed
in different ASs. Then, the subscription should be forked to the
different ASs, if there are more than one AS do div-services for the
specific service user.=20

We once considered one way to avoid the forking condition. The way is to
let the user send different subscription for different div-service. But
by the end user's viewpoint, he/she have no idea of the deployment of
the div-servies(in ASs). So, he/she does't know when should he/she send
separated subscription or not.  =20

So, I suggest to allow forking for this Event package.=20

Thanks,=20

Gao=20

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zip    : 210012
Tel    : 87211
Tel2   :(+86)-025-52877211
e_mail : gao.yang2@zte.com.cn
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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=20
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<p class=3DMsoNormal><span =
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color:#1F497D'>Hi<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Yes we did not want to support forking for the CDIV =
package
since the CDIV notification is only intended for the subscriber who =
subscribed
it . <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Regards<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Ranjit<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
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0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
gao.yang2@zte.com.cn
[mailto:gao.yang2@zte.com.cn] <br>
<b>Sent:</b> Monday, October 11, 2010 7:47 AM<br>
<b>To:</b> dispatch@ietf.org<br>
<b>Cc:</b> Avasarala Ranjit-A20990; jbakker@rim.com; =
drsubirsaha@yahoo.com<br>
<b>Subject:</b> Comments on =
draft-avasarala-dispatch-comm-div-notification-05
//About forking<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In our
implementation(the background system is IMS) of this draft, we find one =
problem
about forking.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In =
section
6.9(Handling of Forked Requests) of this draft:</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>This
specification only allows a single dialog to be constructed as a</span> =
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;
&nbsp;result of emitting an initial SUBSCRIBE request. &nbsp;This =
guarantees</span>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;
&nbsp;that only a single notifier is generating notifications for =
a</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;
&nbsp;particular subscription to a particular user.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>we =
could find
that forked subscription is forbidden for this Event package.</span> =
<br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>While =
in real
IMS network, there are cases the div-services are deployed in different =
ASs.
Then, the subscription should be forked to the different ASs, if there =
are more
than one AS do div-services for the specific service user.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We =
once
considered one way to avoid the forking condition. The way is to let the =
user
send different subscription for different div-service. But by the end =
user's
viewpoint, he/she have no idea of the deployment of the div-servies(in =
ASs).
So, he/she does't know when should he/she send separated subscription or =
not.
&nbsp;</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>So, I =
suggest
to allow forking for this Event package.</span> <br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks,</span=
> <br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Gao</span> =
<br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D<br>
Zip &nbsp; &nbsp;: 210012<br>
Tel &nbsp; &nbsp;: 87211<br>
Tel2 &nbsp; :(+86)-025-52877211<br>
e_mail : gao.yang2@zte.com.cn<br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</span><o:p></o:p></p>

<pre><o:p>&nbsp;</o:p></pre><pre>----------------------------------------=
----------------<o:p></o:p></pre><pre>ZTE&nbsp;Information&nbsp;Security&=
nbsp;Notice:&nbsp;The&nbsp;information&nbsp;contained&nbsp;in&nbsp;this&n=
bsp;mail&nbsp;is&nbsp;solely&nbsp;property&nbsp;of&nbsp;the&nbsp;sender's=
&nbsp;organization.&nbsp;This&nbsp;mail&nbsp;communication&nbsp;is&nbsp;c=
onfidential.&nbsp;Recipients&nbsp;named&nbsp;above&nbsp;are&nbsp;obligate=
d&nbsp;to&nbsp;maintain&nbsp;secrecy&nbsp;and&nbsp;are&nbsp;not&nbsp;perm=
itted&nbsp;to&nbsp;disclose&nbsp;the&nbsp;contents&nbsp;of&nbsp;this&nbsp=
;communication&nbsp;to&nbsp;others.<o:p></o:p></pre><pre>This&nbsp;email&=
nbsp;and&nbsp;any&nbsp;files&nbsp;transmitted&nbsp;with&nbsp;it&nbsp;are&=
nbsp;confidential&nbsp;and&nbsp;intended&nbsp;solely&nbsp;for&nbsp;the&nb=
sp;use&nbsp;of&nbsp;the&nbsp;individual&nbsp;or&nbsp;entity&nbsp;to&nbsp;=
whom&nbsp;they&nbsp;are&nbsp;addressed.&nbsp;If&nbsp;you&nbsp;have&nbsp;r=
eceived&nbsp;this&nbsp;email&nbsp;in&nbsp;error&nbsp;please&nbsp;notify&n=
bsp;the&nbsp;originator&nbsp;of&nbsp;the&nbsp;message.&nbsp;Any&nbsp;view=
s&nbsp;expressed&nbsp;in&nbsp;this&nbsp;message&nbsp;are&nbsp;those&nbsp;=
of&nbsp;the&nbsp;individual&nbsp;sender.<o:p></o:p></pre><pre>This&nbsp;m=
essage&nbsp;has&nbsp;been&nbsp;scanned&nbsp;for&nbsp;viruses&nbsp;and&nbs=
p;Spam&nbsp;by&nbsp;ZTE&nbsp;Anti-Spam&nbsp;system.<o:p></o:p></pre></div=
>

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X-MIMETrack: Serialize by Router on notes_smtp/zte_ltd(Release 8.5.1FP4|July 25, 2010) at 2010-10-12 21:01:38, Serialize complete at 2010-10-12 21:01:38
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [dispatch] Comments on the proposed Telepresence Charter
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Hi,

Some colleagues and I have reviewed the two telepresence drafts and
version 5 of the charter and have some comments on the charter.

To make our comments more understandable I think we should explain our
view on telepresence. First we are expecting great variations in the
capabilities of the participating terminals in a session. From personal
terminals for individual users up to multi-display, cameras and
microphone setups. We also expect that more than two end-points being
present in the same session. As we see it it will require good support
for centralized media handling in a mixer (MCU). We also see video
streams not only of high quality to be presented for maximum immersion,
but also lower qualities that can be used to monitor activity in other
rooms that aren't the ones selected for high quality.

Independent if one uses a central mixer or a full mesh there appears to
be a lack in media stream negotiation for how to negotiate what
capabilities the receiving client has in receiving, decoding and
rendering when it comes to number of simultaneous streams. Our
understanding is that RTP mandates any number of streams, while in
reality outside of conferencing that is in fact one single stream.
However with telepresence and conferencing there is a need to negotiate
the actual capability, especially between a mixer and client when it
comes to deliver multiple streams for various usages.

We also think there is missing functionality for expressing who are the
currently active speakers. The RTP CSRC field are not sufficient as they
express which end-points are included in the mix. However, as one likely
include in a mix more rooms than the active speakers to ensure that any
interruption and ambient sound are correctly captured an mechanism for
indicating the currently active speakers are needed. There is also need
to agree on the type of identifier to be used that allow for mapping to
full user names when appropriate.

The charter includes an item to work on control between client and a
mixer what streams should be delivered. We would like to extended this
to allow also the mixer to control the client on when streams are
delivered. In cases where there are more users then available display
area and a particular user is not actually included in any outgoing
mix/selection from the mixer then there is little point in that the
client consumes network resources. Thus we suggest that the stream
selection and control work is extended to allow also the mixer to pause
individual streams from the client to the mixer. Please note that such
control needs to be responsive, thus using the SIP signaling path for
this is to slow, it needs to be on the media path.

Another question to the group is, are there agreement on how streams
should organized into RTP sessions and how they are utilized? If
anything is documented I would appreciate a pointer. The media streams
are for different purposes from different end-points which are then
mixed by a central point and delivered to end-points. My impression for
example is that most proprietary system breaks the RTP semantics and use
individual sessions between the mixer and each end-point, instead of a
joint session. Here also different robustification tools for RTP such as
FEC and Retransmission needs to be considered. This might be work that
belongs in AVT, but are there need for further clarification into this
field?

Comments much appreciated and we can gladly suggest text additions to
the charter proposal.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

From peter.musgrave@magorcorp.com  Tue Oct 12 08:35:52 2010
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi Magnus,

I concur with your viewpoints on highly diverse endpoints, multi-point
(up to and including mesh connections).

I have more specific comments inline.

There is only one point below which I think requires language in the
charter. Please see if you agree.

Regards,

Peter Musgrave

On Tue, Oct 12, 2010 at 9:47 AM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> Hi,
>
> Some colleagues and I have reviewed the two telepresence drafts and
> version 5 of the charter and have some comments on the charter.
>
> To make our comments more understandable I think we should explain our
> view on telepresence. First we are expecting great variations in the
> capabilities of the participating terminals in a session. From personal
> terminals for individual users up to multi-display, cameras and
> microphone setups. We also expect that more than two end-points being
> present in the same session. As we see it it will require good support
> for centralized media handling in a mixer (MCU). We also see video
> streams not only of high quality to be presented for maximum immersion,
> but also lower qualities that can be used to monitor activity in other
> rooms that aren't the ones selected for high quality.
Agreed.

>
> Independent if one uses a central mixer or a full mesh there appears to
> be a lack in media stream negotiation for how to negotiate what
> capabilities the receiving client has in receiving, decoding and
> rendering when it comes to number of simultaneous streams. Our
> understanding is that RTP mandates any number of streams, while in
> reality outside of conferencing that is in fact one single stream.
> However with telepresence and conferencing there is a need to negotiate
> the actual capability, especially between a mixer and client when it
> comes to deliver multiple streams for various usages.
Agreed. The charter does speak to identifying endpoint capabilities
and I view what you describe here as the kind of meta-information
which needs to fit into that framework.

>
> We also think there is missing functionality for expressing who are the
> currently active speakers. The RTP CSRC field are not sufficient as they
> express which end-points are included in the mix. However, as one likely
> include in a mix more rooms than the active speakers to ensure that any
> interruption and ambient sound are correctly captured an mechanism for
> indicating the currently active speakers are needed. There is also need
> to agree on the type of identifier to be used that allow for mapping to
> full user names when appropriate.
I agree with this. In my opinion this is a separable problem since it
applies to generic audio conferencing as well. There is tangential
work in the conference event package and probably in the XCON stuff -
but I lack detailed background in XCON.

In the context of the TP charter I think the work on indicating which
audio stream is associated with which speaker (or part of the room) is
a slightly different issue - since it's about which stream and not
which speaker within a stream.

>
> The charter includes an item to work on control between client and a
> mixer what streams should be delivered. We would like to extended this
> to allow also the mixer to control the client on when streams are
> delivered. In cases where there are more users then available display
> area and a particular user is not actually included in any outgoing
> mix/selection from the mixer then there is little point in that the
> client consumes network resources. Thus we suggest that the stream
> selection and control work is extended to allow also the mixer to pause
> individual streams from the client to the mixer. Please note that such
> control needs to be responsive, thus using the SIP signaling path for
> this is to slow, it needs to be on the media path.
I concur.

The charter sentence:
 "The WG will create specifications for SIP-based conferencing systems
to enable communication of enough information about each media stream
so that each receiving system or bridge system can make reasonable
decisions about selecting and rendering media streams. This enables
systems to make display choices that optimize the "just like being
there" experience. "

is specific to selection of streams and not to the ability to control
them. I don't see anything here which prevents a source from not
sending selected streams - but I have no objection to making this more
explicit. How about adding:

"Selection choices will be communicated to the sending party to allow
the sender to optionally present information on which streams are in
use and to optionally suspend streams which are not in use." ??


>
> Another question to the group is, are there agreement on how streams
> should organized into RTP sessions and how they are utilized? If
> anything is documented I would appreciate a pointer. The media streams
> are for different purposes from different end-points which are then
> mixed by a central point and delivered to end-points. My impression for
> example is that most proprietary system breaks the RTP semantics and use
> individual sessions between the mixer and each end-point, instead of a
> joint session. Here also different robustification tools for RTP such as
> FEC and Retransmission needs to be considered. This might be work that
> belongs in AVT, but are there need for further clarification into this
> field?
I think this is something which will have to be thought through and
standardized if TP interoperability is to succeed. I would view the
semantics of how the streams are assigned in scope and the "plumbing"
either comes from the AVT/MMUSIC media cap stuff or any additions are
developed in those WGs.

>
> Comments much appreciated and we can gladly suggest text additions to
> the charter proposal.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From adam@nostrum.com  Tue Oct 12 09:04:10 2010
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Cc: dispatch@ietf.org, drsubirsaha@yahoo.com
Subject: Re: [dispatch] Comments on	draft-avasarala-dispatch-comm-div-notification-05 //About forking
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  Ranjit:

I would strongly suggest that you adopt Gao's proposal to allow multiple 
subscriptions to result from a fork.

The more event packages are defined, the more I think that any package 
that specifies that only one subscription can be established as the 
result of forking simply reflects a lack of imagination on the part of 
the draft author.

For example: the registration event package in RFC 3680 also chose to 
specify that only one subscription can be established. The explanation 
(in section 4.9) literally evaluates to "we can't imagine why this might 
be useful, so we're going to prevent it."

Fast-forward 4 years. We're doing work in MARTINI that is predicated on 
a network in which multiple registrars for an AOR are possible. We're 
having to update RFC3680 in the solution document to say, "whoops, that 
was wrong, we really do need multiple subscriptions sometimes."

I think RFC 3265 actually got this wrong. The suggestion that whole 
classes of events never make sense in the context of forking is looking 
increasingly like a failure to envision novel uses of the protocol. And, 
in fact, with GRUUs, it's no longer necessary to even have this kind of 
behavior. If a user doesn't want a SUBSCRIBE to fork, they use a GRUU.

In particular, since Gao has apparently identified a system right now 
that would benefit from multiple subscriptions, it doesn't even require 
that much imagination to forsee a system that might require this.

/a

On 10/12/10 06:51, Oct 12, Avasarala Ranjit-A20990 wrote:
>
> Hi
>
> Yes we did not want to support forking for the CDIV package since the 
> CDIV notification is only intended for the subscriber who subscribed it .
>
> Regards
>
> Ranjit
>
> *From:* gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]
> *Sent:* Monday, October 11, 2010 7:47 AM
> *To:* dispatch@ietf.org
> *Cc:* Avasarala Ranjit-A20990; jbakker@rim.com; drsubirsaha@yahoo.com
> *Subject:* Comments on 
> draft-avasarala-dispatch-comm-div-notification-05 //About forking
>
>
> In our implementation(the background system is IMS) of this draft, we 
> find one problem about forking.
>
> In section 6.9(Handling of Forked Requests) of this draft:
>
> This specification only allows a single dialog to be constructed as a
>    result of emitting an initial SUBSCRIBE request.  This guarantees
>    that only a single notifier is generating notifications for a
>    particular subscription to a particular user.
>
> we could find that forked subscription is forbidden for this Event 
> package.
>
> While in real IMS network, there are cases the div-services are 
> deployed in different ASs. Then, the subscription should be forked to 
> the different ASs, if there are more than one AS do div-services for 
> the specific service user.
>
> We once considered one way to avoid the forking condition. The way is 
> to let the user send different subscription for different div-service. 
> But by the end user's viewpoint, he/she have no idea of the deployment 
> of the div-servies(in ASs). So, he/she does't know when should he/she 
> send separated subscription or not.
>
> So, I suggest to allow forking for this Event package.
>
> Thanks,
>
> Gao
>
> ===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
> ===================================
>
>   
> --------------------------------------------------------
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    Ranjit:<br>
    <br>
    I would strongly suggest that you adopt Gao's proposal to allow
    multiple subscriptions to result from a fork.<br>
    <br>
    The more event packages are defined, the more I think that any
    package that specifies that only one subscription can be established
    as the result of forking simply reflects a lack of imagination on
    the part of the draft author.<br>
    <br>
    For example: the registration event package in RFC 3680 also chose
    to specify that only one subscription can be established. The
    explanation (in section 4.9) literally evaluates to "we can't
    imagine why this might be useful, so we're going to prevent it."<br>
    <br>
    Fast-forward 4 years. We're doing work in MARTINI that is predicated
    on a network in which multiple registrars for an AOR are possible.
    We're having to update RFC3680 in the solution document to say,
    "whoops, that was wrong, we really do need multiple subscriptions
    sometimes."<br>
    <br>
    I think RFC 3265 actually got this wrong. The suggestion that whole
    classes of events never make sense in the context of forking is
    looking increasingly like a failure to envision novel uses of the
    protocol. And, in fact, with GRUUs, it's no longer necessary to even
    have this kind of behavior. If a user doesn't want a SUBSCRIBE to
    fork, they use a GRUU.<br>
    <br>
    In particular, since Gao has apparently identified a system right
    now that would benefit from multiple subscriptions, it doesn't even
    require that much imagination to forsee a system that might require
    this.<br>
    <br>
    /a<br>
    <br>
    On 10/12/10 06:51, Oct 12, Avasarala Ranjit-A20990 wrote:
    <blockquote
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        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">Hi<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">Yes we did not want to support forking for the
            CDIV package
            since the CDIV notification is only intended for the
            subscriber who subscribed
            it . <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">Regards<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">Ranjit<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
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            73, 125);"><o:p>&nbsp;</o:p></span></p>
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          <p class="MsoNormal"><b><span style="font-size: 10pt;
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              style="font-size: 10pt; font-family:
              &quot;Tahoma&quot;,&quot;sans-serif&quot;;">
              <a class="moz-txt-link-abbreviated" href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a>
              [<a class="moz-txt-link-freetext" href="mailto:gao.yang2@zte.com.cn">mailto:gao.yang2@zte.com.cn</a>] <br>
              <b>Sent:</b> Monday, October 11, 2010 7:47 AM<br>
              <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
              <b>Cc:</b> Avasarala Ranjit-A20990; <a class="moz-txt-link-abbreviated" href="mailto:jbakker@rim.com">jbakker@rim.com</a>;
              <a class="moz-txt-link-abbreviated" href="mailto:drsubirsaha@yahoo.com">drsubirsaha@yahoo.com</a><br>
              <b>Subject:</b> Comments on
              draft-avasarala-dispatch-comm-div-notification-05
              //About forking<o:p></o:p></span></p>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal"><br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">In our
            implementation(the background system is IMS) of this draft,
            we find one problem
            about forking.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">In section
            6.9(Handling of Forked Requests) of this draft:</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">This
            specification only allows a single dialog to be constructed
            as a</span> <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">&nbsp;
            &nbsp;result of emitting an initial SUBSCRIBE request. &nbsp;This
            guarantees</span>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">&nbsp;
            &nbsp;that only a single notifier is generating notifications for
            a</span> <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">&nbsp;
            &nbsp;particular subscription to a particular user.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">we could find
            that forked subscription is forbidden for this Event
            package.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">While in real
            IMS network, there are cases the div-services are deployed
            in different ASs.
            Then, the subscription should be forked to the different
            ASs, if there are more
            than one AS do div-services for the specific service user.</span>
          <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">We once
            considered one way to avoid the forking condition. The way
            is to let the user
            send different subscription for different div-service. But
            by the end user's
            viewpoint, he/she have no idea of the deployment of the
            div-servies(in ASs).
            So, he/she does't know when should he/she send separated
            subscription or not.
            &nbsp;</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">So, I suggest
            to allow forking for this Event package.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">Thanks,</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">Gao</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">===================================<br>
            Zip &nbsp; &nbsp;: 210012<br>
            Tel &nbsp; &nbsp;: 87211<br>
            Tel2 &nbsp; :(+86)-025-52877211<br>
            e_mail : <a class="moz-txt-link-abbreviated" href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a><br>
            ===================================</span><o:p></o:p></p>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre>--------------------------------------------------------<o:p></o:p></pre>
        <pre>ZTE&nbsp;Information&nbsp;Security&nbsp;Notice:&nbsp;The&nbsp;information&nbsp;contained&nbsp;in&nbsp;this&nbsp;mail&nbsp;is&nbsp;solely&nbsp;property&nbsp;of&nbsp;the&nbsp;sender's&nbsp;organization.&nbsp;This&nbsp;mail&nbsp;communication&nbsp;is&nbsp;confidential.&nbsp;Recipients&nbsp;named&nbsp;above&nbsp;are&nbsp;obligated&nbsp;to&nbsp;maintain&nbsp;secrecy&nbsp;and&nbsp;are&nbsp;not&nbsp;permitted&nbsp;to&nbsp;disclose&nbsp;the&nbsp;contents&nbsp;of&nbsp;this&nbsp;communication&nbsp;to&nbsp;others.<o:p></o:p></pre>
        <pre>This&nbsp;email&nbsp;and&nbsp;any&nbsp;files&nbsp;transmitted&nbsp;with&nbsp;it&nbsp;are&nbsp;confidential&nbsp;and&nbsp;intended&nbsp;solely&nbsp;for&nbsp;the&nbsp;use&nbsp;of&nbsp;the&nbsp;individual&nbsp;or&nbsp;entity&nbsp;to&nbsp;whom&nbsp;they&nbsp;are&nbsp;addressed.&nbsp;If&nbsp;you&nbsp;have&nbsp;received&nbsp;this&nbsp;email&nbsp;in&nbsp;error&nbsp;please&nbsp;notify&nbsp;the&nbsp;originator&nbsp;of&nbsp;the&nbsp;message.&nbsp;Any&nbsp;views&nbsp;expressed&nbsp;in&nbsp;this&nbsp;message&nbsp;are&nbsp;those&nbsp;of&nbsp;the&nbsp;individual&nbsp;sender.<o:p></o:p></pre>
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      </div>
      <pre wrap="">
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Adam Roach" <adam@nostrum.com>
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Subject: Re: [dispatch] Comments on	draft-avasarala-dispatch-comm-div-notification-05 //About forking
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Hi Adam

=20

So will RFC 3265 be updated to reflect the change of SUBSCRIBE request
getting forked?

=20

=20

Regards

Ranjit

=20

=20

From: Adam Roach [mailto:adam@nostrum.com]=20
Sent: Tuesday, October 12, 2010 9:35 PM
To: Avasarala Ranjit-A20990
Cc: gao.yang2@zte.com.cn; dispatch@ietf.org; drsubirsaha@yahoo.com
Subject: Re: [dispatch] Comments on
draft-avasarala-dispatch-comm-div-notification-05 //About forking

=20

Ranjit:

I would strongly suggest that you adopt Gao's proposal to allow multiple
subscriptions to result from a fork.

The more event packages are defined, the more I think that any package
that specifies that only one subscription can be established as the
result of forking simply reflects a lack of imagination on the part of
the draft author.

For example: the registration event package in RFC 3680 also chose to
specify that only one subscription can be established. The explanation
(in section 4.9) literally evaluates to "we can't imagine why this might
be useful, so we're going to prevent it."

Fast-forward 4 years. We're doing work in MARTINI that is predicated on
a network in which multiple registrars for an AOR are possible. We're
having to update RFC3680 in the solution document to say, "whoops, that
was wrong, we really do need multiple subscriptions sometimes."

I think RFC 3265 actually got this wrong. The suggestion that whole
classes of events never make sense in the context of forking is looking
increasingly like a failure to envision novel uses of the protocol. And,
in fact, with GRUUs, it's no longer necessary to even have this kind of
behavior. If a user doesn't want a SUBSCRIBE to fork, they use a GRUU.

In particular, since Gao has apparently identified a system right now
that would benefit from multiple subscriptions, it doesn't even require
that much imagination to forsee a system that might require this.

/a

On 10/12/10 06:51, Oct 12, Avasarala Ranjit-A20990 wrote:=20

Hi

=20

Yes we did not want to support forking for the CDIV package since the
CDIV notification is only intended for the subscriber who subscribed it
.=20

=20

Regards

Ranjit

=20

From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
Sent: Monday, October 11, 2010 7:47 AM
To: dispatch@ietf.org
Cc: Avasarala Ranjit-A20990; jbakker@rim.com; drsubirsaha@yahoo.com
Subject: Comments on draft-avasarala-dispatch-comm-div-notification-05
//About forking

=20


In our implementation(the background system is IMS) of this draft, we
find one problem about forking.=20

In section 6.9(Handling of Forked Requests) of this draft:=20

This specification only allows a single dialog to be constructed as a=20
   result of emitting an initial SUBSCRIBE request.  This guarantees=20
   that only a single notifier is generating notifications for a=20
   particular subscription to a particular user.=20

we could find that forked subscription is forbidden for this Event
package.=20

While in real IMS network, there are cases the div-services are deployed
in different ASs. Then, the subscription should be forked to the
different ASs, if there are more than one AS do div-services for the
specific service user.=20

We once considered one way to avoid the forking condition. The way is to
let the user send different subscription for different div-service. But
by the end user's viewpoint, he/she have no idea of the deployment of
the div-servies(in ASs). So, he/she does't know when should he/she send
separated subscription or not.  =20

So, I suggest to allow forking for this Event package.=20

Thanks,=20

Gao=20

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zip    : 210012
Tel    : 87211
Tel2   :(+86)-025-52877211
e_mail : gao.yang2@zte.com.cn
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

=20
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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Adam<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>So will RFC 3265 be updated to reflect the change of =
SUBSCRIBE
request getting forked?<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Regards</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Ranjit</span><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> Adam Roach =
[mailto:adam@nostrum.com] <br>
<b>Sent:</b> Tuesday, October 12, 2010 9:35 PM<br>
<b>To:</b> Avasarala Ranjit-A20990<br>
<b>Cc:</b> gao.yang2@zte.com.cn; dispatch@ietf.org; =
drsubirsaha@yahoo.com<br>
<b>Subject:</b> Re: [dispatch] Comments on
draft-avasarala-dispatch-comm-div-notification-05 //About =
forking<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Ranjit:<br>
<br>
I would strongly suggest that you adopt Gao's proposal to allow multiple
subscriptions to result from a fork.<br>
<br>
The more event packages are defined, the more I think that any package =
that
specifies that only one subscription can be established as the result of
forking simply reflects a lack of imagination on the part of the draft =
author.<br>
<br>
For example: the registration event package in RFC 3680 also chose to =
specify
that only one subscription can be established. The explanation (in =
section 4.9)
literally evaluates to &quot;we can't imagine why this might be useful, =
so
we're going to prevent it.&quot;<br>
<br>
Fast-forward 4 years. We're doing work in MARTINI that is predicated on =
a
network in which multiple registrars for an AOR are possible. We're =
having to
update RFC3680 in the solution document to say, &quot;whoops, that was =
wrong,
we really do need multiple subscriptions sometimes.&quot;<br>
<br>
I think RFC 3265 actually got this wrong. The suggestion that whole =
classes of
events never make sense in the context of forking is looking =
increasingly like
a failure to envision novel uses of the protocol. And, in fact, with =
GRUUs,
it's no longer necessary to even have this kind of behavior. If a user =
doesn't
want a SUBSCRIBE to fork, they use a GRUU.<br>
<br>
In particular, since Gao has apparently identified a system right now =
that
would benefit from multiple subscriptions, it doesn't even require that =
much
imagination to forsee a system that might require this.<br>
<br>
/a<br>
<br>
On 10/12/10 06:51, Oct 12, Avasarala Ranjit-A20990 wrote: =
<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Hi</span><o=
:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>&nbsp;</spa=
n><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Yes
we did not want to support forking for the CDIV package since the CDIV
notification is only intended for the subscriber who subscribed it . =
</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>&nbsp;</spa=
n><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Regards</sp=
an><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>Ranjit</spa=
n><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>&nbsp;</spa=
n><o:p></o:p></p>

<div style=3D'border:none;border-top:solid windowtext =
1.0pt;padding:3.0pt 0in 0in 0in;
border-color:rgb(181,&#13;&#10;          196, 223) -moz-use-text-color =
-moz-use-text-color'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <a
href=3D"mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a> [<a
href=3D"mailto:gao.yang2@zte.com.cn">mailto:gao.yang2@zte.com.cn</a>] =
<br>
<b>Sent:</b> Monday, October 11, 2010 7:47 AM<br>
<b>To:</b> <a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<b>Cc:</b> Avasarala Ranjit-A20990; <a =
href=3D"mailto:jbakker@rim.com">jbakker@rim.com</a>;
<a href=3D"mailto:drsubirsaha@yahoo.com">drsubirsaha@yahoo.com</a><br>
<b>Subject:</b> Comments on =
draft-avasarala-dispatch-comm-div-notification-05
//About forking</span><o:p></o:p></p>

</div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In our
implementation(the background system is IMS) of this draft, we find one =
problem
about forking.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In =
section
6.9(Handling of Forked Requests) of this draft:</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>This
specification only allows a single dialog to be constructed as a</span> =
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;
&nbsp;result of emitting an initial SUBSCRIBE request. &nbsp;This =
guarantees</span>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;
&nbsp;that only a single notifier is generating notifications for =
a</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;
&nbsp;particular subscription to a particular user.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>we =
could find
that forked subscription is forbidden for this Event package.</span> =
<br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>While =
in real
IMS network, there are cases the div-services are deployed in different =
ASs.
Then, the subscription should be forked to the different ASs, if there =
are more
than one AS do div-services for the specific service user.</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We =
once
considered one way to avoid the forking condition. The way is to let the =
user
send different subscription for different div-service. But by the end =
user's
viewpoint, he/she have no idea of the deployment of the div-servies(in =
ASs).
So, he/she does't know when should he/she send separated subscription or =
not.
&nbsp;</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>So, I =
suggest
to allow forking for this Event package.</span> <br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks,</span=
> <br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Gao</span> =
<br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D<br>
Zip &nbsp; &nbsp;: 210012<br>
Tel &nbsp; &nbsp;: 87211<br>
Tel2 &nbsp; :(+86)-025-52877211<br>
e_mail : <a =
href=3D"mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a><br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</span><o:p></o:p></p>

<pre>&nbsp;<o:p></o:p></pre><pre>----------------------------------------=
----------------<o:p></o:p></pre><pre>ZTE&nbsp;Information&nbsp;Security&=
nbsp;Notice:&nbsp;The&nbsp;information&nbsp;contained&nbsp;in&nbsp;this&n=
bsp;mail&nbsp;is&nbsp;solely&nbsp;property&nbsp;of&nbsp;the&nbsp;sender's=
&nbsp;organization.&nbsp;This&nbsp;mail&nbsp;communication&nbsp;is&nbsp;c=
onfidential.&nbsp;Recipients&nbsp;named&nbsp;above&nbsp;are&nbsp;obligate=
d&nbsp;to&nbsp;maintain&nbsp;secrecy&nbsp;and&nbsp;are&nbsp;not&nbsp;perm=
itted&nbsp;to&nbsp;disclose&nbsp;the&nbsp;contents&nbsp;of&nbsp;this&nbsp=
;communication&nbsp;to&nbsp;others.<o:p></o:p></pre><pre>This&nbsp;email&=
nbsp;and&nbsp;any&nbsp;files&nbsp;transmitted&nbsp;with&nbsp;it&nbsp;are&=
nbsp;confidential&nbsp;and&nbsp;intended&nbsp;solely&nbsp;for&nbsp;the&nb=
sp;use&nbsp;of&nbsp;the&nbsp;individual&nbsp;or&nbsp;entity&nbsp;to&nbsp;=
whom&nbsp;they&nbsp;are&nbsp;addressed.&nbsp;If&nbsp;you&nbsp;have&nbsp;r=
eceived&nbsp;this&nbsp;email&nbsp;in&nbsp;error&nbsp;please&nbsp;notify&n=
bsp;the&nbsp;originator&nbsp;of&nbsp;the&nbsp;message.&nbsp;Any&nbsp;view=
s&nbsp;expressed&nbsp;in&nbsp;this&nbsp;message&nbsp;are&nbsp;those&nbsp;=
of&nbsp;the&nbsp;individual&nbsp;sender.<o:p></o:p></pre><pre>This&nbsp;m=
essage&nbsp;has&nbsp;been&nbsp;scanned&nbsp;for&nbsp;viruses&nbsp;and&nbs=
p;Spam&nbsp;by&nbsp;ZTE&nbsp;Anti-Spam&nbsp;system.<o:p></o:p></pre><pre>=
<o:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>_________________=
______________________________<o:p></o:p></pre><pre>dispatch mailing =
list<o:p></o:p></pre><pre><a
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><o:p></o:p></pre><=
pre><a
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</a><o:p></o:p></pre>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

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From adam@nostrum.com  Tue Oct 12 21:26:44 2010
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Date: Tue, 12 Oct 2010 23:27:55 -0500
From: Adam Roach <adam@nostrum.com>
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References: <OF984D775C.5A08B7FC-ON482577B9.000A03BA-482577B9.000CB1A0@zte.com.cn> <750BBC72E178114F9DC4872EBFF29A5B0ABAB06D@ZMY16EXM66.ds.mot.com> <4CB4873F.9030101@nostrum.com> <750BBC72E178114F9DC4872EBFF29A5B0ABAB13C@ZMY16EXM66.ds.mot.com>
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Cc: dispatch@ietf.org, drsubirsaha@yahoo.com
Subject: Re: [dispatch] Comments on	draft-avasarala-dispatch-comm-div-notification-05 //About forking
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  The 3265bis document already requires the use of GRUUs. I'm going to 
look carefully at the section regarding forking to see what reasonable 
changes we can make there to prevent this kind of problem in the future.

But, unless you're going to wait for 3265bis to conclude, 
div-notification will be based on 3265. Your best bet is to clearly 
specify that multiple subscriptions may legally be established by a 
single SUBSCRIBE request.

/a

On 10/12/10 23:20, Oct 12, Avasarala Ranjit-A20990 wrote:
>
> Hi Adam
>
> So will RFC 3265 be updated to reflect the change of SUBSCRIBE request 
> getting forked?
>
> Regards
>
> Ranjit
>
> *From:* Adam Roach [mailto:adam@nostrum.com]
> *Sent:* Tuesday, October 12, 2010 9:35 PM
> *To:* Avasarala Ranjit-A20990
> *Cc:* gao.yang2@zte.com.cn; dispatch@ietf.org; drsubirsaha@yahoo.com
> *Subject:* Re: [dispatch] Comments on 
> draft-avasarala-dispatch-comm-div-notification-05 //About forking
>
> Ranjit:
>
> I would strongly suggest that you adopt Gao's proposal to allow 
> multiple subscriptions to result from a fork.
>
> The more event packages are defined, the more I think that any package 
> that specifies that only one subscription can be established as the 
> result of forking simply reflects a lack of imagination on the part of 
> the draft author.
>
> For example: the registration event package in RFC 3680 also chose to 
> specify that only one subscription can be established. The explanation 
> (in section 4.9) literally evaluates to "we can't imagine why this 
> might be useful, so we're going to prevent it."
>
> Fast-forward 4 years. We're doing work in MARTINI that is predicated 
> on a network in which multiple registrars for an AOR are possible. 
> We're having to update RFC3680 in the solution document to say, 
> "whoops, that was wrong, we really do need multiple subscriptions 
> sometimes."
>
> I think RFC 3265 actually got this wrong. The suggestion that whole 
> classes of events never make sense in the context of forking is 
> looking increasingly like a failure to envision novel uses of the 
> protocol. And, in fact, with GRUUs, it's no longer necessary to even 
> have this kind of behavior. If a user doesn't want a SUBSCRIBE to 
> fork, they use a GRUU.
>
> In particular, since Gao has apparently identified a system right now 
> that would benefit from multiple subscriptions, it doesn't even 
> require that much imagination to forsee a system that might require this.
>
> /a
>
> On 10/12/10 06:51, Oct 12, Avasarala Ranjit-A20990 wrote:
>
> Hi
>
> Yes we did not want to support forking for the CDIV package since the 
> CDIV notification is only intended for the subscriber who subscribed it .
>
> Regards
>
> Ranjit
>
> *From:* gao.yang2@zte.com.cn <mailto:gao.yang2@zte.com.cn> 
> [mailto:gao.yang2@zte.com.cn]
> *Sent:* Monday, October 11, 2010 7:47 AM
> *To:* dispatch@ietf.org <mailto:dispatch@ietf.org>
> *Cc:* Avasarala Ranjit-A20990; jbakker@rim.com 
> <mailto:jbakker@rim.com>; drsubirsaha@yahoo.com 
> <mailto:drsubirsaha@yahoo.com>
> *Subject:* Comments on 
> draft-avasarala-dispatch-comm-div-notification-05 //About forking
>
>
> In our implementation(the background system is IMS) of this draft, we 
> find one problem about forking.
>
> In section 6.9(Handling of Forked Requests) of this draft:
>
> This specification only allows a single dialog to be constructed as a
>    result of emitting an initial SUBSCRIBE request.  This guarantees
>    that only a single notifier is generating notifications for a
>    particular subscription to a particular user.
>
> we could find that forked subscription is forbidden for this Event 
> package.
>
> While in real IMS network, there are cases the div-services are 
> deployed in different ASs. Then, the subscription should be forked to 
> the different ASs, if there are more than one AS do div-services for 
> the specific service user.
>
> We once considered one way to avoid the forking condition. The way is 
> to let the user send different subscription for different div-service. 
> But by the end user's viewpoint, he/she have no idea of the deployment 
> of the div-servies(in ASs). So, he/she does't know when should he/she 
> send separated subscription or not.
>
> So, I suggest to allow forking for this Event package.
>
> Thanks,
>
> Gao
>
> ===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn <mailto:gao.yang2@zte.com.cn>
> ===================================
>
>   
> --------------------------------------------------------
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> https://www.ietf.org/mailman/listinfo/dispatch
>


--------------070509030508030900060807
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    The 3265bis document already requires the use of GRUUs. I'm going to
    look carefully at the section regarding forking to see what
    reasonable changes we can make there to prevent this kind of problem
    in the future.<br>
    <br>
    But, unless you're going to wait for 3265bis to conclude,
    div-notification will be based on 3265. Your best bet is to clearly
    specify that multiple subscriptions may legally be established by a
    single SUBSCRIBE request.<br>
    <br>
    /a<br>
    <br>
    On 10/12/10 23:20, Oct 12, Avasarala Ranjit-A20990 wrote:
    <blockquote
cite="mid:750BBC72E178114F9DC4872EBFF29A5B0ABAB13C@ZMY16EXM66.ds.mot.com"
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      <div class="Section1">
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">Hi Adam<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">So will RFC 3265 be updated to reflect the change
            of SUBSCRIBE
            request getting forked?<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;; color: rgb(31, 73,
            125);">Regards</span><span style="color: rgb(31, 73, 125);"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;; color: rgb(31, 73,
            125);">Ranjit</span><span style="font-size: 11pt;
            font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
            color: rgb(31, 73, 125);"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border-right: medium none; border-width: 1pt
            medium medium; border-style: solid none none; border-color:
            rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color;
            padding: 3pt 0in 0in;">
            <p class="MsoNormal"><b><span style="font-size: 10pt;
                  font-family:
                  &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                  windowtext;">From:</span></b><span style="font-size:
                10pt; font-family:
                &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                windowtext;"> Adam Roach [<a class="moz-txt-link-freetext" href="mailto:adam@nostrum.com">mailto:adam@nostrum.com</a>] <br>
                <b>Sent:</b> Tuesday, October 12, 2010 9:35 PM<br>
                <b>To:</b> Avasarala Ranjit-A20990<br>
                <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a>; <a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a>;
                <a class="moz-txt-link-abbreviated" href="mailto:drsubirsaha@yahoo.com">drsubirsaha@yahoo.com</a><br>
                <b>Subject:</b> Re: [dispatch] Comments on
                draft-avasarala-dispatch-comm-div-notification-05
                //About forking<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">Ranjit:<br>
          <br>
          I would strongly suggest that you adopt Gao's proposal to
          allow multiple
          subscriptions to result from a fork.<br>
          <br>
          The more event packages are defined, the more I think that any
          package that
          specifies that only one subscription can be established as the
          result of
          forking simply reflects a lack of imagination on the part of
          the draft author.<br>
          <br>
          For example: the registration event package in RFC 3680 also
          chose to specify
          that only one subscription can be established. The explanation
          (in section 4.9)
          literally evaluates to "we can't imagine why this might be
          useful, so
          we're going to prevent it."<br>
          <br>
          Fast-forward 4 years. We're doing work in MARTINI that is
          predicated on a
          network in which multiple registrars for an AOR are possible.
          We're having to
          update RFC3680 in the solution document to say, "whoops, that
          was wrong,
          we really do need multiple subscriptions sometimes."<br>
          <br>
          I think RFC 3265 actually got this wrong. The suggestion that
          whole classes of
          events never make sense in the context of forking is looking
          increasingly like
          a failure to envision novel uses of the protocol. And, in
          fact, with GRUUs,
          it's no longer necessary to even have this kind of behavior.
          If a user doesn't
          want a SUBSCRIBE to fork, they use a GRUU.<br>
          <br>
          In particular, since Gao has apparently identified a system
          right now that
          would benefit from multiple subscriptions, it doesn't even
          require that much
          imagination to forsee a system that might require this.<br>
          <br>
          /a<br>
          <br>
          On 10/12/10 06:51, Oct 12, Avasarala Ranjit-A20990 wrote: <o:p></o:p></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;;">Hi</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;;">&nbsp;</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;;">Yes
            we did not want to support forking for the CDIV package
            since the CDIV
            notification is only intended for the subscriber who
            subscribed it . </span><o:p></o:p></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;;">&nbsp;</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;;">Regards</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;;">Ranjit</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;;">&nbsp;</span><o:p></o:p></p>
        <div style="border-right: medium none; border-width: 1pt medium
          medium; border-style: solid none none; padding: 3pt 0in 0in;
          border-color: rgb(181, 196, 223) -moz-use-text-color
          -moz-use-text-color;">
          <p class="MsoNormal"><b><span style="font-size: 10pt;
                font-family: &quot;Tahoma&quot;,&quot;sans-serif&quot;;">From:</span></b><span
              style="font-size: 10pt; font-family:
              &quot;Tahoma&quot;,&quot;sans-serif&quot;;"> <a
                moz-do-not-send="true"
                href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a>
              [<a moz-do-not-send="true"
                href="mailto:gao.yang2@zte.com.cn">mailto:gao.yang2@zte.com.cn</a>]
              <br>
              <b>Sent:</b> Monday, October 11, 2010 7:47 AM<br>
              <b>To:</b> <a moz-do-not-send="true"
                href="mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
              <b>Cc:</b> Avasarala Ranjit-A20990; <a
                moz-do-not-send="true" href="mailto:jbakker@rim.com">jbakker@rim.com</a>;
              <a moz-do-not-send="true"
                href="mailto:drsubirsaha@yahoo.com">drsubirsaha@yahoo.com</a><br>
              <b>Subject:</b> Comments on
              draft-avasarala-dispatch-comm-div-notification-05
              //About forking</span><o:p></o:p></p>
        </div>
        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
        <p class="MsoNormal"><br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">In our
            implementation(the background system is IMS) of this draft,
            we find one problem
            about forking.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">In section
            6.9(Handling of Forked Requests) of this draft:</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">This
            specification only allows a single dialog to be constructed
            as a</span> <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">&nbsp;
            &nbsp;result of emitting an initial SUBSCRIBE request. &nbsp;This
            guarantees</span>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">&nbsp;
            &nbsp;that only a single notifier is generating notifications for
            a</span> <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">&nbsp;
            &nbsp;particular subscription to a particular user.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">we could find
            that forked subscription is forbidden for this Event
            package.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">While in real
            IMS network, there are cases the div-services are deployed
            in different ASs.
            Then, the subscription should be forked to the different
            ASs, if there are more
            than one AS do div-services for the specific service user.</span>
          <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">We once
            considered one way to avoid the forking condition. The way
            is to let the user
            send different subscription for different div-service. But
            by the end user's
            viewpoint, he/she have no idea of the deployment of the
            div-servies(in ASs).
            So, he/she does't know when should he/she send separated
            subscription or not.
            &nbsp;</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">So, I suggest
            to allow forking for this Event package.</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">Thanks,</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">Gao</span> <br>
          <br>
          <span style="font-size: 10pt; font-family:
            &quot;Arial&quot;,&quot;sans-serif&quot;;">===================================<br>
            Zip &nbsp; &nbsp;: 210012<br>
            Tel &nbsp; &nbsp;: 87211<br>
            Tel2 &nbsp; :(+86)-025-52877211<br>
            e_mail : <a moz-do-not-send="true"
              href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a><br>
            ===================================</span><o:p></o:p></p>
        <pre>&nbsp;<o:p></o:p></pre>
        <pre>--------------------------------------------------------<o:p></o:p></pre>
        <pre>ZTE&nbsp;Information&nbsp;Security&nbsp;Notice:&nbsp;The&nbsp;information&nbsp;contained&nbsp;in&nbsp;this&nbsp;mail&nbsp;is&nbsp;solely&nbsp;property&nbsp;of&nbsp;the&nbsp;sender's&nbsp;organization.&nbsp;This&nbsp;mail&nbsp;communication&nbsp;is&nbsp;confidential.&nbsp;Recipients&nbsp;named&nbsp;above&nbsp;are&nbsp;obligated&nbsp;to&nbsp;maintain&nbsp;secrecy&nbsp;and&nbsp;are&nbsp;not&nbsp;permitted&nbsp;to&nbsp;disclose&nbsp;the&nbsp;contents&nbsp;of&nbsp;this&nbsp;communication&nbsp;to&nbsp;others.<o:p></o:p></pre>
        <pre>This&nbsp;email&nbsp;and&nbsp;any&nbsp;files&nbsp;transmitted&nbsp;with&nbsp;it&nbsp;are&nbsp;confidential&nbsp;and&nbsp;intended&nbsp;solely&nbsp;for&nbsp;the&nbsp;use&nbsp;of&nbsp;the&nbsp;individual&nbsp;or&nbsp;entity&nbsp;to&nbsp;whom&nbsp;they&nbsp;are&nbsp;addressed.&nbsp;If&nbsp;you&nbsp;have&nbsp;received&nbsp;this&nbsp;email&nbsp;in&nbsp;error&nbsp;please&nbsp;notify&nbsp;the&nbsp;originator&nbsp;of&nbsp;the&nbsp;message.&nbsp;Any&nbsp;views&nbsp;expressed&nbsp;in&nbsp;this&nbsp;message&nbsp;are&nbsp;those&nbsp;of&nbsp;the&nbsp;individual&nbsp;sender.<o:p></o:p></pre>
        <pre>This&nbsp;message&nbsp;has&nbsp;been&nbsp;scanned&nbsp;for&nbsp;viruses&nbsp;and&nbsp;Spam&nbsp;by&nbsp;ZTE&nbsp;Anti-Spam&nbsp;system.<o:p></o:p></pre>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre>_______________________________________________<o:p></o:p></pre>
        <pre>dispatch mailing list<o:p></o:p></pre>
        <pre><a moz-do-not-send="true" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a><o:p></o:p></pre>
        <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a><o:p></o:p></pre>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------070509030508030900060807--

From magnus.westerlund@ericsson.com  Wed Oct 13 02:09:59 2010
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Date: Wed, 13 Oct 2010 11:11:11 +0200
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Hi Peter,

Good that we are in agreement on a number of things. Response and
clarifications inline.

Peter Musgrave skrev 2010-10-12 17:37:
>>
>> Independent if one uses a central mixer or a full mesh there appears to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.

> Agreed. The charter does speak to identifying endpoint capabilities
> and I view what you describe here as the kind of meta-information
> which needs to fit into that framework.

Okay, as long as this is included in the thinking about capabilities I
am fine. However, the charter proposal isn't particular detailed in this
regards. But I guess the following charter text is matching this:

"Information between sources and sinks about media stream capabilities
will be exchanged."

>>
>> We also think there is missing functionality for expressing who are the
>> currently active speakers. The RTP CSRC field are not sufficient as they
>> express which end-points are included in the mix. However, as one likely
>> include in a mix more rooms than the active speakers to ensure that any
>> interruption and ambient sound are correctly captured an mechanism for
>> indicating the currently active speakers are needed. There is also need
>> to agree on the type of identifier to be used that allow for mapping to
>> full user names when appropriate.

> I agree with this. In my opinion this is a separable problem since it
> applies to generic audio conferencing as well. There is tangential
> work in the conference event package and probably in the XCON stuff -
> but I lack detailed background in XCON.
> 
> In the context of the TP charter I think the work on indicating which
> audio stream is associated with which speaker (or part of the room) is
> a slightly different issue - since it's about which stream and not
> which speaker within a stream.

I should probably have been clearer. I am concerned that even on media
stream level there is missing functionality to indicate which streams
that contains active speakers and has been included in a mix produced by
the mixer. From our side we see a difference between being included in
the mix and being tagged as currently an active speaker.

I think this is currently missing functionality for which requirements
needs to be agreed on and then a solution developed. However, the
solution might better fit another WG, like AVT. But I think that is for
later determination. I think the requirements phase makes sense to have
in the Telepresence group.

I do agree that such functionality do have a wider usage scope than just
telepresence and can be used in any case of centralized mixers where
special activity needs to be pointed out, with audio being the foremost
example of such media.

> 
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to pause
>> individual streams from the client to the mixer. Please note that such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
> I concur.
> 
> The charter sentence:
>  "The WG will create specifications for SIP-based conferencing systems
> to enable communication of enough information about each media stream
> so that each receiving system or bridge system can make reasonable
> decisions about selecting and rendering media streams. This enables
> systems to make display choices that optimize the "just like being
> there" experience. "
> 
> is specific to selection of streams and not to the ability to control
> them. I don't see anything here which prevents a source from not
> sending selected streams - but I have no objection to making this more
> explicit. How about adding:
> 
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
> 

When I wrote the above comment I was mostly considering the following
chart text bullet:

* Which sources a receiver wants to receive.  For example, it might want
the source for the left camera, or might want the source chosen by VAD
(Voice Activity Detection).

I think your proposed text combined with the above bullet gives all the
room I see needed for media stream control. However, maybe a
modification of the above to make it clear.

* Which sources a receiver or mixer wants to have transmitted.  For
example, it might want the source for the left camera, or might want the
source chosen by VAD (Voice Activity Detection).

> 
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression for
>> example is that most proprietary system breaks the RTP semantics and use
>> individual sessions between the mixer and each end-point, instead of a
>> joint session. Here also different robustification tools for RTP such as
>> FEC and Retransmission needs to be considered. This might be work that
>> belongs in AVT, but are there need for further clarification into this
>> field?

> I think this is something which will have to be thought through and
> standardized if TP interoperability is to succeed. I would view the
> semantics of how the streams are assigned in scope and the "plumbing"
> either comes from the AVT/MMUSIC media cap stuff or any additions are
> developed in those WGs.
> 

I agree with that the right place to do specification work is likely
these two groups. However, I think telepresence needs to provide
requirements for such work. Making such requirement development explicit
in the charter would make it clear that we do expect work to happen
somewhere else. At the same time it is after all possible to request
that another WG's charter is updated to cover such work.

The WG may identify interoperability obstacles in existing open
standards. The WG may develop requirements to be communicated to other
IETF WGs or Standards Forums as a kind request to try to meet these
requirements.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

From peter.musgrave@magorcorp.com  Wed Oct 13 03:28:48 2010
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HI Magnus,

I have no issues with your proposed changes to the wording.

As for the rest I believe we are in agreement.

Regards,

Peter

On Wed, Oct 13, 2010 at 5:11 AM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> Hi Peter,
>
> Good that we are in agreement on a number of things. Response and
> clarifications inline.
>
> Peter Musgrave skrev 2010-10-12 17:37:
>>>
>>> Independent if one uses a central mixer or a full mesh there appears to
>>> be a lack in media stream negotiation for how to negotiate what
>>> capabilities the receiving client has in receiving, decoding and
>>> rendering when it comes to number of simultaneous streams. Our
>>> understanding is that RTP mandates any number of streams, while in
>>> reality outside of conferencing that is in fact one single stream.
>>> However with telepresence and conferencing there is a need to negotiate
>>> the actual capability, especially between a mixer and client when it
>>> comes to deliver multiple streams for various usages.
>
>> Agreed. The charter does speak to identifying endpoint capabilities
>> and I view what you describe here as the kind of meta-information
>> which needs to fit into that framework.
>
> Okay, as long as this is included in the thinking about capabilities I
> am fine. However, the charter proposal isn't particular detailed in this
> regards. But I guess the following charter text is matching this:
>
> "Information between sources and sinks about media stream capabilities
> will be exchanged."
>
>>>
>>> We also think there is missing functionality for expressing who are the
>>> currently active speakers. The RTP CSRC field are not sufficient as the=
y
>>> express which end-points are included in the mix. However, as one likel=
y
>>> include in a mix more rooms than the active speakers to ensure that any
>>> interruption and ambient sound are correctly captured an mechanism for
>>> indicating the currently active speakers are needed. There is also need
>>> to agree on the type of identifier to be used that allow for mapping to
>>> full user names when appropriate.
>
>> I agree with this. In my opinion this is a separable problem since it
>> applies to generic audio conferencing as well. There is tangential
>> work in the conference event package and probably in the XCON stuff -
>> but I lack detailed background in XCON.
>>
>> In the context of the TP charter I think the work on indicating which
>> audio stream is associated with which speaker (or part of the room) is
>> a slightly different issue - since it's about which stream and not
>> which speaker within a stream.
>
> I should probably have been clearer. I am concerned that even on media
> stream level there is missing functionality to indicate which streams
> that contains active speakers and has been included in a mix produced by
> the mixer. From our side we see a difference between being included in
> the mix and being tagged as currently an active speaker.
>
> I think this is currently missing functionality for which requirements
> needs to be agreed on and then a solution developed. However, the
> solution might better fit another WG, like AVT. But I think that is for
> later determination. I think the requirements phase makes sense to have
> in the Telepresence group.
>
> I do agree that such functionality do have a wider usage scope than just
> telepresence and can be used in any case of centralized mixers where
> special activity needs to be pointed out, with audio being the foremost
> example of such media.
>
>>
>>>
>>> The charter includes an item to work on control between client and a
>>> mixer what streams should be delivered. We would like to extended this
>>> to allow also the mixer to control the client on when streams are
>>> delivered. In cases where there are more users then available display
>>> area and a particular user is not actually included in any outgoing
>>> mix/selection from the mixer then there is little point in that the
>>> client consumes network resources. Thus we suggest that the stream
>>> selection and control work is extended to allow also the mixer to pause
>>> individual streams from the client to the mixer. Please note that such
>>> control needs to be responsive, thus using the SIP signaling path for
>>> this is to slow, it needs to be on the media path.
>> I concur.
>>
>> The charter sentence:
>> =A0"The WG will create specifications for SIP-based conferencing systems
>> to enable communication of enough information about each media stream
>> so that each receiving system or bridge system can make reasonable
>> decisions about selecting and rendering media streams. This enables
>> systems to make display choices that optimize the "just like being
>> there" experience. "
>>
>> is specific to selection of streams and not to the ability to control
>> them. I don't see anything here which prevents a source from not
>> sending selected streams - but I have no objection to making this more
>> explicit. How about adding:
>>
>> "Selection choices will be communicated to the sending party to allow
>> the sender to optionally present information on which streams are in
>> use and to optionally suspend streams which are not in use." ??
>>
>
> When I wrote the above comment I was mostly considering the following
> chart text bullet:
>
> * Which sources a receiver wants to receive. =A0For example, it might wan=
t
> the source for the left camera, or might want the source chosen by VAD
> (Voice Activity Detection).
>
> I think your proposed text combined with the above bullet gives all the
> room I see needed for media stream control. However, maybe a
> modification of the above to make it clear.
>
> * Which sources a receiver or mixer wants to have transmitted. =A0For
> example, it might want the source for the left camera, or might want the
> source chosen by VAD (Voice Activity Detection).
>
>>
>>>
>>> Another question to the group is, are there agreement on how streams
>>> should organized into RTP sessions and how they are utilized? If
>>> anything is documented I would appreciate a pointer. The media streams
>>> are for different purposes from different end-points which are then
>>> mixed by a central point and delivered to end-points. My impression for
>>> example is that most proprietary system breaks the RTP semantics and us=
e
>>> individual sessions between the mixer and each end-point, instead of a
>>> joint session. Here also different robustification tools for RTP such a=
s
>>> FEC and Retransmission needs to be considered. This might be work that
>>> belongs in AVT, but are there need for further clarification into this
>>> field?
>
>> I think this is something which will have to be thought through and
>> standardized if TP interoperability is to succeed. I would view the
>> semantics of how the streams are assigned in scope and the "plumbing"
>> either comes from the AVT/MMUSIC media cap stuff or any additions are
>> developed in those WGs.
>>
>
> I agree with that the right place to do specification work is likely
> these two groups. However, I think telepresence needs to provide
> requirements for such work. Making such requirement development explicit
> in the charter would make it clear that we do expect work to happen
> somewhere else. At the same time it is after all possible to request
> that another WG's charter is updated to cover such work.
>
> The WG may identify interoperability obstacles in existing open
> standards. The WG may develop requirements to be communicated to other
> IETF WGs or Standards Forums as a kind request to try to meet these
> requirements.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>

From allyn@cisco.com  Wed Oct 13 09:46:23 2010
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From: "Allyn Romanow (allyn)" <allyn@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi Magnus,
I was really glad that you had a chance to carefully consider the =
proposed work.
Like Peter, I too agree with most of your comments.
Specific points below.

Best regards,
Allyn

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
Sent: Tuesday, October 12, 2010 6:47 AM
To: dispatch@ietf.org
Subject: [dispatch] Comments on the proposed Telepresence Charter

Hi,

Some colleagues and I have reviewed the two telepresence drafts and
version 5 of the charter and have some comments on the charter.

To make our comments more understandable I think we should explain our
view on telepresence. First we are expecting great variations in the
capabilities of the participating terminals in a session. From personal
terminals for individual users up to multi-display, cameras and
microphone setups. We also expect that more than two end-points being
present in the same session. As we see it it will require good support
for centralized media handling in a mixer (MCU). We also see video
streams not only of high quality to be presented for maximum immersion,
but also lower qualities that can be used to monitor activity in other
rooms that aren't the ones selected for high quality.

[AR] Yes, absolutely.

Independent if one uses a central mixer or a full mesh there appears to
be a lack in media stream negotiation for how to negotiate what
capabilities the receiving client has in receiving, decoding and
rendering when it comes to number of simultaneous streams. Our
understanding is that RTP mandates any number of streams, while in
reality outside of conferencing that is in fact one single stream.
However with telepresence and conferencing there is a need to negotiate
the actual capability, especially between a mixer and client when it
comes to deliver multiple streams for various usages.

[AR] Yes, agree - as you saw, this is covered in the charter.

We also think there is missing functionality for expressing who are the
currently active speakers. The RTP CSRC field are not sufficient as they
express which end-points are included in the mix. However, as one likely
include in a mix more rooms than the active speakers to ensure that any
interruption and ambient sound are correctly captured an mechanism for
indicating the currently active speakers are needed. There is also need
to agree on the type of identifier to be used that allow for mapping to
full user names when appropriate.

[AR] Yes.. Both of these points are covered in the charter too, right?

The charter includes an item to work on control between client and a
mixer what streams should be delivered. We would like to extended this
to allow also the mixer to control the client on when streams are
delivered. In cases where there are more users then available display
area and a particular user is not actually included in any outgoing
mix/selection from the mixer then there is little point in that the
client consumes network resources. Thus we suggest that the stream
selection and control work is extended to allow also the mixer to pause
individual streams from the client to the mixer. Please note that such
control needs to be responsive, thus using the SIP signaling path for
this is to slow, it needs to be on the media path.

[AR} I agree with your point that it would be good to pause the sender, =
but I think that falls
 under the category of "continuous conference control" which is out of =
the scope of this charter.
 Such a feedback mechanism falls probably under AVT.
I think it would be good to add a sentence saying something like, As =
control issues=20
necessary for interoperability arise, they will be directed =
appropriately.
What do you think?


Another question to the group is, are there agreement on how streams
should organized into RTP sessions and how they are utilized? If
anything is documented I would appreciate a pointer. The media streams
are for different purposes from different end-points which are then
mixed by a central point and delivered to end-points. My impression for
example is that most proprietary system breaks the RTP semantics and use
individual sessions between the mixer and each end-point, instead of a
joint session. Here also different robustification tools for RTP such as
FEC and Retransmission needs to be considered. This might be work that
belongs in AVT, but are there need for further clarification into this
field?

[AR} Agree this needs to be worked out for Telepresence, and that any =
activity about it would go to AVT and/or MMUSIC.

Comments much appreciated and we can gladly suggest text additions to
the charter proposal.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From peter.musgrave@magorcorp.com  Thu Oct 14 02:24:11 2010
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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HI Allyn/Magnus,

W.R.T

"[AR} I agree with your point that it would be good to pause the
sender, but I think that falls
 under the category of "continuous conference control" which is out of
the scope of this charter.
 Such a feedback mechanism falls probably under AVT.
I think it would be good to add a sentence saying something like, As
control issues
necessary for interoperability arise, they will be directed appropriately.
What do you think?"

I think these words leave room to *not* do what Magnus is suggesting.
e.g. If I have a 1 screen home TP system getting three streams from a
3 camera system and selecting only one it is not really "necessary" to
tell the 3 camera system to pause two streams - but it is a really
good idea to have a way to that! I don't view this is a continuous
conference control issue (since that gets into floor control which I
agree is out of scope) but more like onhold/offhold type negotiation.
I think this is to some extent a plumbing issue (and based on the
desired response time SIP or media path messages will be required).

So I would still like to see a statement which makes it clear that it
not just an issue of receiver selection but there is a need to
communicate that selection back to the sender. The current text does
not say this. Hence my suggestion:

"Selection choices will be communicated to the sending party to allow
the sender to optionally present information on which streams are in
use and to optionally suspend streams which are not in use." ??

I'm open to other words or some indication of which language in the
current charter ensures this is in scope.

Regards,

Peter Musgrave




On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) <allyn@cisco.com> w=
rote:
> Hi Magnus,
> I was really glad that you had a chance to carefully consider the propose=
d work.
> Like Peter, I too agree with most of your comments.
> Specific points below.
>
> Best regards,
> Allyn
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf Of Magnus Westerlund
> Sent: Tuesday, October 12, 2010 6:47 AM
> To: dispatch@ietf.org
> Subject: [dispatch] Comments on the proposed Telepresence Charter
>
> Hi,
>
> Some colleagues and I have reviewed the two telepresence drafts and
> version 5 of the charter and have some comments on the charter.
>
> To make our comments more understandable I think we should explain our
> view on telepresence. First we are expecting great variations in the
> capabilities of the participating terminals in a session. From personal
> terminals for individual users up to multi-display, cameras and
> microphone setups. We also expect that more than two end-points being
> present in the same session. As we see it it will require good support
> for centralized media handling in a mixer (MCU). We also see video
> streams not only of high quality to be presented for maximum immersion,
> but also lower qualities that can be used to monitor activity in other
> rooms that aren't the ones selected for high quality.
>
> [AR] Yes, absolutely.
>
> Independent if one uses a central mixer or a full mesh there appears to
> be a lack in media stream negotiation for how to negotiate what
> capabilities the receiving client has in receiving, decoding and
> rendering when it comes to number of simultaneous streams. Our
> understanding is that RTP mandates any number of streams, while in
> reality outside of conferencing that is in fact one single stream.
> However with telepresence and conferencing there is a need to negotiate
> the actual capability, especially between a mixer and client when it
> comes to deliver multiple streams for various usages.
>
> [AR] Yes, agree - as you saw, this is covered in the charter.
>
> We also think there is missing functionality for expressing who are the
> currently active speakers. The RTP CSRC field are not sufficient as they
> express which end-points are included in the mix. However, as one likely
> include in a mix more rooms than the active speakers to ensure that any
> interruption and ambient sound are correctly captured an mechanism for
> indicating the currently active speakers are needed. There is also need
> to agree on the type of identifier to be used that allow for mapping to
> full user names when appropriate.
>
> [AR] Yes.. Both of these points are covered in the charter too, right?
>
> The charter includes an item to work on control between client and a
> mixer what streams should be delivered. We would like to extended this
> to allow also the mixer to control the client on when streams are
> delivered. In cases where there are more users then available display
> area and a particular user is not actually included in any outgoing
> mix/selection from the mixer then there is little point in that the
> client consumes network resources. Thus we suggest that the stream
> selection and control work is extended to allow also the mixer to pause
> individual streams from the client to the mixer. Please note that such
> control needs to be responsive, thus using the SIP signaling path for
> this is to slow, it needs to be on the media path.
>
> [AR} I agree with your point that it would be good to pause the sender, b=
ut I think that falls
> =A0under the category of "continuous conference control" which is out of =
the scope of this charter.
> =A0Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As cont=
rol issues
> necessary for interoperability arise, they will be directed appropriately=
.
> What do you think?
>
>
> Another question to the group is, are there agreement on how streams
> should organized into RTP sessions and how they are utilized? If
> anything is documented I would appreciate a pointer. The media streams
> are for different purposes from different end-points which are then
> mixed by a central point and delivered to end-points. My impression for
> example is that most proprietary system breaks the RTP semantics and use
> individual sessions between the mixer and each end-point, instead of a
> joint session. Here also different robustification tools for RTP such as
> FEC and Retransmission needs to be considered. This might be work that
> belongs in AVT, but are there need for further clarification into this
> field?
>
> [AR} Agree this needs to be worked out for Telepresence, and that any act=
ivity about it would go to AVT and/or MMUSIC.
>
> Comments much appreciated and we can gladly suggest text additions to
> the charter proposal.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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From: "Mike Hammer (hmmr)" <hmmr@cisco.com>
To: "Peter Musgrave" <peter.musgrave@magorcorp.com>, "Allyn Romanow (allyn)" <allyn@cisco.com>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Peter,

I have not issue with your proposed words, since they are essentially =
asking for a "mute" button for all types of communications used by a =
sender.  This may complicate the algorithm for all the receive systems =
to know what to display, if for example, you continue to speak, but turn =
off the video transmission.  A "mute all" option might be more polite.

I did have issue with this comment:  "more like onhold/offhold type =
negotiation"

I am not sure I would agree with this. =20

There needs to be a distinction between when a call starts and when a =
call stops, the call being *all* communications (sessions) between two =
points.  Granted putting a call on hold seems like floor control, but =
the participants of the call on hold are effectively out of the call =
(left the room) until brought off-hold.  This is SIP level action.

There needs to be a clear distinction between negotiating capabilities, =
aka knowing what capabilities an endpoint has, and using those =
capabilities during a call.  That learning process is done via SDP.

The latter part seems to be floor control, continuous conference =
control, or whatever you want to call it.  There seems to be splitting =
of hairs over whether the floor control is manual or automatic =
(determined by the loudest speaker).  This may be AVT.

In summary, you have room access, you define capabilities, you have =
"speaker" conflict resolution control.  (Note, point-to-point calls are =
devolution since resolution of conflict is a non-problem with only one =
opposing party.)

Did I miss some other nuance?

Mike


-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Peter Musgrave
Sent: Thursday, October 14, 2010 5:25 AM
To: Allyn Romanow (allyn)
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Comments on the proposed Telepresence Charter

HI Allyn/Magnus,

W.R.T

"[AR} I agree with your point that it would be good to pause the
sender, but I think that falls
 under the category of "continuous conference control" which is out of
the scope of this charter.
 Such a feedback mechanism falls probably under AVT.
I think it would be good to add a sentence saying something like, As
control issues
necessary for interoperability arise, they will be directed =
appropriately.
What do you think?"

I think these words leave room to *not* do what Magnus is suggesting.
e.g. If I have a 1 screen home TP system getting three streams from a
3 camera system and selecting only one it is not really "necessary" to
tell the 3 camera system to pause two streams - but it is a really
good idea to have a way to that! I don't view this is a continuous
conference control issue (since that gets into floor control which I
agree is out of scope) but more like onhold/offhold type negotiation.
I think this is to some extent a plumbing issue (and based on the
desired response time SIP or media path messages will be required).

So I would still like to see a statement which makes it clear that it
not just an issue of receiver selection but there is a need to
communicate that selection back to the sender. The current text does
not say this. Hence my suggestion:

"Selection choices will be communicated to the sending party to allow
the sender to optionally present information on which streams are in
use and to optionally suspend streams which are not in use." ??

I'm open to other words or some indication of which language in the
current charter ensures this is in scope.

Regards,

Peter Musgrave




On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) =
<allyn@cisco.com> wrote:
> Hi Magnus,
> I was really glad that you had a chance to carefully consider the =
proposed work.
> Like Peter, I too agree with most of your comments.
> Specific points below.
>
> Best regards,
> Allyn
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
> Sent: Tuesday, October 12, 2010 6:47 AM
> To: dispatch@ietf.org
> Subject: [dispatch] Comments on the proposed Telepresence Charter
>
> Hi,
>
> Some colleagues and I have reviewed the two telepresence drafts and
> version 5 of the charter and have some comments on the charter.
>
> To make our comments more understandable I think we should explain our
> view on telepresence. First we are expecting great variations in the
> capabilities of the participating terminals in a session. From =
personal
> terminals for individual users up to multi-display, cameras and
> microphone setups. We also expect that more than two end-points being
> present in the same session. As we see it it will require good support
> for centralized media handling in a mixer (MCU). We also see video
> streams not only of high quality to be presented for maximum =
immersion,
> but also lower qualities that can be used to monitor activity in other
> rooms that aren't the ones selected for high quality.
>
> [AR] Yes, absolutely.
>
> Independent if one uses a central mixer or a full mesh there appears =
to
> be a lack in media stream negotiation for how to negotiate what
> capabilities the receiving client has in receiving, decoding and
> rendering when it comes to number of simultaneous streams. Our
> understanding is that RTP mandates any number of streams, while in
> reality outside of conferencing that is in fact one single stream.
> However with telepresence and conferencing there is a need to =
negotiate
> the actual capability, especially between a mixer and client when it
> comes to deliver multiple streams for various usages.
>
> [AR] Yes, agree - as you saw, this is covered in the charter.
>
> We also think there is missing functionality for expressing who are =
the
> currently active speakers. The RTP CSRC field are not sufficient as =
they
> express which end-points are included in the mix. However, as one =
likely
> include in a mix more rooms than the active speakers to ensure that =
any
> interruption and ambient sound are correctly captured an mechanism for
> indicating the currently active speakers are needed. There is also =
need
> to agree on the type of identifier to be used that allow for mapping =
to
> full user names when appropriate.
>
> [AR] Yes.. Both of these points are covered in the charter too, right?
>
> The charter includes an item to work on control between client and a
> mixer what streams should be delivered. We would like to extended this
> to allow also the mixer to control the client on when streams are
> delivered. In cases where there are more users then available display
> area and a particular user is not actually included in any outgoing
> mix/selection from the mixer then there is little point in that the
> client consumes network resources. Thus we suggest that the stream
> selection and control work is extended to allow also the mixer to =
pause
> individual streams from the client to the mixer. Please note that such
> control needs to be responsive, thus using the SIP signaling path for
> this is to slow, it needs to be on the media path.
>
> [AR} I agree with your point that it would be good to pause the =
sender, but I think that falls
> =A0under the category of "continuous conference control" which is out =
of the scope of this charter.
> =A0Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As =
control issues
> necessary for interoperability arise, they will be directed =
appropriately.
> What do you think?
>
>
> Another question to the group is, are there agreement on how streams
> should organized into RTP sessions and how they are utilized? If
> anything is documented I would appreciate a pointer. The media streams
> are for different purposes from different end-points which are then
> mixed by a central point and delivered to end-points. My impression =
for
> example is that most proprietary system breaks the RTP semantics and =
use
> individual sessions between the mixer and each end-point, instead of a
> joint session. Here also different robustification tools for RTP such =
as
> FEC and Retransmission needs to be considered. This might be work that
> belongs in AVT, but are there need for further clarification into this
> field?
>
> [AR} Agree this needs to be worked out for Telepresence, and that any =
activity about it would go to AVT and/or MMUSIC.
>
> Comments much appreciated and we can gladly suggest text additions to
> the charter proposal.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
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Subject: [dispatch] Fwd: New Non-WG Mailing List: maitai -- Multi-stream Attributes for Improving Telepresence Application Interoperability
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FYI...this is for discussion (as noted below) of technical issues and
not for charter discussion. Please continue to post comments wrt the
charter on the DISPATCH WG mailing list.

Regards,
Mary.
DISPATCH WG co-chair


---------- Forwarded message ----------
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Date: Wed, Oct 13, 2010 at 6:55 PM
Subject: New Non-WG Mailing List: maitai -- Multi-stream Attributes
for          Improving Telepresence Application Interoperability
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Cc: maitai@ietf.org, allyn@cisco.com, mary.ietf.barnes@gmail.com


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From spromano@unina.it  Thu Oct 14 07:48:35 2010
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi,

let me jump into the "mute" (aka on-hold/off-hold) discussion. I would 
suggest that BFCP can be fruitfully employed here. BFCP can surely 
provide the mixer with a chance to add/remove streams to the mixed flow 
by using a policy-based approach. With respect to the issue raised in 
this thread, which I report here for brevity:

> We would like to extended this
> to allow also the mixer to control the client on when streams are
> delivered. In cases where there are more users then available display
> area and a particular user is not actually included in any outgoing
> mix/selection from the mixer then there is little point in that the
> client consumes network resources. Thus we suggest that the stream
> selection and control work is extended to allow also the mixer to pause
> individual streams from the client to the mixer. Please note that such
> control needs to be responsive, thus using the SIP signaling path for
> this is to slow, it needs to be on the media path.

I would say that also here BFCP might play a role. If the mixer, acting 
as a floor control server, removes the floor from a BFCP-aware client, 
such client is expected (even though it is not required) to stop sending 
data to the mixer, since it can be sure that such contribution will not 
be taken into account when building up the mix.

Simon

Il 14/10/2010 15.46, Mike Hammer (hmmr) ha scritto:
> Peter,
>
> I have not issue with your proposed words, since they are essentially asking for a "mute" button for all types of communications used by a sender.  This may complicate the algorithm for all the receive systems to know what to display, if for example, you continue to speak, but turn off the video transmission.  A "mute all" option might be more polite.
>
> I did have issue with this comment:  "more like onhold/offhold type negotiation"
>
> I am not sure I would agree with this.
>
> There needs to be a distinction between when a call starts and when a call stops, the call being *all* communications (sessions) between two points.  Granted putting a call on hold seems like floor control, but the participants of the call on hold are effectively out of the call (left the room) until brought off-hold.  This is SIP level action.
>
> There needs to be a clear distinction between negotiating capabilities, aka knowing what capabilities an endpoint has, and using those capabilities during a call.  That learning process is done via SDP.
>
> The latter part seems to be floor control, continuous conference control, or whatever you want to call it.  There seems to be splitting of hairs over whether the floor control is manual or automatic (determined by the loudest speaker).  This may be AVT.
>
> In summary, you have room access, you define capabilities, you have "speaker" conflict resolution control.  (Note, point-to-point calls are devolution since resolution of conflict is a non-problem with only one opposing party.)
>
> Did I miss some other nuance?
>
> Mike
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Musgrave
> Sent: Thursday, October 14, 2010 5:25 AM
> To: Allyn Romanow (allyn)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> HI Allyn/Magnus,
>
> W.R.T
>
> "[AR} I agree with your point that it would be good to pause the
> sender, but I think that falls
>   under the category of "continuous conference control" which is out of
> the scope of this charter.
>   Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As
> control issues
> necessary for interoperability arise, they will be directed appropriately.
> What do you think?"
>
> I think these words leave room to *not* do what Magnus is suggesting.
> e.g. If I have a 1 screen home TP system getting three streams from a
> 3 camera system and selecting only one it is not really "necessary" to
> tell the 3 camera system to pause two streams - but it is a really
> good idea to have a way to that! I don't view this is a continuous
> conference control issue (since that gets into floor control which I
> agree is out of scope) but more like onhold/offhold type negotiation.
> I think this is to some extent a plumbing issue (and based on the
> desired response time SIP or media path messages will be required).
>
> So I would still like to see a statement which makes it clear that it
> not just an issue of receiver selection but there is a need to
> communicate that selection back to the sender. The current text does
> not say this. Hence my suggestion:
>
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
>
> I'm open to other words or some indication of which language in the
> current charter ensures this is in scope.
>
> Regards,
>
> Peter Musgrave
>
>
>
>
> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn)<allyn@cisco.com>  wrote:
>    
>> Hi Magnus,
>> I was really glad that you had a chance to carefully consider the proposed work.
>> Like Peter, I too agree with most of your comments.
>> Specific points below.
>>
>> Best regards,
>> Allyn
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Magnus Westerlund
>> Sent: Tuesday, October 12, 2010 6:47 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>
>> Hi,
>>
>> Some colleagues and I have reviewed the two telepresence drafts and
>> version 5 of the charter and have some comments on the charter.
>>
>> To make our comments more understandable I think we should explain our
>> view on telepresence. First we are expecting great variations in the
>> capabilities of the participating terminals in a session. From personal
>> terminals for individual users up to multi-display, cameras and
>> microphone setups. We also expect that more than two end-points being
>> present in the same session. As we see it it will require good support
>> for centralized media handling in a mixer (MCU). We also see video
>> streams not only of high quality to be presented for maximum immersion,
>> but also lower qualities that can be used to monitor activity in other
>> rooms that aren't the ones selected for high quality.
>>
>> [AR] Yes, absolutely.
>>
>> Independent if one uses a central mixer or a full mesh there appears to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.
>>
>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>
>> We also think there is missing functionality for expressing who are the
>> currently active speakers. The RTP CSRC field are not sufficient as they
>> express which end-points are included in the mix. However, as one likely
>> include in a mix more rooms than the active speakers to ensure that any
>> interruption and ambient sound are correctly captured an mechanism for
>> indicating the currently active speakers are needed. There is also need
>> to agree on the type of identifier to be used that allow for mapping to
>> full user names when appropriate.
>>
>> [AR] Yes.. Both of these points are covered in the charter too, right?
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to pause
>> individual streams from the client to the mixer. Please note that such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
>>
>> [AR} I agree with your point that it would be good to pause the sender, but I think that falls
>>   under the category of "continuous conference control" which is out of the scope of this charter.
>>   Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As control issues
>> necessary for interoperability arise, they will be directed appropriately.
>> What do you think?
>>
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression for
>> example is that most proprietary system breaks the RTP semantics and use
>> individual sessions between the mixer and each end-point, instead of a
>> joint session. Here also different robustification tools for RTP such as
>> FEC and Retransmission needs to be considered. This might be work that
>> belongs in AVT, but are there need for further clarification into this
>> field?
>>
>> [AR} Agree this needs to be worked out for Telepresence, and that any activity about it would go to AVT and/or MMUSIC.
>>
>> Comments much appreciated and we can gladly suggest text additions to
>> the charter proposal.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>      
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>    

-- 
                             _\\|//_
                             ( O-O )
    ~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~
                     Simon Pietro Romano
               Universita' di Napoli Federico II
                  Computer Science Department
         Phone: +39 081 7683823 -- Fax: +39 081 7684219
                 e-mail: spromano@unina.it
           http://www.comics.unina.it/simonpietro.romano

     <<Molti mi dicono che lo scoraggiamento è l'alibi degli
    idioti. Ci rifletto un istante; e mi scoraggio>>. Magritte.
                          oooO
    ~~~~~~~~~~~~~~~~~~~~~~(   )~~ Oooo~~~~~~~~~~~~~~~~~~~~~~~~~
                           \ (    (   )
                            \_)    ) /
                                  (_/



From peter.musgrave@magorcorp.com  Thu Oct 14 07:53:16 2010
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi Mike,

I agree the onhld/offhold analogy is a poor one. It is a suspension of
a call - which is not intended.

Let me re-phrase and see if we agree:

A sends B three video streams and one audio stream .
B has one display and opts to display one of the three streams.
There is a mechanism for B to tell A that this has happened (and to
say when it wishes to display some other stream from A).

Do we think that the work to do this type of stream-suspension is
floor control? (My opinion is no. I view floor control more as who is
the source for a given stream and not a selection of an agreed upon
stream).

Do we think this work is in scope for Telepresence Interoperability? I
think this is an essential element to deployable TP operation
especially as in expands to "at home" use. If an architecture in which
a room sends all video all the time (even though it's not being used)
is viewed as necessary to avoid complexity/meet responsiveness
requirements is thought to be ok then I'd like to make sure we are
clear that that these limitations exist upfront.

My preference is obviously to permit this feedback and tweak the
proposed charter.

Other opinions?

Regards,


Peter Musgrave

On Thu, Oct 14, 2010 at 9:46 AM, Mike Hammer (hmmr) <hmmr@cisco.com> wrote:
> Peter,
>
> I have not issue with your proposed words, since they are essentially ask=
ing for a "mute" button for all types of communications used by a sender. =
=A0This may complicate the algorithm for all the receive systems to know wh=
at to display, if for example, you continue to speak, but turn off the vide=
o transmission. =A0A "mute all" option might be more polite.
>
> I did have issue with this comment: =A0"more like onhold/offhold type neg=
otiation"
>
> I am not sure I would agree with this.
>
> There needs to be a distinction between when a call starts and when a cal=
l stops, the call being *all* communications (sessions) between two points.=
 =A0Granted putting a call on hold seems like floor control, but the partic=
ipants of the call on hold are effectively out of the call (left the room) =
until brought off-hold. =A0This is SIP level action.
>
> There needs to be a clear distinction between negotiating capabilities, a=
ka knowing what capabilities an endpoint has, and using those capabilities =
during a call. =A0That learning process is done via SDP.
>
> The latter part seems to be floor control, continuous conference control,=
 or whatever you want to call it. =A0There seems to be splitting of hairs o=
ver whether the floor control is manual or automatic (determined by the lou=
dest speaker). =A0This may be AVT.
>
> In summary, you have room access, you define capabilities, you have "spea=
ker" conflict resolution control. =A0(Note, point-to-point calls are devolu=
tion since resolution of conflict is a non-problem with only one opposing p=
arty.)
>
> Did I miss some other nuance?
>
> Mike
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf Of Peter Musgrave
> Sent: Thursday, October 14, 2010 5:25 AM
> To: Allyn Romanow (allyn)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> HI Allyn/Magnus,
>
> W.R.T
>
> "[AR} I agree with your point that it would be good to pause the
> sender, but I think that falls
> =A0under the category of "continuous conference control" which is out of
> the scope of this charter.
> =A0Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As
> control issues
> necessary for interoperability arise, they will be directed appropriately=
.
> What do you think?"
>
> I think these words leave room to *not* do what Magnus is suggesting.
> e.g. If I have a 1 screen home TP system getting three streams from a
> 3 camera system and selecting only one it is not really "necessary" to
> tell the 3 camera system to pause two streams - but it is a really
> good idea to have a way to that! I don't view this is a continuous
> conference control issue (since that gets into floor control which I
> agree is out of scope) but more like onhold/offhold type negotiation.
> I think this is to some extent a plumbing issue (and based on the
> desired response time SIP or media path messages will be required).
>
> So I would still like to see a statement which makes it clear that it
> not just an issue of receiver selection but there is a need to
> communicate that selection back to the sender. The current text does
> not say this. Hence my suggestion:
>
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
>
> I'm open to other words or some indication of which language in the
> current charter ensures this is in scope.
>
> Regards,
>
> Peter Musgrave
>
>
>
>
> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) <allyn@cisco.com>=
 wrote:
>> Hi Magnus,
>> I was really glad that you had a chance to carefully consider the propos=
ed work.
>> Like Peter, I too agree with most of your comments.
>> Specific points below.
>>
>> Best regards,
>> Allyn
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Be=
half Of Magnus Westerlund
>> Sent: Tuesday, October 12, 2010 6:47 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>
>> Hi,
>>
>> Some colleagues and I have reviewed the two telepresence drafts and
>> version 5 of the charter and have some comments on the charter.
>>
>> To make our comments more understandable I think we should explain our
>> view on telepresence. First we are expecting great variations in the
>> capabilities of the participating terminals in a session. From personal
>> terminals for individual users up to multi-display, cameras and
>> microphone setups. We also expect that more than two end-points being
>> present in the same session. As we see it it will require good support
>> for centralized media handling in a mixer (MCU). We also see video
>> streams not only of high quality to be presented for maximum immersion,
>> but also lower qualities that can be used to monitor activity in other
>> rooms that aren't the ones selected for high quality.
>>
>> [AR] Yes, absolutely.
>>
>> Independent if one uses a central mixer or a full mesh there appears to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.
>>
>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>
>> We also think there is missing functionality for expressing who are the
>> currently active speakers. The RTP CSRC field are not sufficient as they
>> express which end-points are included in the mix. However, as one likely
>> include in a mix more rooms than the active speakers to ensure that any
>> interruption and ambient sound are correctly captured an mechanism for
>> indicating the currently active speakers are needed. There is also need
>> to agree on the type of identifier to be used that allow for mapping to
>> full user names when appropriate.
>>
>> [AR] Yes.. Both of these points are covered in the charter too, right?
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to pause
>> individual streams from the client to the mixer. Please note that such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
>>
>> [AR} I agree with your point that it would be good to pause the sender, =
but I think that falls
>> =A0under the category of "continuous conference control" which is out of=
 the scope of this charter.
>> =A0Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As con=
trol issues
>> necessary for interoperability arise, they will be directed appropriatel=
y.
>> What do you think?
>>
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression for
>> example is that most proprietary system breaks the RTP semantics and use
>> individual sessions between the mixer and each end-point, instead of a
>> joint session. Here also different robustification tools for RTP such as
>> FEC and Retransmission needs to be considered. This might be work that
>> belongs in AVT, but are there need for further clarification into this
>> field?
>>
>> [AR} Agree this needs to be worked out for Telepresence, and that any ac=
tivity about it would go to AVT and/or MMUSIC.
>>
>> Comments much appreciated and we can gladly suggest text additions to
>> the charter proposal.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
>> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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From: "Mike Hammer (hmmr)" <hmmr@cisco.com>
To: "Simon Pietro Romano" <spromano@unina.it>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Simon,

Agree that SIP is likely too slow.  Plus my concern that it's connection =
into other back-end systems (e.g. accounting) might affect those =
secondary systems.

I agree that BFCP might be able to provide needed capabilities, and thus =
should not be excluded by the charter.

However, when studying the problem space, I would hope we would not be =
too constrained by earlier designs, if more recent experience might =
suggest a different approach. =20

For example, we gained advantages by separating the signaling and media =
planes for VoIP.  Here too, we might find that it is useful to separate =
out into different protocols the aspects of:
- declaring media control powers (who is capable of MCU mixing)
- selection/assignment of powers to a particular device
- passing of powers (e.g. host or presenter) from one device to another
- execution of powers at a particular point
- response to power assertion by non-power devices.

Note, that some of the above could be done in SDP with Offer/Answer, =
some could be power agreements in some signaling protocol, and some =
could be media plane controls.  In essence, the (floor control) policy =
control and policy execution could be split into two or three planes.

My point here is not to agree to any such thing at this time, only the =
recognition that the existing protocols may prove deficient, and though =
overlapping, the WG should be allowed the latitude to do the smart thing =
and not be overly constrained.  I am sure the AD's can help guide the =
WG, if they see it getting too far out of bounds.

Thanks,
Mike


-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Simon Pietro Romano
Sent: Thursday, October 14, 2010 10:49 AM
To: dispatch@ietf.org
Subject: Re: [dispatch] Comments on the proposed Telepresence Charter

Hi,

let me jump into the "mute" (aka on-hold/off-hold) discussion. I would=20
suggest that BFCP can be fruitfully employed here. BFCP can surely=20
provide the mixer with a chance to add/remove streams to the mixed flow=20
by using a policy-based approach. With respect to the issue raised in=20
this thread, which I report here for brevity:

> We would like to extended this
> to allow also the mixer to control the client on when streams are
> delivered. In cases where there are more users then available display
> area and a particular user is not actually included in any outgoing
> mix/selection from the mixer then there is little point in that the
> client consumes network resources. Thus we suggest that the stream
> selection and control work is extended to allow also the mixer to =
pause
> individual streams from the client to the mixer. Please note that such
> control needs to be responsive, thus using the SIP signaling path for
> this is to slow, it needs to be on the media path.

I would say that also here BFCP might play a role. If the mixer, acting=20
as a floor control server, removes the floor from a BFCP-aware client,=20
such client is expected (even though it is not required) to stop sending =

data to the mixer, since it can be sure that such contribution will not=20
be taken into account when building up the mix.

Simon

Il 14/10/2010 15.46, Mike Hammer (hmmr) ha scritto:
> Peter,
>
> I have not issue with your proposed words, since they are essentially =
asking for a "mute" button for all types of communications used by a =
sender.  This may complicate the algorithm for all the receive systems =
to know what to display, if for example, you continue to speak, but turn =
off the video transmission.  A "mute all" option might be more polite.
>
> I did have issue with this comment:  "more like onhold/offhold type =
negotiation"
>
> I am not sure I would agree with this.
>
> There needs to be a distinction between when a call starts and when a =
call stops, the call being *all* communications (sessions) between two =
points.  Granted putting a call on hold seems like floor control, but =
the participants of the call on hold are effectively out of the call =
(left the room) until brought off-hold.  This is SIP level action.
>
> There needs to be a clear distinction between negotiating =
capabilities, aka knowing what capabilities an endpoint has, and using =
those capabilities during a call.  That learning process is done via =
SDP.
>
> The latter part seems to be floor control, continuous conference =
control, or whatever you want to call it.  There seems to be splitting =
of hairs over whether the floor control is manual or automatic =
(determined by the loudest speaker).  This may be AVT.

>
> In summary, you have room access, you define capabilities, you have =
"speaker" conflict resolution control.  (Note, point-to-point calls are =
devolution since resolution of conflict is a non-problem with only one =
opposing party.)
>
> Did I miss some other nuance?
>
> Mike
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Peter Musgrave
> Sent: Thursday, October 14, 2010 5:25 AM
> To: Allyn Romanow (allyn)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> HI Allyn/Magnus,
>
> W.R.T
>
> "[AR} I agree with your point that it would be good to pause the
> sender, but I think that falls
>   under the category of "continuous conference control" which is out =
of
> the scope of this charter.
>   Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As
> control issues
> necessary for interoperability arise, they will be directed =
appropriately.
> What do you think?"
>
> I think these words leave room to *not* do what Magnus is suggesting.
> e.g. If I have a 1 screen home TP system getting three streams from a
> 3 camera system and selecting only one it is not really "necessary" to
> tell the 3 camera system to pause two streams - but it is a really
> good idea to have a way to that! I don't view this is a continuous
> conference control issue (since that gets into floor control which I
> agree is out of scope) but more like onhold/offhold type negotiation.
> I think this is to some extent a plumbing issue (and based on the
> desired response time SIP or media path messages will be required).
>
> So I would still like to see a statement which makes it clear that it
> not just an issue of receiver selection but there is a need to
> communicate that selection back to the sender. The current text does
> not say this. Hence my suggestion:
>
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
>
> I'm open to other words or some indication of which language in the
> current charter ensures this is in scope.
>
> Regards,
>
> Peter Musgrave
>
>
>
>
> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow =
(allyn)<allyn@cisco.com>  wrote:
>   =20
>> Hi Magnus,
>> I was really glad that you had a chance to carefully consider the =
proposed work.
>> Like Peter, I too agree with most of your comments.
>> Specific points below.
>>
>> Best regards,
>> Allyn
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
>> Sent: Tuesday, October 12, 2010 6:47 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>
>> Hi,
>>
>> Some colleagues and I have reviewed the two telepresence drafts and
>> version 5 of the charter and have some comments on the charter.
>>
>> To make our comments more understandable I think we should explain =
our
>> view on telepresence. First we are expecting great variations in the
>> capabilities of the participating terminals in a session. From =
personal
>> terminals for individual users up to multi-display, cameras and
>> microphone setups. We also expect that more than two end-points being
>> present in the same session. As we see it it will require good =
support
>> for centralized media handling in a mixer (MCU). We also see video
>> streams not only of high quality to be presented for maximum =
immersion,
>> but also lower qualities that can be used to monitor activity in =
other
>> rooms that aren't the ones selected for high quality.
>>
>> [AR] Yes, absolutely.
>>
>> Independent if one uses a central mixer or a full mesh there appears =
to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to =
negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.
>>
>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>
>> We also think there is missing functionality for expressing who are =
the
>> currently active speakers. The RTP CSRC field are not sufficient as =
they
>> express which end-points are included in the mix. However, as one =
likely
>> include in a mix more rooms than the active speakers to ensure that =
any
>> interruption and ambient sound are correctly captured an mechanism =
for
>> indicating the currently active speakers are needed. There is also =
need
>> to agree on the type of identifier to be used that allow for mapping =
to
>> full user names when appropriate.
>>
>> [AR] Yes.. Both of these points are covered in the charter too, =
right?
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended =
this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to =
pause
>> individual streams from the client to the mixer. Please note that =
such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
>>
>> [AR} I agree with your point that it would be good to pause the =
sender, but I think that falls
>>   under the category of "continuous conference control" which is out =
of the scope of this charter.
>>   Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As =
control issues
>> necessary for interoperability arise, they will be directed =
appropriately.
>> What do you think?
>>
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media =
streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression =
for
>> example is that most proprietary system breaks the RTP semantics and =
use
>> individual sessions between the mixer and each end-point, instead of =
a
>> joint session. Here also different robustification tools for RTP such =
as
>> FEC and Retransmission needs to be considered. This might be work =
that
>> belongs in AVT, but are there need for further clarification into =
this
>> field?
>>
>> [AR} Agree this needs to be worked out for Telepresence, and that any =
activity about it would go to AVT and/or MMUSIC.
>>
>> Comments much appreciated and we can gladly suggest text additions to
>> the charter proposal.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> =
----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> =
----------------------------------------------------------------------
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>     =20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>   =20

--=20
                             _\\|//_
                             ( O-O )
    ~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~
                     Simon Pietro Romano
               Universita' di Napoli Federico II
                  Computer Science Department
         Phone: +39 081 7683823 -- Fax: +39 081 7684219
                 e-mail: spromano@unina.it
           http://www.comics.unina.it/simonpietro.romano

     <<Molti mi dicono che lo scoraggiamento =E8 l'alibi degli
    idioti. Ci rifletto un istante; e mi scoraggio>>. Magritte.
                          oooO
    ~~~~~~~~~~~~~~~~~~~~~~(   )~~ Oooo~~~~~~~~~~~~~~~~~~~~~~~~~
                           \ (    (   )
                            \_)    ) /
                                  (_/


_______________________________________________
dispatch mailing list
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From lorenzo@meetecho.com  Thu Oct 14 08:57:11 2010
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Date: Thu, 14 Oct 2010 17:55:49 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Peter Musgrave <peter.musgrave@magorcorp.com>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi Peter,

I also see it as an important use case to have in the charter. I agree
that floor control (e.g. BFCP) may not be needed in here (not an
authorization scenario) even though it may come in handy and as such
could at least be considered when we'll be talking about solutions.

>From a "technical" point of view, I guess that, if we find a way to
identify streams (what is what) then even just SDP updates may do the
trick then: e.g., video 1 recvonly, video 2 and 3 inactive. But this
would imply implicit feedback (I'm just telling I want one video out
of three, and not why), and so may be restrictive for the use case.

L.


On Thu, 14 Oct 2010 10:54:30 -0400
Peter Musgrave <peter.musgrave@magorcorp.com> wrote:

> Hi Mike,
>=20
> I agree the onhld/offhold analogy is a poor one. It is a suspension of
> a call - which is not intended.
>=20
> Let me re-phrase and see if we agree:
>=20
> A sends B three video streams and one audio stream .
> B has one display and opts to display one of the three streams.
> There is a mechanism for B to tell A that this has happened (and to
> say when it wishes to display some other stream from A).
>=20
> Do we think that the work to do this type of stream-suspension is
> floor control? (My opinion is no. I view floor control more as who is
> the source for a given stream and not a selection of an agreed upon
> stream).
>=20
> Do we think this work is in scope for Telepresence Interoperability? I
> think this is an essential element to deployable TP operation
> especially as in expands to "at home" use. If an architecture in which
> a room sends all video all the time (even though it's not being used)
> is viewed as necessary to avoid complexity/meet responsiveness
> requirements is thought to be ok then I'd like to make sure we are
> clear that that these limitations exist upfront.
>=20
> My preference is obviously to permit this feedback and tweak the
> proposed charter.
>=20
> Other opinions?
>=20
> Regards,
>=20
>=20
> Peter Musgrave
>=20
> On Thu, Oct 14, 2010 at 9:46 AM, Mike Hammer (hmmr) <hmmr@cisco.com> wrot=
e:
> > Peter,
> >
> > I have not issue with your proposed words, since they are essentially a=
sking for a "mute" button for all types of communications used by a sender.=
 =A0This may complicate the algorithm for all the receive systems to know w=
hat to display, if for example, you continue to speak, but turn off the vid=
eo transmission. =A0A "mute all" option might be more polite.
> >
> > I did have issue with this comment: =A0"more like onhold/offhold type n=
egotiation"
> >
> > I am not sure I would agree with this.
> >
> > There needs to be a distinction between when a call starts and when a c=
all stops, the call being *all* communications (sessions) between two point=
s. =A0Granted putting a call on hold seems like floor control, but the part=
icipants of the call on hold are effectively out of the call (left the room=
) until brought off-hold. =A0This is SIP level action.
> >
> > There needs to be a clear distinction between negotiating capabilities,=
 aka knowing what capabilities an endpoint has, and using those capabilitie=
s during a call. =A0That learning process is done via SDP.
> >
> > The latter part seems to be floor control, continuous conference contro=
l, or whatever you want to call it. =A0There seems to be splitting of hairs=
 over whether the floor control is manual or automatic (determined by the l=
oudest speaker). =A0This may be AVT.
> >
> > In summary, you have room access, you define capabilities, you have "sp=
eaker" conflict resolution control. =A0(Note, point-to-point calls are devo=
lution since resolution of conflict is a non-problem with only one opposing=
 party.)
> >
> > Did I miss some other nuance?
> >
> > Mike
> >
> >
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On B=
ehalf Of Peter Musgrave
> > Sent: Thursday, October 14, 2010 5:25 AM
> > To: Allyn Romanow (allyn)
> > Cc: dispatch@ietf.org
> > Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
> >
> > HI Allyn/Magnus,
> >
> > W.R.T
> >
> > "[AR} I agree with your point that it would be good to pause the
> > sender, but I think that falls
> > =A0under the category of "continuous conference control" which is out of
> > the scope of this charter.
> > =A0Such a feedback mechanism falls probably under AVT.
> > I think it would be good to add a sentence saying something like, As
> > control issues
> > necessary for interoperability arise, they will be directed appropriate=
ly.
> > What do you think?"
> >
> > I think these words leave room to *not* do what Magnus is suggesting.
> > e.g. If I have a 1 screen home TP system getting three streams from a
> > 3 camera system and selecting only one it is not really "necessary" to
> > tell the 3 camera system to pause two streams - but it is a really
> > good idea to have a way to that! I don't view this is a continuous
> > conference control issue (since that gets into floor control which I
> > agree is out of scope) but more like onhold/offhold type negotiation.
> > I think this is to some extent a plumbing issue (and based on the
> > desired response time SIP or media path messages will be required).
> >
> > So I would still like to see a statement which makes it clear that it
> > not just an issue of receiver selection but there is a need to
> > communicate that selection back to the sender. The current text does
> > not say this. Hence my suggestion:
> >
> > "Selection choices will be communicated to the sending party to allow
> > the sender to optionally present information on which streams are in
> > use and to optionally suspend streams which are not in use." ??
> >
> > I'm open to other words or some indication of which language in the
> > current charter ensures this is in scope.
> >
> > Regards,
> >
> > Peter Musgrave
> >
> >
> >
> >
> > On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) <allyn@cisco.co=
m> wrote:
> >> Hi Magnus,
> >> I was really glad that you had a chance to carefully consider the prop=
osed work.
> >> Like Peter, I too agree with most of your comments.
> >> Specific points below.
> >>
> >> Best regards,
> >> Allyn
> >>
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
> >> Sent: Tuesday, October 12, 2010 6:47 AM
> >> To: dispatch@ietf.org
> >> Subject: [dispatch] Comments on the proposed Telepresence Charter
> >>
> >> Hi,
> >>
> >> Some colleagues and I have reviewed the two telepresence drafts and
> >> version 5 of the charter and have some comments on the charter.
> >>
> >> To make our comments more understandable I think we should explain our
> >> view on telepresence. First we are expecting great variations in the
> >> capabilities of the participating terminals in a session. From personal
> >> terminals for individual users up to multi-display, cameras and
> >> microphone setups. We also expect that more than two end-points being
> >> present in the same session. As we see it it will require good support
> >> for centralized media handling in a mixer (MCU). We also see video
> >> streams not only of high quality to be presented for maximum immersion,
> >> but also lower qualities that can be used to monitor activity in other
> >> rooms that aren't the ones selected for high quality.
> >>
> >> [AR] Yes, absolutely.
> >>
> >> Independent if one uses a central mixer or a full mesh there appears to
> >> be a lack in media stream negotiation for how to negotiate what
> >> capabilities the receiving client has in receiving, decoding and
> >> rendering when it comes to number of simultaneous streams. Our
> >> understanding is that RTP mandates any number of streams, while in
> >> reality outside of conferencing that is in fact one single stream.
> >> However with telepresence and conferencing there is a need to negotiate
> >> the actual capability, especially between a mixer and client when it
> >> comes to deliver multiple streams for various usages.
> >>
> >> [AR] Yes, agree - as you saw, this is covered in the charter.
> >>
> >> We also think there is missing functionality for expressing who are the
> >> currently active speakers. The RTP CSRC field are not sufficient as th=
ey
> >> express which end-points are included in the mix. However, as one like=
ly
> >> include in a mix more rooms than the active speakers to ensure that any
> >> interruption and ambient sound are correctly captured an mechanism for
> >> indicating the currently active speakers are needed. There is also need
> >> to agree on the type of identifier to be used that allow for mapping to
> >> full user names when appropriate.
> >>
> >> [AR] Yes.. Both of these points are covered in the charter too, right?
> >>
> >> The charter includes an item to work on control between client and a
> >> mixer what streams should be delivered. We would like to extended this
> >> to allow also the mixer to control the client on when streams are
> >> delivered. In cases where there are more users then available display
> >> area and a particular user is not actually included in any outgoing
> >> mix/selection from the mixer then there is little point in that the
> >> client consumes network resources. Thus we suggest that the stream
> >> selection and control work is extended to allow also the mixer to pause
> >> individual streams from the client to the mixer. Please note that such
> >> control needs to be responsive, thus using the SIP signaling path for
> >> this is to slow, it needs to be on the media path.
> >>
> >> [AR} I agree with your point that it would be good to pause the sender=
, but I think that falls
> >> =A0under the category of "continuous conference control" which is out =
of the scope of this charter.
> >> =A0Such a feedback mechanism falls probably under AVT.
> >> I think it would be good to add a sentence saying something like, As c=
ontrol issues
> >> necessary for interoperability arise, they will be directed appropriat=
ely.
> >> What do you think?
> >>
> >>
> >> Another question to the group is, are there agreement on how streams
> >> should organized into RTP sessions and how they are utilized? If
> >> anything is documented I would appreciate a pointer. The media streams
> >> are for different purposes from different end-points which are then
> >> mixed by a central point and delivered to end-points. My impression for
> >> example is that most proprietary system breaks the RTP semantics and u=
se
> >> individual sessions between the mixer and each end-point, instead of a
> >> joint session. Here also different robustification tools for RTP such =
as
> >> FEC and Retransmission needs to be considered. This might be work that
> >> belongs in AVT, but are there need for further clarification into this
> >> field?
> >>
> >> [AR} Agree this needs to be worked out for Telepresence, and that any =
activity about it would go to AVT and/or MMUSIC.
> >>
> >> Comments much appreciated and we can gladly suggest text additions to
> >> the charter proposal.
> >>
> >> Cheers
> >>
> >> Magnus Westerlund
> >>
> >> ----------------------------------------------------------------------
> >> Multimedia Technologies, Ericsson Research EAB/TVM
> >> ----------------------------------------------------------------------
> >> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> >> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> >> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> >> ----------------------------------------------------------------------
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


--=20
Lorenzo Miniero
Meetecho s.r.l.
http://www.meetecho.com

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From: "Mike Hammer (hmmr)" <hmmr@cisco.com>
To: "Peter Musgrave" <peter.musgrave@magorcorp.com>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Peter,

I see who is the source of a given stream as something that is a =
declared capability at the beginning of the call.

Floor control IMHO is the "green light" to transmit content onto the =
wire (floor).

In the 2-way case you describe, the floor controller is the receiver, in =
this case the end-consumer versus the MCU.

I don't see these models as being incompatible and do think discussion =
should be allowed in the charter.

I also agree that it is goodness *not* to waste network bandwidth if no =
one will render it at a receive point.

So, hope the charter allows that.

Thanks,
Mike


-----Original Message-----
From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
Sent: Thursday, October 14, 2010 10:55 AM
To: Mike Hammer (hmmr)
Cc: Allyn Romanow (allyn); dispatch@ietf.org
Subject: Re: [dispatch] Comments on the proposed Telepresence Charter

Hi Mike,

I agree the onhld/offhold analogy is a poor one. It is a suspension of
a call - which is not intended.

Let me re-phrase and see if we agree:

A sends B three video streams and one audio stream .
B has one display and opts to display one of the three streams.
There is a mechanism for B to tell A that this has happened (and to
say when it wishes to display some other stream from A).

Do we think that the work to do this type of stream-suspension is
floor control? (My opinion is no. I view floor control more as who is
the source for a given stream and not a selection of an agreed upon
stream).

Do we think this work is in scope for Telepresence Interoperability? I
think this is an essential element to deployable TP operation
especially as in expands to "at home" use. If an architecture in which
a room sends all video all the time (even though it's not being used)
is viewed as necessary to avoid complexity/meet responsiveness
requirements is thought to be ok then I'd like to make sure we are
clear that that these limitations exist upfront.

My preference is obviously to permit this feedback and tweak the
proposed charter.

Other opinions?

Regards,


Peter Musgrave

On Thu, Oct 14, 2010 at 9:46 AM, Mike Hammer (hmmr) <hmmr@cisco.com> =
wrote:
> Peter,
>
> I have not issue with your proposed words, since they are essentially =
asking for a "mute" button for all types of communications used by a =
sender. =A0This may complicate the algorithm for all the receive systems =
to know what to display, if for example, you continue to speak, but turn =
off the video transmission. =A0A "mute all" option might be more polite.
>
> I did have issue with this comment: =A0"more like onhold/offhold type =
negotiation"
>
> I am not sure I would agree with this.
>
> There needs to be a distinction between when a call starts and when a =
call stops, the call being *all* communications (sessions) between two =
points. =A0Granted putting a call on hold seems like floor control, but =
the participants of the call on hold are effectively out of the call =
(left the room) until brought off-hold. =A0This is SIP level action.
>
> There needs to be a clear distinction between negotiating =
capabilities, aka knowing what capabilities an endpoint has, and using =
those capabilities during a call. =A0That learning process is done via =
SDP.
>
> The latter part seems to be floor control, continuous conference =
control, or whatever you want to call it. =A0There seems to be splitting =
of hairs over whether the floor control is manual or automatic =
(determined by the loudest speaker). =A0This may be AVT.

>
> In summary, you have room access, you define capabilities, you have =
"speaker" conflict resolution control. =A0(Note, point-to-point calls =
are devolution since resolution of conflict is a non-problem with only =
one opposing party.)
>
> Did I miss some other nuance?
>
> Mike
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Peter Musgrave
> Sent: Thursday, October 14, 2010 5:25 AM
> To: Allyn Romanow (allyn)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> HI Allyn/Magnus,
>
> W.R.T
>
> "[AR} I agree with your point that it would be good to pause the
> sender, but I think that falls
> =A0under the category of "continuous conference control" which is out =
of
> the scope of this charter.
> =A0Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As
> control issues
> necessary for interoperability arise, they will be directed =
appropriately.
> What do you think?"
>
> I think these words leave room to *not* do what Magnus is suggesting.
> e.g. If I have a 1 screen home TP system getting three streams from a
> 3 camera system and selecting only one it is not really "necessary" to
> tell the 3 camera system to pause two streams - but it is a really
> good idea to have a way to that! I don't view this is a continuous
> conference control issue (since that gets into floor control which I
> agree is out of scope) but more like onhold/offhold type negotiation.
> I think this is to some extent a plumbing issue (and based on the
> desired response time SIP or media path messages will be required).
>
> So I would still like to see a statement which makes it clear that it
> not just an issue of receiver selection but there is a need to
> communicate that selection back to the sender. The current text does
> not say this. Hence my suggestion:
>
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
>
> I'm open to other words or some indication of which language in the
> current charter ensures this is in scope.
>
> Regards,
>
> Peter Musgrave
>
>
>
>
> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) =
<allyn@cisco.com> wrote:
>> Hi Magnus,
>> I was really glad that you had a chance to carefully consider the =
proposed work.
>> Like Peter, I too agree with most of your comments.
>> Specific points below.
>>
>> Best regards,
>> Allyn
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
>> Sent: Tuesday, October 12, 2010 6:47 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>
>> Hi,
>>
>> Some colleagues and I have reviewed the two telepresence drafts and
>> version 5 of the charter and have some comments on the charter.
>>
>> To make our comments more understandable I think we should explain =
our
>> view on telepresence. First we are expecting great variations in the
>> capabilities of the participating terminals in a session. From =
personal
>> terminals for individual users up to multi-display, cameras and
>> microphone setups. We also expect that more than two end-points being
>> present in the same session. As we see it it will require good =
support
>> for centralized media handling in a mixer (MCU). We also see video
>> streams not only of high quality to be presented for maximum =
immersion,
>> but also lower qualities that can be used to monitor activity in =
other
>> rooms that aren't the ones selected for high quality.
>>
>> [AR] Yes, absolutely.
>>
>> Independent if one uses a central mixer or a full mesh there appears =
to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to =
negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.
>>
>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>
>> We also think there is missing functionality for expressing who are =
the
>> currently active speakers. The RTP CSRC field are not sufficient as =
they
>> express which end-points are included in the mix. However, as one =
likely
>> include in a mix more rooms than the active speakers to ensure that =
any
>> interruption and ambient sound are correctly captured an mechanism =
for
>> indicating the currently active speakers are needed. There is also =
need
>> to agree on the type of identifier to be used that allow for mapping =
to
>> full user names when appropriate.
>>
>> [AR] Yes.. Both of these points are covered in the charter too, =
right?
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended =
this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to =
pause
>> individual streams from the client to the mixer. Please note that =
such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
>>
>> [AR} I agree with your point that it would be good to pause the =
sender, but I think that falls
>> =A0under the category of "continuous conference control" which is out =
of the scope of this charter.
>> =A0Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As =
control issues
>> necessary for interoperability arise, they will be directed =
appropriately.
>> What do you think?
>>
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media =
streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression =
for
>> example is that most proprietary system breaks the RTP semantics and =
use
>> individual sessions between the mixer and each end-point, instead of =
a
>> joint session. Here also different robustification tools for RTP such =
as
>> FEC and Retransmission needs to be considered. This might be work =
that
>> belongs in AVT, but are there need for further clarification into =
this
>> field?
>>
>> [AR} Agree this needs to be worked out for Telepresence, and that any =
activity about it would go to AVT and/or MMUSIC.
>>
>> Comments much appreciated and we can gladly suggest text additions to
>> the charter proposal.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> =
----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
>> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 =
0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> =
----------------------------------------------------------------------
>> _______________________________________________
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From allyn@cisco.com  Thu Oct 14 16:22:38 2010
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From: "Allyn Romanow (allyn)" <allyn@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "Peter Musgrave" <peter.musgrave@magorcorp.com>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi Magnus,
You have made some very helpful points. In particular you've pointed out =
that the charter talks about the receiver's ability to respond, but not =
the sender's. Also the charter is not explicit about necessary =
discussion of Telepresence interoperability issues that are not within =
the charter of the multi-stream WG to do anything about.  =20

I'd like to propose some wording in the charter that I think will =
address your concerns. See what you think- some of the wording is yours =
:)..

Instead of

	The WG will create specifications for SIP-based conferencing systems to =
enable 	communication of enough information about each media stream so =
that each receiving system 	or bridge system can make reasonable =
decisions about selecting and rendering media 	streams. This enables =
systems to make display choices that optimize the "just like being 	=
there" experience.
I propose
	The WG will create specifications for SIP-based conferencing systems to =
enable 	communication of enough information about each media stream so =
that each sending system, 	receiving system, or bridge system can make =
reasonable decisions about transmitting, 	selecting, and rendering media =
streams. This enables systems to make choices that 	optimize the "just =
like being there" experience.

I would not like to change:
	* Which sources a receiver wants to receive.  For example, it might =
want the source for 	the left camera, or might want the source chosen by =
VAD (Voice Activity Detection).
To the proposed:
	* Which sources a receiver wants transmitted.  For example, it might =
want the source for 	the left camera, or might want the source chosen by =
VAD (Voice Activity Detection).

Because there are multiple receivers. A receiver cannot say what should =
be transmitted, it can only say what itself wants to get, or receive. =
But I think your concern about sender behavior is covered by the changes =
to the paragraph above.

I also propose the following addition:
	This working group is not currently chartered to work on issues of =
continuous conference 	control including: far end camera control, =
indication of fast frame update for video 	codecs or other rapid =
switches, floor control, conference roster. The working group may 	=
identify interoperability obstacles in existing openstandards. If so, =
the WG will develp 	requirements to be communicated to other IETF WGs or =
Standards Forums as a kind request 	to try to meet these requirements.

What do you think?

Best regards,
Allyn

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
Sent: Wednesday, October 13, 2010 2:11 AM
To: Peter Musgrave
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Comments on the proposed Telepresence Charter

Hi Peter,

Good that we are in agreement on a number of things. Response and
clarifications inline.

Peter Musgrave skrev 2010-10-12 17:37:
>>
>> Independent if one uses a central mixer or a full mesh there appears =
to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to =
negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.

> Agreed. The charter does speak to identifying endpoint capabilities
> and I view what you describe here as the kind of meta-information
> which needs to fit into that framework.

Okay, as long as this is included in the thinking about capabilities I
am fine. However, the charter proposal isn't particular detailed in this
regards. But I guess the following charter text is matching this:

"Information between sources and sinks about media stream capabilities
will be exchanged."

>>
>> We also think there is missing functionality for expressing who are =
the
>> currently active speakers. The RTP CSRC field are not sufficient as =
they
>> express which end-points are included in the mix. However, as one =
likely
>> include in a mix more rooms than the active speakers to ensure that =
any
>> interruption and ambient sound are correctly captured an mechanism =
for
>> indicating the currently active speakers are needed. There is also =
need
>> to agree on the type of identifier to be used that allow for mapping =
to
>> full user names when appropriate.

> I agree with this. In my opinion this is a separable problem since it
> applies to generic audio conferencing as well. There is tangential
> work in the conference event package and probably in the XCON stuff -
> but I lack detailed background in XCON.
>=20
> In the context of the TP charter I think the work on indicating which
> audio stream is associated with which speaker (or part of the room) is
> a slightly different issue - since it's about which stream and not
> which speaker within a stream.

I should probably have been clearer. I am concerned that even on media
stream level there is missing functionality to indicate which streams
that contains active speakers and has been included in a mix produced by
the mixer. From our side we see a difference between being included in
the mix and being tagged as currently an active speaker.

I think this is currently missing functionality for which requirements
needs to be agreed on and then a solution developed. However, the
solution might better fit another WG, like AVT. But I think that is for
later determination. I think the requirements phase makes sense to have
in the Telepresence group.

I do agree that such functionality do have a wider usage scope than just
telepresence and can be used in any case of centralized mixers where
special activity needs to be pointed out, with audio being the foremost
example of such media.

>=20
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended =
this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to =
pause
>> individual streams from the client to the mixer. Please note that =
such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
> I concur.
>=20
> The charter sentence:
>  "The WG will create specifications for SIP-based conferencing systems
> to enable communication of enough information about each media stream
> so that each receiving system or bridge system can make reasonable
> decisions about selecting and rendering media streams. This enables
> systems to make display choices that optimize the "just like being
> there" experience. "
>=20
> is specific to selection of streams and not to the ability to control
> them. I don't see anything here which prevents a source from not
> sending selected streams - but I have no objection to making this more
> explicit. How about adding:
>=20
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
>=20

When I wrote the above comment I was mostly considering the following
chart text bullet:

* Which sources a receiver wants to receive.  For example, it might want
the source for the left camera, or might want the source chosen by VAD
(Voice Activity Detection).

I think your proposed text combined with the above bullet gives all the
room I see needed for media stream control. However, maybe a
modification of the above to make it clear.

* Which sources a receiver or mixer wants to have transmitted.  For
example, it might want the source for the left camera, or might want the
source chosen by VAD (Voice Activity Detection).

>=20
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media =
streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression =
for
>> example is that most proprietary system breaks the RTP semantics and =
use
>> individual sessions between the mixer and each end-point, instead of =
a
>> joint session. Here also different robustification tools for RTP such =
as
>> FEC and Retransmission needs to be considered. This might be work =
that
>> belongs in AVT, but are there need for further clarification into =
this
>> field?

> I think this is something which will have to be thought through and
> standardized if TP interoperability is to succeed. I would view the
> semantics of how the streams are assigned in scope and the "plumbing"
> either comes from the AVT/MMUSIC media cap stuff or any additions are
> developed in those WGs.
>=20

I agree with that the right place to do specification work is likely
these two groups. However, I think telepresence needs to provide
requirements for such work. Making such requirement development explicit
in the charter would make it clear that we do expect work to happen
somewhere else. At the same time it is after all possible to request
that another WG's charter is updated to cover such work.

The WG may identify interoperability obstacles in existing open
standards. The WG may develop requirements to be communicated to other
IETF WGs or Standards Forums as a kind request to try to meet these
requirements.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------
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From peter.musgrave@magorcorp.com  Fri Oct 15 02:48:40 2010
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
To: "Allyn Romanow (allyn)" <allyn@cisco.com>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi Allyn,

This text works for me.

Thanks,

Peter Musgrave

On Thu, Oct 14, 2010 at 7:23 PM, Allyn Romanow (allyn) <allyn@cisco.com> wr=
ote:
> Hi Magnus,
> You have made some very helpful points. In particular you've pointed out =
that the charter talks about the receiver's ability to respond, but not the=
 sender's. Also the charter is not explicit about necessary discussion of T=
elepresence interoperability issues that are not within the charter of the =
multi-stream WG to do anything about.
>
> I'd like to propose some wording in the charter that I think will address=
 your concerns. See what you think- some of the wording is yours :)..
>
> Instead of
>
> =A0 =A0 =A0 =A0The WG will create specifications for SIP-based conferenci=
ng systems to enable =A0communication of enough information about each medi=
a stream so that each receiving system =A0 =A0 =A0 or bridge system can mak=
e reasonable decisions about selecting and rendering media =A0 =A0 =A0strea=
ms. This enables systems to make display choices that optimize the "just li=
ke being =A0 =A0 =A0 =A0there" experience.
> I propose
> =A0 =A0 =A0 =A0The WG will create specifications for SIP-based conferenci=
ng systems to enable =A0communication of enough information about each medi=
a stream so that each sending system, =A0 =A0 =A0 =A0receiving system, or b=
ridge system can make reasonable decisions about transmitting, =A0 =A0selec=
ting, and rendering media streams. This enables systems to make choices tha=
t =A0 =A0 =A0 optimize the "just like being there" experience.
>
> I would not like to change:
> =A0 =A0 =A0 =A0* Which sources a receiver wants to receive. =A0For exampl=
e, it might want the source for =A0 =A0 =A0 =A0 the left camera, or might w=
ant the source chosen by VAD (Voice Activity Detection).
> To the proposed:
> =A0 =A0 =A0 =A0* Which sources a receiver wants transmitted. =A0For examp=
le, it might want the source for =A0 =A0 =A0 =A0the left camera, or might w=
ant the source chosen by VAD (Voice Activity Detection).
>
> Because there are multiple receivers. A receiver cannot say what should b=
e transmitted, it can only say what itself wants to get, or receive. But I =
think your concern about sender behavior is covered by the changes to the p=
aragraph above.
>
> I also propose the following addition:
> =A0 =A0 =A0 =A0This working group is not currently chartered to work on i=
ssues of continuous conference =A0 =A0 =A0 =A0control including: far end ca=
mera control, indication of fast frame update for video =A0 =A0codecs or ot=
her rapid switches, floor control, conference roster. The working group may=
 =A0 =A0 =A0 =A0 identify interoperability obstacles in existing openstanda=
rds. If so, the WG will develp =A0 =A0 =A0 =A0requirements to be communicat=
ed to other IETF WGs or Standards Forums as a kind request =A0 =A0 =A0 =A0 =
to try to meet these requirements.
>
> What do you think?
>
> Best regards,
> Allyn
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf Of Magnus Westerlund
> Sent: Wednesday, October 13, 2010 2:11 AM
> To: Peter Musgrave
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> Hi Peter,
>
> Good that we are in agreement on a number of things. Response and
> clarifications inline.
>
> Peter Musgrave skrev 2010-10-12 17:37:
>>>
>>> Independent if one uses a central mixer or a full mesh there appears to
>>> be a lack in media stream negotiation for how to negotiate what
>>> capabilities the receiving client has in receiving, decoding and
>>> rendering when it comes to number of simultaneous streams. Our
>>> understanding is that RTP mandates any number of streams, while in
>>> reality outside of conferencing that is in fact one single stream.
>>> However with telepresence and conferencing there is a need to negotiate
>>> the actual capability, especially between a mixer and client when it
>>> comes to deliver multiple streams for various usages.
>
>> Agreed. The charter does speak to identifying endpoint capabilities
>> and I view what you describe here as the kind of meta-information
>> which needs to fit into that framework.
>
> Okay, as long as this is included in the thinking about capabilities I
> am fine. However, the charter proposal isn't particular detailed in this
> regards. But I guess the following charter text is matching this:
>
> "Information between sources and sinks about media stream capabilities
> will be exchanged."
>
>>>
>>> We also think there is missing functionality for expressing who are the
>>> currently active speakers. The RTP CSRC field are not sufficient as the=
y
>>> express which end-points are included in the mix. However, as one likel=
y
>>> include in a mix more rooms than the active speakers to ensure that any
>>> interruption and ambient sound are correctly captured an mechanism for
>>> indicating the currently active speakers are needed. There is also need
>>> to agree on the type of identifier to be used that allow for mapping to
>>> full user names when appropriate.
>
>> I agree with this. In my opinion this is a separable problem since it
>> applies to generic audio conferencing as well. There is tangential
>> work in the conference event package and probably in the XCON stuff -
>> but I lack detailed background in XCON.
>>
>> In the context of the TP charter I think the work on indicating which
>> audio stream is associated with which speaker (or part of the room) is
>> a slightly different issue - since it's about which stream and not
>> which speaker within a stream.
>
> I should probably have been clearer. I am concerned that even on media
> stream level there is missing functionality to indicate which streams
> that contains active speakers and has been included in a mix produced by
> the mixer. From our side we see a difference between being included in
> the mix and being tagged as currently an active speaker.
>
> I think this is currently missing functionality for which requirements
> needs to be agreed on and then a solution developed. However, the
> solution might better fit another WG, like AVT. But I think that is for
> later determination. I think the requirements phase makes sense to have
> in the Telepresence group.
>
> I do agree that such functionality do have a wider usage scope than just
> telepresence and can be used in any case of centralized mixers where
> special activity needs to be pointed out, with audio being the foremost
> example of such media.
>
>>
>>>
>>> The charter includes an item to work on control between client and a
>>> mixer what streams should be delivered. We would like to extended this
>>> to allow also the mixer to control the client on when streams are
>>> delivered. In cases where there are more users then available display
>>> area and a particular user is not actually included in any outgoing
>>> mix/selection from the mixer then there is little point in that the
>>> client consumes network resources. Thus we suggest that the stream
>>> selection and control work is extended to allow also the mixer to pause
>>> individual streams from the client to the mixer. Please note that such
>>> control needs to be responsive, thus using the SIP signaling path for
>>> this is to slow, it needs to be on the media path.
>> I concur.
>>
>> The charter sentence:
>> =A0"The WG will create specifications for SIP-based conferencing systems
>> to enable communication of enough information about each media stream
>> so that each receiving system or bridge system can make reasonable
>> decisions about selecting and rendering media streams. This enables
>> systems to make display choices that optimize the "just like being
>> there" experience. "
>>
>> is specific to selection of streams and not to the ability to control
>> them. I don't see anything here which prevents a source from not
>> sending selected streams - but I have no objection to making this more
>> explicit. How about adding:
>>
>> "Selection choices will be communicated to the sending party to allow
>> the sender to optionally present information on which streams are in
>> use and to optionally suspend streams which are not in use." ??
>>
>
> When I wrote the above comment I was mostly considering the following
> chart text bullet:
>
> * Which sources a receiver wants to receive. =A0For example, it might wan=
t
> the source for the left camera, or might want the source chosen by VAD
> (Voice Activity Detection).
>
> I think your proposed text combined with the above bullet gives all the
> room I see needed for media stream control. However, maybe a
> modification of the above to make it clear.
>
> * Which sources a receiver or mixer wants to have transmitted. =A0For
> example, it might want the source for the left camera, or might want the
> source chosen by VAD (Voice Activity Detection).
>
>>
>>>
>>> Another question to the group is, are there agreement on how streams
>>> should organized into RTP sessions and how they are utilized? If
>>> anything is documented I would appreciate a pointer. The media streams
>>> are for different purposes from different end-points which are then
>>> mixed by a central point and delivered to end-points. My impression for
>>> example is that most proprietary system breaks the RTP semantics and us=
e
>>> individual sessions between the mixer and each end-point, instead of a
>>> joint session. Here also different robustification tools for RTP such a=
s
>>> FEC and Retransmission needs to be considered. This might be work that
>>> belongs in AVT, but are there need for further clarification into this
>>> field?
>
>> I think this is something which will have to be thought through and
>> standardized if TP interoperability is to succeed. I would view the
>> semantics of how the streams are assigned in scope and the "plumbing"
>> either comes from the AVT/MMUSIC media cap stuff or any additions are
>> developed in those WGs.
>>
>
> I agree with that the right place to do specification work is likely
> these two groups. However, I think telepresence needs to provide
> requirements for such work. Making such requirement development explicit
> in the charter would make it clear that we do expect work to happen
> somewhere else. At the same time it is after all possible to request
> that another WG's charter is updated to cover such work.
>
> The WG may identify interoperability obstacles in existing open
> standards. The WG may develop requirements to be communicated to other
> IETF WGs or Standards Forums as a kind request to try to meet these
> requirements.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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A new Internet Draft draft-allen-dispatch-imei-urn-as-instanceid-00 has
been submitted. It can be found at:

 

http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-00.tx
t

 

The referenced URN definition draft can also be found at:

 

http://www.ietf.org/id/draft-montemurro-gsma-imei-urn-05.txt

 

 

The draft defines how the IMEI (International Mobile Equipment
Identifier) encoded as a URN can be used as an Instance ID as defined in
Outbound (RFC 5626) and as used by GRUU (RFC 5627). 3GPP have a need to
use the IMEI as an Instance ID as explained in the draft.

 

Please provide comments and feedback.

 

Thanks

Andrew


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<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>A new Internet Draft
draft-allen-dispatch-imei-urn-as-instanceid-00 has been submitted. It can be
found at:<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><a
href=3D"http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-0=
0.txt">http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-00=
.txt</a><o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal>The referenced URN definition draft can also be found a=
t:<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><a
href=3D"http://www.ietf.org/id/draft-montemurro-gsma-imei-urn-05.txt">http:/=
/www.ietf.org/id/draft-montemurro-gsma-imei-urn-05.txt</a><o:p></o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>The draft defines how the=
 IMEI
(International Mobile Equipment Identifier) encoded as a URN can be used as=
 an
Instance ID as defined in Outbound (RFC 5626) and as used by GRUU (RFC 5627)=
.
3GPP have a need to use the IMEI as an Instance ID as explained in the draft=
.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Please provide comments a=
nd
feedback.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Andrew<o:p></o:p></p>

</div>

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From pkyzivat@cisco.com  Fri Oct 15 09:32:21 2010
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Subject: Re: [dispatch] draft-allen-dispatch-imei-urn-as-instanceid-00
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Andrew,

I don't understand the point of this draft.

The sip.instance feature tag takes a urn as a value. Either the imei urn 
works with that or it doesn't. If it does, no draft is needed. If it 
doesn't, then I don't think a draft will help, unless it is an extension 
to 5626.

5626 is a little hazy on whether all URNs are ok or not. It implies that 
it only works for ones that can be compared as strings for equality. 
Operationally I think that must be the requirement.

It would be very bad if implementors of gruu and outbound had to 
understand each type of URN that might be used as the instance id.

I gather that the imei urn might not always work with simple string 
comparison as the identity test. Hence you have included registrar 
procedures to ignore parameters when comparing the imei urns.

ISTM that you should not be imposing *any* new requirements on 
registrars. I guess there could be a purpose for such a draft if it was 
to impose requirements on user agents, about how they format their imei 
URNs so that they will work with generic comparison procedures at the 
registrar.

	Thanks,
	Paul

On 10/15/2010 6:34 AM, Andrew Allen wrote:
> A new Internet Draft draft-allen-dispatch-imei-urn-as-instanceid-00 has
> been submitted. It can be found at:
>
> http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-00.txt
>
> The referenced URN definition draft can also be found at:
>
> http://www.ietf.org/id/draft-montemurro-gsma-imei-urn-05.txt
>
> The draft defines how the IMEI (International Mobile Equipment
> Identifier) encoded as a URN can be used as an Instance ID as defined in
> Outbound (RFC 5626) and as used by GRUU (RFC 5627). 3GPP have a need to
> use the IMEI as an Instance ID as explained in the draft.
>
> Please provide comments and feedback.
>
> Thanks
>
> Andrew
>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Discussions of putting something "on-hold/off-hold" in the context of 
sip are necessarily imprecise. SIP really has no concept of hold - it is 
an application level issue.

SIP has mechanisms for negotiating how media streams are used, and those 
are often *used* to accomplish a hold feature. But the recipient of an 
offer that manipulates a=sendonly/... does not know if this is because 
of "hold" or something else. And those mechanisms are per-media-stream. 
Most of the examples people are familiar with are on calls that only 
have one media stream, so on can be sloppy and think it is about the 
call as a whole, but that isn't so. As soon as you add a 2nd media 
stream, you have the potential to negotiation the directionality of the 
media independently for each m-line.

Bottom line - IMO its perfectly fine for telepresence devices to use the 
same mechanism to negotiate the directionality (or inactivity) of 
individual media streams. That mechanism *might* be used as part of 
floor control policy (if done by a focus), or not (if done by a regular 
endpoint that just doesn't want to receive stuff it is ignoring.)

	Thanks,
	Paul

On 10/14/2010 9:46 AM, Mike Hammer (hmmr) wrote:
> Peter,
>
> I have not issue with your proposed words, since they are essentially asking for a "mute" button for all types of communications used by a sender.  This may complicate the algorithm for all the receive systems to know what to display, if for example, you continue to speak, but turn off the video transmission.  A "mute all" option might be more polite.
>
> I did have issue with this comment:  "more like onhold/offhold type negotiation"
>
> I am not sure I would agree with this.
>
> There needs to be a distinction between when a call starts and when a call stops, the call being *all* communications (sessions) between two points.  Granted putting a call on hold seems like floor control, but the participants of the call on hold are effectively out of the call (left the room) until brought off-hold.  This is SIP level action.
>
> There needs to be a clear distinction between negotiating capabilities, aka knowing what capabilities an endpoint has, and using those capabilities during a call.  That learning process is done via SDP.
>
> The latter part seems to be floor control, continuous conference control, or whatever you want to call it.  There seems to be splitting of hairs over whether the floor control is manual or automatic (determined by the loudest speaker).  This may be AVT.
>
> In summary, you have room access, you define capabilities, you have "speaker" conflict resolution control.  (Note, point-to-point calls are devolution since resolution of conflict is a non-problem with only one opposing party.)
>
> Did I miss some other nuance?
>
> Mike
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Musgrave
> Sent: Thursday, October 14, 2010 5:25 AM
> To: Allyn Romanow (allyn)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> HI Allyn/Magnus,
>
> W.R.T
>
> "[AR} I agree with your point that it would be good to pause the
> sender, but I think that falls
>   under the category of "continuous conference control" which is out of
> the scope of this charter.
>   Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As
> control issues
> necessary for interoperability arise, they will be directed appropriately.
> What do you think?"
>
> I think these words leave room to *not* do what Magnus is suggesting.
> e.g. If I have a 1 screen home TP system getting three streams from a
> 3 camera system and selecting only one it is not really "necessary" to
> tell the 3 camera system to pause two streams - but it is a really
> good idea to have a way to that! I don't view this is a continuous
> conference control issue (since that gets into floor control which I
> agree is out of scope) but more like onhold/offhold type negotiation.
> I think this is to some extent a plumbing issue (and based on the
> desired response time SIP or media path messages will be required).
>
> So I would still like to see a statement which makes it clear that it
> not just an issue of receiver selection but there is a need to
> communicate that selection back to the sender. The current text does
> not say this. Hence my suggestion:
>
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
>
> I'm open to other words or some indication of which language in the
> current charter ensures this is in scope.
>
> Regards,
>
> Peter Musgrave
>
>
>
>
> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn)<allyn@cisco.com>  wrote:
>> Hi Magnus,
>> I was really glad that you had a chance to carefully consider the proposed work.
>> Like Peter, I too agree with most of your comments.
>> Specific points below.
>>
>> Best regards,
>> Allyn
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Magnus Westerlund
>> Sent: Tuesday, October 12, 2010 6:47 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>
>> Hi,
>>
>> Some colleagues and I have reviewed the two telepresence drafts and
>> version 5 of the charter and have some comments on the charter.
>>
>> To make our comments more understandable I think we should explain our
>> view on telepresence. First we are expecting great variations in the
>> capabilities of the participating terminals in a session. From personal
>> terminals for individual users up to multi-display, cameras and
>> microphone setups. We also expect that more than two end-points being
>> present in the same session. As we see it it will require good support
>> for centralized media handling in a mixer (MCU). We also see video
>> streams not only of high quality to be presented for maximum immersion,
>> but also lower qualities that can be used to monitor activity in other
>> rooms that aren't the ones selected for high quality.
>>
>> [AR] Yes, absolutely.
>>
>> Independent if one uses a central mixer or a full mesh there appears to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.
>>
>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>
>> We also think there is missing functionality for expressing who are the
>> currently active speakers. The RTP CSRC field are not sufficient as they
>> express which end-points are included in the mix. However, as one likely
>> include in a mix more rooms than the active speakers to ensure that any
>> interruption and ambient sound are correctly captured an mechanism for
>> indicating the currently active speakers are needed. There is also need
>> to agree on the type of identifier to be used that allow for mapping to
>> full user names when appropriate.
>>
>> [AR] Yes.. Both of these points are covered in the charter too, right?
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to pause
>> individual streams from the client to the mixer. Please note that such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
>>
>> [AR} I agree with your point that it would be good to pause the sender, but I think that falls
>>   under the category of "continuous conference control" which is out of the scope of this charter.
>>   Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As control issues
>> necessary for interoperability arise, they will be directed appropriately.
>> What do you think?
>>
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression for
>> example is that most proprietary system breaks the RTP semantics and use
>> individual sessions between the mixer and each end-point, instead of a
>> joint session. Here also different robustification tools for RTP such as
>> FEC and Retransmission needs to be considered. This might be work that
>> belongs in AVT, but are there need for further clarification into this
>> field?
>>
>> [AR} Agree this needs to be worked out for Telepresence, and that any activity about it would go to AVT and/or MMUSIC.
>>
>> Comments much appreciated and we can gladly suggest text additions to
>> the charter proposal.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Paul,

I can also drive a screw with a hammer.  That doesn't mean it is the =
right tool.  :)

In this case, I am not so certain that the use of a 3-way handshake is =
necessary.

In any case, I think the charter allows us the flexibility to make an =
informed decision.

Cheers,
Mike



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Paul Kyzivat (pkyzivat)
Sent: Friday, October 15, 2010 1:17 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] Comments on the proposed Telepresence Charter

Discussions of putting something "on-hold/off-hold" in the context of=20
sip are necessarily imprecise. SIP really has no concept of hold - it is =

an application level issue.

SIP has mechanisms for negotiating how media streams are used, and those =

are often *used* to accomplish a hold feature. But the recipient of an=20
offer that manipulates a=3Dsendonly/... does not know if this is because =

of "hold" or something else. And those mechanisms are per-media-stream.=20
Most of the examples people are familiar with are on calls that only=20
have one media stream, so on can be sloppy and think it is about the=20
call as a whole, but that isn't so. As soon as you add a 2nd media=20
stream, you have the potential to negotiation the directionality of the=20
media independently for each m-line.

Bottom line - IMO its perfectly fine for telepresence devices to use the =

same mechanism to negotiate the directionality (or inactivity) of=20
individual media streams. That mechanism *might* be used as part of=20
floor control policy (if done by a focus), or not (if done by a regular=20
endpoint that just doesn't want to receive stuff it is ignoring.)

	Thanks,
	Paul

On 10/14/2010 9:46 AM, Mike Hammer (hmmr) wrote:
> Peter,
>
> I have not issue with your proposed words, since they are essentially =
asking for a "mute" button for all types of communications used by a =
sender.  This may complicate the algorithm for all the receive systems =
to know what to display, if for example, you continue to speak, but turn =
off the video transmission.  A "mute all" option might be more polite.
>
> I did have issue with this comment:  "more like onhold/offhold type =
negotiation"
>
> I am not sure I would agree with this.
>
> There needs to be a distinction between when a call starts and when a =
call stops, the call being *all* communications (sessions) between two =
points.  Granted putting a call on hold seems like floor control, but =
the participants of the call on hold are effectively out of the call =
(left the room) until brought off-hold.  This is SIP level action.
>
> There needs to be a clear distinction between negotiating =
capabilities, aka knowing what capabilities an endpoint has, and using =
those capabilities during a call.  That learning process is done via =
SDP.
>
> The latter part seems to be floor control, continuous conference =
control, or whatever you want to call it.  There seems to be splitting =
of hairs over whether the floor control is manual or automatic =
(determined by the loudest speaker).  This may be AVT.

>
> In summary, you have room access, you define capabilities, you have =
"speaker" conflict resolution control.  (Note, point-to-point calls are =
devolution since resolution of conflict is a non-problem with only one =
opposing party.)
>
> Did I miss some other nuance?
>
> Mike
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Peter Musgrave
> Sent: Thursday, October 14, 2010 5:25 AM
> To: Allyn Romanow (allyn)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> HI Allyn/Magnus,
>
> W.R.T
>
> "[AR} I agree with your point that it would be good to pause the
> sender, but I think that falls
>   under the category of "continuous conference control" which is out =
of
> the scope of this charter.
>   Such a feedback mechanism falls probably under AVT.
> I think it would be good to add a sentence saying something like, As
> control issues
> necessary for interoperability arise, they will be directed =
appropriately.
> What do you think?"
>
> I think these words leave room to *not* do what Magnus is suggesting.
> e.g. If I have a 1 screen home TP system getting three streams from a
> 3 camera system and selecting only one it is not really "necessary" to
> tell the 3 camera system to pause two streams - but it is a really
> good idea to have a way to that! I don't view this is a continuous
> conference control issue (since that gets into floor control which I
> agree is out of scope) but more like onhold/offhold type negotiation.
> I think this is to some extent a plumbing issue (and based on the
> desired response time SIP or media path messages will be required).
>
> So I would still like to see a statement which makes it clear that it
> not just an issue of receiver selection but there is a need to
> communicate that selection back to the sender. The current text does
> not say this. Hence my suggestion:
>
> "Selection choices will be communicated to the sending party to allow
> the sender to optionally present information on which streams are in
> use and to optionally suspend streams which are not in use." ??
>
> I'm open to other words or some indication of which language in the
> current charter ensures this is in scope.
>
> Regards,
>
> Peter Musgrave
>
>
>
>
> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow =
(allyn)<allyn@cisco.com>  wrote:
>> Hi Magnus,
>> I was really glad that you had a chance to carefully consider the =
proposed work.
>> Like Peter, I too agree with most of your comments.
>> Specific points below.
>>
>> Best regards,
>> Allyn
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
>> Sent: Tuesday, October 12, 2010 6:47 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>
>> Hi,
>>
>> Some colleagues and I have reviewed the two telepresence drafts and
>> version 5 of the charter and have some comments on the charter.
>>
>> To make our comments more understandable I think we should explain =
our
>> view on telepresence. First we are expecting great variations in the
>> capabilities of the participating terminals in a session. From =
personal
>> terminals for individual users up to multi-display, cameras and
>> microphone setups. We also expect that more than two end-points being
>> present in the same session. As we see it it will require good =
support
>> for centralized media handling in a mixer (MCU). We also see video
>> streams not only of high quality to be presented for maximum =
immersion,
>> but also lower qualities that can be used to monitor activity in =
other
>> rooms that aren't the ones selected for high quality.
>>
>> [AR] Yes, absolutely.
>>
>> Independent if one uses a central mixer or a full mesh there appears =
to
>> be a lack in media stream negotiation for how to negotiate what
>> capabilities the receiving client has in receiving, decoding and
>> rendering when it comes to number of simultaneous streams. Our
>> understanding is that RTP mandates any number of streams, while in
>> reality outside of conferencing that is in fact one single stream.
>> However with telepresence and conferencing there is a need to =
negotiate
>> the actual capability, especially between a mixer and client when it
>> comes to deliver multiple streams for various usages.
>>
>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>
>> We also think there is missing functionality for expressing who are =
the
>> currently active speakers. The RTP CSRC field are not sufficient as =
they
>> express which end-points are included in the mix. However, as one =
likely
>> include in a mix more rooms than the active speakers to ensure that =
any
>> interruption and ambient sound are correctly captured an mechanism =
for
>> indicating the currently active speakers are needed. There is also =
need
>> to agree on the type of identifier to be used that allow for mapping =
to
>> full user names when appropriate.
>>
>> [AR] Yes.. Both of these points are covered in the charter too, =
right?
>>
>> The charter includes an item to work on control between client and a
>> mixer what streams should be delivered. We would like to extended =
this
>> to allow also the mixer to control the client on when streams are
>> delivered. In cases where there are more users then available display
>> area and a particular user is not actually included in any outgoing
>> mix/selection from the mixer then there is little point in that the
>> client consumes network resources. Thus we suggest that the stream
>> selection and control work is extended to allow also the mixer to =
pause
>> individual streams from the client to the mixer. Please note that =
such
>> control needs to be responsive, thus using the SIP signaling path for
>> this is to slow, it needs to be on the media path.
>>
>> [AR} I agree with your point that it would be good to pause the =
sender, but I think that falls
>>   under the category of "continuous conference control" which is out =
of the scope of this charter.
>>   Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As =
control issues
>> necessary for interoperability arise, they will be directed =
appropriately.
>> What do you think?
>>
>>
>> Another question to the group is, are there agreement on how streams
>> should organized into RTP sessions and how they are utilized? If
>> anything is documented I would appreciate a pointer. The media =
streams
>> are for different purposes from different end-points which are then
>> mixed by a central point and delivered to end-points. My impression =
for
>> example is that most proprietary system breaks the RTP semantics and =
use
>> individual sessions between the mixer and each end-point, instead of =
a
>> joint session. Here also different robustification tools for RTP such =
as
>> FEC and Retransmission needs to be considered. This might be work =
that
>> belongs in AVT, but are there need for further clarification into =
this
>> field?
>>
>> [AR} Agree this needs to be worked out for Telepresence, and that any =
activity about it would go to AVT and/or MMUSIC.
>>
>> Comments much appreciated and we can gladly suggest text additions to
>> the charter proposal.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> =
----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> =
----------------------------------------------------------------------
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
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> https://www.ietf.org/mailman/listinfo/dispatch
>
_______________________________________________
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Fri, 15 Oct 2010 16:45:43 -0400
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________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Pa=
ul Kyzivat [pkyzivat@cisco.com]

I don't understand the point of this draft.

The sip.instance feature tag takes a urn as a value. Either the imei urn
works with that or it doesn't. If it does, no draft is needed. If it
doesn't, then I don't think a draft will help, unless it is an extension
to 5626.
_______________________________________________

As I see it, the point of the draft is the definition/registration of the "=
gsma:" scheme for URNs.  In that regard, it is a proper step to enable use =
of "gsma:imea" URNs as instance identifiers.

In that regard, though, I don't see the function of section 4.  It seems to=
 describe now IMEAs can be encoded via BCD, which seems to have nothing to =
do with the gsma:imea URNs within SIP.

Also, in the Abstract, the expansion of the acronym GSMA should have been d=
one the first time it is used, rather than the last time.

Dale

From pkyzivat@cisco.com  Fri Oct 15 15:50:23 2010
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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On 10/15/2010 4:22 PM, Mike Hammer (hmmr) wrote:
> Paul,
>
> I can also drive a screw with a hammer.  That doesn't mean it is the right tool.  :)
>
> In this case, I am not so certain that the use of a 3-way handshake is necessary.
>
> In any case, I think the charter allows us the flexibility to make an informed decision.

I'm not saying its the only way, or the best way for the purpose, but 
its certainly a candidate.

And it can be 2-way rather than 3-way (with UPDATE).

	Thanks,	
	Paul

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat (pkyzivat)
> Sent: Friday, October 15, 2010 1:17 PM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>
> Discussions of putting something "on-hold/off-hold" in the context of
> sip are necessarily imprecise. SIP really has no concept of hold - it is
> an application level issue.
>
> SIP has mechanisms for negotiating how media streams are used, and those
> are often *used* to accomplish a hold feature. But the recipient of an
> offer that manipulates a=sendonly/... does not know if this is because
> of "hold" or something else. And those mechanisms are per-media-stream.
> Most of the examples people are familiar with are on calls that only
> have one media stream, so on can be sloppy and think it is about the
> call as a whole, but that isn't so. As soon as you add a 2nd media
> stream, you have the potential to negotiation the directionality of the
> media independently for each m-line.
>
> Bottom line - IMO its perfectly fine for telepresence devices to use the
> same mechanism to negotiate the directionality (or inactivity) of
> individual media streams. That mechanism *might* be used as part of
> floor control policy (if done by a focus), or not (if done by a regular
> endpoint that just doesn't want to receive stuff it is ignoring.)
>
> 	Thanks,
> 	Paul
>
> On 10/14/2010 9:46 AM, Mike Hammer (hmmr) wrote:
>> Peter,
>>
>> I have not issue with your proposed words, since they are essentially asking for a "mute" button for all types of communications used by a sender.  This may complicate the algorithm for all the receive systems to know what to display, if for example, you continue to speak, but turn off the video transmission.  A "mute all" option might be more polite.
>>
>> I did have issue with this comment:  "more like onhold/offhold type negotiation"
>>
>> I am not sure I would agree with this.
>>
>> There needs to be a distinction between when a call starts and when a call stops, the call being *all* communications (sessions) between two points.  Granted putting a call on hold seems like floor control, but the participants of the call on hold are effectively out of the call (left the room) until brought off-hold.  This is SIP level action.
>>
>> There needs to be a clear distinction between negotiating capabilities, aka knowing what capabilities an endpoint has, and using those capabilities during a call.  That learning process is done via SDP.
>>
>> The latter part seems to be floor control, continuous conference control, or whatever you want to call it.  There seems to be splitting of hairs over whether the floor control is manual or automatic (determined by the loudest speaker).  This may be AVT.
>
>>
>> In summary, you have room access, you define capabilities, you have "speaker" conflict resolution control.  (Note, point-to-point calls are devolution since resolution of conflict is a non-problem with only one opposing party.)
>>
>> Did I miss some other nuance?
>>
>> Mike
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Musgrave
>> Sent: Thursday, October 14, 2010 5:25 AM
>> To: Allyn Romanow (allyn)
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>>
>> HI Allyn/Magnus,
>>
>> W.R.T
>>
>> "[AR} I agree with your point that it would be good to pause the
>> sender, but I think that falls
>>    under the category of "continuous conference control" which is out of
>> the scope of this charter.
>>    Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As
>> control issues
>> necessary for interoperability arise, they will be directed appropriately.
>> What do you think?"
>>
>> I think these words leave room to *not* do what Magnus is suggesting.
>> e.g. If I have a 1 screen home TP system getting three streams from a
>> 3 camera system and selecting only one it is not really "necessary" to
>> tell the 3 camera system to pause two streams - but it is a really
>> good idea to have a way to that! I don't view this is a continuous
>> conference control issue (since that gets into floor control which I
>> agree is out of scope) but more like onhold/offhold type negotiation.
>> I think this is to some extent a plumbing issue (and based on the
>> desired response time SIP or media path messages will be required).
>>
>> So I would still like to see a statement which makes it clear that it
>> not just an issue of receiver selection but there is a need to
>> communicate that selection back to the sender. The current text does
>> not say this. Hence my suggestion:
>>
>> "Selection choices will be communicated to the sending party to allow
>> the sender to optionally present information on which streams are in
>> use and to optionally suspend streams which are not in use." ??
>>
>> I'm open to other words or some indication of which language in the
>> current charter ensures this is in scope.
>>
>> Regards,
>>
>> Peter Musgrave
>>
>>
>>
>>
>> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn)<allyn@cisco.com>   wrote:
>>> Hi Magnus,
>>> I was really glad that you had a chance to carefully consider the proposed work.
>>> Like Peter, I too agree with most of your comments.
>>> Specific points below.
>>>
>>> Best regards,
>>> Allyn
>>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>> Sent: Tuesday, October 12, 2010 6:47 AM
>>> To: dispatch@ietf.org
>>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>>
>>> Hi,
>>>
>>> Some colleagues and I have reviewed the two telepresence drafts and
>>> version 5 of the charter and have some comments on the charter.
>>>
>>> To make our comments more understandable I think we should explain our
>>> view on telepresence. First we are expecting great variations in the
>>> capabilities of the participating terminals in a session. From personal
>>> terminals for individual users up to multi-display, cameras and
>>> microphone setups. We also expect that more than two end-points being
>>> present in the same session. As we see it it will require good support
>>> for centralized media handling in a mixer (MCU). We also see video
>>> streams not only of high quality to be presented for maximum immersion,
>>> but also lower qualities that can be used to monitor activity in other
>>> rooms that aren't the ones selected for high quality.
>>>
>>> [AR] Yes, absolutely.
>>>
>>> Independent if one uses a central mixer or a full mesh there appears to
>>> be a lack in media stream negotiation for how to negotiate what
>>> capabilities the receiving client has in receiving, decoding and
>>> rendering when it comes to number of simultaneous streams. Our
>>> understanding is that RTP mandates any number of streams, while in
>>> reality outside of conferencing that is in fact one single stream.
>>> However with telepresence and conferencing there is a need to negotiate
>>> the actual capability, especially between a mixer and client when it
>>> comes to deliver multiple streams for various usages.
>>>
>>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>>
>>> We also think there is missing functionality for expressing who are the
>>> currently active speakers. The RTP CSRC field are not sufficient as they
>>> express which end-points are included in the mix. However, as one likely
>>> include in a mix more rooms than the active speakers to ensure that any
>>> interruption and ambient sound are correctly captured an mechanism for
>>> indicating the currently active speakers are needed. There is also need
>>> to agree on the type of identifier to be used that allow for mapping to
>>> full user names when appropriate.
>>>
>>> [AR] Yes.. Both of these points are covered in the charter too, right?
>>>
>>> The charter includes an item to work on control between client and a
>>> mixer what streams should be delivered. We would like to extended this
>>> to allow also the mixer to control the client on when streams are
>>> delivered. In cases where there are more users then available display
>>> area and a particular user is not actually included in any outgoing
>>> mix/selection from the mixer then there is little point in that the
>>> client consumes network resources. Thus we suggest that the stream
>>> selection and control work is extended to allow also the mixer to pause
>>> individual streams from the client to the mixer. Please note that such
>>> control needs to be responsive, thus using the SIP signaling path for
>>> this is to slow, it needs to be on the media path.
>>>
>>> [AR} I agree with your point that it would be good to pause the sender, but I think that falls
>>>    under the category of "continuous conference control" which is out of the scope of this charter.
>>>    Such a feedback mechanism falls probably under AVT.
>>> I think it would be good to add a sentence saying something like, As control issues
>>> necessary for interoperability arise, they will be directed appropriately.
>>> What do you think?
>>>
>>>
>>> Another question to the group is, are there agreement on how streams
>>> should organized into RTP sessions and how they are utilized? If
>>> anything is documented I would appreciate a pointer. The media streams
>>> are for different purposes from different end-points which are then
>>> mixed by a central point and delivered to end-points. My impression for
>>> example is that most proprietary system breaks the RTP semantics and use
>>> individual sessions between the mixer and each end-point, instead of a
>>> joint session. Here also different robustification tools for RTP such as
>>> FEC and Retransmission needs to be considered. This might be work that
>>> belongs in AVT, but are there need for further clarification into this
>>> field?
>>>
>>> [AR} Agree this needs to be worked out for Telepresence, and that any activity about it would go to AVT and/or MMUSIC.
>>>
>>> Comments much appreciated and we can gladly suggest text additions to
>>> the charter proposal.
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> ----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> ----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> Färögatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> ----------------------------------------------------------------------
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
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>> https://www.ietf.org/mailman/listinfo/dispatch
>>
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>

From pkyzivat@cisco.com  Fri Oct 15 17:05:27 2010
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On 10/15/2010 4:45 PM, Worley, Dale R (Dale) wrote:
> ________________________________________
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat [pkyzivat@cisco.com]
>
> I don't understand the point of this draft.
>
> The sip.instance feature tag takes a urn as a value. Either the imei urn
> works with that or it doesn't. If it does, no draft is needed. If it
> doesn't, then I don't think a draft will help, unless it is an extension
> to 5626.
> _______________________________________________
>
> As I see it, the point of the draft is the definition/registration of the "gsma:" scheme for URNs.  In that regard, it is a proper step to enable use of "gsma:imea" URNs as instance identifiers.

This draft doesn't define a new urn scheme. IIUC the one referenced has 
previously been defined.

However, I went and checked 5656 again, and I see that it does have a 
normative requirement on an RFC for use of something other than a UUID 
urn. So there is point to this draft.

But my other points hold. As I interpret things, registrars should not 
have to change to accommodate new URNs in this role.

IMO 5626 should simply have spelled out the required characteristics for 
a URN that can be used, rather than requiring an RFC. But that is water 
over the dam.

	Thanks,
	Paul

> In that regard, though, I don't see the function of section 4.  It seems to describe now IMEAs can be encoded via BCD, which seems to have nothing to do with the gsma:imea URNs within SIP.
>
> Also, in the Abstract, the expansion of the acronym GSMA should have been done the first time it is used, rather than the last time.
>
> Dale
>

From aallen@rim.com  Sat Oct 16 12:06:59 2010
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Paul

The reason for the draft is that RFC 5626 requires an RFC for use of URNs ot=
her than UUID as instance IDs.

With regard to your comments about putting requirements on the registrar may=
be any requirements for gr parameter generation can be left to 3GPP since th=
at is domain specific policy in any case.

Andrew

----- Original Message -----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
Sent: Friday, October 15, 2010 12:33 PM
To: dispatch@ietf.org <dispatch@ietf.org>
Subject: Re: [dispatch] draft-allen-dispatch-imei-urn-as-instanceid-00

Andrew,

I don't understand the point of this draft.

The sip.instance feature tag takes a urn as a value. Either the imei urn 
works with that or it doesn't. If it does, no draft is needed. If it 
doesn't, then I don't think a draft will help, unless it is an extension 
to 5626.

5626 is a little hazy on whether all URNs are ok or not. It implies that 
it only works for ones that can be compared as strings for equality. 
Operationally I think that must be the requirement.

It would be very bad if implementors of gruu and outbound had to 
understand each type of URN that might be used as the instance id.

I gather that the imei urn might not always work with simple string 
comparison as the identity test. Hence you have included registrar 
procedures to ignore parameters when comparing the imei urns.

ISTM that you should not be imposing *any* new requirements on 
registrars. I guess there could be a purpose for such a draft if it was 
to impose requirements on user agents, about how they format their imei 
URNs so that they will work with generic comparison procedures at the 
registrar.

	Thanks,
	Paul

On 10/15/2010 6:34 AM, Andrew Allen wrote:
> A new Internet Draft draft-allen-dispatch-imei-urn-as-instanceid-00 has
> been submitted. It can be found at:
>
> http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-00.txt
>
> The referenced URN definition draft can also be found at:
>
> http://www.ietf.org/id/draft-montemurro-gsma-imei-urn-05.txt
>
> The draft defines how the IMEI (International Mobile Equipment
> Identifier) encoded as a URN can be used as an Instance ID as defined in
> Outbound (RFC 5626) and as used by GRUU (RFC 5627). 3GPP have a need to
> use the IMEI as an Instance ID as explained in the draft.
>
> Please provide comments and feedback.
>
> Thanks
>
> Andrew
>
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_______________________________________
From: Paul Kyzivat [pkyzivat@cisco.com]

This draft doesn't define a new urn scheme. IIUC the one referenced has
previously been defined.
_______________________________________

I don't see "gsma" listed in the IANA registry (http://www.iana.org/assignm=
ents/uri-schemes.html).

Dale

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On 10/16/2010 3:08 PM, Andrew Allen wrote:
>
> Paul
>
> The reason for the draft is that RFC 5626 requires an RFC for use of URNs other than UUID as instance IDs.

Yes, my bad.

> With regard to your comments about putting requirements on the registrar maybe any requirements for gr parameter generation can be left to 3GPP since that is domain specific policy in any case.

My concern was mostly with imposing particular URN comparison 
requirements on the registrar. IMO the registrar must not be required to 
understand this particular URN or apply any new comparison rules when it 
is used. The registrar should apply the same procedures regardless of 
URN type.

	Thanks,
	Paul

> Andrew
>
> ----- Original Message -----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Friday, October 15, 2010 12:33 PM
> To: dispatch@ietf.org<dispatch@ietf.org>
> Subject: Re: [dispatch] draft-allen-dispatch-imei-urn-as-instanceid-00
>
> Andrew,
>
> I don't understand the point of this draft.
>
> The sip.instance feature tag takes a urn as a value. Either the imei urn
> works with that or it doesn't. If it does, no draft is needed. If it
> doesn't, then I don't think a draft will help, unless it is an extension
> to 5626.
>
> 5626 is a little hazy on whether all URNs are ok or not. It implies that
> it only works for ones that can be compared as strings for equality.
> Operationally I think that must be the requirement.
>
> It would be very bad if implementors of gruu and outbound had to
> understand each type of URN that might be used as the instance id.
>
> I gather that the imei urn might not always work with simple string
> comparison as the identity test. Hence you have included registrar
> procedures to ignore parameters when comparing the imei urns.
>
> ISTM that you should not be imposing *any* new requirements on
> registrars. I guess there could be a purpose for such a draft if it was
> to impose requirements on user agents, about how they format their imei
> URNs so that they will work with generic comparison procedures at the
> registrar.
>
> 	Thanks,
> 	Paul
>
> On 10/15/2010 6:34 AM, Andrew Allen wrote:
>> A new Internet Draft draft-allen-dispatch-imei-urn-as-instanceid-00 has
>> been submitted. It can be found at:
>>
>> http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-00.txt
>>
>> The referenced URN definition draft can also be found at:
>>
>> http://www.ietf.org/id/draft-montemurro-gsma-imei-urn-05.txt
>>
>> The draft defines how the IMEI (International Mobile Equipment
>> Identifier) encoded as a URN can be used as an Instance ID as defined in
>> Outbound (RFC 5626) and as used by GRUU (RFC 5627). 3GPP have a need to
>> use the IMEI as an Instance ID as explained in the draft.
>>
>> Please provide comments and feedback.
>>
>> Thanks
>>
>> Andrew
>>
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On 10/17/2010 8:25 PM, Worley, Dale R (Dale) wrote:
> _______________________________________
> From: Paul Kyzivat [pkyzivat@cisco.com]
>
> This draft doesn't define a new urn scheme. IIUC the one referenced has
> previously been defined.
> _______________________________________
>
> I don't see "gsma" listed in the IANA registry (http://www.iana.org/assignments/uri-schemes.html).

Maybe its in progress or something. I didn't see anything in this draft 
that actually registers a new urn scheme.

	Thanks,
	Paul

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To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 18 Oct 2010 11:14:43 -0400
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________________________________________
From: Paul Kyzivat [pkyzivat@cisco.com]

On 10/17/2010 8:25 PM, Worley, Dale R (Dale) wrote:
> I don't see "gsma" listed in the IANA registry (http://www.iana.org/assig=
nments/uri-schemes.html).

Maybe its in progress or something. I didn't see anything in this draft
that actually registers a new urn scheme.
________________________________________

OK, there it is:  https://datatracker.ietf.org/doc/draft-montemurro-gsma-im=
ei-urn/
That draft has been kicking around for several years.

Dale

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Andrew Allen <aallen@rim.com>
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________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Pa=
ul Kyzivat [pkyzivat@cisco.com]

My concern was mostly with imposing particular URN comparison
requirements on the registrar. IMO the registrar must not be required to
understand this particular URN or apply any new comparison rules when it
is used. The registrar should apply the same procedures regardless of
URN type.
_______________________________________________

The sensible rules are spelled out in RFC 5626 section 4.1:

      Thus, equality comparisons are performed using the rules for
      URN equality that are specific to the scheme in the URN.  If the
      element performing the comparisons does not understand the URN
      scheme, it performs the comparisons using the lexical equality
      rules defined in [RFC2141].  Lexical equality could result in two
      URNs being considered unequal when they are actually equal.  In
      this specific usage of URNs, the only element that provides the
      URN is the SIP UA instance identified by that URN.  As a result,
      the UA instance has to provide lexically equivalent URNs in each
      registration it generates.  This is likely to be normal behavior
      in any case; clients are not likely to modify the value of the
      instance ID so that it remains functionally equivalent to (yet
      lexicographically different from) previous registrations.

The problem isn't that the URN scheme definition provides an elaborate defi=
nition of equality that is suitable to its application, but that UAs must a=
lways provide instance ids that are the same character string, even if the =
URN scheme allows them flexibility in how to represent the URN.

Dale

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On 10/18/2010 11:16 AM, Worley, Dale R (Dale) wrote:
> ________________________________________
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat [pkyzivat@cisco.com]
>
> My concern was mostly with imposing particular URN comparison
> requirements on the registrar. IMO the registrar must not be required to
> understand this particular URN or apply any new comparison rules when it
> is used. The registrar should apply the same procedures regardless of
> URN type.
> _______________________________________________
>
> The sensible rules are spelled out in RFC 5626 section 4.1:
>
>        Thus, equality comparisons are performed using the rules for
>        URN equality that are specific to the scheme in the URN.  If the
>        element performing the comparisons does not understand the URN
>        scheme, it performs the comparisons using the lexical equality
>        rules defined in [RFC2141].  Lexical equality could result in two
>        URNs being considered unequal when they are actually equal.  In
>        this specific usage of URNs, the only element that provides the
>        URN is the SIP UA instance identified by that URN.  As a result,
>        the UA instance has to provide lexically equivalent URNs in each
>        registration it generates.  This is likely to be normal behavior
>        in any case; clients are not likely to modify the value of the
>        instance ID so that it remains functionally equivalent to (yet
>        lexicographically different from) previous registrations.
>
> The problem isn't that the URN scheme definition provides an elaborate definition of equality that is suitable to its application, but that UAs must always provide instance ids that are the same character string, even if the URN scheme allows them flexibility in how to represent the URN.

IMO the problem is that this draft expects the registrar to use special 
comparison rules for this sort of URN.

If it removed that requirement, and instead put a requirement on the UA 
to construct its URN in such a way that the lexical equality rule of 
2141 is sufficient, then I think all would be well.

	Thanks,
	Paul

From dworley@avaya.com  Mon Oct 18 14:21:56 2010
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 18 Oct 2010 17:19:48 -0400
Thread-Topic: [dispatch] draft-allen-dispatch-imei-urn-as-instanceid-00
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> From: Paul Kyzivat [pkyzivat@cisco.com]
>=20
> IMO the problem is that this draft expects the registrar to use special
> comparison rules for this sort of URN.
>=20
> If it removed that requirement, and instead put a requirement on the UA
> to construct its URN in such a way that the lexical equality rule of
> 2141 is sufficient, then I think all would be well.

Yes, clearly, having the UAs depend on special behavior in the
registrar for this URN scheme would mean that only specialized
registrars could cope with UAs that used these URNs as their instance
id.  That would minimize compatibility.  The draft needs to be revised
to follow RFC 5626 section 4.1 (which is referenced from the GRUU
RFC).

Dale

From john.elwell@siemens-enterprise.com  Tue Oct 19 04:02:41 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: DISPATCH list <dispatch@ietf.org>
Date: Tue, 19 Oct 2010 13:04:08 +0200
Thread-Topic: Comment on draft-kaplan-dispatch-gruu-problematic-00
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Subject: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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An interesting draft.

One particular point I noticed, in section 6:
"Furthermore, due to the proliferation of B2BUAs in the signaling=20
   path, it's not very common that a Contact URI is not routable-to by=20
   a UA."

The problem is out-of-dialog requests, when the contact URI supplied by a B=
2BUA does not have a lifetime beyond that of the original dialog. So you mi=
ght be able to route back to the B2BUA, but whether it will be interpreted =
correctly when you get there is another matter.

Apart from call transfer, another problem might be the routing of a callbac=
k to a device that initiated an emergency call. The "Global Contacts" solut=
ion would work only if a global contact has a lifetime beyond a given dialo=
g.

John=

From pkyzivat@cisco.com  Tue Oct 19 06:40:59 2010
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Subject: Re: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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I haven't seen the announcement for this draft yet, so this was my first 
introduction to it.

While there are some real problems identified in here, for the most part 
this is calling out cases where things have been done wrong or poorly, 
and blaming gruu.

I do think it will be useful to go through this point by point and 
discuss plausible remedies.

IMO it is the responsibility of those deploying B2BUAs to do so in a way 
that doesn't break things. I realize that in some cases the *purpose* of 
the B2BUA is to break things, and that can be a valid thing to do, as 
long as it only breaks the things that should be broken.

Also, ISTM that its about time to start doing something about access 
limited networks. Access limiting sip networks makes as little sense as 
access limiting email networks. In spite of spam problems, the world 
would not work if email was restricted the way sip is.

	Thanks,
	Paul

On 10/19/2010 7:04 AM, Elwell, John wrote:
> An interesting draft.
>
> One particular point I noticed, in section 6:
> "Furthermore, due to the proliferation of B2BUAs in the signaling
>     path, it's not very common that a Contact URI is not routable-to by
>     a UA."
>
> The problem is out-of-dialog requests, when the contact URI supplied by a B2BUA does not have a lifetime beyond that of the original dialog. So you might be able to route back to the B2BUA, but whether it will be interpreted correctly when you get there is another matter.
>
> Apart from call transfer, another problem might be the routing of a callback to a device that initiated an emergency call. The "Global Contacts" solution would work only if a global contact has a lifetime beyond a given dialog.
>
> John
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From HKaplan@acmepacket.com  Tue Oct 19 08:50:34 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Tue, 19 Oct 2010 11:51:42 -0400
Thread-Topic: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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Sorry, I didn't announce it because it's not my final version - I just subm=
itted what I had as a placeholder due to the deadline. (there's a loophole =
in the whole I-D submission thing that you can't submit draft-00 after yest=
erday, but you can submit a 00 yesterday as a placeholder and then a 01 nex=
t week, effectively letting you submit a brand new I-D a week late. :)

With regards to below, yes absolutely - if the Global Contact doesn't survi=
ve a dialog's lifetime, then by definition it wont' be usable later.  I don=
't know of any B2BUA's that generate Global Contacts that have only a lifet=
ime of a dialog *on purpose*, but since the original Contact-URI they gener=
ated the Global based on may only have a lifetime of a dialog, the Global C=
ontact would inherit that behavior as well in effect.  And I do know of som=
e devices which generate Contact-URIs which only last a dialog (because the=
y encode call-specific info in their contact).  So if their request goes th=
rough a B2BUA, the Global Contact created by the B2BUA for them only lasts =
that long too, from a usefulness perspective.

-hadriel


On Oct 19, 2010, at 7:04 AM, Elwell, John wrote:

> An interesting draft.
>=20
> One particular point I noticed, in section 6:
> "Furthermore, due to the proliferation of B2BUAs in the signaling=20
>   path, it's not very common that a Contact URI is not routable-to by=20
>   a UA."
>=20
> The problem is out-of-dialog requests, when the contact URI supplied by a=
 B2BUA does not have a lifetime beyond that of the original dialog. So you =
might be able to route back to the B2BUA, but whether it will be interprete=
d correctly when you get there is another matter.
>=20
> Apart from call transfer, another problem might be the routing of a callb=
ack to a device that initiated an emergency call. The "Global Contacts" sol=
ution would work only if a global contact has a lifetime beyond a given dia=
log.
>=20
> John
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From adam@nostrum.com  Tue Oct 19 09:32:04 2010
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Subject: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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  On 10/19/10 10:51, Oct 19, Hadriel Kaplan wrote:
> Sorry, I didn't announce it because it's not my final version - I just submitted what I had as a placeholder due to the deadline. (there's a loophole in the whole I-D submission thing that you can't submit draft-00 after yesterday, but you can submit a 00 yesterday as a placeholder and then a 01 next week, effectively letting you submit a brand new I-D a week late. :)

I don't think that's the intention. In fact, historically (prior to the 
tools), the i-d repository manager would reject such submissions as 
being in violation of the deadlines. You can see, for example, 
discussion of the "no placeholder" rule in this email exchange:

https://www.ietf.org/ibin/c5i?mid=6&rid=49&gid=0&k1=933&k2=21546&tid=1259394188

I can't find any current policy documents that outline this, and I'm not 
sure whether the tools allow you to bypass the deadlines the way you 
intend (I guess you'll find out). But I believe that omitting discussion 
of the "no placeholder" rule in the i-d author guidelines document is an 
oversight.

The idea, of course, is that people need additional time to digest new 
ideas than they do to process updates to existing documents. Submitting 
an unfinished placeholder to subvert this intention seems outside the 
spirit of the deadlines.

I'm copying Russ here as current author of the "Guidelines to Authors of 
Internet Drafts," and Henrik as an Important Person in the tools 
pantheon to see if they can comment on official policy and tools 
behavior on this matter.

/a

From stpeter@stpeter.im  Tue Oct 19 09:39:22 2010
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Date: Tue, 19 Oct 2010 10:40:50 -0600
From: Peter Saint-Andre <stpeter@stpeter.im>
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Cc: dispatch@ietf.org, Russ Housley <housley@vigilsec.com>, Henrik Levkowetz <henrik@levkowetz.com>
Subject: Re: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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On 10/19/10 10:33 AM, Adam Roach wrote:
>  On 10/19/10 10:51, Oct 19, Hadriel Kaplan wrote:
>> Sorry, I didn't announce it because it's not my final version - I just=

>> submitted what I had as a placeholder due to the deadline. (there's a
>> loophole in the whole I-D submission thing that you can't submit
>> draft-00 after yesterday, but you can submit a 00 yesterday as a
>> placeholder and then a 01 next week, effectively letting you submit a
>> brand new I-D a week late. :)
>=20
> I don't think that's the intention. In fact, historically (prior to the=

> tools), the i-d repository manager would reject such submissions as
> being in violation of the deadlines. You can see, for example,
> discussion of the "no placeholder" rule in this email exchange:
>=20
> https://www.ietf.org/ibin/c5i?mid=3D6&rid=3D49&gid=3D0&k1=3D933&k2=3D21=
546&tid=3D1259394188
>=20
>=20
> I can't find any current policy documents that outline this, and I'm no=
t
> sure whether the tools allow you to bypass the deadlines the way you
> intend (I guess you'll find out). But I believe that omitting discussio=
n
> of the "no placeholder" rule in the i-d author guidelines document is a=
n
> oversight.
>=20
> The idea, of course, is that people need additional time to digest new
> ideas than they do to process updates to existing documents. Submitting=

> an unfinished placeholder to subvert this intention seems outside the
> spirit of the deadlines.

However, the matter is not always clear-cut. Sometimes an author has
found the time to roughly sketch out a proposal (resulting in -00) and
wants to get that out for discussion, but hasn't yet had time to create
examples, track down all the references, etc. It's not always easy to
draw a sharp line between "placeholder", "stub", "rough draft", etc.

Do we have examples of true placeholders that were then replaced with
something totally different?

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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Cc: dispatch@ietf.org, Russ Housley <housley@vigilsec.com>, Henrik Levkowetz <henrik@levkowetz.com>
Subject: Re: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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  On 10/19/10 11:40, Oct 19, Peter Saint-Andre wrote:
> On 10/19/10 10:51, Oct 19, Hadriel Kaplan wrote:
>> >>  Sorry, I didn't announce it because it's not my final version - I just
>> >>  submitted what I had as a placeholder due to the deadline.
...
> However, the matter is not always clear-cut. Sometimes an author has 
> found the time to roughly sketch out a proposal (resulting in -00) and 
> wants to get that out for discussion, but hasn't yet had time to 
> create examples, track down all the references, etc. It's not always 
> easy to draw a sharp line between "placeholder", "stub", "rough 
> draft", etc.

Perhaps I'm being naÃ¯ve, but I would suspect an author calling their own 
draft a "placeholder" actually puts a pretty bright line around it.

More to the point: if you have something together by the -00 deadline 
that is sufficient to foster discussion, there's no real rush to get the 
i's dotted and the t's crossed by the -01 deadline. Just hold off on the 
examples, references, and other polish until after the meeting. On the 
other hand, if it's not in good enough shape that people can use it as 
the basis of a technical conversation, it is exactly equivalent to an 
empty placeholder.

/a

From HKaplan@acmepacket.com  Tue Oct 19 09:51:16 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 19 Oct 2010 12:52:37 -0400
Thread-Topic: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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Subject: Re: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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Hi Paul,
comments inline...

On Oct 19, 2010, at 9:42 AM, Paul Kyzivat wrote:

> I haven't seen the announcement for this draft yet, so this was my first=
=20
> introduction to it.
>=20
> While there are some real problems identified in here, for the most part=
=20
> this is calling out cases where things have been done wrong or poorly,=20
> and blaming gruu.

I don't think I'm "blaming gruu" (well... except for the argument that the =
RFC should have made the UA insert a Record-Route of its real contact).  In=
 general the draft isn't trying to blame gruu - it's trying to explain why =
gruu isn't a magic cure-all.


> I do think it will be useful to go through this point by point and=20
> discuss plausible remedies.

How would you like to do that?  Email in dispatch, or separate mailing list=
, or future IETF Dispatch meeting, or BOF?


> IMO it is the responsibility of those deploying B2BUAs to do so in a way=
=20
> that doesn't break things. I realize that in some cases the *purpose* of=
=20
> the B2BUA is to break things, and that can be a valid thing to do, as=20
> long as it only breaks the things that should be broken.

I think the hard part here is figuring out which behavior doesn't break thi=
ngs.  I've convinced myself that "passing on" the GRUU will break things mo=
re than replacing it, but I'm not sure whether replacing it with a self-mad=
e GRUU is actually good or bad.  I'm not sure that a B2BUA or any system ev=
er really knows whether its Contact URI is globally routable to, or whether=
 it actually matters or not.


> Also, ISTM that its about time to start doing something about access=20
> limited networks. Access limiting sip networks makes as little sense as=20
> access limiting email networks. In spite of spam problems, the world=20
> would not work if email was restricted the way sip is.

Good luck with that. :)
Email's spam level would destroy voip's viability, never-mind the regulator=
y, billing, and interop issues.  There are free SIP domains without access =
restrictions, but they're the minority afaict.
Over time this may well change, but I doubt it'll be soon.

-hadriel


From stpeter@stpeter.im  Tue Oct 19 09:52:01 2010
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Cc: dispatch@ietf.org, Russ Housley <housley@vigilsec.com>, Henrik Levkowetz <henrik@levkowetz.com>
Subject: Re: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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On 10/19/10 10:48 AM, Adam Roach wrote:
>  On 10/19/10 11:40, Oct 19, Peter Saint-Andre wrote:
>> On 10/19/10 10:51, Oct 19, Hadriel Kaplan wrote:
>>> >>  Sorry, I didn't announce it because it's not my final version - I=

>>> just
>>> >>  submitted what I had as a placeholder due to the deadline.
> ...
>> However, the matter is not always clear-cut. Sometimes an author has
>> found the time to roughly sketch out a proposal (resulting in -00) and=

>> wants to get that out for discussion, but hasn't yet had time to
>> create examples, track down all the references, etc. It's not always
>> easy to draw a sharp line between "placeholder", "stub", "rough
>> draft", etc.
>=20
> Perhaps I'm being na=C3=AFve, but I would suspect an author calling the=
ir own
> draft a "placeholder" actually puts a pretty bright line around it.

Given that we don't have a clear definition of "placeholder", I'm not so
sure. Is it a sketch, a stub, something that you submitted only to get
draft-smith-foobar-00 and then you're going to completely change the
meaning of "foobar" next week?

> More to the point: if you have something together by the -00 deadline
> that is sufficient to foster discussion, there's no real rush to get th=
e
> i's dotted and the t's crossed by the -01 deadline. Just hold off on th=
e
> examples, references, and other polish until after the meeting.=20

Some people are perfectionists and it pains them to submit anything
that's merely a rough draft.

> On the
> other hand, if it's not in good enough shape that people can use it as
> the basis of a technical conversation, it is exactly equivalent to an
> empty placeholder.

Ah, you didn't say "empty placeholder" before. That makes all the
difference. :)

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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From HKaplan@acmepacket.com  Tue Oct 19 10:03:33 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>
Date: Tue, 19 Oct 2010 13:04:40 -0400
Thread-Topic: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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Subject: Re: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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Oh the tool will let you do it unless it's been changed recently - I've don=
e it several times before.  Although in my defense the draft's weren't real=
ly "placeholders" in the sense of being empty boilerplates, but more like I=
 hadn't completely finished them.

But I don't think the tool should be changed to prevent it - the real purpo=
se as you say is to provide people with time to read, and if people feel a =
late draft is too late, then they won't read it and it wont' get time at th=
e meeting anyway.  And in some of the RAI working groups in particular you =
couldn't expect meeting time for a draft submitted *before* the deadline, l=
et alone after, because our agenda is fixed long before the submission dead=
line. (September 6th in Dispatch's case) =20
I didn't submit this I-D for meeting time.

-hadriel

On Oct 19, 2010, at 12:33 PM, Adam Roach wrote:

>  On 10/19/10 10:51, Oct 19, Hadriel Kaplan wrote:
>> Sorry, I didn't announce it because it's not my final version - I just s=
ubmitted what I had as a placeholder due to the deadline. (there's a loopho=
le in the whole I-D submission thing that you can't submit draft-00 after y=
esterday, but you can submit a 00 yesterday as a placeholder and then a 01 =
next week, effectively letting you submit a brand new I-D a week late. :)
>=20
> I don't think that's the intention. In fact, historically (prior to the=20
> tools), the i-d repository manager would reject such submissions as=20
> being in violation of the deadlines. You can see, for example,=20
> discussion of the "no placeholder" rule in this email exchange:
>=20
> https://www.ietf.org/ibin/c5i?mid=3D6&rid=3D49&gid=3D0&k1=3D933&k2=3D2154=
6&tid=3D1259394188
>=20
> I can't find any current policy documents that outline this, and I'm not=
=20
> sure whether the tools allow you to bypass the deadlines the way you=20
> intend (I guess you'll find out). But I believe that omitting discussion=
=20
> of the "no placeholder" rule in the i-d author guidelines document is an=
=20
> oversight.
>=20
> The idea, of course, is that people need additional time to digest new=20
> ideas than they do to process updates to existing documents. Submitting=20
> an unfinished placeholder to subvert this intention seems outside the=20
> spirit of the deadlines.
>=20
> I'm copying Russ here as current author of the "Guidelines to Authors of=
=20
> Internet Drafts," and Henrik as an Important Person in the tools=20
> pantheon to see if they can comment on official policy and tools=20
> behavior on this matter.
>=20
> /a
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From HKaplan@acmepacket.com  Tue Oct 19 10:10:32 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>
Date: Tue, 19 Oct 2010 13:11:31 -0400
Thread-Topic: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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Subject: Re: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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Ahh - sorry for the confusion - it's not a "placeholder" in the sense of em=
pty boilerplate - more like I wanted to tidy it up a bit before announcing =
it because I'm a nit-picker.

I'm not asking for any meeting time for this topic - I just wanted to get i=
t in before IETF-79 because I had promised I would submit such a draft at t=
he face-to-face meeting of Martini in Maastricht.

-hadriel

On Oct 19, 2010, at 12:48 PM, Adam Roach wrote:

> Perhaps I'm being na=EFve, but I would suspect an author calling their ow=
n=20
> draft a "placeholder" actually puts a pretty bright line around it.
>=20
> More to the point: if you have something together by the -00 deadline=20
> that is sufficient to foster discussion, there's no real rush to get the=
=20
> i's dotted and the t's crossed by the -01 deadline. Just hold off on the=
=20
> examples, references, and other polish until after the meeting. On the=20
> other hand, if it's not in good enough shape that people can use it as=20
> the basis of a technical conversation, it is exactly equivalent to an=20
> empty placeholder.
>=20




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Subject: Re: [dispatch] Placeholder Drafts to Circumvent Deadlines (was Re: Comment on draft-kaplan-dispatch-gruu-problematic-00)
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On 10/19/10 11:11 AM, Hadriel Kaplan wrote:

> Ahh - sorry for the confusion - it's not a "placeholder" in the sense
> of empty boilerplate - more like I wanted to tidy it up a bit before
> announcing it because I'm a nit-picker.

Tempest, meet teapot. :)



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From dworley@avaya.com  Tue Oct 19 10:25:16 2010
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Tue, 19 Oct 2010 13:21:50 -0400
Thread-Topic: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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My first impression is "Don't *do* that!".

At the very least, the element that mints GRUUs, or rather the administrati=
ve structure that is responsible for it, is responsible for ensuring that t=
he GRUUs are *globally routable*.  If the SIP domain is part of an access-l=
imited network, then they need to construct proper globally-advertised and =
globally-accessible incoming gateways.

If they're minting URIs that claim to be GRUUs but are not, the problem isn=
't with GRUUs per se but rather that they're not minting GRUUs.  More subtl=
y, the problem may be that there are situations where it is *not possible* =
to provide a globally-routable URI for a UA that has a indefinite lifetime.=
  We may need to provide alternative solutions for that problem.

Dale

From adam@nostrum.com  Tue Oct 19 10:29:32 2010
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Subject: Re: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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  On 10/19/10 12:21, Oct 19, Worley, Dale R (Dale) wrote:
> More subtly, the problem may be that there are situations where it is*not possible*  to provide a globally-routable URI for a UA that has a indefinite lifetime.

Huh?

Do you mean GRUU or temp GRUU?

/a

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    On 10/19/10 12:21, Oct 19, Worley, Dale R (Dale) wrote:
    <blockquote
cite="mid:CD5674C3CD99574EBA7432465FC13C1B2202288954@DC-US1MBEX4.global.avaya.com"
      type="cite">
      <pre wrap="">More subtly, the problem may be that there are situations where it is <b class="moz-txt-star"><span class="moz-txt-tag">*</span>not possible<span class="moz-txt-tag">*</span></b> to provide a globally-routable URI for a UA that has a indefinite lifetime.</pre>
    </blockquote>
    <br>
    Huh?<br>
    <br>
    Do you mean GRUU or temp GRUU?<br>
    <br>
    /a<br>
  </body>
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Adam Roach <adam@nostrum.com>
Date: Tue, 19 Oct 2010 13:33:07 -0400
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________________________________________
From: Adam Roach [adam@nostrum.com]

> More subtly, the problem may be that there are situations where it is *no=
t possible*
> to provide a globally-routable URI for a UA that has a indefinite lifetim=
e.

Huh?

Do you mean GRUU or temp GRUU?
________________________________________

If the only way for global access to a UA is through some sort of border SB=
C, and if the border SBC won't keep any information around much longer than=
 the length of a call, then it is impossible to provide a globally-routable=
, long-term URI to the UA.  (Or at least, it appears so to me at this momen=
t.  There may be tricks to avoid the problem.)  Which means that it is impo=
ssible to provide a GRUU for the UA.

Dale

From HKaplan@acmepacket.com  Tue Oct 19 11:16:25 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Worley, Dale R (Dale)" <dworley@avaya.com>
Date: Tue, 19 Oct 2010 14:17:38 -0400
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On Oct 19, 2010, at 1:21 PM, Worley, Dale R (Dale) wrote:

> My first impression is "Don't *do* that!".
> At the very least, the element that mints GRUUs, or rather the administra=
tive structure that is responsible for it, is responsible for ensuring that=
 the GRUUs are *globally routable*.  If the SIP domain is part of an access=
-limited network, then they need to construct proper globally-advertised an=
d globally-accessible incoming gateways.

Then by definition they're *not* an access-restricted domain.  There's no d=
ebate that if everyone can successfully send a request into your domain, fr=
om anywhere on the public Internets (v4 and v6), you can use GRUUs.  The dr=
aft doesn't say otherwise.  What it says is that most domains don't fit tha=
t bill, so they shouldn't mint GRUUs. =20

And before you say "well, duh!", I should point out 3GPP specs think they c=
an use GRUUs, when they really can't.  There isn't a single 3GPP deployment=
 I can think of that could really use a GRUU as intended. =20
And at least one Enterprise-based vendor provides GRUUs by default, even th=
ough in every deployment I've seen them used in they shouldn't be. =20

I just don't think it's clear to folks what the ramifications are of mintin=
g a GRUU.   At least if we assume UAs really would change behavior when get=
ting a GRUU contact. (for example, if they switch to using out-of-dialog RE=
FER, or a b2bua decides to pass the gruu contact on)

-hadriel


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To: "Worley, Dale R (Dale)" <dworley@avaya.com>
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On Oct 19, 2010, at 1:33 PM, Worley, Dale R (Dale) wrote:
> If the only way for global access to a UA is through some sort of border =
SBC, and if the border SBC won't keep any information around much longer th=
an the length of a call, then it is impossible to provide a globally-routab=
le, long-term URI to the UA.  (Or at least, it appears so to me at this mom=
ent.  There may be tricks to avoid the problem.)  Which means that it is im=
possible to provide a GRUU for the UA.

It's not the "SBC" that isn't keeping the info around - afaik most SBC's ha=
ve the ability to generate "Global Contacts" which last exactly as long as =
the Contact they generated it based on. (which itself may or may not be pas=
t the duration of a call)  For example, they generate an encrypted version =
of the original.  The problem would be if the original contact they generat=
ed it based on didn't last long, which does happen in practice.  But that's=
 not a "problem" the draft really focuses on, or is meant to. (and arguably=
 it's not a "problem" to begin with)

-hadriel


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Dale, Paul

Thank you for your comments. I plan to make a revision based on them and hop=
efully resubmit before the 2nd deadline. Hopefully we can discuss in Beijing=
.

Andrew

----- Original Message -----
From: Worley, Dale R (Dale) [mailto:dworley@avaya.com]
Sent: Monday, October 18, 2010 05:19 PM
To: Paul Kyzivat <pkyzivat@cisco.com>
Cc: Andrew Allen; dispatch@ietf.org <dispatch@ietf.org>
Subject: RE: [dispatch] draft-allen-dispatch-imei-urn-as-instanceid-00

> From: Paul Kyzivat [pkyzivat@cisco.com]
> 
> IMO the problem is that this draft expects the registrar to use special
> comparison rules for this sort of URN.
> 
> If it removed that requirement, and instead put a requirement on the UA
> to construct its URN in such a way that the lexical equality rule of
> 2141 is sufficient, then I think all would be well.

Yes, clearly, having the UAs depend on special behavior in the
registrar for this URN scheme would mean that only specialized
registrars could cope with UAs that used these URNs as their instance
id.  That would minimize compatibility.  The draft needs to be revised
to follow RFC 5626 section 4.1 (which is referenced from the GRUU
RFC).

Dale

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Date: Tue, 19 Oct 2010 14:16:56 -0600
From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: [dispatch] proposed SIXPAC charter
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Earlier this year, some folks in the RAI area proposed an initiative to
define a few small extensions to both SIP and XMPP that would make it
easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on
feedback provided on the DISPATCH list and received from the RAI ADs,
I've taken the liberty of rewriting the proposed charter, in the hopes
that fresh text will spur a more conclusive discussion. I'm mostly just
trying to help the proponents put their best foot forward, so if folks
here have more feedback I'd expect that people like Simo Veikkolainen
and Emil Ivov will be able to engage in further discussion.

/psa

###

SIXPAC (SIP Integration with XMPP in Presence Aware Clients)
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Problem Statement

Both the Session Initiation Protocol (SIP) and the Extensible Messaging
and Presence Protocol (XMPP) are widely deployed technologies for
real-time communication over the Internet.  In order to offer a complete
suite of features as well as communication across multiple networks,
several user-oriented software applications support both SIP and XMPP,
and more software developers have expressed interest in building such
"dual-stack" solutions.  Unfortunately, it is difficult to provide a
good end-user experience in such applications because SIP and XMPP are
not aware of each other.  For example:

- XMPP presence does not include availability states related to a SIP
  voice call or video call (e.g., "on the phone"), thus preventing an
  a dual-stack endpoint from showing presence-based communication hints

- There is no correlation between an XMPP IM session and a SIP voice
  or video session, thus preventing a dual-stack endpoint from providing
  integrated user interfaces and communications history

- SIP accounts and XMPP accounts are not directly correlated in contact
  lists or vCards (and not all deployed services support storage of such
  information), thus preventing a dual-stack endpoint from knowing that
  a contact has both SIP and XMPP capabilities

Although some proprietary solutions exist to the foregoing problems, it
would be preferable to define standardized solutions for the sake of
improved interoperability.

Objectives

Because both SIP and XMPP are easily extended through new SIP headers
and XMPP elements, it should be possible to provide tighter integration
within dual-stack SIP/XMPP user agents to improve the user experience.

Any such extensions should meet the following criteria:

- Be completely optional and backwards-compatible for all endpoints

- Work without changes to deployed infrastructure such as existing
  SIP and XMPP servers, B2BUAs, firewalls, etc.

The SIXPAC WG will define a small number of SIP and XMPP extensions to
solve the following use cases in dual-stack endpoints:

- Including SIP-based availability states in XMPP presence (limited to
  basic presence and availability states only, not the full range of
  PIDF extensions)

- Correlating an XMPP IM session with a SIP voice/video session, and
  vice-versa

- Advertising a SIP account address over XMPP and an XMPP account
  address over SIP

Additional use cases are out of scope, including anything related to or
requiring server integration, multiparty communication, SIP-based IM
and presence, XMPP-based voice and video, file transfer, generalized
service discovery and capabilities exchange, full protocol translation
in communication gateways, shared credentials across both SIP and XMPP
accounts, rich presence extensions for features such as geolocation,
etc.  Although such topics are important and interesting, they are out
of scope for this group.

However, in addition to the protocol extensions explicitly mentioned
above, the group may also define best practices related to the
implementation and deployment of dual-stack SIP/XMPP endpoints,
including topics such as user agent configuration.

Deliverables

- Use cases and protocol requirements

- XMPP presence extension for SIP-based availability states

- Media session correlation extensions for SIP and XMPP

- Contact address advertisement extensions for SIP and XMPP

- Best practices for implementation and deployment of dual-stack
  endpoints

Milestones

Feb 2011  Submit use cases and protocol requirements document to
          IESG (Informational)

Oct 2011  Submit XMPP presence extension document to IESG
          (Standards Track)

Oct 2011  Submit media session correlation extensions document to IESG
          (Standards Track)

Oct 2011  Submit contact address advertisement extension document to
          IESG (Standards Track)

Oct 2011  Submit best practices document to IESG (Informational)

###



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From HKaplan@acmepacket.com  Tue Oct 19 13:45:32 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "dispatch@ietf.org list" <dispatch@ietf.org>
Date: Tue, 19 Oct 2010 16:46:40 -0400
Thread-Topic: New I-D:  draft-kaplan-dispatch-gruu-problematic-00
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So I suppose there's no point in waiting for a dotting of i's and crossing =
of t's...

During the Martini WG meeting in Maastricht, and a debate about mandating S=
SPs mint GRUUs in GIN, I mentioned that GRUU's weren't as useful as we thin=
k they are, and that they have some issues.  I was then told to submit an I=
-D, which is what this I-D announcement is about.

Below is the link to an I-D discussing some issues with GRUUs discovered in=
 deployments, cases in which using GRUUs may make things worse, and conside=
rations for B2BUAs.

-hadriel
p.s. and yes, the draft name is meant to be humorous

...........................

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

	Title           : Problems with the SIP Globally Routable User Agent URIs =
(GRUUs)
	Author(s)       : H. Kaplan
	Filename        : draft-kaplan-dispatch-gruu-problematic-00.txt
	Pages           : 11
	Date            : 2010-10-18

This document identifies some issues discovered in deployments of=20
the SIP GRUU mechanism defined in [RFC5627], cases in which GRUUs=20
may be harmful, and considerations for B2BUAs handling GRUUs.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-kaplan-dispatch-gruu-problematic-=
00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From pkyzivat@cisco.com  Tue Oct 19 17:20:10 2010
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On 10/19/2010 12:52 PM, Hadriel Kaplan wrote:
> Hi Paul,
> comments inline...
>
> On Oct 19, 2010, at 9:42 AM, Paul Kyzivat wrote:
>
>> I haven't seen the announcement for this draft yet, so this was my first
>> introduction to it.
>>
>> While there are some real problems identified in here, for the most part
>> this is calling out cases where things have been done wrong or poorly,
>> and blaming gruu.
>
> I don't think I'm "blaming gruu" (well... except for the argument that the RFC should have made the UA insert a Record-Route of its real contact).  In general the draft isn't trying to blame gruu - it's trying to explain why gruu isn't a magic cure-all.

Well, I *took* it as blame. :-)

I don't really want to end up as an apologist for gruu. Its not a 
wonderful thing. But its what we have, and I think we should try to make 
it work before giving up on it.

>> I do think it will be useful to go through this point by point and
>> discuss plausible remedies.
>
> How would you like to do that?  Email in dispatch, or separate mailing list, or future IETF Dispatch meeting, or BOF?

I don't think it warrants its own mailing list, or scheduled time at the 
meeting. So I think either discussion on some existing list (dispatch or 
sipcore would do), and/or informal time in Beijing.

>> IMO it is the responsibility of those deploying B2BUAs to do so in a way
>> that doesn't break things. I realize that in some cases the *purpose* of
>> the B2BUA is to break things, and that can be a valid thing to do, as
>> long as it only breaks the things that should be broken.
>
> I think the hard part here is figuring out which behavior doesn't break things.  I've convinced myself that "passing on" the GRUU will break things more than replacing it, but I'm not sure whether replacing it with a self-made GRUU is actually good or bad.  I'm not sure that a B2BUA or any system ever really knows whether its Contact URI is globally routable to, or whether it actually matters or not.

I think the hard part here is understanding the environments where 
things will break. In every case it seems to come down to:

- operators of some system/domain want to restrict use in some way
- B2BUAs (or maybe just firewalls, NATs, etc.) are inserted to
   accomplish that

- are the restrictions reasonable, or simply wrong headed?
- which things *should* work given reasonable restrictions
- do GRUUs contribute to a solution, or just make it worse?
- how must GRUUs be managed so that they do work in the
   cases where they should?

This is in some sense a revisiting of the "what is an SBC" question.

>> Also, ISTM that its about time to start doing something about access
>> limited networks. Access limiting sip networks makes as little sense as
>> access limiting email networks. In spite of spam problems, the world
>> would not work if email was restricted the way sip is.
>
> Good luck with that. :)

I realize it seems like an impossible task.
But it also once seemed like an impossible task to federate all the 
disparate networks in the world. That pretty much got done.

> Email's spam level would destroy voip's viability,

Certainly. But if email were being deployed for the first time today it 
would have a lot of the things that have now been invented to address 
the problem but haven't been widely adopted.

> never-mind the regulatory, billing, and interop issues.  There are free SIP domains without access restrictions, but they're the minority afaict.
> Over time this may well change, but I doubt it'll be soon.

It'll change or else everybody will just use Skype and the PSTN will die.

	Thanks,
	Paul

From pkyzivat@cisco.com  Tue Oct 19 17:26:28 2010
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Subject: Re: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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On the surface this seems something like the ICE problem. ICE assumes 
that the UA can't really know if it has a public address.

Its a bit different here in that we aren't talking about individual UAs 
- its really a domain administration problem. It should be possible for 
the administrator of a domain to know if it has publicly routable 
addresses or not, and then to provision those elements that will be 
minting GRUUs (or not) accordingly.

A registrar that doesn't have a public address should not support 
handing out GRUUs. That will limit the functionality of the devices it 
services.

The fits in with Martini - which potentially gives a registrar without 
its own public address a way to mint gruus for its devices.

	Thanks,
	Paul

On 10/19/2010 1:21 PM, Worley, Dale R (Dale) wrote:
> My first impression is "Don't *do* that!".
>
> At the very least, the element that mints GRUUs, or rather the administrative structure that is responsible for it, is responsible for ensuring that the GRUUs are *globally routable*.  If the SIP domain is part of an access-limited network, then they need to construct proper globally-advertised and globally-accessible incoming gateways.
>
> If they're minting URIs that claim to be GRUUs but are not, the problem isn't with GRUUs per se but rather that they're not minting GRUUs.  More subtly, the problem may be that there are situations where it is *not possible* to provide a globally-routable URI for a UA that has a indefinite lifetime.  We may need to provide alternative solutions for that problem.
>
> Dale
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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Subject: Re: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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I think we need to separate out the ipv6 issue from the gruu issue.
If I have a publicly routable URI that is only v4, then I think it still 
can qualify as a GRUU, even though v6-only nodes cannot use it.

We need some sort of mechanism to solve the v4/v6 problem for such cases.

	Thanks,
	Paul

On 10/19/2010 2:17 PM, Hadriel Kaplan wrote:
>
> On Oct 19, 2010, at 1:21 PM, Worley, Dale R (Dale) wrote:
>
>> My first impression is "Don't *do* that!".
>> At the very least, the element that mints GRUUs, or rather the administrative structure that is responsible for it, is responsible for ensuring that the GRUUs are *globally routable*.  If the SIP domain is part of an access-limited network, then they need to construct proper globally-advertised and globally-accessible incoming gateways.
>
> Then by definition they're *not* an access-restricted domain.  There's no debate that if everyone can successfully send a request into your domain, from anywhere on the public Internets (v4 and v6), you can use GRUUs.  The draft doesn't say otherwise.  What it says is that most domains don't fit that bill, so they shouldn't mint GRUUs.
>
> And before you say "well, duh!", I should point out 3GPP specs think they can use GRUUs, when they really can't.  There isn't a single 3GPP deployment I can think of that could really use a GRUU as intended.
> And at least one Enterprise-based vendor provides GRUUs by default, even though in every deployment I've seen them used in they shouldn't be.
>
> I just don't think it's clear to folks what the ramifications are of minting a GRUU.   At least if we assume UAs really would change behavior when getting a GRUU contact. (for example, if they switch to using out-of-dialog REFER, or a b2bua decides to pass the gruu contact on)
>
> -hadriel
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From john.elwell@siemens-enterprise.com  Wed Oct 20 00:44:21 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Peter Saint-Andre <stpeter@stpeter.im>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 20 Oct 2010 09:45:49 +0200
Thread-Topic: [dispatch] proposed SIXPAC charter
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I find this a worthwhile topic to pursue. I had been wondering whether this=
 activity would turn out to be more of a profiling exercise, and whether th=
e IETF might not be the best choice of venue for such work. From the curren=
t draft charter it looks like there will be at least some protocol extensio=
n work, for which I believe the IETF is the correct venue. On the other han=
d, the Unified Communications Interoperability Forum (UCIF) is seeking to a=
dvance the state of play on XMPP interoperability, and if we were just talk=
ing about a profile or BCP, that might have been a better venue. Perhaps th=
e IETF should focus on requirements and protocol extensions, and consider w=
hether the BCP work would be better done elsewhere. Or at least, there shou=
ld be some coordination with other activities relating to XMPP interoperabi=
lity.

John =20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre
> Sent: 19 October 2010 21:17
> To: dispatch@ietf.org
> Subject: [dispatch] proposed SIXPAC charter
>=20
> Earlier this year, some folks in the RAI area proposed an=20
> initiative to
> define a few small extensions to both SIP and XMPP that would make it
> easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on
> feedback provided on the DISPATCH list and received from the RAI ADs,
> I've taken the liberty of rewriting the proposed charter, in the hopes
> that fresh text will spur a more conclusive discussion. I'm=20
> mostly just
> trying to help the proponents put their best foot forward, so if folks
> here have more feedback I'd expect that people like Simo Veikkolainen
> and Emil Ivov will be able to engage in further discussion.
>=20
> /psa
>=20
> ###
>=20
> SIXPAC (SIP Integration with XMPP in Presence Aware Clients)
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Problem Statement
>=20
> Both the Session Initiation Protocol (SIP) and the Extensible=20
> Messaging
> and Presence Protocol (XMPP) are widely deployed technologies for
> real-time communication over the Internet.  In order to offer=20
> a complete
> suite of features as well as communication across multiple networks,
> several user-oriented software applications support both SIP and XMPP,
> and more software developers have expressed interest in building such
> "dual-stack" solutions.  Unfortunately, it is difficult to provide a
> good end-user experience in such applications because SIP and XMPP are
> not aware of each other.  For example:
>=20
> - XMPP presence does not include availability states related to a SIP
>   voice call or video call (e.g., "on the phone"), thus preventing an
>   a dual-stack endpoint from showing presence-based=20
> communication hints
>=20
> - There is no correlation between an XMPP IM session and a SIP voice
>   or video session, thus preventing a dual-stack endpoint=20
> from providing
>   integrated user interfaces and communications history
>=20
> - SIP accounts and XMPP accounts are not directly correlated=20
> in contact
>   lists or vCards (and not all deployed services support=20
> storage of such
>   information), thus preventing a dual-stack endpoint from=20
> knowing that
>   a contact has both SIP and XMPP capabilities
>=20
> Although some proprietary solutions exist to the foregoing=20
> problems, it
> would be preferable to define standardized solutions for the sake of
> improved interoperability.
>=20
> Objectives
>=20
> Because both SIP and XMPP are easily extended through new SIP headers
> and XMPP elements, it should be possible to provide tighter=20
> integration
> within dual-stack SIP/XMPP user agents to improve the user experience.
>=20
> Any such extensions should meet the following criteria:
>=20
> - Be completely optional and backwards-compatible for all endpoints
>=20
> - Work without changes to deployed infrastructure such as existing
>   SIP and XMPP servers, B2BUAs, firewalls, etc.
>=20
> The SIXPAC WG will define a small number of SIP and XMPP extensions to
> solve the following use cases in dual-stack endpoints:
>=20
> - Including SIP-based availability states in XMPP presence (limited to
>   basic presence and availability states only, not the full range of
>   PIDF extensions)
>=20
> - Correlating an XMPP IM session with a SIP voice/video session, and
>   vice-versa
>=20
> - Advertising a SIP account address over XMPP and an XMPP account
>   address over SIP
>=20
> Additional use cases are out of scope, including anything=20
> related to or
> requiring server integration, multiparty communication, SIP-based IM
> and presence, XMPP-based voice and video, file transfer, generalized
> service discovery and capabilities exchange, full protocol translation
> in communication gateways, shared credentials across both SIP and XMPP
> accounts, rich presence extensions for features such as geolocation,
> etc.  Although such topics are important and interesting, they are out
> of scope for this group.
>=20
> However, in addition to the protocol extensions explicitly mentioned
> above, the group may also define best practices related to the
> implementation and deployment of dual-stack SIP/XMPP endpoints,
> including topics such as user agent configuration.
>=20
> Deliverables
>=20
> - Use cases and protocol requirements
>=20
> - XMPP presence extension for SIP-based availability states
>=20
> - Media session correlation extensions for SIP and XMPP
>=20
> - Contact address advertisement extensions for SIP and XMPP
>=20
> - Best practices for implementation and deployment of dual-stack
>   endpoints
>=20
> Milestones
>=20
> Feb 2011  Submit use cases and protocol requirements document to
>           IESG (Informational)
>=20
> Oct 2011  Submit XMPP presence extension document to IESG
>           (Standards Track)
>=20
> Oct 2011  Submit media session correlation extensions document to IESG
>           (Standards Track)
>=20
> Oct 2011  Submit contact address advertisement extension document to
>           IESG (Standards Track)
>=20
> Oct 2011  Submit best practices document to IESG (Informational)
>=20
> ###
>=20
>=20
> =

From peter.musgrave@magorcorp.com  Wed Oct 20 03:37:07 2010
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [dispatch] proposed SIXPAC charter
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I also think this is a worthwhile topic.

I understand John's reservation about just doing profile work - although I
would prefer if this was done elsewhere it be be in an open forum (it's my
understanding UCIF is a 'paying members only' organization).

Peter Musgrave

On Wed, Oct 20, 2010 at 3:45 AM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> I find this a worthwhile topic to pursue. I had been wondering whether this
> activity would turn out to be more of a profiling exercise, and whether the
> IETF might not be the best choice of venue for such work. From the current
> draft charter it looks like there will be at least some protocol extension
> work, for which I believe the IETF is the correct venue. On the other hand,
> the Unified Communications Interoperability Forum (UCIF) is seeking to
> advance the state of play on XMPP interoperability, and if we were just
> talking about a profile or BCP, that might have been a better venue. Perhaps
> the IETF should focus on requirements and protocol extensions, and consider
> whether the BCP work would be better done elsewhere. Or at least, there
> should be some coordination with other activities relating to XMPP
> interoperability.
>
> John
>
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre
> > Sent: 19 October 2010 21:17
> > To: dispatch@ietf.org
> > Subject: [dispatch] proposed SIXPAC charter
> >
> > Earlier this year, some folks in the RAI area proposed an
> > initiative to
> > define a few small extensions to both SIP and XMPP that would make it
> > easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on
> > feedback provided on the DISPATCH list and received from the RAI ADs,
> > I've taken the liberty of rewriting the proposed charter, in the hopes
> > that fresh text will spur a more conclusive discussion. I'm
> > mostly just
> > trying to help the proponents put their best foot forward, so if folks
> > here have more feedback I'd expect that people like Simo Veikkolainen
> > and Emil Ivov will be able to engage in further discussion.
> >
> > /psa
> >
> > ###
> >
> > SIXPAC (SIP Integration with XMPP in Presence Aware Clients)
> > ==============================================================
> > ===========
> >
> > Problem Statement
> >
> > Both the Session Initiation Protocol (SIP) and the Extensible
> > Messaging
> > and Presence Protocol (XMPP) are widely deployed technologies for
> > real-time communication over the Internet.  In order to offer
> > a complete
> > suite of features as well as communication across multiple networks,
> > several user-oriented software applications support both SIP and XMPP,
> > and more software developers have expressed interest in building such
> > "dual-stack" solutions.  Unfortunately, it is difficult to provide a
> > good end-user experience in such applications because SIP and XMPP are
> > not aware of each other.  For example:
> >
> > - XMPP presence does not include availability states related to a SIP
> >   voice call or video call (e.g., "on the phone"), thus preventing an
> >   a dual-stack endpoint from showing presence-based
> > communication hints
> >
> > - There is no correlation between an XMPP IM session and a SIP voice
> >   or video session, thus preventing a dual-stack endpoint
> > from providing
> >   integrated user interfaces and communications history
> >
> > - SIP accounts and XMPP accounts are not directly correlated
> > in contact
> >   lists or vCards (and not all deployed services support
> > storage of such
> >   information), thus preventing a dual-stack endpoint from
> > knowing that
> >   a contact has both SIP and XMPP capabilities
> >
> > Although some proprietary solutions exist to the foregoing
> > problems, it
> > would be preferable to define standardized solutions for the sake of
> > improved interoperability.
> >
> > Objectives
> >
> > Because both SIP and XMPP are easily extended through new SIP headers
> > and XMPP elements, it should be possible to provide tighter
> > integration
> > within dual-stack SIP/XMPP user agents to improve the user experience.
> >
> > Any such extensions should meet the following criteria:
> >
> > - Be completely optional and backwards-compatible for all endpoints
> >
> > - Work without changes to deployed infrastructure such as existing
> >   SIP and XMPP servers, B2BUAs, firewalls, etc.
> >
> > The SIXPAC WG will define a small number of SIP and XMPP extensions to
> > solve the following use cases in dual-stack endpoints:
> >
> > - Including SIP-based availability states in XMPP presence (limited to
> >   basic presence and availability states only, not the full range of
> >   PIDF extensions)
> >
> > - Correlating an XMPP IM session with a SIP voice/video session, and
> >   vice-versa
> >
> > - Advertising a SIP account address over XMPP and an XMPP account
> >   address over SIP
> >
> > Additional use cases are out of scope, including anything
> > related to or
> > requiring server integration, multiparty communication, SIP-based IM
> > and presence, XMPP-based voice and video, file transfer, generalized
> > service discovery and capabilities exchange, full protocol translation
> > in communication gateways, shared credentials across both SIP and XMPP
> > accounts, rich presence extensions for features such as geolocation,
> > etc.  Although such topics are important and interesting, they are out
> > of scope for this group.
> >
> > However, in addition to the protocol extensions explicitly mentioned
> > above, the group may also define best practices related to the
> > implementation and deployment of dual-stack SIP/XMPP endpoints,
> > including topics such as user agent configuration.
> >
> > Deliverables
> >
> > - Use cases and protocol requirements
> >
> > - XMPP presence extension for SIP-based availability states
> >
> > - Media session correlation extensions for SIP and XMPP
> >
> > - Contact address advertisement extensions for SIP and XMPP
> >
> > - Best practices for implementation and deployment of dual-stack
> >   endpoints
> >
> > Milestones
> >
> > Feb 2011  Submit use cases and protocol requirements document to
> >           IESG (Informational)
> >
> > Oct 2011  Submit XMPP presence extension document to IESG
> >           (Standards Track)
> >
> > Oct 2011  Submit media session correlation extensions document to IESG
> >           (Standards Track)
> >
> > Oct 2011  Submit contact address advertisement extension document to
> >           IESG (Standards Track)
> >
> > Oct 2011  Submit best practices document to IESG (Informational)
> >
> > ###
> >
> >
> >
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

--00163630f4bbdde96504930a02a3
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I also think this is a worthwhile topic.=A0<div><br></div><div>I understand=
 John&#39;s reservation about just doing profile work - although I would pr=
efer if this was done elsewhere it be be in an open forum (it&#39;s my unde=
rstanding UCIF is a &#39;paying members only&#39; organization).=A0</div>
<div><br></div><div>Peter Musgrave<br><br><div class=3D"gmail_quote">On Wed=
, Oct 20, 2010 at 3:45 AM, Elwell, John <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:john.elwell@siemens-enterprise.com">john.elwell@siemens-enterprise.com=
</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">I find this a worthwhile topic to pursue. I=
 had been wondering whether this activity would turn out to be more of a pr=
ofiling exercise, and whether the IETF might not be the best choice of venu=
e for such work. From the current draft charter it looks like there will be=
 at least some protocol extension work, for which I believe the IETF is the=
 correct venue. On the other hand, the Unified Communications Interoperabil=
ity Forum (UCIF) is seeking to advance the state of play on XMPP interopera=
bility, and if we were just talking about a profile or BCP, that might have=
 been a better venue. Perhaps the IETF should focus on requirements and pro=
tocol extensions, and consider whether the BCP work would be better done el=
sewhere. Or at least, there should be some coordination with other activiti=
es relating to XMPP interoperability.<br>

<br>
John<br>
<div><div></div><div class=3D"h5"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ie=
tf.org</a><br>
&gt; [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@=
ietf.org</a>] On Behalf Of Peter Saint-Andre<br>
&gt; Sent: 19 October 2010 21:17<br>
&gt; To: <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt; Subject: [dispatch] proposed SIXPAC charter<br>
&gt;<br>
&gt; Earlier this year, some folks in the RAI area proposed an<br>
&gt; initiative to<br>
&gt; define a few small extensions to both SIP and XMPP that would make it<=
br>
&gt; easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on<b=
r>
&gt; feedback provided on the DISPATCH list and received from the RAI ADs,<=
br>
&gt; I&#39;ve taken the liberty of rewriting the proposed charter, in the h=
opes<br>
&gt; that fresh text will spur a more conclusive discussion. I&#39;m<br>
&gt; mostly just<br>
&gt; trying to help the proponents put their best foot forward, so if folks=
<br>
&gt; here have more feedback I&#39;d expect that people like Simo Veikkolai=
nen<br>
&gt; and Emil Ivov will be able to engage in further discussion.<br>
&gt;<br>
&gt; /psa<br>
&gt;<br>
&gt; ###<br>
&gt;<br>
&gt; SIXPAC (SIP Integration with XMPP in Presence Aware Clients)<br>
&gt; =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<br>
&gt; =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<br>
&gt;<br>
&gt; Problem Statement<br>
&gt;<br>
&gt; Both the Session Initiation Protocol (SIP) and the Extensible<br>
&gt; Messaging<br>
&gt; and Presence Protocol (XMPP) are widely deployed technologies for<br>
&gt; real-time communication over the Internet. =A0In order to offer<br>
&gt; a complete<br>
&gt; suite of features as well as communication across multiple networks,<b=
r>
&gt; several user-oriented software applications support both SIP and XMPP,=
<br>
&gt; and more software developers have expressed interest in building such<=
br>
&gt; &quot;dual-stack&quot; solutions. =A0Unfortunately, it is difficult to=
 provide a<br>
&gt; good end-user experience in such applications because SIP and XMPP are=
<br>
&gt; not aware of each other. =A0For example:<br>
&gt;<br>
&gt; - XMPP presence does not include availability states related to a SIP<=
br>
&gt; =A0 voice call or video call (e.g., &quot;on the phone&quot;), thus pr=
eventing an<br>
&gt; =A0 a dual-stack endpoint from showing presence-based<br>
&gt; communication hints<br>
&gt;<br>
&gt; - There is no correlation between an XMPP IM session and a SIP voice<b=
r>
&gt; =A0 or video session, thus preventing a dual-stack endpoint<br>
&gt; from providing<br>
&gt; =A0 integrated user interfaces and communications history<br>
&gt;<br>
&gt; - SIP accounts and XMPP accounts are not directly correlated<br>
&gt; in contact<br>
&gt; =A0 lists or vCards (and not all deployed services support<br>
&gt; storage of such<br>
&gt; =A0 information), thus preventing a dual-stack endpoint from<br>
&gt; knowing that<br>
&gt; =A0 a contact has both SIP and XMPP capabilities<br>
&gt;<br>
&gt; Although some proprietary solutions exist to the foregoing<br>
&gt; problems, it<br>
&gt; would be preferable to define standardized solutions for the sake of<b=
r>
&gt; improved interoperability.<br>
&gt;<br>
&gt; Objectives<br>
&gt;<br>
&gt; Because both SIP and XMPP are easily extended through new SIP headers<=
br>
&gt; and XMPP elements, it should be possible to provide tighter<br>
&gt; integration<br>
&gt; within dual-stack SIP/XMPP user agents to improve the user experience.=
<br>
&gt;<br>
&gt; Any such extensions should meet the following criteria:<br>
&gt;<br>
&gt; - Be completely optional and backwards-compatible for all endpoints<br=
>
&gt;<br>
&gt; - Work without changes to deployed infrastructure such as existing<br>
&gt; =A0 SIP and XMPP servers, B2BUAs, firewalls, etc.<br>
&gt;<br>
&gt; The SIXPAC WG will define a small number of SIP and XMPP extensions to=
<br>
&gt; solve the following use cases in dual-stack endpoints:<br>
&gt;<br>
&gt; - Including SIP-based availability states in XMPP presence (limited to=
<br>
&gt; =A0 basic presence and availability states only, not the full range of=
<br>
&gt; =A0 PIDF extensions)<br>
&gt;<br>
&gt; - Correlating an XMPP IM session with a SIP voice/video session, and<b=
r>
&gt; =A0 vice-versa<br>
&gt;<br>
&gt; - Advertising a SIP account address over XMPP and an XMPP account<br>
&gt; =A0 address over SIP<br>
&gt;<br>
&gt; Additional use cases are out of scope, including anything<br>
&gt; related to or<br>
&gt; requiring server integration, multiparty communication, SIP-based IM<b=
r>
&gt; and presence, XMPP-based voice and video, file transfer, generalized<b=
r>
&gt; service discovery and capabilities exchange, full protocol translation=
<br>
&gt; in communication gateways, shared credentials across both SIP and XMPP=
<br>
&gt; accounts, rich presence extensions for features such as geolocation,<b=
r>
&gt; etc. =A0Although such topics are important and interesting, they are o=
ut<br>
&gt; of scope for this group.<br>
&gt;<br>
&gt; However, in addition to the protocol extensions explicitly mentioned<b=
r>
&gt; above, the group may also define best practices related to the<br>
&gt; implementation and deployment of dual-stack SIP/XMPP endpoints,<br>
&gt; including topics such as user agent configuration.<br>
&gt;<br>
&gt; Deliverables<br>
&gt;<br>
&gt; - Use cases and protocol requirements<br>
&gt;<br>
&gt; - XMPP presence extension for SIP-based availability states<br>
&gt;<br>
&gt; - Media session correlation extensions for SIP and XMPP<br>
&gt;<br>
&gt; - Contact address advertisement extensions for SIP and XMPP<br>
&gt;<br>
&gt; - Best practices for implementation and deployment of dual-stack<br>
&gt; =A0 endpoints<br>
&gt;<br>
&gt; Milestones<br>
&gt;<br>
&gt; Feb 2011 =A0Submit use cases and protocol requirements document to<br>
&gt; =A0 =A0 =A0 =A0 =A0 IESG (Informational)<br>
&gt;<br>
&gt; Oct 2011 =A0Submit XMPP presence extension document to IESG<br>
&gt; =A0 =A0 =A0 =A0 =A0 (Standards Track)<br>
&gt;<br>
&gt; Oct 2011 =A0Submit media session correlation extensions document to IE=
SG<br>
&gt; =A0 =A0 =A0 =A0 =A0 (Standards Track)<br>
&gt;<br>
&gt; Oct 2011 =A0Submit contact address advertisement extension document to=
<br>
&gt; =A0 =A0 =A0 =A0 =A0 IESG (Standards Track)<br>
&gt;<br>
&gt; Oct 2011 =A0Submit best practices document to IESG (Informational)<br>
&gt;<br>
&gt; ###<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div></div>_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div><br></div>

--00163630f4bbdde96504930a02a3--

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Wed, 20 Oct 2010 10:16:05 -0400
Thread-Topic: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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________________________________________
From: Hadriel Kaplan [HKaplan@acmepacket.com]

I should point out 3GPP specs think they can use GRUUs, when they really ca=
n't.
________________________________________

That doesn't surprise me.  But is that a problem with GRUU or with 3GPP?  I=
n a world where people consider it normal for a network to have *no* global=
ly-accessible ingress points, it's not surprising that they have trouble im=
plementing internet protocols.

________________________________________
From: Hadriel Kaplan [HKaplan@acmepacket.com]

I don't think I'm "blaming gruu"
________________________________________

You have titled the I-D, "Problems with the SIP Globally Routable User Agen=
t URIs (GRUUs)".

Dale

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Wed, 20 Oct 2010 10:19:16 -0400
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________________________________________
From: Hadriel Kaplan [HKaplan@acmepacket.com]

I just don't think it's clear to folks what the ramifications are of mintin=
g a GRUU.
________________________________________

Strictly speaking, what you mean is "I don't think it's clear to folks what=
 the requirements are on what constitutes a GRUU."  If something constructs=
 a URI and it's not globally-routable, it's not a GRUU, regardless of wheth=
er it is claimed to be one.

Dale

From HKaplan@acmepacket.com  Wed Oct 20 08:28:37 2010
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To: "Worley, Dale R (Dale)" <dworley@avaya.com>
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On Oct 20, 2010, at 10:16 AM, Worley, Dale R (Dale) wrote:
> ________________________________________
> From: Hadriel Kaplan [HKaplan@acmepacket.com]
>=20
> I don't think I'm "blaming gruu"
> ________________________________________
>=20
> You have titled the I-D, "Problems with the SIP Globally Routable User Ag=
ent URIs (GRUUs)".

So?  If I titled it "Problems with SDP", would it imply I'm blaming SDP for=
 the problems?

-hadriel=

From HKaplan@acmepacket.com  Wed Oct 20 08:34:12 2010
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To: "Worley, Dale R (Dale)" <dworley@avaya.com>
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On Oct 20, 2010, at 10:19 AM, Worley, Dale R (Dale) wrote:

> From: Hadriel Kaplan [HKaplan@acmepacket.com]
>=20
> I just don't think it's clear to folks what the ramifications are of mint=
ing a GRUU.
> ________________________________________
>=20
> Strictly speaking, what you mean is "I don't think it's clear to folks wh=
at the requirements are on what constitutes a GRUU."  If something construc=
ts a URI and it's not globally-routable, it's not a GRUU, regardless of whe=
ther it is claimed to be one.

Yup.=20

Note, though, that global reachability is only one of the "problems" discus=
sed in the draft (although a big section).

-hadriel=

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Thread-Topic: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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References: <A444A0F8084434499206E78C106220CA0232CD96F4@MCHP058A.global-ad.net><4CBDA041.5080804@cisco.com>, <044360F3-B0E3-46AC-8C94-532428AFDD41@acmepacket.com><CD5674C3CD99574EBA7432465FC13C1B2202288954@DC-US1MBEX4.global.avaya.com>, <0B2570E8-F3AE-4A8A-A460-12FA4711CF92@acmepacket.com><CD5674C3CD99574EBA7432465FC13C1B220228895B@DC-US1MBEX4.global.avaya.com> <271ED498-1D0C-4DC7-916B-E58B3FF78A9F@acmepacket.com>
From: "Mike Hammer (hmmr)" <hmmr@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Worley, Dale R (Dale)" <dworley@avaya.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Comment on draft-kaplan-dispatch-gruu-problematic-00
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Maybe you should title it:  "Best Common Practices for using GRUU
Correctly"

Then you could describe various problems:  X hurts, don't do that.  Or,
be sure you do this instead.

Mike
 =20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Hadriel Kaplan
Sent: Wednesday, October 20, 2010 11:30 AM
To: Worley, Dale R (Dale)
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Comment on
draft-kaplan-dispatch-gruu-problematic-00


On Oct 20, 2010, at 10:16 AM, Worley, Dale R (Dale) wrote:
> ________________________________________
> From: Hadriel Kaplan [HKaplan@acmepacket.com]
>=20
> I don't think I'm "blaming gruu"
> ________________________________________
>=20
> You have titled the I-D, "Problems with the SIP Globally Routable User
Agent URIs (GRUUs)".

So?  If I titled it "Problems with SDP", would it imply I'm blaming SDP
for the problems?

-hadriel
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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________________________________________
From: Hadriel Kaplan [HKaplan@acmepacket.com]

So?  If I titled it "Problems with SDP", would it imply I'm blaming SDP for=
 the problems?
________________________________________

It sounds like that to me -- you're saying the problem is with the design o=
f SDP.

If you said, "Problems using SDP", that would have a very different meaning=
.

Consider "Problems with Hadriel's work" vs. "Problems applying Hadriel's wo=
rk".  Which would you rather have your boss say?

And the abstract doesn't make it clear that the core of the problem is syst=
ems attempting to create GRUUs that are not, in fact, globally routable.  A=
 better description is "There are many SIP architectures in which GRUUs can=
not be constructed.  Thus, alternative means need to be found to solve the =
problems for which GRUUs are commonly considered to be the solution."

Dale

From emcho@sip-communicator.org  Thu Oct 21 09:52:27 2010
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Subject: Re: [dispatch] proposed SIXPAC charter
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Hey folks,

=D0=9D=D0=B0 20.10.10 09:45, Elwell, John =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=
=B0:
> I find this a worthwhile topic to pursue. I had been wondering
> whether this activity would turn out to be more of a profiling
> exercise, and whether the IETF might not be the best choice of venue
> for such work. From the current draft charter it looks like there
> will be at least some protocol extension work, for which I believe
> the IETF is the correct venue. On the other hand, the Unified
> Communications Interoperability Forum (UCIF) is seeking to advance
> the state of play on XMPP interoperability, and if we were just
> talking about a profile or BCP, that might have been a better venue.

As you mention yourself, it is likely that the charter would involve
extension work that could lead to, for example, new SIP headers. These
are undoubtedly best handled here.

Cheers,
Emil



> Perhaps the IETF should focus on requirements and protocol
> extensions, and consider whether the BCP work would be better done
> elsewhere. Or at least, there should be some coordination with other
> activities relating to XMPP interoperability.
>=20
> John
>=20
>> -----Original Message----- From: dispatch-bounces@ietf.org=20
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre=20
>> Sent: 19 October 2010 21:17 To: dispatch@ietf.org Subject:
>> [dispatch] proposed SIXPAC charter
>>=20
>> Earlier this year, some folks in the RAI area proposed an=20
>> initiative to define a few small extensions to both SIP and XMPP
>> that would make it easier to develop and deploy dual-stack SIP+XMPP
>> endpoints. Based on feedback provided on the DISPATCH list and
>> received from the RAI ADs, I've taken the liberty of rewriting the
>> proposed charter, in the hopes that fresh text will spur a more
>> conclusive discussion. I'm mostly just trying to help the
>> proponents put their best foot forward, so if folks here have more
>> feedback I'd expect that people like Simo Veikkolainen and Emil
>> Ivov will be able to engage in further discussion.
>>=20
>> /psa
>>=20
>> ###
>>=20
>> SIXPAC (SIP Integration with XMPP in Presence Aware Clients)=20
>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=20
>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>=20
>> Problem Statement
>>=20
>> Both the Session Initiation Protocol (SIP) and the Extensible=20
>> Messaging and Presence Protocol (XMPP) are widely deployed
>> technologies for real-time communication over the Internet.  In
>> order to offer a complete suite of features as well as
>> communication across multiple networks, several user-oriented
>> software applications support both SIP and XMPP, and more software
>> developers have expressed interest in building such "dual-stack"
>> solutions.  Unfortunately, it is difficult to provide a good
>> end-user experience in such applications because SIP and XMPP are=20
>> not aware of each other.  For example:
>>=20
>> - XMPP presence does not include availability states related to a
>> SIP voice call or video call (e.g., "on the phone"), thus
>> preventing an a dual-stack endpoint from showing presence-based=20
>> communication hints
>>=20
>> - There is no correlation between an XMPP IM session and a SIP
>> voice or video session, thus preventing a dual-stack endpoint from
>> providing integrated user interfaces and communications history
>>=20
>> - SIP accounts and XMPP accounts are not directly correlated in
>> contact lists or vCards (and not all deployed services support=20
>> storage of such information), thus preventing a dual-stack endpoint
>> from knowing that a contact has both SIP and XMPP capabilities
>>=20
>> Although some proprietary solutions exist to the foregoing=20
>> problems, it would be preferable to define standardized solutions
>> for the sake of improved interoperability.
>>=20
>> Objectives
>>=20
>> Because both SIP and XMPP are easily extended through new SIP
>> headers and XMPP elements, it should be possible to provide tighter
>>  integration within dual-stack SIP/XMPP user agents to improve the
>> user experience.
>>=20
>> Any such extensions should meet the following criteria:
>>=20
>> - Be completely optional and backwards-compatible for all
>> endpoints
>>=20
>> - Work without changes to deployed infrastructure such as existing=20
>> SIP and XMPP servers, B2BUAs, firewalls, etc.
>>=20
>> The SIXPAC WG will define a small number of SIP and XMPP extensions
>> to solve the following use cases in dual-stack endpoints:
>>=20
>> - Including SIP-based availability states in XMPP presence (limited
>> to basic presence and availability states only, not the full range
>> of PIDF extensions)
>>=20
>> - Correlating an XMPP IM session with a SIP voice/video session,
>> and vice-versa
>>=20
>> - Advertising a SIP account address over XMPP and an XMPP account=20
>> address over SIP
>>=20
>> Additional use cases are out of scope, including anything related
>> to or requiring server integration, multiparty communication,
>> SIP-based IM and presence, XMPP-based voice and video, file
>> transfer, generalized service discovery and capabilities exchange,
>> full protocol translation in communication gateways, shared
>> credentials across both SIP and XMPP accounts, rich presence
>> extensions for features such as geolocation, etc.  Although such
>> topics are important and interesting, they are out of scope for
>> this group.
>>=20
>> However, in addition to the protocol extensions explicitly
>> mentioned above, the group may also define best practices related
>> to the implementation and deployment of dual-stack SIP/XMPP
>> endpoints, including topics such as user agent configuration.
>>=20
>> Deliverables
>>=20
>> - Use cases and protocol requirements
>>=20
>> - XMPP presence extension for SIP-based availability states
>>=20
>> - Media session correlation extensions for SIP and XMPP
>>=20
>> - Contact address advertisement extensions for SIP and XMPP
>>=20
>> - Best practices for implementation and deployment of dual-stack=20
>> endpoints
>>=20
>> Milestones
>>=20
>> Feb 2011  Submit use cases and protocol requirements document to=20
>> IESG (Informational)
>>=20
>> Oct 2011  Submit XMPP presence extension document to IESG=20
>> (Standards Track)
>>=20
>> Oct 2011  Submit media session correlation extensions document to
>> IESG (Standards Track)
>>=20
>> Oct 2011  Submit contact address advertisement extension document
>> to IESG (Standards Track)
>>=20
>> Oct 2011  Submit best practices document to IESG (Informational)
>>=20
>> ###
>>=20
>>=20
>>=20
> _______________________________________________ dispatch mailing
> list dispatch@ietf.org=20
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

--=20
Emil Ivov, Ph.D.                               67000 Strasbourg,
Project Lead                                   France
SIP Communicator
emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
http://sip-communicator.org                    FAX:   +33.1.77.62.47.31


From fluffy@cisco.com  Sat Oct 23 06:54:06 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Christer, can you send me a bit more info about this - perhaps a pointer =
to the 3GPP spec. Do you know if anyone is following up with the =
registration of this?

Thanks, Cullen


On Aug 23, 2010, at 6:53 PM, Christer Holmberg wrote:

>=20
> Hi,
>=20
> A while ago, 3GPP also defined a DTMF package. For some reason it =
hasn't been registered with IANA yet, but I have requested to ensure =
that it happens.
>=20
> Regards,
>=20
> Christer


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From: Cullen Jennings <fluffy@cisco.com>
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References: <430FC6BDED356B4C8498F634416644A92694381B86@mail><1D062974A4845E4D8A343C6538049202036C3B5F@XMB-BGL-414.cisco.com><430FC6BDED356B4C8498F634416644A92694381C2C@mail><1D062974A4845E4D8A343C6538049202036C3C98@XMB-BGL-414.cisco.com><430FC6BDED356B4C8498F634416644A92694381E9F@mail><4C729FEF.6020509@cisco.com><430FC6BDED356B4C8498F634416644A92694381F16@mail><4C72C9CA.3070304@cisco.com><430FC6BDED356B4C8498F634416644A92694381FDF@mail><4C72EFCC.2000107@cisco.com><430FC6BDED356B4C8498F634416644A92694382179@mail> <4C73EA66.4000305@cisco.com> <7F2072F1E0DE894DA4B517B93C6A05851B8637@ESESSCMS0356.eemea.ericsson.se> <4C7413D9.80804@cisco.com> <A11921905DA1564D9BCF64A6430A62390293A4A8@XMB-BGL-411.cisco.com> <7F2072F1E0DE894DA4B517B93C6A05851B8650@ESESSCMS0356.eemea.ericsson.se> <A11921905DA1564D9BCF64A6430A62390293A4A9@XMB-BGL-411.cisco.com> <7F2072F1E0DE894DA4B517B93C6A05851B8652@ESESSCMS0356.eemea.ericsson.se> <1D062974A4845E4D8A343C6538049202036C3F6D@XMB-BGL-414.cisco.com> <7F2 072F1E0DE894DA4B517B93C6A05851B8654@ESESSCMS0356.eemea.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
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Uh, same as Paul, sorry, ignore my previous email. Don't know how I =
missed this on first search. Thank you for sending the pointer to this.=20=


What is the sate of this 3GPP doc and what is going on with the =
registration?

Thanks, Cullen


On Aug 24, 2010, at 9:57 PM, Christer Holmberg wrote:

>=20
> Hi,
>=20
> The package definition can be found in Annex P of 3GPP TS 24.229:
>=20
> http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-a00.zip
>=20
> The MIME body used by the package to transport the digits is defined =
in Annex G of 3GPP TS 29.163 (it re-uses a body used for overlap =
dialling, in case the wording confuses):=20
>=20
> http://www.3gpp.org/ftp/Specs/archive/29_series/29.163/29163-920.zip
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>> -----Original Message-----
>> From: Muthu ArulMozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20=

>> Sent: 25. elokuuta 2010 6:47
>> To: Christer Holmberg; Parthasarathi R (partr); Paul Kyzivat=20
>> (pkyzivat)
>> Cc: dispatch@ietf.org; Hadriel Kaplan
>> Subject: RE:=20
>> [dispatch]I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
>>=20
>> Can you please provide the pointer to the 3GPP spec=20
>> describing this DTMF INFO package?
>>=20
>> thanks,
>> Muthu
>>=20
>> |-----Original Message-----
>> |From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> |Sent: Wednesday, August 25, 2010 9:05 AM
>> |To: Parthasarathi R (partr); Paul Kyzivat (pkyzivat)
>> |Cc: dispatch@ietf.org; Hadriel Kaplan; Muthu ArulMozhi Perumal
>> (mperumal)
>> |Subject: RE:
>> [dispatch]I-DAction:draft-kaplan-dispatch-info-dtmf-package-
>> |00.txt
>> |
>> |
>> |Hi,
>> |
>> |>1) whether 3GPP DTMF Info package is in the state to be accepted as
>> IETF
>> |standards.
>> |
>> |The Info Package registration procedure requires an expert review,
>> which
>> |purpose is to ensure that the Info Package does not break the SIP
>> protocol
>> |etc. I assume such review will take place when the package is
>> registered
>> |with IANA.
>> |
>> |Regards,
>> |
>> |Christer
>> |
>> |
>> |
>> |
>> |________________________________
>> |
>> |	From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> |	Sent: Wed 8/25/2010 8:45 AM
>> |	To: Parthasarathi R (partr); Paul Kyzivat (pkyzivat)
>> |	Cc: dispatch@ietf.org; Hadriel Kaplan; Muthu ArulMozhi Perumal
>> |(mperumal)
>> |	Subject: RE:
>> [dispatch]I-DAction:draft-kaplan-dispatch-info-dtmf-
>> |package-00.txt
>> |
>> |
>> |
>> |
>> |	Hi,
>> |
>> |	>In case IMS needs new INFO based DTMF relay mechanism, I agree
>> that
>> |it is the valid requirement and we need to evaluate it. Before
>> concluding
>> |about new
>> |	>INFO based mechanism in IMS, Please throw the light for not
>> selecting
>> |KPML in IMS. I'm comparing KPML vs INFO based mechanism as both are
>> based on
>> |SIP
>> |	>signaling based mechanism. KPML has limited deployment as
>> mentioned
>> |in Hadriel Info draft but new INFO package is yet to be=20
>> defined. It is=20
>> |better to have
>> |	>the single active IETF SIP signaling based dtmf-relay=20
>> mechanism=20
>> |rather than producing more standards wherein implementers will
>> completely
>> |confused which
>> |	>one to select.
>> |	>
>> |	>As Muthu mentioned, I also heard about KPML deployment issues
>> but the
>> |legacy INFO based dtmf-relay has lot more caveats. I guess that Info
>> package
>> |is
>> |	>designed as generic as KPML, and then both KPML and INFO may
>> look
>> |heavyweight.
>> |
>> |	The 3GPP DTMF Info Package is very simple.
>> |
>> |	Regards,
>> |
>> |	Christer
>> |
>> |
>> |
>> |
>> |	________________________________
>> |
>> |	        From: dispatch-bounces@ietf.org on behalf of Paul
>> Kyzivat
>> |(pkyzivat)
>> |	        Sent: Wed 8/25/2010 12:17 AM
>> |	        To: Christer Holmberg
>> |	        Cc: dispatch@ietf.org; Hadriel Kaplan
>> |	        Subject: Re: [dispatch]I-D
>> Action:draft-kaplan-dispatch-info-
>> |dtmf-package-00.txt
>> |
>> |
>> |
>> |
>> |
>> |	        Christer Holmberg wrote:
>> |	        > Hi,
>> |	        >
>> |	        > I think it is strange to reject something based on a
>> claim
>> |there already exists a number of legacy ways of there. Not=20
>> everyone has=20
>> |implemented the same (if any) of them, and the whole purpose of the
>> Info
>> |Package framework is to come up with a mechanism for which the usage
>> can be
>> |negotiated.
>> |	        >
>> |	        > And, as I said earlier, at least in IMS an Info
>> Package is
>> |being used for DTMF transport. It is currently defined in 3GPP=20
>> |specifications, but there shouldn't be anything IMS specific=20
>> about the=20
>> |package itself.
>> |
>> |	        I wasn't asserting a general principle.
>> |	        I was talking about this particular mechanism.
>> |
>> |	        If there is yet another dtmf package, then I guess we
>> can
>> |consider the
>> |	        pros/cons of it. Those are probably best assessed by
>> those who
>> |use the
>> |	        existing one and/or might use whatever info package you
>> might
>> |propose to
>> |	        replace it.
>> |
>> |	        We clearly suffer from a surplus of ways to convey dtmf.
>> |	        While its not impossible to argue than another=20
>> one would=20
>> |improve the
>> |	        situation, it is going to be a hard sell.
>> |
>> |	        OTOH, the more there are the more case it makes for
>> having an
>> |SBC in the
>> |	        path to "fix" the interoperability. We sell them, so
>> maybe I
>> |should be
>> |	        cheering on every new one as a way to boost sales.
>> |
>> |	                Thanks,
>> |	                Paul
>> |
>> |	        > Regards,
>> |	        >
>> |	        > Christer
>> |	        >
>> |	        >
>> |	        >
>> |	        >> -----Original Message-----
>> |	        >> From: dispatch-bounces@ietf.org
>> |	        >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul=20=

>> |Kyzivat
>> |	        >> Sent: 24. elokuuta 2010 18:51
>> |	        >> To: Hadriel Kaplan
>> |	        >> Cc: dispatch@ietf.org
>> |	        >> Subject: Re: [dispatch] I-D
>> |	        >> Action:draft-kaplan-dispatch-info-dtmf-package-00.txt
>> |	        >>
>> |	        >> Maybe we should ask our ADs if they have an opinion
>> about
>> |this.
>> |	        >>
>> |	        >>      Thanks,
>> |	        >>      Paul
>> |	        >>
>> |	        >> Hadriel Kaplan wrote:
>> |	        >>>> -----Original Message-----
>> |	        >>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> |	        >>>> Sent: Monday, August 23, 2010 6:02 PM
>> |	        >>>> To: Hadriel Kaplan
>> |	        >>>>
>> |	        >>>> That reduces things somewhat. But if=20
>> everybody that=20
>> |supports the
>> |	        >>>> package also supports the legacy approach, what is
>> the
>> |win?
>> |	        >>> The legacy mode has no published standards document
>> |	        >> defining it.  I thought when the info-packages work
>> was
>> |	        >> started, DTMF-in-info was one of the main drivers.
>> But I
>> |	        >> take your point - if people would prefer to just
>> document
>> |it
>> |	        >> as it is today (sans info-packages), I can change the
>> draft
>> |	        >> to just be that.  I'm cool with either way (defining
>> the
>> |	        >> current dtmf-relay as a legacy mode was Option-2).
>> |	        >>> -hadriel
>> |	        >>>
>> |	        >> _______________________________________________
>> |	        >> dispatch mailing list
>> |	        >> dispatch@ietf.org
>> |	        >> https://www.ietf.org/mailman/listinfo/dispatch
>> |	        >>
>> |	        _______________________________________________
>> |	        dispatch mailing list
>> |	        dispatch@ietf.org
>> |	        https://www.ietf.org/mailman/listinfo/dispatch
>> |
>> |
>> |
>>=20
>>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From magnus.westerlund@ericsson.com  Sun Oct 24 03:53:16 2010
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: "Allyn Romanow (allyn)" <allyn@cisco.com>
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Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
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Hi Allyn,

That way of getting it into the charter is fine with me.

Magnus

Allyn Romanow (allyn) skrev 2010-10-15 01:23:
> Hi Magnus,
> You have made some very helpful points. In particular you've pointed out that the charter talks about the receiver's ability to respond, but not the sender's. Also the charter is not explicit about necessary discussion of Telepresence interoperability issues that are not within the charter of the multi-stream WG to do anything about.   
> 
> I'd like to propose some wording in the charter that I think will address your concerns. See what you think- some of the wording is yours :)..
> 
> Instead of
> 
> 	The WG will create specifications for SIP-based conferencing systems to enable 	communication of enough information about each media stream so that each receiving system 	or bridge system can make reasonable decisions about selecting and rendering media 	streams. This enables systems to make display choices that optimize the "just like being 	there" experience.
> I propose
> 	The WG will create specifications for SIP-based conferencing systems to enable 	communication of enough information about each media stream so that each sending system, 	receiving system, or bridge system can make reasonable decisions about transmitting, 	selecting, and rendering media streams. This enables systems to make choices that 	optimize the "just like being there" experience.
> 
> I would not like to change:
> 	* Which sources a receiver wants to receive.  For example, it might want the source for 	the left camera, or might want the source chosen by VAD (Voice Activity Detection).
> To the proposed:
> 	* Which sources a receiver wants transmitted.  For example, it might want the source for 	the left camera, or might want the source chosen by VAD (Voice Activity Detection).
> 
> Because there are multiple receivers. A receiver cannot say what should be transmitted, it can only say what itself wants to get, or receive. But I think your concern about sender behavior is covered by the changes to the paragraph above.
> 
> I also propose the following addition:
> 	This working group is not currently chartered to work on issues of continuous conference 	control including: far end camera control, indication of fast frame update for video 	codecs or other rapid switches, floor control, conference roster. The working group may 	identify interoperability obstacles in existing openstandards. If so, the WG will develp 	requirements to be communicated to other IETF WGs or Standards Forums as a kind request 	to try to meet these requirements.
> 
> What do you think?
> 
> Best regards,
> Allyn
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Magnus Westerlund
> Sent: Wednesday, October 13, 2010 2:11 AM
> To: Peter Musgrave
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
> 
> Hi Peter,
> 
> Good that we are in agreement on a number of things. Response and
> clarifications inline.
> 
> Peter Musgrave skrev 2010-10-12 17:37:
>>>
>>> Independent if one uses a central mixer or a full mesh there appears to
>>> be a lack in media stream negotiation for how to negotiate what
>>> capabilities the receiving client has in receiving, decoding and
>>> rendering when it comes to number of simultaneous streams. Our
>>> understanding is that RTP mandates any number of streams, while in
>>> reality outside of conferencing that is in fact one single stream.
>>> However with telepresence and conferencing there is a need to negotiate
>>> the actual capability, especially between a mixer and client when it
>>> comes to deliver multiple streams for various usages.
> 
>> Agreed. The charter does speak to identifying endpoint capabilities
>> and I view what you describe here as the kind of meta-information
>> which needs to fit into that framework.
> 
> Okay, as long as this is included in the thinking about capabilities I
> am fine. However, the charter proposal isn't particular detailed in this
> regards. But I guess the following charter text is matching this:
> 
> "Information between sources and sinks about media stream capabilities
> will be exchanged."
> 
>>>
>>> We also think there is missing functionality for expressing who are the
>>> currently active speakers. The RTP CSRC field are not sufficient as they
>>> express which end-points are included in the mix. However, as one likely
>>> include in a mix more rooms than the active speakers to ensure that any
>>> interruption and ambient sound are correctly captured an mechanism for
>>> indicating the currently active speakers are needed. There is also need
>>> to agree on the type of identifier to be used that allow for mapping to
>>> full user names when appropriate.
> 
>> I agree with this. In my opinion this is a separable problem since it
>> applies to generic audio conferencing as well. There is tangential
>> work in the conference event package and probably in the XCON stuff -
>> but I lack detailed background in XCON.
>>
>> In the context of the TP charter I think the work on indicating which
>> audio stream is associated with which speaker (or part of the room) is
>> a slightly different issue - since it's about which stream and not
>> which speaker within a stream.
> 
> I should probably have been clearer. I am concerned that even on media
> stream level there is missing functionality to indicate which streams
> that contains active speakers and has been included in a mix produced by
> the mixer. From our side we see a difference between being included in
> the mix and being tagged as currently an active speaker.
> 
> I think this is currently missing functionality for which requirements
> needs to be agreed on and then a solution developed. However, the
> solution might better fit another WG, like AVT. But I think that is for
> later determination. I think the requirements phase makes sense to have
> in the Telepresence group.
> 
> I do agree that such functionality do have a wider usage scope than just
> telepresence and can be used in any case of centralized mixers where
> special activity needs to be pointed out, with audio being the foremost
> example of such media.
> 
>>
>>>
>>> The charter includes an item to work on control between client and a
>>> mixer what streams should be delivered. We would like to extended this
>>> to allow also the mixer to control the client on when streams are
>>> delivered. In cases where there are more users then available display
>>> area and a particular user is not actually included in any outgoing
>>> mix/selection from the mixer then there is little point in that the
>>> client consumes network resources. Thus we suggest that the stream
>>> selection and control work is extended to allow also the mixer to pause
>>> individual streams from the client to the mixer. Please note that such
>>> control needs to be responsive, thus using the SIP signaling path for
>>> this is to slow, it needs to be on the media path.
>> I concur.
>>
>> The charter sentence:
>>  "The WG will create specifications for SIP-based conferencing systems
>> to enable communication of enough information about each media stream
>> so that each receiving system or bridge system can make reasonable
>> decisions about selecting and rendering media streams. This enables
>> systems to make display choices that optimize the "just like being
>> there" experience. "
>>
>> is specific to selection of streams and not to the ability to control
>> them. I don't see anything here which prevents a source from not
>> sending selected streams - but I have no objection to making this more
>> explicit. How about adding:
>>
>> "Selection choices will be communicated to the sending party to allow
>> the sender to optionally present information on which streams are in
>> use and to optionally suspend streams which are not in use." ??
>>
> 
> When I wrote the above comment I was mostly considering the following
> chart text bullet:
> 
> * Which sources a receiver wants to receive.  For example, it might want
> the source for the left camera, or might want the source chosen by VAD
> (Voice Activity Detection).
> 
> I think your proposed text combined with the above bullet gives all the
> room I see needed for media stream control. However, maybe a
> modification of the above to make it clear.
> 
> * Which sources a receiver or mixer wants to have transmitted.  For
> example, it might want the source for the left camera, or might want the
> source chosen by VAD (Voice Activity Detection).
> 
>>
>>>
>>> Another question to the group is, are there agreement on how streams
>>> should organized into RTP sessions and how they are utilized? If
>>> anything is documented I would appreciate a pointer. The media streams
>>> are for different purposes from different end-points which are then
>>> mixed by a central point and delivered to end-points. My impression for
>>> example is that most proprietary system breaks the RTP semantics and use
>>> individual sessions between the mixer and each end-point, instead of a
>>> joint session. Here also different robustification tools for RTP such as
>>> FEC and Retransmission needs to be considered. This might be work that
>>> belongs in AVT, but are there need for further clarification into this
>>> field?
> 
>> I think this is something which will have to be thought through and
>> standardized if TP interoperability is to succeed. I would view the
>> semantics of how the streams are assigned in scope and the "plumbing"
>> either comes from the AVT/MMUSIC media cap stuff or any additions are
>> developed in those WGs.
>>
> 
> I agree with that the right place to do specification work is likely
> these two groups. However, I think telepresence needs to provide
> requirements for such work. Making such requirement development explicit
> in the charter would make it clear that we do expect work to happen
> somewhere else. At the same time it is after all possible to request
> that another WG's charter is updated to cover such work.
> 
> The WG may identify interoperability obstacles in existing open
> standards. The WG may develop requirements to be communicated to other
> IETF WGs or Standards Forums as a kind request to try to meet these
> requirements.
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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Hi,

Sorry for the delay in responding.

When it comes to controlling the mix, that is clearly a dynamic control.
A user must be able to basically click or select a particular user as
the one it currently finds most important, and then at some point later
do another selection or go back to letting the mixer do the selection
based on activity.

I don't know the background to the charter paragraph:
This working group is not currently chartered to work on issues of
continuous conference control including: far end camera control,
indication of fast frame update for video codecs or other rapid
switches, floor control, conference roster.

But, I guess what I believe is needed will fall under the part of
continuous conference control. I don't see the SIP signalling path work
as this is something that has the potential for being updated often
during a session's duration, and also should be quickly responsive.

If we can't change the charter to perform work on stream selection in
mixers, then can the Telepresence group at least work on requirements
that can be thrown over the wall to some other WG? I also think we need
to have some discussion where this would be appropriate, or if even a
new WG is needed for that topic.

I know that we within Ericsson has been discussing something that is an
extension to BFCP.

Cheers

Magnus

Peter Musgrave skrev 2010-10-14 16:54:
> Hi Mike,
> 
> I agree the onhld/offhold analogy is a poor one. It is a suspension of
> a call - which is not intended.
> 
> Let me re-phrase and see if we agree:
> 
> A sends B three video streams and one audio stream .
> B has one display and opts to display one of the three streams.
> There is a mechanism for B to tell A that this has happened (and to
> say when it wishes to display some other stream from A).
> 
> Do we think that the work to do this type of stream-suspension is
> floor control? (My opinion is no. I view floor control more as who is
> the source for a given stream and not a selection of an agreed upon
> stream).
> 
> Do we think this work is in scope for Telepresence Interoperability? I
> think this is an essential element to deployable TP operation
> especially as in expands to "at home" use. If an architecture in which
> a room sends all video all the time (even though it's not being used)
> is viewed as necessary to avoid complexity/meet responsiveness
> requirements is thought to be ok then I'd like to make sure we are
> clear that that these limitations exist upfront.
> 
> My preference is obviously to permit this feedback and tweak the
> proposed charter.
> 
> Other opinions?
> 
> Regards,
> 
> 
> Peter Musgrave
> 
> On Thu, Oct 14, 2010 at 9:46 AM, Mike Hammer (hmmr) <hmmr@cisco.com> wrote:
>> Peter,
>>
>> I have not issue with your proposed words, since they are essentially asking for a "mute" button for all types of communications used by a sender.  This may complicate the algorithm for all the receive systems to know what to display, if for example, you continue to speak, but turn off the video transmission.  A "mute all" option might be more polite.
>>
>> I did have issue with this comment:  "more like onhold/offhold type negotiation"
>>
>> I am not sure I would agree with this.
>>
>> There needs to be a distinction between when a call starts and when a call stops, the call being *all* communications (sessions) between two points.  Granted putting a call on hold seems like floor control, but the participants of the call on hold are effectively out of the call (left the room) until brought off-hold.  This is SIP level action.
>>
>> There needs to be a clear distinction between negotiating capabilities, aka knowing what capabilities an endpoint has, and using those capabilities during a call.  That learning process is done via SDP.
>>
>> The latter part seems to be floor control, continuous conference control, or whatever you want to call it.  There seems to be splitting of hairs over whether the floor control is manual or automatic (determined by the loudest speaker).  This may be AVT.
>>
>> In summary, you have room access, you define capabilities, you have "speaker" conflict resolution control.  (Note, point-to-point calls are devolution since resolution of conflict is a non-problem with only one opposing party.)
>>
>> Did I miss some other nuance?
>>
>> Mike
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Musgrave
>> Sent: Thursday, October 14, 2010 5:25 AM
>> To: Allyn Romanow (allyn)
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>>
>> HI Allyn/Magnus,
>>
>> W.R.T
>>
>> "[AR} I agree with your point that it would be good to pause the
>> sender, but I think that falls
>>  under the category of "continuous conference control" which is out of
>> the scope of this charter.
>>  Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As
>> control issues
>> necessary for interoperability arise, they will be directed appropriately.
>> What do you think?"
>>
>> I think these words leave room to *not* do what Magnus is suggesting.
>> e.g. If I have a 1 screen home TP system getting three streams from a
>> 3 camera system and selecting only one it is not really "necessary" to
>> tell the 3 camera system to pause two streams - but it is a really
>> good idea to have a way to that! I don't view this is a continuous
>> conference control issue (since that gets into floor control which I
>> agree is out of scope) but more like onhold/offhold type negotiation.
>> I think this is to some extent a plumbing issue (and based on the
>> desired response time SIP or media path messages will be required).
>>
>> So I would still like to see a statement which makes it clear that it
>> not just an issue of receiver selection but there is a need to
>> communicate that selection back to the sender. The current text does
>> not say this. Hence my suggestion:
>>
>> "Selection choices will be communicated to the sending party to allow
>> the sender to optionally present information on which streams are in
>> use and to optionally suspend streams which are not in use." ??
>>
>> I'm open to other words or some indication of which language in the
>> current charter ensures this is in scope.
>>
>> Regards,
>>
>> Peter Musgrave
>>
>>
>>
>>
>> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) <allyn@cisco.com> wrote:
>>> Hi Magnus,
>>> I was really glad that you had a chance to carefully consider the proposed work.
>>> Like Peter, I too agree with most of your comments.
>>> Specific points below.
>>>
>>> Best regards,
>>> Allyn
>>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>> Sent: Tuesday, October 12, 2010 6:47 AM
>>> To: dispatch@ietf.org
>>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>>
>>> Hi,
>>>
>>> Some colleagues and I have reviewed the two telepresence drafts and
>>> version 5 of the charter and have some comments on the charter.
>>>
>>> To make our comments more understandable I think we should explain our
>>> view on telepresence. First we are expecting great variations in the
>>> capabilities of the participating terminals in a session. From personal
>>> terminals for individual users up to multi-display, cameras and
>>> microphone setups. We also expect that more than two end-points being
>>> present in the same session. As we see it it will require good support
>>> for centralized media handling in a mixer (MCU). We also see video
>>> streams not only of high quality to be presented for maximum immersion,
>>> but also lower qualities that can be used to monitor activity in other
>>> rooms that aren't the ones selected for high quality.
>>>
>>> [AR] Yes, absolutely.
>>>
>>> Independent if one uses a central mixer or a full mesh there appears to
>>> be a lack in media stream negotiation for how to negotiate what
>>> capabilities the receiving client has in receiving, decoding and
>>> rendering when it comes to number of simultaneous streams. Our
>>> understanding is that RTP mandates any number of streams, while in
>>> reality outside of conferencing that is in fact one single stream.
>>> However with telepresence and conferencing there is a need to negotiate
>>> the actual capability, especially between a mixer and client when it
>>> comes to deliver multiple streams for various usages.
>>>
>>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>>
>>> We also think there is missing functionality for expressing who are the
>>> currently active speakers. The RTP CSRC field are not sufficient as they
>>> express which end-points are included in the mix. However, as one likely
>>> include in a mix more rooms than the active speakers to ensure that any
>>> interruption and ambient sound are correctly captured an mechanism for
>>> indicating the currently active speakers are needed. There is also need
>>> to agree on the type of identifier to be used that allow for mapping to
>>> full user names when appropriate.
>>>
>>> [AR] Yes.. Both of these points are covered in the charter too, right?
>>>
>>> The charter includes an item to work on control between client and a
>>> mixer what streams should be delivered. We would like to extended this
>>> to allow also the mixer to control the client on when streams are
>>> delivered. In cases where there are more users then available display
>>> area and a particular user is not actually included in any outgoing
>>> mix/selection from the mixer then there is little point in that the
>>> client consumes network resources. Thus we suggest that the stream
>>> selection and control work is extended to allow also the mixer to pause
>>> individual streams from the client to the mixer. Please note that such
>>> control needs to be responsive, thus using the SIP signaling path for
>>> this is to slow, it needs to be on the media path.
>>>
>>> [AR} I agree with your point that it would be good to pause the sender, but I think that falls
>>>  under the category of "continuous conference control" which is out of the scope of this charter.
>>>  Such a feedback mechanism falls probably under AVT.
>>> I think it would be good to add a sentence saying something like, As control issues
>>> necessary for interoperability arise, they will be directed appropriately.
>>> What do you think?
>>>
>>>
>>> Another question to the group is, are there agreement on how streams
>>> should organized into RTP sessions and how they are utilized? If
>>> anything is documented I would appreciate a pointer. The media streams
>>> are for different purposes from different end-points which are then
>>> mixed by a central point and delivered to end-points. My impression for
>>> example is that most proprietary system breaks the RTP semantics and use
>>> individual sessions between the mixer and each end-point, instead of a
>>> joint session. Here also different robustification tools for RTP such as
>>> FEC and Retransmission needs to be considered. This might be work that
>>> belongs in AVT, but are there need for further clarification into this
>>> field?
>>>
>>> [AR} Agree this needs to be worked out for Telepresence, and that any activity about it would go to AVT and/or MMUSIC.
>>>
>>> Comments much appreciated and we can gladly suggest text additions to
>>> the charter proposal.
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> ----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> ----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> Färögatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> ----------------------------------------------------------------------
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
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>>
> _______________________________________________
> dispatch mailing list
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> https://www.ietf.org/mailman/listinfo/dispatch
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

From peter.musgrave@magorcorp.com  Sun Oct 24 04:30:15 2010
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi Magnus/Mike:

Allyn had previously proposed:
" also propose the following addition:
       This working group is not currently chartered to work on issues of
continuous conference        control including: far end camera control,
indication of fast frame update for video    codecs or other rapid switches=
,
floor control, conference roster. The working group may identify
interoperability obstacles in existing open standards. If so, the WG will
develp        requirements to be communicated to other IETF WGs or Standard=
s
Forums as a kind request to try to meet these requirements.
"

Here the "..will define requirements..." applies to interoperability
obstacles.  I would regard a single panel system having a way of determinin=
g
dynamically which video stream it gets from a multi-video source as an
interoperability issue. Hence I would view this as "in scope".

I agree that on-demand media source selection will be a critical
requirement. Personally, I view it as quite different from floor control so
I'm not sure I'd start with BFCP - but I think that discussion makes more
sense when the requirements have been well defined.

Regards,

Peter Musgrave

On Sun, Oct 24, 2010 at 7:03 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> Sorry for the delay in responding.
>
> When it comes to controlling the mix, that is clearly a dynamic control.
> A user must be able to basically click or select a particular user as
> the one it currently finds most important, and then at some point later
> do another selection or go back to letting the mixer do the selection
> based on activity.
>
> I don't know the background to the charter paragraph:
> This working group is not currently chartered to work on issues of
> continuous conference control including: far end camera control,
> indication of fast frame update for video codecs or other rapid
> switches, floor control, conference roster.
>
> But, I guess what I believe is needed will fall under the part of
> continuous conference control. I don't see the SIP signalling path work
> as this is something that has the potential for being updated often
> during a session's duration, and also should be quickly responsive.
>
> If we can't change the charter to perform work on stream selection in
> mixers, then can the Telepresence group at least work on requirements
> that can be thrown over the wall to some other WG? I also think we need
> to have some discussion where this would be appropriate, or if even a
> new WG is needed for that topic.
>
> I know that we within Ericsson has been discussing something that is an
> extension to BFCP.
>
> Cheers
>
> Magnus
>
> Peter Musgrave skrev 2010-10-14 16:54:
> > Hi Mike,
> >
> > I agree the onhld/offhold analogy is a poor one. It is a suspension of
> > a call - which is not intended.
> >
> > Let me re-phrase and see if we agree:
> >
> > A sends B three video streams and one audio stream .
> > B has one display and opts to display one of the three streams.
> > There is a mechanism for B to tell A that this has happened (and to
> > say when it wishes to display some other stream from A).
> >
> > Do we think that the work to do this type of stream-suspension is
> > floor control? (My opinion is no. I view floor control more as who is
> > the source for a given stream and not a selection of an agreed upon
> > stream).
> >
> > Do we think this work is in scope for Telepresence Interoperability? I
> > think this is an essential element to deployable TP operation
> > especially as in expands to "at home" use. If an architecture in which
> > a room sends all video all the time (even though it's not being used)
> > is viewed as necessary to avoid complexity/meet responsiveness
> > requirements is thought to be ok then I'd like to make sure we are
> > clear that that these limitations exist upfront.
> >
> > My preference is obviously to permit this feedback and tweak the
> > proposed charter.
> >
> > Other opinions?
> >
> > Regards,
> >
> >
> > Peter Musgrave
> >
> > On Thu, Oct 14, 2010 at 9:46 AM, Mike Hammer (hmmr) <hmmr@cisco.com>
> wrote:
> >> Peter,
> >>
> >> I have not issue with your proposed words, since they are essentially
> asking for a "mute" button for all types of communications used by a send=
er.
>  This may complicate the algorithm for all the receive systems to know wh=
at
> to display, if for example, you continue to speak, but turn off the video
> transmission.  A "mute all" option might be more polite.
> >>
> >> I did have issue with this comment:  "more like onhold/offhold type
> negotiation"
> >>
> >> I am not sure I would agree with this.
> >>
> >> There needs to be a distinction between when a call starts and when a
> call stops, the call being *all* communications (sessions) between two
> points.  Granted putting a call on hold seems like floor control, but the
> participants of the call on hold are effectively out of the call (left th=
e
> room) until brought off-hold.  This is SIP level action.
> >>
> >> There needs to be a clear distinction between negotiating capabilities=
,
> aka knowing what capabilities an endpoint has, and using those capabiliti=
es
> during a call.  That learning process is done via SDP.
> >>
> >> The latter part seems to be floor control, continuous conference
> control, or whatever you want to call it.  There seems to be splitting of
> hairs over whether the floor control is manual or automatic (determined b=
y
> the loudest speaker).  This may be AVT.
> >>
> >> In summary, you have room access, you define capabilities, you have
> "speaker" conflict resolution control.  (Note, point-to-point calls are
> devolution since resolution of conflict is a non-problem with only one
> opposing party.)
> >>
> >> Did I miss some other nuance?
> >>
> >> Mike
> >>
> >>
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Peter Musgrave
> >> Sent: Thursday, October 14, 2010 5:25 AM
> >> To: Allyn Romanow (allyn)
> >> Cc: dispatch@ietf.org
> >> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
> >>
> >> HI Allyn/Magnus,
> >>
> >> W.R.T
> >>
> >> "[AR} I agree with your point that it would be good to pause the
> >> sender, but I think that falls
> >>  under the category of "continuous conference control" which is out of
> >> the scope of this charter.
> >>  Such a feedback mechanism falls probably under AVT.
> >> I think it would be good to add a sentence saying something like, As
> >> control issues
> >> necessary for interoperability arise, they will be directed
> appropriately.
> >> What do you think?"
> >>
> >> I think these words leave room to *not* do what Magnus is suggesting.
> >> e.g. If I have a 1 screen home TP system getting three streams from a
> >> 3 camera system and selecting only one it is not really "necessary" to
> >> tell the 3 camera system to pause two streams - but it is a really
> >> good idea to have a way to that! I don't view this is a continuous
> >> conference control issue (since that gets into floor control which I
> >> agree is out of scope) but more like onhold/offhold type negotiation.
> >> I think this is to some extent a plumbing issue (and based on the
> >> desired response time SIP or media path messages will be required).
> >>
> >> So I would still like to see a statement which makes it clear that it
> >> not just an issue of receiver selection but there is a need to
> >> communicate that selection back to the sender. The current text does
> >> not say this. Hence my suggestion:
> >>
> >> "Selection choices will be communicated to the sending party to allow
> >> the sender to optionally present information on which streams are in
> >> use and to optionally suspend streams which are not in use." ??
> >>
> >> I'm open to other words or some indication of which language in the
> >> current charter ensures this is in scope.
> >>
> >> Regards,
> >>
> >> Peter Musgrave
> >>
> >>
> >>
> >>
> >> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) <
> allyn@cisco.com> wrote:
> >>> Hi Magnus,
> >>> I was really glad that you had a chance to carefully consider the
> proposed work.
> >>> Like Peter, I too agree with most of your comments.
> >>> Specific points below.
> >>>
> >>> Best regards,
> >>> Allyn
> >>>
> >>> -----Original Message-----
> >>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Magnus Westerlund
> >>> Sent: Tuesday, October 12, 2010 6:47 AM
> >>> To: dispatch@ietf.org
> >>> Subject: [dispatch] Comments on the proposed Telepresence Charter
> >>>
> >>> Hi,
> >>>
> >>> Some colleagues and I have reviewed the two telepresence drafts and
> >>> version 5 of the charter and have some comments on the charter.
> >>>
> >>> To make our comments more understandable I think we should explain ou=
r
> >>> view on telepresence. First we are expecting great variations in the
> >>> capabilities of the participating terminals in a session. From person=
al
> >>> terminals for individual users up to multi-display, cameras and
> >>> microphone setups. We also expect that more than two end-points being
> >>> present in the same session. As we see it it will require good suppor=
t
> >>> for centralized media handling in a mixer (MCU). We also see video
> >>> streams not only of high quality to be presented for maximum immersio=
n,
> >>> but also lower qualities that can be used to monitor activity in othe=
r
> >>> rooms that aren't the ones selected for high quality.
> >>>
> >>> [AR] Yes, absolutely.
> >>>
> >>> Independent if one uses a central mixer or a full mesh there appears =
to
> >>> be a lack in media stream negotiation for how to negotiate what
> >>> capabilities the receiving client has in receiving, decoding and
> >>> rendering when it comes to number of simultaneous streams. Our
> >>> understanding is that RTP mandates any number of streams, while in
> >>> reality outside of conferencing that is in fact one single stream.
> >>> However with telepresence and conferencing there is a need to negotia=
te
> >>> the actual capability, especially between a mixer and client when it
> >>> comes to deliver multiple streams for various usages.
> >>>
> >>> [AR] Yes, agree - as you saw, this is covered in the charter.
> >>>
> >>> We also think there is missing functionality for expressing who are t=
he
> >>> currently active speakers. The RTP CSRC field are not sufficient as
> they
> >>> express which end-points are included in the mix. However, as one
> likely
> >>> include in a mix more rooms than the active speakers to ensure that a=
ny
> >>> interruption and ambient sound are correctly captured an mechanism fo=
r
> >>> indicating the currently active speakers are needed. There is also ne=
ed
> >>> to agree on the type of identifier to be used that allow for mapping =
to
> >>> full user names when appropriate.
> >>>
> >>> [AR] Yes.. Both of these points are covered in the charter too, right=
?
> >>>
> >>> The charter includes an item to work on control between client and a
> >>> mixer what streams should be delivered. We would like to extended thi=
s
> >>> to allow also the mixer to control the client on when streams are
> >>> delivered. In cases where there are more users then available display
> >>> area and a particular user is not actually included in any outgoing
> >>> mix/selection from the mixer then there is little point in that the
> >>> client consumes network resources. Thus we suggest that the stream
> >>> selection and control work is extended to allow also the mixer to pau=
se
> >>> individual streams from the client to the mixer. Please note that suc=
h
> >>> control needs to be responsive, thus using the SIP signaling path for
> >>> this is to slow, it needs to be on the media path.
> >>>
> >>> [AR} I agree with your point that it would be good to pause the sende=
r,
> but I think that falls
> >>>  under the category of "continuous conference control" which is out o=
f
> the scope of this charter.
> >>>  Such a feedback mechanism falls probably under AVT.
> >>> I think it would be good to add a sentence saying something like, As
> control issues
> >>> necessary for interoperability arise, they will be directed
> appropriately.
> >>> What do you think?
> >>>
> >>>
> >>> Another question to the group is, are there agreement on how streams
> >>> should organized into RTP sessions and how they are utilized? If
> >>> anything is documented I would appreciate a pointer. The media stream=
s
> >>> are for different purposes from different end-points which are then
> >>> mixed by a central point and delivered to end-points. My impression f=
or
> >>> example is that most proprietary system breaks the RTP semantics and
> use
> >>> individual sessions between the mixer and each end-point, instead of =
a
> >>> joint session. Here also different robustification tools for RTP such
> as
> >>> FEC and Retransmission needs to be considered. This might be work tha=
t
> >>> belongs in AVT, but are there need for further clarification into thi=
s
> >>> field?
> >>>
> >>> [AR} Agree this needs to be worked out for Telepresence, and that any
> activity about it would go to AVT and/or MMUSIC.
> >>>
> >>> Comments much appreciated and we can gladly suggest text additions to
> >>> the charter proposal.
> >>>
> >>> Cheers
> >>>
> >>> Magnus Westerlund
> >>>
> >>> ---------------------------------------------------------------------=
-
> >>> Multimedia Technologies, Ericsson Research EAB/TVM
> >>> ---------------------------------------------------------------------=
-
> >>> Ericsson AB                | Phone  +46 10 7148287
> >>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> >>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> >>> ---------------------------------------------------------------------=
-
> >>> _______________________________________________
> >>> dispatch mailing list
> >>> dispatch@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dispatch
> >>> _______________________________________________
> >>> dispatch mailing list
> >>> dispatch@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dispatch
> >>>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>

--0016363b83f84c21f904935b38fb
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Magnus/Mike:<div><br></div><div>Allyn had previously proposed:</div><div=
>&quot;<span class=3D"Apple-style-span" style=3D"font-family: arial, sans-s=
erif; font-size: 11px; border-collapse: collapse; ">=A0also propose the fol=
lowing addition:<br>
=A0 =A0 =A0 =A0This working group is not currently chartered to work on iss=
ues of continuous conference =A0 =A0 =A0 =A0control including: far end came=
ra control, indication of fast frame update for video =A0 =A0codecs or othe=
r rapid switches, floor control, conference roster. The working group may i=
dentify interoperability obstacles in existing open standards. If so, the W=
G will develp =A0 =A0 =A0 =A0requirements to be communicated to other IETF =
WGs or Standards Forums as a kind request to try to meet these requirements=
.<br>
</span>&quot;</div><div><br></div><div>Here the &quot;..will define require=
ments...&quot; applies to interoperability obstacles. =A0I would regard a s=
ingle panel system having a way of determining dynamically which video stre=
am it gets from a multi-video source as an interoperability issue. Hence I =
would view this as &quot;in scope&quot;.=A0</div>
<div><br></div><div>I agree that on-demand media source selection will be a=
 critical requirement. Personally, I view it as quite different from floor =
control so I&#39;m not sure I&#39;d start with BFCP - but I think that disc=
ussion makes more sense when the requirements have been well defined.=A0</d=
iv>
<div><br></div><div>Regards,=A0</div><div><br></div><div>Peter Musgrave</di=
v><div><br><div class=3D"gmail_quote">On Sun, Oct 24, 2010 at 7:03 AM, Magn=
us Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@eri=
csson.com">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">Hi,<br>
<br>
Sorry for the delay in responding.<br>
<br>
When it comes to controlling the mix, that is clearly a dynamic control.<br=
>
A user must be able to basically click or select a particular user as<br>
the one it currently finds most important, and then at some point later<br>
do another selection or go back to letting the mixer do the selection<br>
based on activity.<br>
<br>
I don&#39;t know the background to the charter paragraph:<br>
<div class=3D"im">This working group is not currently chartered to work on =
issues of<br>
continuous conference control including: far end camera control,<br>
indication of fast frame update for video codecs or other rapid<br>
switches, floor control, conference roster.<br>
<br>
</div>But, I guess what I believe is needed will fall under the part of<br>
continuous conference control. I don&#39;t see the SIP signalling path work=
<br>
as this is something that has the potential for being updated often<br>
during a session&#39;s duration, and also should be quickly responsive.<br>
<br>
If we can&#39;t change the charter to perform work on stream selection in<b=
r>
mixers, then can the Telepresence group at least work on requirements<br>
that can be thrown over the wall to some other WG? I also think we need<br>
to have some discussion where this would be appropriate, or if even a<br>
new WG is needed for that topic.<br>
<br>
I know that we within Ericsson has been discussing something that is an<br>
extension to BFCP.<br>
<br>
Cheers<br>
<br>
Magnus<br>
<br>
Peter Musgrave skrev 2010-10-14 16:54:<br>
<div><div></div><div class=3D"h5">&gt; Hi Mike,<br>
&gt;<br>
&gt; I agree the onhld/offhold analogy is a poor one. It is a suspension of=
<br>
&gt; a call - which is not intended.<br>
&gt;<br>
&gt; Let me re-phrase and see if we agree:<br>
&gt;<br>
&gt; A sends B three video streams and one audio stream .<br>
&gt; B has one display and opts to display one of the three streams.<br>
&gt; There is a mechanism for B to tell A that this has happened (and to<br=
>
&gt; say when it wishes to display some other stream from A).<br>
&gt;<br>
&gt; Do we think that the work to do this type of stream-suspension is<br>
&gt; floor control? (My opinion is no. I view floor control more as who is<=
br>
&gt; the source for a given stream and not a selection of an agreed upon<br=
>
&gt; stream).<br>
&gt;<br>
&gt; Do we think this work is in scope for Telepresence Interoperability? I=
<br>
&gt; think this is an essential element to deployable TP operation<br>
&gt; especially as in expands to &quot;at home&quot; use. If an architectur=
e in which<br>
&gt; a room sends all video all the time (even though it&#39;s not being us=
ed)<br>
&gt; is viewed as necessary to avoid complexity/meet responsiveness<br>
&gt; requirements is thought to be ok then I&#39;d like to make sure we are=
<br>
&gt; clear that that these limitations exist upfront.<br>
&gt;<br>
&gt; My preference is obviously to permit this feedback and tweak the<br>
&gt; proposed charter.<br>
&gt;<br>
&gt; Other opinions?<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt;<br>
&gt; Peter Musgrave<br>
&gt;<br>
&gt; On Thu, Oct 14, 2010 at 9:46 AM, Mike Hammer (hmmr) &lt;<a href=3D"mai=
lto:hmmr@cisco.com">hmmr@cisco.com</a>&gt; wrote:<br>
&gt;&gt; Peter,<br>
&gt;&gt;<br>
&gt;&gt; I have not issue with your proposed words, since they are essentia=
lly asking for a &quot;mute&quot; button for all types of communications us=
ed by a sender. =A0This may complicate the algorithm for all the receive sy=
stems to know what to display, if for example, you continue to speak, but t=
urn off the video transmission. =A0A &quot;mute all&quot; option might be m=
ore polite.<br>

&gt;&gt;<br>
&gt;&gt; I did have issue with this comment: =A0&quot;more like onhold/offh=
old type negotiation&quot;<br>
&gt;&gt;<br>
&gt;&gt; I am not sure I would agree with this.<br>
&gt;&gt;<br>
&gt;&gt; There needs to be a distinction between when a call starts and whe=
n a call stops, the call being *all* communications (sessions) between two =
points. =A0Granted putting a call on hold seems like floor control, but the=
 participants of the call on hold are effectively out of the call (left the=
 room) until brought off-hold. =A0This is SIP level action.<br>

&gt;&gt;<br>
&gt;&gt; There needs to be a clear distinction between negotiating capabili=
ties, aka knowing what capabilities an endpoint has, and using those capabi=
lities during a call. =A0That learning process is done via SDP.<br>
&gt;&gt;<br>
&gt;&gt; The latter part seems to be floor control, continuous conference c=
ontrol, or whatever you want to call it. =A0There seems to be splitting of =
hairs over whether the floor control is manual or automatic (determined by =
the loudest speaker). =A0This may be AVT.<br>

&gt;&gt;<br>
&gt;&gt; In summary, you have room access, you define capabilities, you hav=
e &quot;speaker&quot; conflict resolution control. =A0(Note, point-to-point=
 calls are devolution since resolution of conflict is a non-problem with on=
ly one opposing party.)<br>

&gt;&gt;<br>
&gt;&gt; Did I miss some other nuance?<br>
&gt;&gt;<br>
&gt;&gt; Mike<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: <a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounce=
s@ietf.org</a> [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org">dispatc=
h-bounces@ietf.org</a>] On Behalf Of Peter Musgrave<br>
&gt;&gt; Sent: Thursday, October 14, 2010 5:25 AM<br>
&gt;&gt; To: Allyn Romanow (allyn)<br>
&gt;&gt; Cc: <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt;&gt; Subject: Re: [dispatch] Comments on the proposed Telepresence Char=
ter<br>
&gt;&gt;<br>
&gt;&gt; HI Allyn/Magnus,<br>
&gt;&gt;<br>
&gt;&gt; W.R.T<br>
&gt;&gt;<br>
&gt;&gt; &quot;[AR} I agree with your point that it would be good to pause =
the<br>
&gt;&gt; sender, but I think that falls<br>
&gt;&gt; =A0under the category of &quot;continuous conference control&quot;=
 which is out of<br>
&gt;&gt; the scope of this charter.<br>
&gt;&gt; =A0Such a feedback mechanism falls probably under AVT.<br>
&gt;&gt; I think it would be good to add a sentence saying something like, =
As<br>
&gt;&gt; control issues<br>
&gt;&gt; necessary for interoperability arise, they will be directed approp=
riately.<br>
&gt;&gt; What do you think?&quot;<br>
&gt;&gt;<br>
&gt;&gt; I think these words leave room to *not* do what Magnus is suggesti=
ng.<br>
&gt;&gt; e.g. If I have a 1 screen home TP system getting three streams fro=
m a<br>
&gt;&gt; 3 camera system and selecting only one it is not really &quot;nece=
ssary&quot; to<br>
&gt;&gt; tell the 3 camera system to pause two streams - but it is a really=
<br>
&gt;&gt; good idea to have a way to that! I don&#39;t view this is a contin=
uous<br>
&gt;&gt; conference control issue (since that gets into floor control which=
 I<br>
&gt;&gt; agree is out of scope) but more like onhold/offhold type negotiati=
on.<br>
&gt;&gt; I think this is to some extent a plumbing issue (and based on the<=
br>
&gt;&gt; desired response time SIP or media path messages will be required)=
.<br>
&gt;&gt;<br>
&gt;&gt; So I would still like to see a statement which makes it clear that=
 it<br>
&gt;&gt; not just an issue of receiver selection but there is a need to<br>
&gt;&gt; communicate that selection back to the sender. The current text do=
es<br>
&gt;&gt; not say this. Hence my suggestion:<br>
&gt;&gt;<br>
&gt;&gt; &quot;Selection choices will be communicated to the sending party =
to allow<br>
&gt;&gt; the sender to optionally present information on which streams are =
in<br>
&gt;&gt; use and to optionally suspend streams which are not in use.&quot; =
??<br>
&gt;&gt;<br>
&gt;&gt; I&#39;m open to other words or some indication of which language i=
n the<br>
&gt;&gt; current charter ensures this is in scope.<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;<br>
&gt;&gt; Peter Musgrave<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) &lt;<a hre=
f=3D"mailto:allyn@cisco.com">allyn@cisco.com</a>&gt; wrote:<br>
&gt;&gt;&gt; Hi Magnus,<br>
&gt;&gt;&gt; I was really glad that you had a chance to carefully consider =
the proposed work.<br>
&gt;&gt;&gt; Like Peter, I too agree with most of your comments.<br>
&gt;&gt;&gt; Specific points below.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Best regards,<br>
&gt;&gt;&gt; Allyn<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -----Original Message-----<br>
&gt;&gt;&gt; From: <a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bo=
unces@ietf.org</a> [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org">dis=
patch-bounces@ietf.org</a>] On Behalf Of Magnus Westerlund<br>
&gt;&gt;&gt; Sent: Tuesday, October 12, 2010 6:47 AM<br>
&gt;&gt;&gt; To: <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a>=
<br>
&gt;&gt;&gt; Subject: [dispatch] Comments on the proposed Telepresence Char=
ter<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Some colleagues and I have reviewed the two telepresence draft=
s and<br>
&gt;&gt;&gt; version 5 of the charter and have some comments on the charter=
.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; To make our comments more understandable I think we should exp=
lain our<br>
&gt;&gt;&gt; view on telepresence. First we are expecting great variations =
in the<br>
&gt;&gt;&gt; capabilities of the participating terminals in a session. From=
 personal<br>
&gt;&gt;&gt; terminals for individual users up to multi-display, cameras an=
d<br>
&gt;&gt;&gt; microphone setups. We also expect that more than two end-point=
s being<br>
&gt;&gt;&gt; present in the same session. As we see it it will require good=
 support<br>
&gt;&gt;&gt; for centralized media handling in a mixer (MCU). We also see v=
ideo<br>
&gt;&gt;&gt; streams not only of high quality to be presented for maximum i=
mmersion,<br>
&gt;&gt;&gt; but also lower qualities that can be used to monitor activity =
in other<br>
&gt;&gt;&gt; rooms that aren&#39;t the ones selected for high quality.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; [AR] Yes, absolutely.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Independent if one uses a central mixer or a full mesh there a=
ppears to<br>
&gt;&gt;&gt; be a lack in media stream negotiation for how to negotiate wha=
t<br>
&gt;&gt;&gt; capabilities the receiving client has in receiving, decoding a=
nd<br>
&gt;&gt;&gt; rendering when it comes to number of simultaneous streams. Our=
<br>
&gt;&gt;&gt; understanding is that RTP mandates any number of streams, whil=
e in<br>
&gt;&gt;&gt; reality outside of conferencing that is in fact one single str=
eam.<br>
&gt;&gt;&gt; However with telepresence and conferencing there is a need to =
negotiate<br>
&gt;&gt;&gt; the actual capability, especially between a mixer and client w=
hen it<br>
&gt;&gt;&gt; comes to deliver multiple streams for various usages.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; [AR] Yes, agree - as you saw, this is covered in the charter.<=
br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; We also think there is missing functionality for expressing wh=
o are the<br>
&gt;&gt;&gt; currently active speakers. The RTP CSRC field are not sufficie=
nt as they<br>
&gt;&gt;&gt; express which end-points are included in the mix. However, as =
one likely<br>
&gt;&gt;&gt; include in a mix more rooms than the active speakers to ensure=
 that any<br>
&gt;&gt;&gt; interruption and ambient sound are correctly captured an mecha=
nism for<br>
&gt;&gt;&gt; indicating the currently active speakers are needed. There is =
also need<br>
&gt;&gt;&gt; to agree on the type of identifier to be used that allow for m=
apping to<br>
&gt;&gt;&gt; full user names when appropriate.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; [AR] Yes.. Both of these points are covered in the charter too=
, right?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The charter includes an item to work on control between client=
 and a<br>
&gt;&gt;&gt; mixer what streams should be delivered. We would like to exten=
ded this<br>
&gt;&gt;&gt; to allow also the mixer to control the client on when streams =
are<br>
&gt;&gt;&gt; delivered. In cases where there are more users then available =
display<br>
&gt;&gt;&gt; area and a particular user is not actually included in any out=
going<br>
&gt;&gt;&gt; mix/selection from the mixer then there is little point in tha=
t the<br>
&gt;&gt;&gt; client consumes network resources. Thus we suggest that the st=
ream<br>
&gt;&gt;&gt; selection and control work is extended to allow also the mixer=
 to pause<br>
&gt;&gt;&gt; individual streams from the client to the mixer. Please note t=
hat such<br>
&gt;&gt;&gt; control needs to be responsive, thus using the SIP signaling p=
ath for<br>
&gt;&gt;&gt; this is to slow, it needs to be on the media path.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; [AR} I agree with your point that it would be good to pause th=
e sender, but I think that falls<br>
&gt;&gt;&gt; =A0under the category of &quot;continuous conference control&q=
uot; which is out of the scope of this charter.<br>
&gt;&gt;&gt; =A0Such a feedback mechanism falls probably under AVT.<br>
&gt;&gt;&gt; I think it would be good to add a sentence saying something li=
ke, As control issues<br>
&gt;&gt;&gt; necessary for interoperability arise, they will be directed ap=
propriately.<br>
&gt;&gt;&gt; What do you think?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Another question to the group is, are there agreement on how s=
treams<br>
&gt;&gt;&gt; should organized into RTP sessions and how they are utilized? =
If<br>
&gt;&gt;&gt; anything is documented I would appreciate a pointer. The media=
 streams<br>
&gt;&gt;&gt; are for different purposes from different end-points which are=
 then<br>
&gt;&gt;&gt; mixed by a central point and delivered to end-points. My impre=
ssion for<br>
&gt;&gt;&gt; example is that most proprietary system breaks the RTP semanti=
cs and use<br>
&gt;&gt;&gt; individual sessions between the mixer and each end-point, inst=
ead of a<br>
&gt;&gt;&gt; joint session. Here also different robustification tools for R=
TP such as<br>
&gt;&gt;&gt; FEC and Retransmission needs to be considered. This might be w=
ork that<br>
&gt;&gt;&gt; belongs in AVT, but are there need for further clarification i=
nto this<br>
&gt;&gt;&gt; field?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; [AR} Agree this needs to be worked out for Telepresence, and t=
hat any activity about it would go to AVT and/or MMUSIC.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Comments much appreciated and we can gladly suggest text addit=
ions to<br>
&gt;&gt;&gt; the charter proposal.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Cheers<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Magnus Westerlund<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; --------------------------------------------------------------=
--------<br>
&gt;&gt;&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt;&gt;&gt; --------------------------------------------------------------=
--------<br>
&gt;&gt;&gt; Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7=
148287<br>
&gt;&gt;&gt; F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73=
 0949079<br>
&gt;&gt;&gt; SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.=
westerlund@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt;&gt;&gt; --------------------------------------------------------------=
--------<br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; dispatch mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; dispatch mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" tar=
get=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
&gt;&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; dispatch mailing list<br>
&gt;&gt; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
&gt;&gt;<br>
&gt; _______________________________________________<br>
&gt; dispatch mailing list<br>
&gt; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_=
blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
&gt;<br>
<br>
<br>
</div></div>--<br>
<div><div></div><div class=3D"h5"><br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287<br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079<br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
</div></div></blockquote></div><br></div>

--0016363b83f84c21f904935b38fb--

From allyn@cisco.com  Sun Oct 24 17:22:26 2010
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From: "Allyn Romanow (allyn)" <allyn@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, "Peter Musgrave" <peter.musgrave@magorcorp.com>
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Hi Magnus -=20

I agree with you that the issues you bring up need to be dealt with, and =
I feel they should potentially result in requirements from this working =
group.

With respect in particular to your next to last paragraph below, did you =
not think that my proposed change to the charter (as Peter quoted), =
covered your concern?

"also propose the following addition:
       This working group is not currently chartered to work on issues =
of continuous conference        control including: far end camera =
control, indication of fast frame update for video codecs or other rapid =
switches, floor control, conference roster. The working group may =
identify interoperability obstacles in existing open standards. If so, =
the WG will develop        requirements to be communicated to other IETF =
WGs or Standards Forums as a kind request to try to meet these =
requirements."

Best regards,
Allyn

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Magnus Westerlund
Sent: Sunday, October 24, 2010 4:04 AM
To: Peter Musgrave
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Comments on the proposed Telepresence Charter

Hi,

Sorry for the delay in responding.

When it comes to controlling the mix, that is clearly a dynamic control.
A user must be able to basically click or select a particular user as
the one it currently finds most important, and then at some point later
do another selection or go back to letting the mixer do the selection
based on activity.

I don't know the background to the charter paragraph:
This working group is not currently chartered to work on issues of
continuous conference control including: far end camera control,
indication of fast frame update for video codecs or other rapid
switches, floor control, conference roster.

But, I guess what I believe is needed will fall under the part of
continuous conference control. I don't see the SIP signalling path work
as this is something that has the potential for being updated often
during a session's duration, and also should be quickly responsive.

If we can't change the charter to perform work on stream selection in
mixers, then can the Telepresence group at least work on requirements
that can be thrown over the wall to some other WG? I also think we need
to have some discussion where this would be appropriate, or if even a
new WG is needed for that topic.

I know that we within Ericsson has been discussing something that is an
extension to BFCP.

Cheers

Magnus

Peter Musgrave skrev 2010-10-14 16:54:
> Hi Mike,
>=20
> I agree the onhld/offhold analogy is a poor one. It is a suspension of
> a call - which is not intended.
>=20
> Let me re-phrase and see if we agree:
>=20
> A sends B three video streams and one audio stream .
> B has one display and opts to display one of the three streams.
> There is a mechanism for B to tell A that this has happened (and to
> say when it wishes to display some other stream from A).
>=20
> Do we think that the work to do this type of stream-suspension is
> floor control? (My opinion is no. I view floor control more as who is
> the source for a given stream and not a selection of an agreed upon
> stream).
>=20
> Do we think this work is in scope for Telepresence Interoperability? I
> think this is an essential element to deployable TP operation
> especially as in expands to "at home" use. If an architecture in which
> a room sends all video all the time (even though it's not being used)
> is viewed as necessary to avoid complexity/meet responsiveness
> requirements is thought to be ok then I'd like to make sure we are
> clear that that these limitations exist upfront.
>=20
> My preference is obviously to permit this feedback and tweak the
> proposed charter.
>=20
> Other opinions?
>=20
> Regards,
>=20
>=20
> Peter Musgrave
>=20
> On Thu, Oct 14, 2010 at 9:46 AM, Mike Hammer (hmmr) <hmmr@cisco.com> =
wrote:
>> Peter,
>>
>> I have not issue with your proposed words, since they are essentially =
asking for a "mute" button for all types of communications used by a =
sender.  This may complicate the algorithm for all the receive systems =
to know what to display, if for example, you continue to speak, but turn =
off the video transmission.  A "mute all" option might be more polite.
>>
>> I did have issue with this comment:  "more like onhold/offhold type =
negotiation"
>>
>> I am not sure I would agree with this.
>>
>> There needs to be a distinction between when a call starts and when a =
call stops, the call being *all* communications (sessions) between two =
points.  Granted putting a call on hold seems like floor control, but =
the participants of the call on hold are effectively out of the call =
(left the room) until brought off-hold.  This is SIP level action.
>>
>> There needs to be a clear distinction between negotiating =
capabilities, aka knowing what capabilities an endpoint has, and using =
those capabilities during a call.  That learning process is done via =
SDP.
>>
>> The latter part seems to be floor control, continuous conference =
control, or whatever you want to call it.  There seems to be splitting =
of hairs over whether the floor control is manual or automatic =
(determined by the loudest speaker).  This may be AVT.
>>
>> In summary, you have room access, you define capabilities, you have =
"speaker" conflict resolution control.  (Note, point-to-point calls are =
devolution since resolution of conflict is a non-problem with only one =
opposing party.)
>>
>> Did I miss some other nuance?
>>
>> Mike
>>
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of Peter Musgrave
>> Sent: Thursday, October 14, 2010 5:25 AM
>> To: Allyn Romanow (allyn)
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Comments on the proposed Telepresence Charter
>>
>> HI Allyn/Magnus,
>>
>> W.R.T
>>
>> "[AR} I agree with your point that it would be good to pause the
>> sender, but I think that falls
>>  under the category of "continuous conference control" which is out =
of
>> the scope of this charter.
>>  Such a feedback mechanism falls probably under AVT.
>> I think it would be good to add a sentence saying something like, As
>> control issues
>> necessary for interoperability arise, they will be directed =
appropriately.
>> What do you think?"
>>
>> I think these words leave room to *not* do what Magnus is suggesting.
>> e.g. If I have a 1 screen home TP system getting three streams from a
>> 3 camera system and selecting only one it is not really "necessary" =
to
>> tell the 3 camera system to pause two streams - but it is a really
>> good idea to have a way to that! I don't view this is a continuous
>> conference control issue (since that gets into floor control which I
>> agree is out of scope) but more like onhold/offhold type negotiation.
>> I think this is to some extent a plumbing issue (and based on the
>> desired response time SIP or media path messages will be required).
>>
>> So I would still like to see a statement which makes it clear that it
>> not just an issue of receiver selection but there is a need to
>> communicate that selection back to the sender. The current text does
>> not say this. Hence my suggestion:
>>
>> "Selection choices will be communicated to the sending party to allow
>> the sender to optionally present information on which streams are in
>> use and to optionally suspend streams which are not in use." ??
>>
>> I'm open to other words or some indication of which language in the
>> current charter ensures this is in scope.
>>
>> Regards,
>>
>> Peter Musgrave
>>
>>
>>
>>
>> On Wed, Oct 13, 2010 at 12:47 PM, Allyn Romanow (allyn) =
<allyn@cisco.com> wrote:
>>> Hi Magnus,
>>> I was really glad that you had a chance to carefully consider the =
proposed work.
>>> Like Peter, I too agree with most of your comments.
>>> Specific points below.
>>>
>>> Best regards,
>>> Allyn
>>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] =
On Behalf Of Magnus Westerlund
>>> Sent: Tuesday, October 12, 2010 6:47 AM
>>> To: dispatch@ietf.org
>>> Subject: [dispatch] Comments on the proposed Telepresence Charter
>>>
>>> Hi,
>>>
>>> Some colleagues and I have reviewed the two telepresence drafts and
>>> version 5 of the charter and have some comments on the charter.
>>>
>>> To make our comments more understandable I think we should explain =
our
>>> view on telepresence. First we are expecting great variations in the
>>> capabilities of the participating terminals in a session. From =
personal
>>> terminals for individual users up to multi-display, cameras and
>>> microphone setups. We also expect that more than two end-points =
being
>>> present in the same session. As we see it it will require good =
support
>>> for centralized media handling in a mixer (MCU). We also see video
>>> streams not only of high quality to be presented for maximum =
immersion,
>>> but also lower qualities that can be used to monitor activity in =
other
>>> rooms that aren't the ones selected for high quality.
>>>
>>> [AR] Yes, absolutely.
>>>
>>> Independent if one uses a central mixer or a full mesh there appears =
to
>>> be a lack in media stream negotiation for how to negotiate what
>>> capabilities the receiving client has in receiving, decoding and
>>> rendering when it comes to number of simultaneous streams. Our
>>> understanding is that RTP mandates any number of streams, while in
>>> reality outside of conferencing that is in fact one single stream.
>>> However with telepresence and conferencing there is a need to =
negotiate
>>> the actual capability, especially between a mixer and client when it
>>> comes to deliver multiple streams for various usages.
>>>
>>> [AR] Yes, agree - as you saw, this is covered in the charter.
>>>
>>> We also think there is missing functionality for expressing who are =
the
>>> currently active speakers. The RTP CSRC field are not sufficient as =
they
>>> express which end-points are included in the mix. However, as one =
likely
>>> include in a mix more rooms than the active speakers to ensure that =
any
>>> interruption and ambient sound are correctly captured an mechanism =
for
>>> indicating the currently active speakers are needed. There is also =
need
>>> to agree on the type of identifier to be used that allow for mapping =
to
>>> full user names when appropriate.
>>>
>>> [AR] Yes.. Both of these points are covered in the charter too, =
right?
>>>
>>> The charter includes an item to work on control between client and a
>>> mixer what streams should be delivered. We would like to extended =
this
>>> to allow also the mixer to control the client on when streams are
>>> delivered. In cases where there are more users then available =
display
>>> area and a particular user is not actually included in any outgoing
>>> mix/selection from the mixer then there is little point in that the
>>> client consumes network resources. Thus we suggest that the stream
>>> selection and control work is extended to allow also the mixer to =
pause
>>> individual streams from the client to the mixer. Please note that =
such
>>> control needs to be responsive, thus using the SIP signaling path =
for
>>> this is to slow, it needs to be on the media path.
>>>
>>> [AR} I agree with your point that it would be good to pause the =
sender, but I think that falls
>>>  under the category of "continuous conference control" which is out =
of the scope of this charter.
>>>  Such a feedback mechanism falls probably under AVT.
>>> I think it would be good to add a sentence saying something like, As =
control issues
>>> necessary for interoperability arise, they will be directed =
appropriately.
>>> What do you think?
>>>
>>>
>>> Another question to the group is, are there agreement on how streams
>>> should organized into RTP sessions and how they are utilized? If
>>> anything is documented I would appreciate a pointer. The media =
streams
>>> are for different purposes from different end-points which are then
>>> mixed by a central point and delivered to end-points. My impression =
for
>>> example is that most proprietary system breaks the RTP semantics and =
use
>>> individual sessions between the mixer and each end-point, instead of =
a
>>> joint session. Here also different robustification tools for RTP =
such as
>>> FEC and Retransmission needs to be considered. This might be work =
that
>>> belongs in AVT, but are there need for further clarification into =
this
>>> field?
>>>
>>> [AR} Agree this needs to be worked out for Telepresence, and that =
any activity about it would go to AVT and/or MMUSIC.
>>>
>>> Comments much appreciated and we can gladly suggest text additions =
to
>>> the charter proposal.
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> =
----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> =
----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> =
----------------------------------------------------------------------
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20


--=20

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------
_______________________________________________
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Thread-Topic: [dispatch] Telepresence charter version 5
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From: "Allyn Romanow (allyn)" <allyn@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "DISPATCH list" <dispatch@ietf.org>
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Subject: Re: [dispatch] Telepresence charter version 5
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Hi Christer,

In reviewing this, I think it would be fine to just omit the sentence at
issue-
	In addition,  there is no standardized way to exchange semantic
information about what each 	media stream represents. =20

Because I feel that the sentence just prior to it says adequately what
is necessary. It is
	A major factor in the inability of telepresence systems to
interwork is that there is no 	standardized way to describe and
negotiate the use of the multiple streams of audio and video 	that
comprise the media flows.


What do you think?

Best regards,
Allyn
-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Monday, September 20, 2010 10:19 PM
To: Allyn Romanow (allyn); Charles Eckel (eckelcu); Elwell, John;
DISPATCH list
Subject: RE: [dispatch] Telepresence charter version 5


Hi,=20

>Thanks for pointing out that the original sentence is not=20
>strictly accurate, and also not precisely what was meant. I=20
>think it is fine to change the wording along the lines you=20
>suggest, can I get back to you shortly on precise wording? =20

Sure, no problem.

Regards,

Christer

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Monday, September 20, 2010 10:07 PM
> To: Charles Eckel (eckelcu); Allyn Romanow (allyn); Elwell,=20
> John; DISPATCH list
> Subject: RE: [dispatch] Telepresence charter version 5
>=20
>=20
> Hi Charles,=20
>=20
> >I think RFC label attribute may be useful; however, we would=20
> >need to establish and register set of tokens and the=20
> >semantics of those tokens.
> >Once we get to the actual design, that would be worth considering.
> >Without addition specification, I think you would have=20
> >proprietary sets of labels being used at best.
>=20
> Yes, I agree that we would need to specify the labels.
>=20
> My point was that there does exist a mechanism for doing so. So, maybe
> the text could say something like:
>=20
> "In addition, standardization work, in order to be able to exchange
> semantic
> information about what each media stream represents, might be needed."
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> > > -----Original Message-----
> > > From: dispatch-bounces@ietf.org=20
> > [mailto:dispatch-bounces@ietf.org] On
> > Behalf Of Christer Holmberg
> > > Sent: Monday, September 20, 2010 2:18 AM
> > > To: Allyn Romanow (allyn); Elwell, John; DISPATCH list
> > > Subject: Re: [dispatch] Telepresence charter version 5
> > >=20
> > >=20
> > > Hi,
> > >=20
> > > One question for clarification. The text says:
> > >=20
> > > "In addition, there is no standardized way to exchange semantic
> > information about what each media
> > > stream represents."
> > >=20
> > > Isn't that something RFC 4574 could be used for?
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > > -----Original Message-----
> > > From: dispatch-bounces@ietf.org=20
> > [mailto:dispatch-bounces@ietf.org] On
> > Behalf Of Allyn Romanow (allyn)
> > > Sent: 15. syyskuuta 2010 20:42
> > > To: Elwell, John; DISPATCH list
> > > Subject: [dispatch] Telepresence charter version 5
> > >=20
> > > In keeping with John's good catch, the reference to BFCP has been
> > removed.
> > > This was the only suggested change to the charter.
> > >=20
> > >=20
> > > Thanks,
> > > Allyn
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > > MALT - Multi-stream Attributes for Lifelike Telepresence=20
> COCKTAIL -
> > Communication and Correlation of
> > > Key Telepresence Attributes for Interoperable Links MAITAI -
> > Multi-stream Attributes for Improving
> > > Telepresence Application Interoperability TEQUILA - Telepresence
> > Encoding of QUalifiers for
> > > Interoperable Lifelike Applications MOJITO - Multi-stream=20
> > Orientation
> > for Joining of Interoperable
> > > Telepresence Operations
> > >=20
> > >=20
> > > In the context of this WG, the term telepresence is used in=20
> > a general
> > manner to describe systems that
> > > provide high definition, high quality audio/video enabling a
> > "being-there" experience.  One example is
> > > an immersive telepresence system using specially designed=20
> > and special
> > purpose rooms with multiple
> > > displays permitting life size image reproduction using multiple
> > cameras, encoders, decoders,
> > > microphones and loudspeakers.
> > >=20
> > > Current telepresence systems are based on open standards=20
> > such as RTP,
> > SIP, H.264, the H.323 suite,
> > > however, they cannot easily interoperate with each other without
> > operator assistance and expensive
> > > additional equipment which translates from one vendor to=20
> another. A
> > major factor in the inability of
> > > telepresence systems to interwork is that there is no=20
> > standardized way
> > to describe and negotiate the
> > > use of the multiple streams of audio and video that=20
> > comprise the media
> > flows. In addition, there is no
> > > standardized way to exchange semantic information about what each
> > media stream represents.
> > >=20
> > > The WG will create specifications for SIP-based=20
> conferencing systems
> > to enable communication of enough
> > > information about each media stream so that each=20
> receiving system or
> > bridge system can make reasonable
> > > decisions about selecting and rendering media streams.=20
> This enables
> > systems to make display choices
> > > that optimize the "just like being there" experience.
> > >=20
> > > This working group is chartered to specify the information=20
> > about media
> > streams from one entity to
> > > another entity:
> > >=20
> > > * Spatial relationships of cameras, displays, microphones, and
> > >   Speakers - in relation to each other and to likely positions of
> > >   participants
> > >=20
> > > * Specific characteristics such as viewpoint, field of=20
> view/capture
> > >   for camera/microphone/display/speaker - so that senders and
> > >=20
> > >   middleboxes can understand how best to compose streams for
> > >   receivers, and the receivers will know the=20
> characteristics of its
> > >   received streams
> > >=20
> > > *Usage of the stream, for example whether the stream is=20
> > presentation,
> > or document camera output
> > >=20
> > > * Aspect ratio of cameras and displays
> > >=20
> > > * Which sources a receiver wants to receive.  For=20
> example, it might
> > want the source for the left
> > > camera, or might want the source chosen by VAD (Voice Activity
> > Detection).
> > >=20
> > >=20
> > > Information between sources and sinks about media stream=20
> > capabilities
> > will be exchanged.
> > >=20
> > > The working group will define the semantics, syntax,  and=20
> transport
> > mechanism necessary for
> > > communicating the necessary information. It will consider=20
> > whether the
> > existing signaling mechanisms
> > > (e. g., SDP) can be extended, or another messaging method=20
> should be
> > used.
> > >=20
> > > The scope of the work includes describing relatively static=20
> > relations
> > between entities (participants
> > > and devices). It also includes handling more dynamic=20
> relationships,
> > such as identifying the audio and
> > > video streams for the current speaker. The scope includes=20
> > both systems
> > that provide a fully immersive
> > > experience, and systems that interwork with them and=20
> > therefore need to
> > understand the same multiple
> > > stream semantics.
> > >=20
> > > The focus of this work is on multiple audio and video=20
> > streams.  Other
> > media types may be considered,
> > > however development of methodologies for them is not within=20
> > the scope
> > of this work.
> > >=20
> > > Interoperation with SIP and related standards for audio=20
> and video is
> > required.  However, backwards
> > > compatibility with existing non-standards compliant telepresence
> > systems is not required.
> > >=20
> > > This working group is not currently chartered to work on issues of
> > continuous conference control
> > > including: far end camera control, indication of fast frame=20
> > update for
> > video codecs or other rapid
> > > switches, floor control, conference roster.
> > >=20
> > > Reuse of existing protocols and backwards compatibility with
> > SIP-compliant audio/video endpoints  are
> > > important factors for the working group to consider. The work will
> > closely coordinate with the
> > > appropriate areas and working groups including OPS Area,=20
> > AVT, MMUSIC,
> > MEDIACTRL, XCON, and SIPCORE.
> > >=20
> > >  Milestones
> > >=20
> > >  Nov 2010 Submit information draft to IESG on use cases and
> > requirements
> > >=20
> > > Nov 2011 Submit standards track specification to IESG  indicating
> > spatial relationships of  screens
> > > cameras (including variable field of view and=20
> orientation), speakers
> > and microphones; and the "usage"
> > > of a stream as defined in the charter.  Semantics, language and
> > transport mechanism will be specified.
> > >=20
> > >=20
> > >=20
> > >
> > --------------------------------------------------------------
> > ----------
> > > --------
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > > Sent: Thursday, September 09, 2010 12:31 AM
> > > To: Allyn Romanow (allyn); DISPATCH list
> > > Subject: RE: Telepresence charter version 4
> > >=20
> > > Allyn,
> > >=20
> > > Can you explain the following:
> > > "It will consider whether the existing signaling=20
> mechanisms (e. g.,
> > ....
> > > BFCP) can be extended, or another messaging method should be used"
> > >=20
> > > and
> > >=20
> > > "This working group is not currently chartered to work on=20
> issues of=20
> > > continuous conference control including: .... floor control..."
> > >=20
> > > It is unclear to me how BFCP can be within scope yet floor=20
> > control is=20
> > > out of scope.
> > >=20
> > > John
> > > _______________________________________________
> > > dispatch mailing list
> > > dispatch@ietf.org
> > > https://www.ietf.org/mailman/listinfo/dispatch
> > > _______________________________________________
> > > dispatch mailing list
> > > dispatch@ietf.org
> > > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
>=20

From tme@americafree.tv  Sun Oct 24 20:44:27 2010
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To: "Allyn Romanow (allyn)" <allyn@cisco.com>
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Subject: Re: [dispatch] Telepresence charter version 5
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On Oct 24, 2010, at 9:14 PM, Allyn Romanow (allyn) wrote:

> Hi Christer,
>=20
> In reviewing this, I think it would be fine to just omit the sentence =
at
> issue-
> 	In addition,  there is no standardized way to exchange semantic
> information about what each 	media stream represents. =20
>=20
> Because I feel that the sentence just prior to it says adequately what
> is necessary. It is
> 	A major factor in the inability of telepresence systems to
> interwork is that there is no 	standardized way to describe and
> negotiate the use of the multiple streams of audio and video 	that
> comprise the media flows.

That would work for me. This sentence, however, has too many thats.

How about =20

A major factor limiting the interoperability of telepresence systems=20
is the lack of a standardized way to describe and negotiate the use of =
the multiple streams of audio and video comprising the media flows.

Marshall


>=20
>=20
> What do you think?
>=20
> Best regards,
> Allyn
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Monday, September 20, 2010 10:19 PM
> To: Allyn Romanow (allyn); Charles Eckel (eckelcu); Elwell, John;
> DISPATCH list
> Subject: RE: [dispatch] Telepresence charter version 5
>=20
>=20
> Hi,=20
>=20
>> Thanks for pointing out that the original sentence is not=20
>> strictly accurate, and also not precisely what was meant. I=20
>> think it is fine to change the wording along the lines you=20
>> suggest, can I get back to you shortly on precise wording? =20
>=20
> Sure, no problem.
>=20
> Regards,
>=20
> Christer
>=20
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> Sent: Monday, September 20, 2010 10:07 PM
>> To: Charles Eckel (eckelcu); Allyn Romanow (allyn); Elwell,=20
>> John; DISPATCH list
>> Subject: RE: [dispatch] Telepresence charter version 5
>>=20
>>=20
>> Hi Charles,=20
>>=20
>>> I think RFC label attribute may be useful; however, we would=20
>>> need to establish and register set of tokens and the=20
>>> semantics of those tokens.
>>> Once we get to the actual design, that would be worth considering.
>>> Without addition specification, I think you would have=20
>>> proprietary sets of labels being used at best.
>>=20
>> Yes, I agree that we would need to specify the labels.
>>=20
>> My point was that there does exist a mechanism for doing so. So, =
maybe
>> the text could say something like:
>>=20
>> "In addition, standardization work, in order to be able to exchange
>> semantic
>> information about what each media stream represents, might be =
needed."
>>=20
>> Regards,
>>=20
>> Christer
>>=20
>>=20
>>=20
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org=20
>>> [mailto:dispatch-bounces@ietf.org] On
>>> Behalf Of Christer Holmberg
>>>> Sent: Monday, September 20, 2010 2:18 AM
>>>> To: Allyn Romanow (allyn); Elwell, John; DISPATCH list
>>>> Subject: Re: [dispatch] Telepresence charter version 5
>>>>=20
>>>>=20
>>>> Hi,
>>>>=20
>>>> One question for clarification. The text says:
>>>>=20
>>>> "In addition, there is no standardized way to exchange semantic
>>> information about what each media
>>>> stream represents."
>>>>=20
>>>> Isn't that something RFC 4574 could be used for?
>>>>=20
>>>> Regards,
>>>>=20
>>>> Christer
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org=20
>>> [mailto:dispatch-bounces@ietf.org] On
>>> Behalf Of Allyn Romanow (allyn)
>>>> Sent: 15. syyskuuta 2010 20:42
>>>> To: Elwell, John; DISPATCH list
>>>> Subject: [dispatch] Telepresence charter version 5
>>>>=20
>>>> In keeping with John's good catch, the reference to BFCP has been
>>> removed.
>>>> This was the only suggested change to the charter.
>>>>=20
>>>>=20
>>>> Thanks,
>>>> Allyn
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> MALT - Multi-stream Attributes for Lifelike Telepresence=20
>> COCKTAIL -
>>> Communication and Correlation of
>>>> Key Telepresence Attributes for Interoperable Links MAITAI -
>>> Multi-stream Attributes for Improving
>>>> Telepresence Application Interoperability TEQUILA - Telepresence
>>> Encoding of QUalifiers for
>>>> Interoperable Lifelike Applications MOJITO - Multi-stream=20
>>> Orientation
>>> for Joining of Interoperable
>>>> Telepresence Operations
>>>>=20
>>>>=20
>>>> In the context of this WG, the term telepresence is used in=20
>>> a general
>>> manner to describe systems that
>>>> provide high definition, high quality audio/video enabling a
>>> "being-there" experience.  One example is
>>>> an immersive telepresence system using specially designed=20
>>> and special
>>> purpose rooms with multiple
>>>> displays permitting life size image reproduction using multiple
>>> cameras, encoders, decoders,
>>>> microphones and loudspeakers.
>>>>=20
>>>> Current telepresence systems are based on open standards=20
>>> such as RTP,
>>> SIP, H.264, the H.323 suite,
>>>> however, they cannot easily interoperate with each other without
>>> operator assistance and expensive
>>>> additional equipment which translates from one vendor to=20
>> another. A
>>> major factor in the inability of
>>>> telepresence systems to interwork is that there is no=20
>>> standardized way
>>> to describe and negotiate the
>>>> use of the multiple streams of audio and video that=20
>>> comprise the media
>>> flows. In addition, there is no
>>>> standardized way to exchange semantic information about what each
>>> media stream represents.
>>>>=20
>>>> The WG will create specifications for SIP-based=20
>> conferencing systems
>>> to enable communication of enough
>>>> information about each media stream so that each=20
>> receiving system or
>>> bridge system can make reasonable
>>>> decisions about selecting and rendering media streams.=20
>> This enables
>>> systems to make display choices
>>>> that optimize the "just like being there" experience.
>>>>=20
>>>> This working group is chartered to specify the information=20
>>> about media
>>> streams from one entity to
>>>> another entity:
>>>>=20
>>>> * Spatial relationships of cameras, displays, microphones, and
>>>>  Speakers - in relation to each other and to likely positions of
>>>>  participants
>>>>=20
>>>> * Specific characteristics such as viewpoint, field of=20
>> view/capture
>>>>  for camera/microphone/display/speaker - so that senders and
>>>>=20
>>>>  middleboxes can understand how best to compose streams for
>>>>  receivers, and the receivers will know the=20
>> characteristics of its
>>>>  received streams
>>>>=20
>>>> *Usage of the stream, for example whether the stream is=20
>>> presentation,
>>> or document camera output
>>>>=20
>>>> * Aspect ratio of cameras and displays
>>>>=20
>>>> * Which sources a receiver wants to receive.  For=20
>> example, it might
>>> want the source for the left
>>>> camera, or might want the source chosen by VAD (Voice Activity
>>> Detection).
>>>>=20
>>>>=20
>>>> Information between sources and sinks about media stream=20
>>> capabilities
>>> will be exchanged.
>>>>=20
>>>> The working group will define the semantics, syntax,  and=20
>> transport
>>> mechanism necessary for
>>>> communicating the necessary information. It will consider=20
>>> whether the
>>> existing signaling mechanisms
>>>> (e. g., SDP) can be extended, or another messaging method=20
>> should be
>>> used.
>>>>=20
>>>> The scope of the work includes describing relatively static=20
>>> relations
>>> between entities (participants
>>>> and devices). It also includes handling more dynamic=20
>> relationships,
>>> such as identifying the audio and
>>>> video streams for the current speaker. The scope includes=20
>>> both systems
>>> that provide a fully immersive
>>>> experience, and systems that interwork with them and=20
>>> therefore need to
>>> understand the same multiple
>>>> stream semantics.
>>>>=20
>>>> The focus of this work is on multiple audio and video=20
>>> streams.  Other
>>> media types may be considered,
>>>> however development of methodologies for them is not within=20
>>> the scope
>>> of this work.
>>>>=20
>>>> Interoperation with SIP and related standards for audio=20
>> and video is
>>> required.  However, backwards
>>>> compatibility with existing non-standards compliant telepresence
>>> systems is not required.
>>>>=20
>>>> This working group is not currently chartered to work on issues of
>>> continuous conference control
>>>> including: far end camera control, indication of fast frame=20
>>> update for
>>> video codecs or other rapid
>>>> switches, floor control, conference roster.
>>>>=20
>>>> Reuse of existing protocols and backwards compatibility with
>>> SIP-compliant audio/video endpoints  are
>>>> important factors for the working group to consider. The work will
>>> closely coordinate with the
>>>> appropriate areas and working groups including OPS Area,=20
>>> AVT, MMUSIC,
>>> MEDIACTRL, XCON, and SIPCORE.
>>>>=20
>>>> Milestones
>>>>=20
>>>> Nov 2010 Submit information draft to IESG on use cases and
>>> requirements
>>>>=20
>>>> Nov 2011 Submit standards track specification to IESG  indicating
>>> spatial relationships of  screens
>>>> cameras (including variable field of view and=20
>> orientation), speakers
>>> and microphones; and the "usage"
>>>> of a stream as defined in the charter.  Semantics, language and
>>> transport mechanism will be specified.
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>> --------------------------------------------------------------
>>> ----------
>>>> --------
>>>> -----Original Message-----
>>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>>> Sent: Thursday, September 09, 2010 12:31 AM
>>>> To: Allyn Romanow (allyn); DISPATCH list
>>>> Subject: RE: Telepresence charter version 4
>>>>=20
>>>> Allyn,
>>>>=20
>>>> Can you explain the following:
>>>> "It will consider whether the existing signaling=20
>> mechanisms (e. g.,
>>> ....
>>>> BFCP) can be extended, or another messaging method should be used"
>>>>=20
>>>> and
>>>>=20
>>>> "This working group is not currently chartered to work on=20
>> issues of=20
>>>> continuous conference control including: .... floor control..."
>>>>=20
>>>> It is unclear to me how BFCP can be within scope yet floor=20
>>> control is=20
>>>> out of scope.
>>>>=20
>>>> John
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>=20
>>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20


From allyn@cisco.com  Sun Oct 24 22:38:30 2010
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Thread-Topic: [dispatch] Telepresence charter version 5
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References: <9AC2C4348FD86B4BB1F8FA9C5E3A5EDC0254DFFA@xmb-sjc-221.amer.cisco.com><A444A0F8084434499206E78C106220CA01C7FC8309@MCHP058A.global-ad.net><9AC2C4348FD86B4BB1F8FA9C5E3A5EDC02700028@xmb-sjc-221.amer.cisco.com> <7F2072F1E0DE894DA4B517B93C6A058501653F94@ESESSCMS0356.eemea.ericsson.se> <E1CBF4C7095A3D4CAAAEAD09FBB8E08C021BCC62@xmb-sjc-234.amer.cisco.com> <7F2072F1E0DE894DA4B517B93C6A058501703C4C@ESESSCMS0356.eemea.ericsson.se> <9AC2C4348FD86B4BB1F8FA9C5E3A5EDC027B1D20@xmb-sjc-221.amer.cisco.com> <7F2072F1E0DE894DA4B517B93C6A058501703C50@ESESSCMS0356.eemea.ericsson.se> <9AC2C4348FD86B4BB1F8FA9C5E3A5EDC02BB188B@xmb-sjc-221.amer.cisco.com> <383AAA4F-B7A4-40E2-9F37-1772A60B0EDA@americafree.tv>
From: "Allyn Romanow (allyn)" <allyn@cisco.com>
To: "Marshall Eubanks" <tme@americafree.tv>
X-OriginalArrivalTime: 25 Oct 2010 05:40:07.0301 (UTC) FILETIME=[14D0C350:01CB7407]
Cc: DISPATCH list <dispatch@ietf.org>
Subject: Re: [dispatch] Telepresence charter version 5
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Much improved. Will incorporate it in next version, which I hope to send
out Monday night.

Thanks,
Allyn=20

> -----Original Message-----
> From: Marshall Eubanks [mailto:tme@americafree.tv]=20
> Sent: Sunday, October 24, 2010 8:46 PM
> To: Allyn Romanow (allyn)
> Cc: Christer Holmberg; Charles Eckel (eckelcu); Elwell, John;=20
> DISPATCH list
> Subject: Re: [dispatch] Telepresence charter version 5
>=20
>=20
> On Oct 24, 2010, at 9:14 PM, Allyn Romanow (allyn) wrote:
>=20
> > Hi Christer,
> >=20
> > In reviewing this, I think it would be fine to just omit=20
> the sentence=20
> > at
> > issue-
> > 	In addition,  there is no standardized way to exchange semantic
> > information about what each 	media stream represents. =20
> >=20
> > Because I feel that the sentence just prior to it says=20
> adequately what=20
> > is necessary. It is
> > 	A major factor in the inability of telepresence systems to
> > interwork is that there is no 	standardized way to describe and
> > negotiate the use of the multiple streams of audio and=20
> video 	that
> > comprise the media flows.
>=20
> That would work for me. This sentence, however, has too many thats.
>=20
> How about =20
>=20
> A major factor limiting the interoperability of telepresence=20
> systems is the lack of a standardized way to describe and=20
> negotiate the use of the multiple streams of audio and video=20
> comprising the media flows.
>=20
> Marshall
>=20
>=20
> >=20
> >=20
> > What do you think?
> >=20
> > Best regards,
> > Allyn
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: Monday, September 20, 2010 10:19 PM
> > To: Allyn Romanow (allyn); Charles Eckel (eckelcu); Elwell, John;=20
> > DISPATCH list
> > Subject: RE: [dispatch] Telepresence charter version 5
> >=20
> >=20
> > Hi,
> >=20
> >> Thanks for pointing out that the original sentence is not strictly=20
> >> accurate, and also not precisely what was meant. I think=20
> it is fine=20
> >> to change the wording along the lines you suggest, can I=20
> get back to=20
> >> you shortly on precise wording?
> >=20
> > Sure, no problem.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >> -----Original Message-----
> >> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >> Sent: Monday, September 20, 2010 10:07 PM
> >> To: Charles Eckel (eckelcu); Allyn Romanow (allyn); Elwell, John;=20
> >> DISPATCH list
> >> Subject: RE: [dispatch] Telepresence charter version 5
> >>=20
> >>=20
> >> Hi Charles,
> >>=20
> >>> I think RFC label attribute may be useful; however, we=20
> would need to=20
> >>> establish and register set of tokens and the semantics of those=20
> >>> tokens.
> >>> Once we get to the actual design, that would be worth considering.
> >>> Without addition specification, I think you would have=20
> proprietary=20
> >>> sets of labels being used at best.
> >>=20
> >> Yes, I agree that we would need to specify the labels.
> >>=20
> >> My point was that there does exist a mechanism for doing so. So,=20
> >> maybe the text could say something like:
> >>=20
> >> "In addition, standardization work, in order to be able to=20
> exchange=20
> >> semantic information about what each media stream=20
> represents, might=20
> >> be needed."
> >>=20
> >> Regards,
> >>=20
> >> Christer
> >>=20
> >>=20
> >>=20
> >>>> -----Original Message-----
> >>>> From: dispatch-bounces@ietf.org
> >>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Christer Holmberg
> >>>> Sent: Monday, September 20, 2010 2:18 AM
> >>>> To: Allyn Romanow (allyn); Elwell, John; DISPATCH list
> >>>> Subject: Re: [dispatch] Telepresence charter version 5
> >>>>=20
> >>>>=20
> >>>> Hi,
> >>>>=20
> >>>> One question for clarification. The text says:
> >>>>=20
> >>>> "In addition, there is no standardized way to exchange semantic
> >>> information about what each media
> >>>> stream represents."
> >>>>=20
> >>>> Isn't that something RFC 4574 could be used for?
> >>>>=20
> >>>> Regards,
> >>>>=20
> >>>> Christer
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>> -----Original Message-----
> >>>> From: dispatch-bounces@ietf.org
> >>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Allyn Romanow=20
> >>> (allyn)
> >>>> Sent: 15. syyskuuta 2010 20:42
> >>>> To: Elwell, John; DISPATCH list
> >>>> Subject: [dispatch] Telepresence charter version 5
> >>>>=20
> >>>> In keeping with John's good catch, the reference to BFCP has been
> >>> removed.
> >>>> This was the only suggested change to the charter.
> >>>>=20
> >>>>=20
> >>>> Thanks,
> >>>> Allyn
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>>> MALT - Multi-stream Attributes for Lifelike Telepresence
> >> COCKTAIL -
> >>> Communication and Correlation of
> >>>> Key Telepresence Attributes for Interoperable Links MAITAI -
> >>> Multi-stream Attributes for Improving
> >>>> Telepresence Application Interoperability TEQUILA - Telepresence
> >>> Encoding of QUalifiers for
> >>>> Interoperable Lifelike Applications MOJITO - Multi-stream
> >>> Orientation
> >>> for Joining of Interoperable
> >>>> Telepresence Operations
> >>>>=20
> >>>>=20
> >>>> In the context of this WG, the term telepresence is used in
> >>> a general
> >>> manner to describe systems that
> >>>> provide high definition, high quality audio/video enabling a
> >>> "being-there" experience.  One example is
> >>>> an immersive telepresence system using specially designed
> >>> and special
> >>> purpose rooms with multiple
> >>>> displays permitting life size image reproduction using multiple
> >>> cameras, encoders, decoders,
> >>>> microphones and loudspeakers.
> >>>>=20
> >>>> Current telepresence systems are based on open standards
> >>> such as RTP,
> >>> SIP, H.264, the H.323 suite,
> >>>> however, they cannot easily interoperate with each other without
> >>> operator assistance and expensive
> >>>> additional equipment which translates from one vendor to
> >> another. A
> >>> major factor in the inability of
> >>>> telepresence systems to interwork is that there is no
> >>> standardized way
> >>> to describe and negotiate the
> >>>> use of the multiple streams of audio and video that
> >>> comprise the media
> >>> flows. In addition, there is no
> >>>> standardized way to exchange semantic information about what each
> >>> media stream represents.
> >>>>=20
> >>>> The WG will create specifications for SIP-based
> >> conferencing systems
> >>> to enable communication of enough
> >>>> information about each media stream so that each
> >> receiving system or
> >>> bridge system can make reasonable
> >>>> decisions about selecting and rendering media streams.=20
> >> This enables
> >>> systems to make display choices
> >>>> that optimize the "just like being there" experience.
> >>>>=20
> >>>> This working group is chartered to specify the information
> >>> about media
> >>> streams from one entity to
> >>>> another entity:
> >>>>=20
> >>>> * Spatial relationships of cameras, displays, microphones, and =20
> >>>> Speakers - in relation to each other and to likely positions of =20
> >>>> participants
> >>>>=20
> >>>> * Specific characteristics such as viewpoint, field of
> >> view/capture
> >>>>  for camera/microphone/display/speaker - so that senders and
> >>>>=20
> >>>>  middleboxes can understand how best to compose streams for =20
> >>>> receivers, and the receivers will know the
> >> characteristics of its
> >>>>  received streams
> >>>>=20
> >>>> *Usage of the stream, for example whether the stream is
> >>> presentation,
> >>> or document camera output
> >>>>=20
> >>>> * Aspect ratio of cameras and displays
> >>>>=20
> >>>> * Which sources a receiver wants to receive.  For
> >> example, it might
> >>> want the source for the left
> >>>> camera, or might want the source chosen by VAD (Voice Activity
> >>> Detection).
> >>>>=20
> >>>>=20
> >>>> Information between sources and sinks about media stream
> >>> capabilities
> >>> will be exchanged.
> >>>>=20
> >>>> The working group will define the semantics, syntax,  and
> >> transport
> >>> mechanism necessary for
> >>>> communicating the necessary information. It will consider
> >>> whether the
> >>> existing signaling mechanisms
> >>>> (e. g., SDP) can be extended, or another messaging method
> >> should be
> >>> used.
> >>>>=20
> >>>> The scope of the work includes describing relatively static
> >>> relations
> >>> between entities (participants
> >>>> and devices). It also includes handling more dynamic
> >> relationships,
> >>> such as identifying the audio and
> >>>> video streams for the current speaker. The scope includes
> >>> both systems
> >>> that provide a fully immersive
> >>>> experience, and systems that interwork with them and
> >>> therefore need to
> >>> understand the same multiple
> >>>> stream semantics.
> >>>>=20
> >>>> The focus of this work is on multiple audio and video
> >>> streams.  Other
> >>> media types may be considered,
> >>>> however development of methodologies for them is not within
> >>> the scope
> >>> of this work.
> >>>>=20
> >>>> Interoperation with SIP and related standards for audio
> >> and video is
> >>> required.  However, backwards
> >>>> compatibility with existing non-standards compliant telepresence
> >>> systems is not required.
> >>>>=20
> >>>> This working group is not currently chartered to work on=20
> issues of
> >>> continuous conference control
> >>>> including: far end camera control, indication of fast frame
> >>> update for
> >>> video codecs or other rapid
> >>>> switches, floor control, conference roster.
> >>>>=20
> >>>> Reuse of existing protocols and backwards compatibility with
> >>> SIP-compliant audio/video endpoints  are
> >>>> important factors for the working group to consider. The=20
> work will
> >>> closely coordinate with the
> >>>> appropriate areas and working groups including OPS Area,
> >>> AVT, MMUSIC,
> >>> MEDIACTRL, XCON, and SIPCORE.
> >>>>=20
> >>>> Milestones
> >>>>=20
> >>>> Nov 2010 Submit information draft to IESG on use cases and
> >>> requirements
> >>>>=20
> >>>> Nov 2011 Submit standards track specification to IESG  indicating
> >>> spatial relationships of  screens
> >>>> cameras (including variable field of view and
> >> orientation), speakers
> >>> and microphones; and the "usage"
> >>>> of a stream as defined in the charter.  Semantics, language and
> >>> transport mechanism will be specified.
> >>>>=20
> >>>>=20
> >>>>=20
> >>>>=20
> >>> --------------------------------------------------------------
> >>> ----------
> >>>> --------
> >>>> -----Original Message-----
> >>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> >>>> Sent: Thursday, September 09, 2010 12:31 AM
> >>>> To: Allyn Romanow (allyn); DISPATCH list
> >>>> Subject: RE: Telepresence charter version 4
> >>>>=20
> >>>> Allyn,
> >>>>=20
> >>>> Can you explain the following:
> >>>> "It will consider whether the existing signaling
> >> mechanisms (e. g.,
> >>> ....
> >>>> BFCP) can be extended, or another messaging method=20
> should be used"
> >>>>=20
> >>>> and
> >>>>=20
> >>>> "This working group is not currently chartered to work on
> >> issues of
> >>>> continuous conference control including: .... floor control..."
> >>>>=20
> >>>> It is unclear to me how BFCP can be within scope yet floor
> >>> control is
> >>>> out of scope.
> >>>>=20
> >>>> John
> >>>> _______________________________________________
> >>>> dispatch mailing list
> >>>> dispatch@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/dispatch
> >>>> _______________________________________________
> >>>> dispatch mailing list
> >>>> dispatch@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/dispatch
> >>>=20
> >>=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
>=20
>=20

From christer.holmberg@ericsson.com  Sun Oct 24 22:40:54 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@cisco.com>
Date: Mon, 25 Oct 2010 07:40:50 +0200
Thread-Topic: [dispatch] I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
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Hi,

>Uh, same as Paul, sorry, ignore my previous email. Don't know how I missed=
 this on first search. Thank you for sending the pointer to this.
>
>What is the sate of this 3GPP doc and what is going on with the registrati=
on?

The document is stable. A reason the package hasn't been registered yet mig=
ht be that people assume the info-event draft has to become RFC first.

But, if that is not the case, I can inform people that the package can be r=
egistered.

Regards,

Christer



On Aug 24, 2010, at 9:57 PM, Christer Holmberg wrote:

>
> Hi,
>
> The package definition can be found in Annex P of 3GPP TS 24.229:
>
> http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-a00.zip
>
> The MIME body used by the package to transport the digits is defined in A=
nnex G of 3GPP TS 29.163 (it re-uses a body used for overlap dialling, in c=
ase the wording confuses):
>
> http://www.3gpp.org/ftp/Specs/archive/29_series/29.163/29163-920.zip
>
> Regards,
>
> Christer
>
>
>
>> -----Original Message-----
>> From: Muthu ArulMozhi Perumal (mperumal) [mailto:mperumal@cisco.com]
>> Sent: 25. elokuuta 2010 6:47
>> To: Christer Holmberg; Parthasarathi R (partr); Paul Kyzivat
>> (pkyzivat)
>> Cc: dispatch@ietf.org; Hadriel Kaplan
>> Subject: RE:
>> [dispatch]I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
>>
>> Can you please provide the pointer to the 3GPP spec
>> describing this DTMF INFO package?
>>
>> thanks,
>> Muthu
>>
>> |-----Original Message-----
>> |From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> |Sent: Wednesday, August 25, 2010 9:05 AM
>> |To: Parthasarathi R (partr); Paul Kyzivat (pkyzivat)
>> |Cc: dispatch@ietf.org; Hadriel Kaplan; Muthu ArulMozhi Perumal
>> (mperumal)
>> |Subject: RE:
>> [dispatch]I-DAction:draft-kaplan-dispatch-info-dtmf-package-
>> |00.txt
>> |
>> |
>> |Hi,
>> |
>> |>1) whether 3GPP DTMF Info package is in the state to be accepted as
>> IETF
>> |standards.
>> |
>> |The Info Package registration procedure requires an expert review,
>> which
>> |purpose is to ensure that the Info Package does not break the SIP
>> protocol
>> |etc. I assume such review will take place when the package is
>> registered
>> |with IANA.
>> |
>> |Regards,
>> |
>> |Christer
>> |
>> |
>> |
>> |
>> |________________________________
>> |
>> |    From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> |    Sent: Wed 8/25/2010 8:45 AM
>> |    To: Parthasarathi R (partr); Paul Kyzivat (pkyzivat)
>> |    Cc: dispatch@ietf.org; Hadriel Kaplan; Muthu ArulMozhi Perumal
>> |(mperumal)
>> |    Subject: RE:
>> [dispatch]I-DAction:draft-kaplan-dispatch-info-dtmf-
>> |package-00.txt
>> |
>> |
>> |
>> |
>> |    Hi,
>> |
>> |    >In case IMS needs new INFO based DTMF relay mechanism, I agree
>> that
>> |it is the valid requirement and we need to evaluate it. Before
>> concluding
>> |about new
>> |    >INFO based mechanism in IMS, Please throw the light for not
>> selecting
>> |KPML in IMS. I'm comparing KPML vs INFO based mechanism as both are
>> based on
>> |SIP
>> |    >signaling based mechanism. KPML has limited deployment as
>> mentioned
>> |in Hadriel Info draft but new INFO package is yet to be
>> defined. It is
>> |better to have
>> |    >the single active IETF SIP signaling based dtmf-relay
>> mechanism
>> |rather than producing more standards wherein implementers will
>> completely
>> |confused which
>> |    >one to select.
>> |    >
>> |    >As Muthu mentioned, I also heard about KPML deployment issues
>> but the
>> |legacy INFO based dtmf-relay has lot more caveats. I guess that Info
>> package
>> |is
>> |    >designed as generic as KPML, and then both KPML and INFO may
>> look
>> |heavyweight.
>> |
>> |    The 3GPP DTMF Info Package is very simple.
>> |
>> |    Regards,
>> |
>> |    Christer
>> |
>> |
>> |
>> |
>> |    ________________________________
>> |
>> |            From: dispatch-bounces@ietf.org on behalf of Paul
>> Kyzivat
>> |(pkyzivat)
>> |            Sent: Wed 8/25/2010 12:17 AM
>> |            To: Christer Holmberg
>> |            Cc: dispatch@ietf.org; Hadriel Kaplan
>> |            Subject: Re: [dispatch]I-D
>> Action:draft-kaplan-dispatch-info-
>> |dtmf-package-00.txt
>> |
>> |
>> |
>> |
>> |
>> |            Christer Holmberg wrote:
>> |            > Hi,
>> |            >
>> |            > I think it is strange to reject something based on a
>> claim
>> |there already exists a number of legacy ways of there. Not
>> everyone has
>> |implemented the same (if any) of them, and the whole purpose of the
>> Info
>> |Package framework is to come up with a mechanism for which the usage
>> can be
>> |negotiated.
>> |            >
>> |            > And, as I said earlier, at least in IMS an Info
>> Package is
>> |being used for DTMF transport. It is currently defined in 3GPP
>> |specifications, but there shouldn't be anything IMS specific
>> about the
>> |package itself.
>> |
>> |            I wasn't asserting a general principle.
>> |            I was talking about this particular mechanism.
>> |
>> |            If there is yet another dtmf package, then I guess we
>> can
>> |consider the
>> |            pros/cons of it. Those are probably best assessed by
>> those who
>> |use the
>> |            existing one and/or might use whatever info package you
>> might
>> |propose to
>> |            replace it.
>> |
>> |            We clearly suffer from a surplus of ways to convey dtmf.
>> |            While its not impossible to argue than another
>> one would
>> |improve the
>> |            situation, it is going to be a hard sell.
>> |
>> |            OTOH, the more there are the more case it makes for
>> having an
>> |SBC in the
>> |            path to "fix" the interoperability. We sell them, so
>> maybe I
>> |should be
>> |            cheering on every new one as a way to boost sales.
>> |
>> |                    Thanks,
>> |                    Paul
>> |
>> |            > Regards,
>> |            >
>> |            > Christer
>> |            >
>> |            >
>> |            >
>> |            >> -----Original Message-----
>> |            >> From: dispatch-bounces@ietf.org
>> |            >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul
>> |Kyzivat
>> |            >> Sent: 24. elokuuta 2010 18:51
>> |            >> To: Hadriel Kaplan
>> |            >> Cc: dispatch@ietf.org
>> |            >> Subject: Re: [dispatch] I-D
>> |            >> Action:draft-kaplan-dispatch-info-dtmf-package-00.txt
>> |            >>
>> |            >> Maybe we should ask our ADs if they have an opinion
>> about
>> |this.
>> |            >>
>> |            >>      Thanks,
>> |            >>      Paul
>> |            >>
>> |            >> Hadriel Kaplan wrote:
>> |            >>>> -----Original Message-----
>> |            >>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> |            >>>> Sent: Monday, August 23, 2010 6:02 PM
>> |            >>>> To: Hadriel Kaplan
>> |            >>>>
>> |            >>>> That reduces things somewhat. But if
>> everybody that
>> |supports the
>> |            >>>> package also supports the legacy approach, what is
>> the
>> |win?
>> |            >>> The legacy mode has no published standards document
>> |            >> defining it.  I thought when the info-packages work
>> was
>> |            >> started, DTMF-in-info was one of the main drivers.
>> But I
>> |            >> take your point - if people would prefer to just
>> document
>> |it
>> |            >> as it is today (sans info-packages), I can change the
>> draft
>> |            >> to just be that.  I'm cool with either way (defining
>> the
>> |            >> current dtmf-relay as a legacy mode was Option-2).
>> |            >>> -hadriel
>> |            >>>
>> |            >> _______________________________________________
>> |            >> dispatch mailing list
>> |            >> dispatch@ietf.org
>> |            >> https://www.ietf.org/mailman/listinfo/dispatch
>> |            >>
>> |            _______________________________________________
>> |            dispatch mailing list
>> |            dispatch@ietf.org
>> |            https://www.ietf.org/mailman/listinfo/dispatch
>> |
>> |
>> |
>>
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch=

From magnus.westerlund@ericsson.com  Mon Oct 25 01:44:17 2010
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Allyn Romanow (allyn) skrev 2010-10-25 02:24:
> Hi Magnus -
> 
> I agree with you that the issues you bring up need to be dealt with, and I feel they should potentially result in requirements from this working group.
> 
> With respect in particular to your next to last paragraph below, did you not think that my proposed change to the charter (as Peter quoted), covered your concern?
> 
> "also propose the following addition:
>        This working group is not currently chartered to work on issues of continuous conference        control including: far end camera control, indication of fast frame update for video codecs or other rapid switches, floor control, conference roster. The working group may identify interoperability obstacles in existing open standards. If so, the WG will develop        requirements to be communicated to other IETF WGs or Standards Forums as a kind request to try to meet these requirements."

Ok, I missed that proposal. That seems fine, except that I would
possibly add that a recharter is one possible way of realizing the
requirements?

cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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From: "Allyn Romanow (allyn)" <allyn@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Okay, how about

This working group is not currently chartered to work on issues of =
continuous conference        control including: far end camera control, =
indication of fast frame update for video codecs or other rapid =
switches, floor control, conference roster. The working group may =
identify interoperability obstacles in existing open standards. If so, =
the WG will develop        requirements to be communicated to other IETF =
WGs or Standards Forums, or recharter as appropriate."

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: Monday, October 25, 2010 1:46 AM
To: Allyn Romanow (allyn)
Cc: Peter Musgrave; dispatch@ietf.org
Subject: Re: [dispatch] Comments on the proposed Telepresence Charter

Allyn Romanow (allyn) skrev 2010-10-25 02:24:
> Hi Magnus -
>=20
> I agree with you that the issues you bring up need to be dealt with, =
and I feel they should potentially result in requirements from this =
working group.
>=20
> With respect in particular to your next to last paragraph below, did =
you not think that my proposed change to the charter (as Peter quoted), =
covered your concern?
>=20
> "also propose the following addition:
>        This working group is not currently chartered to work on issues =
of continuous conference        control including: far end camera =
control, indication of fast frame update for video codecs or other rapid =
switches, floor control, conference roster. The working group may =
identify interoperability obstacles in existing open standards. If so, =
the WG will develop        requirements to be communicated to other IETF =
WGs or Standards Forums as a kind request to try to meet these =
requirements."

Ok, I missed that proposal. That seems fine, except that I would
possibly add that a recharter is one possible way of realizing the
requirements?

cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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In preparation for discussion in DISPATCH meeting, here are the changes
since the last draft. They include:

=20

Changes from Version 5

*         Editorial comments from Jonathan Lennox

*         Christer Holmberg - "In addition, standardization work, in
order to be able to exchange semantic information about what each media
stream represents, might be needed."

      Propose leave out sentence

*         Magnus Westerlund - include transmitting system, add WG
chartered to make requirements for TP interoperability to distribute or
handle as appropriate

*         Ted Hardie - early milestone, use case without participants
(use case draft)

*         Minor editorial changes

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

=20

MAITAI - Multi-stream Attributes for Improving Telepresence Application
Interoperability

CLUE -- ControLling mUltiple streams for tElepresence

=20

In the context of this WG, the term telepresence is used in a general
manner to describe systems that provide high definition, high quality
audio/video enabling a "being-there" experience.  One example is an
immersive telepresence system using specially designed and special
purpose rooms with multiple displays permitting life size image
reproduction using multiple cameras, encoders, decoders, microphones and
loudspeakers.

=20

Current telepresence systems are based on open standards such as RTP,
SIP, H.264, the H.323 suite. However, they cannot easily interoperate
with each other without operator assistance and expensive additional
equipment which translates from one vendor to another. A major factor
limiting the interoperability of telepresence systems  is the lack of a
standardized way to describe and negotiate the use of the multiple
streams of audio and video comprising the media flows.

=20

The WG will create specifications for SIP-based conferencing systems to
enable communication of  information about media streams so that a
sending system,  receiving system, or intermediate system can make
reasonable decisions about transmitting, selecting, and rendering media
streams. This enables systems to make choices that optimize user
experience.

=20

This working group is chartered to specify  information about media
streams from one entity to another entity:=20

=20

* Spatial relationships of cameras, displays, microphones, and speakers
- relative to each other and to likely positions of participants

=20

* Specific characteristics such as viewpoint, field of view/capture for
camera/microphone/display/speaker - so that senders and intermediate
devices can understand how best to compose streams for receivers, and
the receiver will know the characteristics of its received streams

=20

* Usage of the stream, for example whether the stream is presentation,
or document camera output

=20

* Aspect ratio of cameras and displays=20

=20

*Which sources a receiver wants to receive.  For example, it might  want
the source for the left camera, or might want the source chosen by VAD
(Voice Activity Detection)

=20

Information between sources and sinks about media stream capabilities
will be exchanged.=20

=20

The working group will define the semantics, syntax, and transport
mechanism necessary for communicating the necessary information. It will
consider whether the existing signaling mechanisms (e. g., SDP) can be
extended, or another messaging method should be used. =20

=20

The scope of the work includes describing relatively static relations
between entities (participants and devices). It also includes handling
more dynamic relationships, such as identifying the audio and video
streams for the current speaker. The scope includes both systems that
provide a fully immersive experience, and systems that interwork with
them and therefore need to understand the same multiple stream
semantics. =20

=20

The focus of this work is on multiple audio and video streams.  Other
media types may be considered, however development of methodologies for
them is not within the scope of this work.

=20

Interoperation with SIP and related standards for audio and video is
required.  However, backwards compatibility with existing non-standards
compliant telepresence systems is not required.

=20

This working group is not currently chartered to work on issues of
continuous conference control including: far end camera control,
indication of fast frame update for video codecs or other rapid
switches, floor control, conference roster. The working group may
identify interoperability obstacles in existing open standards. If so,
the WG will develop requirements to be communicated to other IETF WGs or
Standards Forums, or recharter as appropriate.

=20

Reuse of existing protocols and backwards compatibility with
SIP-compliant audio/video endpoints are important factors for the
working group to consider. The work will closely coordinate with the
appropriate areas (e. g., OPS, SEC), and working groups including  AVT,
MMUSIC, MEDIACTRL, XCON, and SIPCORE.

=20

 Milestones =20

=20

July 2011 Submit informational draft to IESG on use cases=20

=20

July 2011 Submit informational draft to IESG on framework and
requirements

=20

Nov 2011 Submit standards track specification(s) to IESG to support
framework and requirements

=20


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DWordSection1>

<p class=3DMsoNormal>In preparation for discussion in DISPATCH meeting, =
here are
the changes since the last draft. They include:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Changes from Version 5<o:p></o:p></p>

<p class=3DMsoListParagraphCxSpFirst =
style=3D'margin-left:.25in;text-indent:-.25in;
mso-list:l2 level1 lfo1'><![if !supportLists]><span =
style=3D'font-family:Symbol'><span
style=3D'mso-list:Ignore'>&middot;<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D'font-family:"Times New =
Roman","serif"'>Editorial
comments from Jonathan Lennox<o:p></o:p></span></p>

<p class=3DMsoListParagraphCxSpLast =
style=3D'margin-left:.25in;text-indent:-.25in;
mso-list:l2 level1 lfo1'><![if !supportLists]><span =
style=3D'font-family:Symbol'><span
style=3D'mso-list:Ignore'>&middot;<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D'font-family:"Times New =
Roman","serif"'>Christer
Holmberg &#8211; &#8220;In addition, standardization work, in order to =
be able
to exchange semantic information about what each media stream =
represents, might
be needed.&#8221;<o:p></o:p></span></p>

<p class=3DMsoPlainText><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Propose leave out sentence<o:p></o:p></span></p>

<p class=3DMsoPlainText =
style=3D'margin-left:.25in;text-indent:-.25in;mso-list:
l0 level1 lfo2'><![if !supportLists]><span =
style=3D'font-size:12.0pt;font-family:
Symbol'><span style=3D'mso-list:Ignore'>&middot;<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>Magnus
Westerlund &#8211; include transmitting system, add WG chartered to make
requirements for TP interoperability to distribute or handle as =
appropriate<o:p></o:p></span></p>

<p class=3DMsoPlainText =
style=3D'margin-left:.25in;text-indent:-.25in;mso-list:
l1 level1 lfo3'><![if !supportLists]><span =
style=3D'font-size:12.0pt;font-family:
Symbol'><span style=3D'mso-list:Ignore'>&middot;<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>Ted
Hardie &#8211; early milestone, use case without participants&nbsp; (use =
case
draft)<o:p></o:p></span></p>

<p class=3DMsoPlainText =
style=3D'margin-left:.25in;text-indent:-.25in;mso-list:
l1 level1 lfo3'><![if !supportLists]><span =
style=3D'font-size:12.0pt;font-family:
Symbol'><span style=3D'mso-list:Ignore'>&middot;<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>Minor
editorial changes<o:p></o:p></span></p>

<p class=3DMsoNormal>------------------------<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>MAITAI &#8211; Multi-stream Attributes for =
Improving
Telepresence Application Interoperability<o:p></o:p></p>

<p class=3DMsoNormal>CLUE -- ControLling mUltiple streams for =
tElepresence<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>In the context of this WG, the term telepresence is =
used in
a general manner to describe systems that provide high definition, high =
quality
audio/video enabling a &quot;being-there&quot; experience. &nbsp;One =
example is
an immersive telepresence system using specially designed and special =
purpose
rooms with multiple displays permitting life size image reproduction =
using
multiple cameras, encoders, decoders, microphones and =
loudspeakers.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Current telepresence systems are based on open =
standards
such as RTP, SIP, H.264, the H.323 suite. However, they cannot easily =
interoperate
with each other without operator assistance and expensive additional =
equipment
which translates from one vendor to another. A major factor limiting the
interoperability of telepresence systems&nbsp; is the lack of a =
standardized
way to describe and negotiate the use of the multiple streams of audio =
and
video comprising the media flows.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The WG will create specifications for SIP-based =
conferencing
systems to enable communication of&nbsp; information about media streams =
so
that a sending system,&nbsp; receiving system, or intermediate system =
can make
reasonable decisions about transmitting, selecting, and rendering media
streams. This enables systems to make choices that optimize <span
style=3D'color:black'>user</span> experience.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>This working group is chartered <span =
style=3D'color:black'>to
specify&nbsp; information about media</span> streams from one entity to =
another
entity: <o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>* Spatial relationships of cameras, displays, =
microphones,
and speakers &#8211; relative to each other and to likely positions of
participants<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>* Specific characteristics such as viewpoint, field =
of
view/capture for camera/microphone/display/speaker &#8211; so that =
senders and intermediate
devices can understand how best to compose streams for receivers, and =
the
receiver will know the characteristics of its received =
streams<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>* <a name=3D"OLE_LINK1"></a><a =
name=3D"OLE_LINK2">Usage of the
stream, for example whether the stream is presentation, </a>or document =
camera
output<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>* Aspect ratio of cameras and displays =
<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>*Which sources a receiver wants to receive.&nbsp; =
For
example, it might&nbsp; want the source for the left camera, or might =
want the
source chosen by VAD (Voice Activity Detection)<span =
style=3D'font-size:10.5pt;
font-family:Consolas'><o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Information between sources and sinks about media =
stream
capabilities will be exchanged.&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The working group will define the semantics, =
syntax, and
transport mechanism necessary for communicating the necessary =
information. It
will consider whether the existing signaling mechanisms (e. g., SDP) can =
be
extended, or another messaging method should be used. =
&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The scope of the work includes describing =
relatively static
relations between entities (participants and devices). It also includes
handling more dynamic relationships, such as identifying the audio and =
video
streams for the current speaker. The scope includes both systems that =
provide a
fully immersive experience, and systems that interwork with them and =
therefore
need to understand the same multiple stream semantics. =
&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The focus of this work is on multiple audio and =
video
streams. &nbsp;Other media types may be considered, however development =
of
methodologies for them is not within the scope of this =
work.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Interoperation with SIP and related standards for =
audio and
video is required. &nbsp;However, backwards compatibility with existing
non-standards compliant telepresence systems is not =
required.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>This working group is not currently chartered to =
work on
issues of continuous conference control including: far end camera =
control,
indication of fast frame update for video codecs or other rapid =
switches, floor
control, conference roster. The working group may identify =
interoperability
obstacles in existing open standards. If so, the WG will develop =
requirements
to be communicated to other IETF WGs or Standards Forums, or recharter =
as
appropriate.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Reuse of existing protocols and backwards =
compatibility with
SIP-compliant audio/video endpoints are important factors for the =
working group
to consider. The work will closely coordinate with the appropriate areas =
(e.
g., OPS, SEC), and working groups including&nbsp; AVT, MMUSIC, =
MEDIACTRL, XCON,
and SIPCORE.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&nbsp;Milestones &nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>July 2011 Submit informational draft to IESG on use =
cases <o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>July 2011 Submit informational draft to IESG on =
framework
and requirements<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Nov 2011 Submit standards track specification(s) to =
IESG to
support framework and requirements<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></p>

</div>

</body>

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Subject: Re: [dispatch] Asking comments for draft-yang-dispatch-sip-connection-address-type
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Lili,

I guess there is growing interest in involving a separate video device
in a call, because I have seen several different proposals for how to do it.

Regarding this draft - from a technical perspective, because it proposes
a change to SDP, it should probably be discussed in MMUSIC. But given
that it competes with other proposed mechanisms that are not entirely
SDP, perhaps dispatch is a good place for it.

I don't understand the premise of this draft - that the party sending
the offer doesn't know the address of the media stream - only the sip
URI, and that therefore the other party should use the sip address to
establish the media stream.

Some UA will presumably have to establish a separate sip dialog in order
to determine the media capabilities and address. The question is which
UA should it be. I see no advantage in delegating this responsibility
from the device that knows the SIP URI to the other unsuspecting party.
Doing so reduces the likelihood of a successful call by imposing
additional requirements on the other party.

I see no particular advantage to this approach over:
- Alice calls Anna's device sip:anna@example2.net
- Alice discovers Anna's device video address and codecs
- Alice combines that video information with her own
  audio information into an offer to Bob.
(I'm glossing over the details here. But its well understood 3pcc.)

In that way, Bob's device need have no particular capabilities other
than the ability to support video and some video codec in common with
Anna's device.

	Thanks,
	Paul


On 10/26/2010 8:04 AM, Lili Yang_lili wrote:
> Hello Paul,
> 
> I am Lili Yang from Huawei.
> 
> Georg Mayer and I wrote an I-D 
> (http://tools.ietf.org/html/draft-yang-dispatch-sip-connection-address-type) 
> , which is related to a current 3GPP Rel-10 feature, namely 
> Inter-UE-Transfer for cases when a so-called collaborative session is 
> established during a SIP session establishment.
> 
> So far we did not receive any feed-back from IETF list on the draft and 
> we would like to hear your point of view.
> 
> I will attend the upcoming IETF#79 Beijing meeting, and it¡¯s appreciated 
> if you have some time to discuss the draft during the meeting.
> 
> Looking forward to your feedback!
> 
> Thanks and best regards,
> 
> Lili
> 
> ------------------------------------------------------------------------
> 
> Lili Yang
> »ªÎª¼¼ÊõÓÐÏÞ¹«Ë¾Huawei Technologies Co., Ltd.
> Company_logo
> 
> Phone: +86-755-28422553
> Fax: +86-755-28420413
> Mobile:+86-13632632065
> Email: Lili.yang@huawei.com
> µØÖ·£ºÉîÛÚÊÐÁú¸ÚÇøÛàÌï»ªÎª»ùµØÓÊ±à£º518129
> Huawei Technologies Co., Ltd.
> Bantian, Longgang District,Shenzhen 518129, P.R.China
> http://www.huawei.com
> 
> ------------------------------------------------------------------------
> 
> ±¾ÓÊ¼þ¼°Æä¸½¼þº¬ÓÐ»ªÎª¹«Ë¾µÄ±£ÃÜÐÅÏ¢£¬½öÏÞÓÚ·¢ËÍ¸øÉÏÃæµØÖ·ÖÐÁÐ³öµÄ¸öÈË»ò 
> Èº×é¡£½û
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From magnus.westerlund@ericsson.com  Tue Oct 26 07:56:04 2010
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Allyn Romanow (allyn) skrev 2010-10-25 21:15:
> Okay, how about
> 
> This working group is not currently chartered to work on issues of continuous conference        control including: far end camera control, indication of fast frame update for video codecs or other rapid switches, floor control, conference roster. The working group may identify interoperability obstacles in existing open standards. If so, the WG will develop        requirements to be communicated to other IETF WGs or Standards Forums, or recharter as appropriate."

Yes, that sounds good.

Cheers

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
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Färögatan 6                | Mobile +46 73 0949079
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On 25 October 2010 23:45,  <Internet-Drafts@ietf.org> wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>
> =A0 =A0 =A0 =A0Title =A0 =A0 =A0 =A0 =A0 : Revision of the Binary Floor C=
ontrol Protocol (BFCP) for use over an unreliable transport
> =A0 =A0 =A0 =A0Author(s) =A0 =A0 =A0 : T. Kristensen, et al.
> =A0 =A0 =A0 =A0Filename =A0 =A0 =A0 =A0: draft-sandbakken-dispatch-bfcp-u=
dp-01.txt
> =A0 =A0 =A0 =A0Pages =A0 =A0 =A0 =A0 =A0 : 26
> =A0 =A0 =A0 =A0Date =A0 =A0 =A0 =A0 =A0 =A0: 2010-10-25
>
> This memo extends the Binary Floor Control Protocol (BFCP) for use
> over an unreliable transport. =A0It details a set of revisions to the
> protocol definition document and the specification of Session
> Description Protocol (SDP) format for BFCP streams.
>
> A URL for this Internet-Draft is:
  http://tools.ietf.org/html/draft-sandbakken-dispatch-bfcp-udp-01

We managed to submit this version before the IETF-79 great
wall surrounded the draft submission tool yesterday. This is
the basis for the planned UDP/BFCP presentation/discussion
at the Dispatch meeting in Beijing:
  http://tools.ietf.org/wg/dispatch/agenda

Changes from -00:
  http://tools.ietf.org/html/draft-sandbakken-dispatch-bfcp-udp-01#appendix=
-A

Most notable changes:
- Removed the - too verbose and not directly related - rationale/
  motivation text describing background and why other approaches
  where not chosen. Focus on the actual extensions.
- Not mandate ICE as a SHALL, but leave it as a non-mandatory
  way of solving the potential need for NAT/FW traversal.
- Emphasized that the reference to DTLS-SRTP are merely
  informational.
- Decision made to not increase the protocol version number as
  a result of this extension.

-- Tom

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I definitely support this work.  I've been involved lately in several =
instances where both SIP and XMPP were included on the client side.  Any =
work that can help make the user experience more positive would be a =
good thing.

Regards,
Dan

On Oct 19, 2010, at 4:16 PM, Peter Saint-Andre wrote:

> Earlier this year, some folks in the RAI area proposed an initiative =
to
> define a few small extensions to both SIP and XMPP that would make it
> easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on
> feedback provided on the DISPATCH list and received from the RAI ADs,
> I've taken the liberty of rewriting the proposed charter, in the hopes
> that fresh text will spur a more conclusive discussion. I'm mostly =
just
> trying to help the proponents put their best foot forward, so if folks
> here have more feedback I'd expect that people like Simo Veikkolainen
> and Emil Ivov will be able to engage in further discussion.
>=20
> /psa
>=20
> ###
>=20
> SIXPAC (SIP Integration with XMPP in Presence Aware Clients)
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Problem Statement
>=20
> Both the Session Initiation Protocol (SIP) and the Extensible =
Messaging
> and Presence Protocol (XMPP) are widely deployed technologies for
> real-time communication over the Internet.  In order to offer a =
complete
> suite of features as well as communication across multiple networks,
> several user-oriented software applications support both SIP and XMPP,
> and more software developers have expressed interest in building such
> "dual-stack" solutions.  Unfortunately, it is difficult to provide a
> good end-user experience in such applications because SIP and XMPP are
> not aware of each other.  For example:
>=20
> - XMPP presence does not include availability states related to a SIP
>  voice call or video call (e.g., "on the phone"), thus preventing an
>  a dual-stack endpoint from showing presence-based communication hints
>=20
> - There is no correlation between an XMPP IM session and a SIP voice
>  or video session, thus preventing a dual-stack endpoint from =
providing
>  integrated user interfaces and communications history
>=20
> - SIP accounts and XMPP accounts are not directly correlated in =
contact
>  lists or vCards (and not all deployed services support storage of =
such
>  information), thus preventing a dual-stack endpoint from knowing that
>  a contact has both SIP and XMPP capabilities
>=20
> Although some proprietary solutions exist to the foregoing problems, =
it
> would be preferable to define standardized solutions for the sake of
> improved interoperability.
>=20
> Objectives
>=20
> Because both SIP and XMPP are easily extended through new SIP headers
> and XMPP elements, it should be possible to provide tighter =
integration
> within dual-stack SIP/XMPP user agents to improve the user experience.
>=20
> Any such extensions should meet the following criteria:
>=20
> - Be completely optional and backwards-compatible for all endpoints
>=20
> - Work without changes to deployed infrastructure such as existing
>  SIP and XMPP servers, B2BUAs, firewalls, etc.
>=20
> The SIXPAC WG will define a small number of SIP and XMPP extensions to
> solve the following use cases in dual-stack endpoints:
>=20
> - Including SIP-based availability states in XMPP presence (limited to
>  basic presence and availability states only, not the full range of
>  PIDF extensions)
>=20
> - Correlating an XMPP IM session with a SIP voice/video session, and
>  vice-versa
>=20
> - Advertising a SIP account address over XMPP and an XMPP account
>  address over SIP
>=20
> Additional use cases are out of scope, including anything related to =
or
> requiring server integration, multiparty communication, SIP-based IM
> and presence, XMPP-based voice and video, file transfer, generalized
> service discovery and capabilities exchange, full protocol translation
> in communication gateways, shared credentials across both SIP and XMPP
> accounts, rich presence extensions for features such as geolocation,
> etc.  Although such topics are important and interesting, they are out
> of scope for this group.
>=20
> However, in addition to the protocol extensions explicitly mentioned
> above, the group may also define best practices related to the
> implementation and deployment of dual-stack SIP/XMPP endpoints,
> including topics such as user agent configuration.
>=20
> Deliverables
>=20
> - Use cases and protocol requirements
>=20
> - XMPP presence extension for SIP-based availability states
>=20
> - Media session correlation extensions for SIP and XMPP
>=20
> - Contact address advertisement extensions for SIP and XMPP
>=20
> - Best practices for implementation and deployment of dual-stack
>  endpoints
>=20
> Milestones
>=20
> Feb 2011  Submit use cases and protocol requirements document to
>          IESG (Informational)
>=20
> Oct 2011  Submit XMPP presence extension document to IESG
>          (Standards Track)
>=20
> Oct 2011  Submit media session correlation extensions document to IESG
>          (Standards Track)
>=20
> Oct 2011  Submit contact address advertisement extension document to
>          IESG (Standards Track)
>=20
> Oct 2011  Submit best practices document to IESG (Informational)
>=20
> ###
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

--=20
Dan York, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-407-455-5859    Skype: danyork =20

Join the Voxeo conversation:
Blogs: http://blogs.voxeo.com
Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
Facebook: http://www.facebook.com/voxeo


--Apple-Mail-1-596152662
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I =
definitely support this work. &nbsp;I've been involved lately in several =
instances where both SIP and XMPP were included on the client side. =
&nbsp;Any work that can help make the user experience more positive =
would be a good =
thing.<div><br></div><div>Regards,</div><div>Dan</div><div><br></div><div>=
<div><div>On Oct 19, 2010, at 4:16 PM, Peter Saint-Andre wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div>Earlier=
 this year, some folks in the RAI area proposed an initiative =
to<br>define a few small extensions to both SIP and XMPP that would make =
it<br>easier to develop and deploy dual-stack SIP+XMPP endpoints. Based =
on<br>feedback provided on the DISPATCH list and received from the RAI =
ADs,<br>I've taken the liberty of rewriting the proposed charter, in the =
hopes<br>that fresh text will spur a more conclusive discussion. I'm =
mostly just<br>trying to help the proponents put their best foot =
forward, so if folks<br>here have more feedback I'd expect that people =
like Simo Veikkolainen<br>and Emil Ivov will be able to engage in =
further discussion.<br><br>/psa<br><br>###<br><br>SIXPAC (SIP =
Integration with XMPP in Presence Aware =
Clients)<br>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D<br><br>Problem Statement<br><br>Both the Session Initiation =
Protocol (SIP) and the Extensible Messaging<br>and Presence Protocol =
(XMPP) are widely deployed technologies for<br>real-time communication =
over the Internet. &nbsp;In order to offer a complete<br>suite of =
features as well as communication across multiple networks,<br>several =
user-oriented software applications support both SIP and XMPP,<br>and =
more software developers have expressed interest in building =
such<br>"dual-stack" solutions. &nbsp;Unfortunately, it is difficult to =
provide a<br>good end-user experience in such applications because SIP =
and XMPP are<br>not aware of each other. &nbsp;For example:<br><br>- =
XMPP presence does not include availability states related to a SIP<br> =
&nbsp;voice call or video call (e.g., "on the phone"), thus preventing =
an<br> &nbsp;a dual-stack endpoint from showing presence-based =
communication hints<br><br>- There is no correlation between an XMPP IM =
session and a SIP voice<br> &nbsp;or video session, thus preventing a =
dual-stack endpoint from providing<br> &nbsp;integrated user interfaces =
and communications history<br><br>- SIP accounts and XMPP accounts are =
not directly correlated in contact<br> &nbsp;lists or vCards (and not =
all deployed services support storage of such<br> &nbsp;information), =
thus preventing a dual-stack endpoint from knowing that<br> &nbsp;a =
contact has both SIP and XMPP capabilities<br><br>Although some =
proprietary solutions exist to the foregoing problems, it<br>would be =
preferable to define standardized solutions for the sake of<br>improved =
interoperability.<br><br>Objectives<br><br>Because both SIP and XMPP are =
easily extended through new SIP headers<br>and XMPP elements, it should =
be possible to provide tighter integration<br>within dual-stack SIP/XMPP =
user agents to improve the user experience.<br><br>Any such extensions =
should meet the following criteria:<br><br>- Be completely optional and =
backwards-compatible for all endpoints<br><br>- Work without changes to =
deployed infrastructure such as existing<br> &nbsp;SIP and XMPP servers, =
B2BUAs, firewalls, etc.<br><br>The SIXPAC WG will define a small number =
of SIP and XMPP extensions to<br>solve the following use cases in =
dual-stack endpoints:<br><br>- Including SIP-based availability states =
in XMPP presence (limited to<br> &nbsp;basic presence and availability =
states only, not the full range of<br> &nbsp;PIDF extensions)<br><br>- =
Correlating an XMPP IM session with a SIP voice/video session, and<br> =
&nbsp;vice-versa<br><br>- Advertising a SIP account address over XMPP =
and an XMPP account<br> &nbsp;address over SIP<br><br>Additional use =
cases are out of scope, including anything related to or<br>requiring =
server integration, multiparty communication, SIP-based IM<br>and =
presence, XMPP-based voice and video, file transfer, =
generalized<br>service discovery and capabilities exchange, full =
protocol translation<br>in communication gateways, shared credentials =
across both SIP and XMPP<br>accounts, rich presence extensions for =
features such as geolocation,<br>etc. &nbsp;Although such topics are =
important and interesting, they are out<br>of scope for this =
group.<br><br>However, in addition to the protocol extensions explicitly =
mentioned<br>above, the group may also define best practices related to =
the<br>implementation and deployment of dual-stack SIP/XMPP =
endpoints,<br>including topics such as user agent =
configuration.<br><br>Deliverables<br><br>- Use cases and protocol =
requirements<br><br>- XMPP presence extension for SIP-based availability =
states<br><br>- Media session correlation extensions for SIP and =
XMPP<br><br>- Contact address advertisement extensions for SIP and =
XMPP<br><br>- Best practices for implementation and deployment of =
dual-stack<br> &nbsp;endpoints<br><br>Milestones<br><br>Feb 2011 =
&nbsp;Submit use cases and protocol requirements document to<br> =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;IESG =
(Informational)<br><br>Oct 2011 &nbsp;Submit XMPP presence extension =
document to IESG<br> =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Standards =
Track)<br><br>Oct 2011 &nbsp;Submit media session correlation extensions =
document to IESG<br> =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Standards =
Track)<br><br>Oct 2011 &nbsp;Submit contact address advertisement =
extension document to<br> =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;IESG (Standards =
Track)<br><br>Oct 2011 &nbsp;Submit best practices document to IESG =
(Informational)<br><br>###<br><br><br>____________________________________=
___________<br>dispatch mailing list<br><a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>https://www.iet=
f.org/mailman/listinfo/dispatch<br></div></blockquote></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
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0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
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class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
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auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
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-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
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-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
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auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
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after-white-space; "><span class=3D"Apple-style-span" =
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0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
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margin-left: 0px; "><span class=3D"Apple-style-span" style=3D"font-size: =
medium; "><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; =
">--&nbsp;</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; ">Dan York, =
Director of Conversations</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font-size: =
12px; ">Voxeo Corporation&nbsp;&nbsp;&nbsp;<a =
href=3D"http://www.voxeo.com">http://www.voxeo.com</a>&nbsp;&nbsp;<a =
href=3D"mailto:dyork@voxeo.com">dyork@voxeo.com</a></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; ">Phone: =
+1-407-455-5859&nbsp;&nbsp;&nbsp;&nbsp;Skype: =
danyork&nbsp;&nbsp;</div><div style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; font-size: 12px; =
"><br></div><div style=3D"margin-top: 0px; margin-right: 0px; =
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margin-left: 0px; "><b><font class=3D"Apple-style-span" =
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><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
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href=3D"http://twitter.com/danyork">http://twitter.com/danyork</a></font><=
/b></div><div style=3D"margin-top: 0px; margin-right: 0px; =
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font></b></div></div></span></div></span></div></span></div></span></div><=
/span></div></span></div></span></div></span></div></span></div></span></s=
pan>
</div>
<br></div></body></html>=

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Subject: Re: [dispatch] Asking comments for draft-yang-dispatch-sip-connection-address-type
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Lili - comments inline.

On 10/27/2010 3:46 AM, Lili Yang_lili wrote:
>
> Hello Paul,
>
> Thanks for your feedback!
>
> The premise of this draft is to fulfill the specific requirement in
> 3GPP. And it's a different case with the 3PCC as you mentioned.
>
> Concurrent with a normal session set-up towards a remote party, UE-1
> wants to establish a Collaborative Session, which is anchored at the
> SCC AS,
>

What is an SCC?

> with a new Media Flow-A on itself and a new Media Flow-B on another
> UE-2. Following is the information flow defined in 3GPP:
>

I'd like to see the intended o/a details for each of steps 1-6 above.

> And in step1, the ession setup request shall include enough
> information for the SCC AS to:
>
> - identify that Media-A shall be established in UE-1;
>
> - identify that Media-B shall be established in UE-2 and the requested
> media type associated with Media-B (e.g. video);
>
> Normally the UE-1 doesn¡¯t know the IP address of the UE-2 but only the
> SIP address, hence we propose UE-1 sending the offer with the UE-2¡¯s
> SIP address, to identify the requested media will be established in UE-2.
>

Having the above detail would allow a more informed discussion of the
possibilities.
(I don't have enough detail to know if this would work, but perhaps UE-1
could simply send a REFER to UE-2, with Refer-To of the SCC and a Join
header.)

I am much happier knowing that its the SCC, which is presumably a
resource under the control of UE-1, that is doing the heavy lifting,
rather than putting the burden on remote-party, which is an innocent
bystander.

Thanks,
Paul

> Hope this clarifies.
>
> Thanks and best regards,
>
> Lili
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, October 26, 2010 10:28 PM
> To: Lili Yang_lili
> Cc: Spencer Dawkins; 'Mayer Georg'; dispatch@ietf.org
> Subject: Re: Asking comments for
> draft-yang-dispatch-sip-connection-address-type
>
> Lili,
>
> I guess there is growing interest in involving a separate video device
>
> in a call, because I have seen several different proposals for how to
> do it.
>
> Regarding this draft - from a technical perspective, because it proposes
>
> a change to SDP, it should probably be discussed in MMUSIC. But given
>
> that it competes with other proposed mechanisms that are not entirely
>
> SDP, perhaps dispatch is a good place for it.
>
> I don't understand the premise of this draft - that the party sending
>
> the offer doesn't know the address of the media stream - only the sip
>
> URI, and that therefore the other party should use the sip address to
>
> establish the media stream.
>
> Some UA will presumably have to establish a separate sip dialog in order
>
> to determine the media capabilities and address. The question is which
>
> UA should it be. I see no advantage in delegating this responsibility
>
> from the device that knows the SIP URI to the other unsuspecting party.
>
> Doing so reduces the likelihood of a successful call by imposing
>
> additional requirements on the other party.
>
> I see no particular advantage to this approach over:
>
> - Alice calls Anna's device sip:anna@example2.net
>
> - Alice discovers Anna's device video address and codecs
>
> - Alice combines that video information with her own
>
> audio information into an offer to Bob.
>
> (I'm glossing over the details here. But its well understood 3pcc.)
>
> In that way, Bob's device need have no particular capabilities other
>
> than the ability to support video and some video codec in common with
>
> Anna's device.
>
> Thanks,
>
> Paul
>
> On 10/26/2010 8:04 AM, Lili Yang_lili wrote:
>
> > Hello Paul,
>
> >
>
> > I am Lili Yang from Huawei.
>
> >
>
> > Georg Mayer and I wrote an I-D
>
> > (http://tools.ietf.org/html/draft-yang-dispatch-sip-connection-address-type)
>
> > , which is related to a current 3GPP Rel-10 feature, namely
>
> > Inter-UE-Transfer for cases when a so-called collaborative session is
>
> > established during a SIP session establishment.
>
> >
>
> > So far we did not receive any feed-back from IETF list on the draft and
>
> > we would like to hear your point of view.
>
> >
>
> > I will attend the upcoming IETF#79 Beijing meeting, and it¡¯s appreciated
>
> > if you have some time to discuss the draft during the meeting.
>
> >
>
> > Looking forward to your feedback!
>
> >
>
> > Thanks and best regards,
>
> >
>
> > Lili
>
> >
>
> > ------------------------------------------------------------------------
>
> >
>
> > Lili Yang
>
> > »ªÎª¼¼ÊõÓÐÏÞ¹«Ë¾Huawei Technologies Co., Ltd.
>
> > Company_logo
>
> >
>
> > Phone: +86-755-28422553
>
> > Fax: +86-755-28420413
>
> > Mobile:+86-13632632065
>
> > Email: Lili.yang@huawei.com
>
> > µØÖ·£ºÉîÛÚÊÐÁú¸ÚÇøÛàÌï»ªÎª»ùµØÓÊ±à£º518129
>
> > Huawei Technologies Co., Ltd.
>
> > Bantian, Longgang District,Shenzhen 518129, P.R.China
>
> > http://www.huawei.com
>
> >
>
> > ------------------------------------------------------------------------
>
> >
>
> > ±¾ÓÊ¼þ¼°Æä¸½¼þº¬ÓÐ»ªÎª¹«Ë¾µÄ±£ÃÜÐÅÏ¢£¬½öÏÞÓÚ·¢ËÍ¸øÉÏÃæµØÖ·ÖÐÁÐ³öµÄ¸öÈË »ò
>
> > Èº×é¡£½û
>
> > Ö¹ÈÎºÎÆäËûÈËÒÔÈÎºÎÐÎÊ½Ê¹ÓÃ£¨°üÀ¨µ«²»ÏÞÓÚÈ«²¿»ò²¿·ÖµØÐ¹Â¶¡¢¸´ÖÆ¡¢»òÉ¢ ·¢£©
>
> > ±¾ÓÊ¼þÖÐ
>
> > µÄÐÅÏ¢¡£Èç¹ûÄú´íÊÕÁË±¾ÓÊ¼þ£¬ÇëÄúÁ¢¼´µç»°»òÓÊ¼þÍ¨Öª·¢¼þÈË²¢É¾³ý±¾ÓÊ¼þ£¡
>
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    Lili - comments inline.<br>
    <br>
    On 10/27/2010 3:46 AM, Lili Yang_lili wrote:
    <blockquote
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        <p class="MsoPlainText"><span lang="EN-US">Hello Paul,<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Thanks for your
            feedback!<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">The premise of this
            draft is to fulfill the specific requirement in 3GPP. And
            it's a different case with the 3PCC as you mentioned.
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Concurrent with a
            normal session set-up towards a remote party, UE-1 wants to
            establish a Collaborative Session, which is anchored at the
            SCC AS,</span></p>
      </div>
    </blockquote>
    <br>
    What is an SCC?<br>
    <br>
    <blockquote
cite="mid:EDD4D02EBE87354591C2DB9D0EA70CB60B1ACC4B@szxeml504-mbx.china.huawei.com"
      type="cite">
      <div class="Section1">
        <p class="MsoPlainText"><span lang="EN-US"> with a new Media
            Flow-A on itself and a new Media Flow-B on another UE-2.
            Following is the information flow defined in 3GPP:<o:p></o:p></span></p>
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        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
      </div>
    </blockquote>
    <br>
    I'd like to see the intended o/a details for each of steps 1-6
    above.<br>
    <br>
    <blockquote
cite="mid:EDD4D02EBE87354591C2DB9D0EA70CB60B1ACC4B@szxeml504-mbx.china.huawei.com"
      type="cite">
      <div class="Section1">
        <p class="MsoPlainText"><span lang="EN-US">And in step1, the </span><span
            lang="EN-GB">ession setup request shall include enough
            information for the SCC AS to:
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB">-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; identify
            that Media-A shall be established in UE-1;<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB">-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; identify
            that Media-B shall be established in UE-2 and the requested
            media type associated with Media-B (e.g. video);<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB">Normally the UE-1
            doesn</span><span style="font-family: &quot;Courier
            New&quot;;" lang="EN-GB">¡¯</span><span lang="EN-GB">t know
            the IP address of the UE-2 but only the SIP address, hence
            we propose UE-1 sending the offer with the UE-2</span><span
            style="font-family: &quot;Courier New&quot;;" lang="EN-GB">¡¯</span><span
            lang="EN-GB">s SIP address, to identify the requested media
            will be established in UE-2.
          </span></p>
      </div>
    </blockquote>
    <br>
    Having the above detail would allow a more informed discussion of
    the possibilities.<br>
    (I don't have enough detail to know if this would work, but perhaps
    UE-1 could simply send a REFER to UE-2, with Refer-To of the SCC and
    a Join header.)<br>
    <br>
    I am much happier knowing that its the SCC, which is presumably a
    resource under the control of UE-1, that is doing the heavy lifting,
    rather than putting the burden on remote-party, which is an innocent
    bystander.<br>
    <br>
    &nbsp;&nbsp;&nbsp; Thanks,<br>
    &nbsp;&nbsp;&nbsp; Paul<br>
    <br>
    <blockquote
cite="mid:EDD4D02EBE87354591C2DB9D0EA70CB60B1ACC4B@szxeml504-mbx.china.huawei.com"
      type="cite">
      <div class="Section1">
        <p class="MsoPlainText"><span lang="EN-GB"><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB">Hope this clarifies.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-GB">Thanks and best
            regards,</span><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Lili <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">-----Original
            Message-----<br>
            From: Paul Kyzivat [<a class="moz-txt-link-freetext" href="mailto:pkyzivat@cisco.com">mailto:pkyzivat@cisco.com</a>] <br>
            Sent: Tuesday, October 26, 2010 10:28 PM<br>
            To: Lili Yang_lili<br>
            Cc: Spencer Dawkins; 'Mayer Georg'; <a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
            Subject: Re: Asking comments for
            draft-yang-dispatch-sip-connection-address-type<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Lili,<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">I guess there is
            growing interest in involving a separate video device<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">in a call, because I
            have seen several different proposals for how to do it.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Regarding this draft
            - from a technical perspective, because it proposes<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">a change to SDP, it
            should probably be discussed in MMUSIC. But given<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">that it competes with
            other proposed mechanisms that are not entirely<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">SDP, perhaps dispatch
            is a good place for it.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">I don't understand
            the premise of this draft - that the party sending<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">the offer doesn't
            know the address of the media stream - only the sip<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">URI, and that
            therefore the other party should use the sip address to<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">establish the media
            stream.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Some UA will
            presumably have to establish a separate sip dialog in order<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">to determine the
            media capabilities and address. The question is which<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">UA should it be. I
            see no advantage in delegating this responsibility<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">from the device that
            knows the SIP URI to the other unsuspecting party.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Doing so reduces the
            likelihood of a successful call by imposing<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">additional
            requirements on the other party.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">I see no particular
            advantage to this approach over:<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">- Alice calls Anna's
            device <a class="moz-txt-link-abbreviated" href="mailto:sip:anna@example2.net">sip:anna@example2.net</a><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">- Alice discovers
            Anna's device video address and codecs<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">- Alice combines that
            video information with her own<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&nbsp; audio information
            into an offer to Bob.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">(I'm glossing over
            the details here. But its well understood 3pcc.)<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">In that way, Bob's
            device need have no particular capabilities other<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">than the ability to
            support video and some video codec in common with<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">Anna's device.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">On 10/26/2010 8:04
            AM, Lili Yang_lili wrote:<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Hello Paul,<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; I am Lili Yang
            from Huawei.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Georg Mayer and
            I wrote an I-D <o:p>
            </o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt;
(<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-yang-dispatch-sip-connection-address-type">http://tools.ietf.org/html/draft-yang-dispatch-sip-connection-address-type</a>)<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; , which is
            related to a current 3GPP Rel-10 feature, namely
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt;
            Inter-UE-Transfer for cases when a so-called collaborative
            session is
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; established
            during a SIP session establishment.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; So far we did
            not receive any feed-back from IETF list on the draft and
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; we would like to
            hear your point of view.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; I will attend
            the upcoming IETF#79 Beijing meeting, and it</span><span
            style="font-family: &quot;Courier New&quot;;" lang="EN-US">¡¯</span><span
            lang="EN-US">s appreciated
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; if you have some
            time to discuss the draft during the meeting.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Looking forward
            to your feedback!<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Thanks and best
            regards,<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Lili<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt;
            ------------------------------------------------------------------------<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Lili Yang<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; </span><span
            style="font-family: ËÎÌå;">»ªÎª¼¼ÊõÓÐÏÞ¹«Ë¾</span><span lang="EN-US">Huawei
            Technologies Co., Ltd.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Company_logo<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Phone:
            +86-755-28422553<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Fax:
            +86-755-28420413<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt;
            Mobile:+86-13632632065<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Email:
            <a class="moz-txt-link-abbreviated" href="mailto:Lili.yang@huawei.com">Lili.yang@huawei.com</a><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; </span><span
            style="font-family: ËÎÌå;">µØÖ·£ºÉîÛÚÊÐÁú¸ÚÇøÛàÌï»ªÎª»ùµØÓÊ±à£º</span><span
            lang="EN-US">518129<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Huawei
            Technologies Co., Ltd.<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Bantian,
            Longgang District,Shenzhen 518129, P.R.China<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt;
            <a class="moz-txt-link-freetext" href="http://www.huawei.com">http://www.huawei.com</a><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt;
            ------------------------------------------------------------------------<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; </span><span
            style="font-family: ËÎÌå;">±¾ÓÊ¼þ¼°Æä¸½¼þº¬ÓÐ»ªÎª¹«Ë¾µÄ±£ÃÜÐÅÏ¢£¬½öÏÞÓÚ·¢ËÍ¸øÉÏÃæµØÖ·ÖÐÁÐ³öµÄ¸öÈË
            »ò</span><span lang="EN-US">
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; </span><span
            style="font-family: ËÎÌå;">Èº×é¡£½û</span><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; </span><span
            style="font-family: ËÎÌå;">Ö¹ÈÎºÎÆäËûÈËÒÔÈÎºÎÐÎÊ½Ê¹ÓÃ£¨°üÀ¨µ«²»ÏÞÓÚÈ«²¿»ò²¿·ÖµØÐ¹Â¶¡¢¸´ÖÆ¡¢»òÉ¢
            ·¢£©</span><span lang="EN-US">
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; </span><span
            style="font-family: ËÎÌå;">±¾ÓÊ¼þÖÐ</span><span lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; </span><span
            style="font-family: ËÎÌå;">µÄÐÅÏ¢¡£Èç¹ûÄú´íÊÕÁË±¾ÓÊ¼þ£¬ÇëÄúÁ¢¼´µç»°»òÓÊ¼þÍ¨Öª·¢¼þÈË²¢É¾³ý±¾ÓÊ¼þ£¡</span><span
            lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; This e-mail and
            its attachments contain confidential information from
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; HUAWEI, which<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; is intended only
            for the person or entity whose address is listed above.
            <o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; Any use of the<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; information
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        <p class="MsoPlainText"><span lang="EN-US">&gt; total or partial<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; disclosure,
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        <p class="MsoPlainText"><span lang="EN-US">&gt; recipient(s) is
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        <p class="MsoPlainText"><span lang="EN-US">&gt; phone or email
            immediately and delete it!<o:p></o:p></span></p>
        <p class="MsoPlainText"><span lang="EN-US">&gt; <o:p></o:p></span></p>
      </div>
    </blockquote>
  </body>
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From Sebastian.Schumann@st.sk  Tue Oct 26 23:58:44 2010
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From: Schumann Sebastian <Sebastian.Schumann@st.sk>
To: Peter Saint-Andre <stpeter@stpeter.im>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 27 Oct 2010 09:00:24 +0200
Thread-Topic: [dispatch] proposed SIXPAC charter
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Dear all, Peter,
I think this initiative is touching some very actual problem, I second it a=
s well.

One question: Would you think that this interaction between SIP and XMPP wo=
uld also be targeted not only by dual-stack clients but also on the network=
 side?
That is, having a SIP based telephony infrastructure (w/o SIP presence) wit=
h certain users and a XMPP based IM/P network  (w/o Jingle) with the same u=
sers? Having a mechanism of mapping the users in both networks, XMPP presen=
ce could be enhanced with SIP based telephony states. Would you see this in=
 scope or rather out of scope of your proposal?

Thanks.
Sebastian

> -----Original Message-----
> From: dispatch-bounces@ietf.org
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre
> Sent: 19 October 2010 21:17
> To: dispatch@ietf.org
> Subject: [dispatch] proposed SIXPAC charter
>
> Earlier this year, some folks in the RAI area proposed an
> initiative to
> define a few small extensions to both SIP and XMPP that would make it
> easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on
> feedback provided on the DISPATCH list and received from the RAI ADs,
> I've taken the liberty of rewriting the proposed charter, in the hopes
> that fresh text will spur a more conclusive discussion. I'm
> mostly just
> trying to help the proponents put their best foot forward, so if folks
> here have more feedback I'd expect that people like Simo Veikkolainen
> and Emil Ivov will be able to engage in further discussion.
>
> /psa
>
> ###
>
> SIXPAC (SIP Integration with XMPP in Presence Aware Clients)
>
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
> Problem Statement
>
> Both the Session Initiation Protocol (SIP) and the Extensible
> Messaging
> and Presence Protocol (XMPP) are widely deployed technologies for
> real-time communication over the Internet.  In order to offer
> a complete
> suite of features as well as communication across multiple networks,
> several user-oriented software applications support both SIP and XMPP,
> and more software developers have expressed interest in building such
> "dual-stack" solutions.  Unfortunately, it is difficult to provide a
> good end-user experience in such applications because SIP and XMPP are
> not aware of each other.  For example:
>
> - XMPP presence does not include availability states related to a SIP
>   voice call or video call (e.g., "on the phone"), thus preventing an
>   a dual-stack endpoint from showing presence-based
> communication hints
>
> - There is no correlation between an XMPP IM session and a SIP voice
>   or video session, thus preventing a dual-stack endpoint
> from providing
>   integrated user interfaces and communications history
>
> - SIP accounts and XMPP accounts are not directly correlated
> in contact
>   lists or vCards (and not all deployed services support
> storage of such
>   information), thus preventing a dual-stack endpoint from
> knowing that
>   a contact has both SIP and XMPP capabilities
>
> Although some proprietary solutions exist to the foregoing
> problems, it
> would be preferable to define standardized solutions for the sake of
> improved interoperability.
>
> Objectives
>
> Because both SIP and XMPP are easily extended through new SIP headers
> and XMPP elements, it should be possible to provide tighter
> integration
> within dual-stack SIP/XMPP user agents to improve the user experience.
>
> Any such extensions should meet the following criteria:
>
> - Be completely optional and backwards-compatible for all endpoints
>
> - Work without changes to deployed infrastructure such as existing
>   SIP and XMPP servers, B2BUAs, firewalls, etc.
>
> The SIXPAC WG will define a small number of SIP and XMPP extensions to
> solve the following use cases in dual-stack endpoints:
>
> - Including SIP-based availability states in XMPP presence (limited to
>   basic presence and availability states only, not the full range of
>   PIDF extensions)
>
> - Correlating an XMPP IM session with a SIP voice/video session, and
>   vice-versa
>
> - Advertising a SIP account address over XMPP and an XMPP account
>   address over SIP
>
> Additional use cases are out of scope, including anything
> related to or
> requiring server integration, multiparty communication, SIP-based IM
> and presence, XMPP-based voice and video, file transfer, generalized
> service discovery and capabilities exchange, full protocol translation
> in communication gateways, shared credentials across both SIP and XMPP
> accounts, rich presence extensions for features such as geolocation,
> etc.  Although such topics are important and interesting, they are out
> of scope for this group.
>
> However, in addition to the protocol extensions explicitly mentioned
> above, the group may also define best practices related to the
> implementation and deployment of dual-stack SIP/XMPP endpoints,
> including topics such as user agent configuration.
>
> Deliverables
>
> - Use cases and protocol requirements
>
> - XMPP presence extension for SIP-based availability states
>
> - Media session correlation extensions for SIP and XMPP
>
> - Contact address advertisement extensions for SIP and XMPP
>
> - Best practices for implementation and deployment of dual-stack
>   endpoints
>
> Milestones
>
> Feb 2011  Submit use cases and protocol requirements document to
>           IESG (Informational)
>
> Oct 2011  Submit XMPP presence extension document to IESG
>           (Standards Track)
>
> Oct 2011  Submit media session correlation extensions document to IESG
>           (Standards Track)
>
> Oct 2011  Submit contact address advertisement extension document to
>           IESG (Standards Track)
>
> Oct 2011  Submit best practices document to IESG (Informational)
>
> ###
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From emcho@sip-communicator.org  Wed Oct 27 10:20:43 2010
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] proposed SIXPAC charter
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Hey Sebastian,

=D0=9D=D0=B0 27.10.10 10:00, Schumann Sebastian =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> Dear all, Peter, I think this initiative is touching some very actual
> problem, I second it as well.
>=20
> One question: Would you think that this interaction between SIP and
> XMPP would also be targeted not only by dual-stack clients but also
> on the network side? That is, having a SIP based telephony
> infrastructure (w/o SIP presence) with certain users and a XMPP based
> IM/P network  (w/o Jingle) with the same users? Having a mechanism of
> mapping the users in both networks, XMPP presence could be enhanced
> with SIP based telephony states. Would you see this in scope or
> rather out of scope of your proposal?

As it is the charter states:

> - Including SIP-based availability states in XMPP presence (limited
> to basic presence and availability states only, not the full range
> of PIDF extensions)

Is this what you mean or did I miss something?

Cheers,
Emil


>=20
> Thanks. Sebastian
>=20
>> -----Original Message----- From: dispatch-bounces@ietf.org=20
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre=20
>> Sent: 19 October 2010 21:17 To: dispatch@ietf.org Subject:
>> [dispatch] proposed SIXPAC charter
>>=20
>> Earlier this year, some folks in the RAI area proposed an=20
>> initiative to define a few small extensions to both SIP and XMPP
>> that would make it easier to develop and deploy dual-stack SIP+XMPP
>> endpoints. Based on feedback provided on the DISPATCH list and
>> received from the RAI ADs, I've taken the liberty of rewriting the
>> proposed charter, in the hopes that fresh text will spur a more
>> conclusive discussion. I'm mostly just trying to help the
>> proponents put their best foot forward, so if folks here have more
>> feedback I'd expect that people like Simo Veikkolainen and Emil
>> Ivov will be able to engage in further discussion.
>>=20
>> /psa
>>=20
>> ###
>>=20
>> SIXPAC (SIP Integration with XMPP in Presence Aware Clients)
>>=20
>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>=20
>> Problem Statement
>>=20
>> Both the Session Initiation Protocol (SIP) and the Extensible=20
>> Messaging and Presence Protocol (XMPP) are widely deployed
>> technologies for real-time communication over the Internet.  In
>> order to offer a complete suite of features as well as
>> communication across multiple networks, several user-oriented
>> software applications support both SIP and XMPP, and more software
>> developers have expressed interest in building such "dual-stack"
>> solutions.  Unfortunately, it is difficult to provide a good
>> end-user experience in such applications because SIP and XMPP are=20
>> not aware of each other.  For example:
>>=20
>> - XMPP presence does not include availability states related to a
>> SIP voice call or video call (e.g., "on the phone"), thus
>> preventing an a dual-stack endpoint from showing presence-based=20
>> communication hints
>>=20
>> - There is no correlation between an XMPP IM session and a SIP
>> voice or video session, thus preventing a dual-stack endpoint from
>> providing integrated user interfaces and communications history
>>=20
>> - SIP accounts and XMPP accounts are not directly correlated in
>> contact lists or vCards (and not all deployed services support=20
>> storage of such information), thus preventing a dual-stack endpoint
>> from knowing that a contact has both SIP and XMPP capabilities
>>=20
>> Although some proprietary solutions exist to the foregoing=20
>> problems, it would be preferable to define standardized solutions
>> for the sake of improved interoperability.
>>=20
>> Objectives
>>=20
>> Because both SIP and XMPP are easily extended through new SIP
>> headers and XMPP elements, it should be possible to provide
>> tighter integration within dual-stack SIP/XMPP user agents to
>> improve the user experience.
>>=20
>> Any such extensions should meet the following criteria:
>>=20
>> - Be completely optional and backwards-compatible for all
>> endpoints
>>=20
>> - Work without changes to deployed infrastructure such as existing=20
>> SIP and XMPP servers, B2BUAs, firewalls, etc.
>>=20
>> The SIXPAC WG will define a small number of SIP and XMPP extensions
>> to solve the following use cases in dual-stack endpoints:
>>=20
>> - Including SIP-based availability states in XMPP presence (limited
>> to basic presence and availability states only, not the full range
>> of PIDF extensions)
>>=20
>> - Correlating an XMPP IM session with a SIP voice/video session,
>> and vice-versa
>>=20
>> - Advertising a SIP account address over XMPP and an XMPP account=20
>> address over SIP
>>=20
>> Additional use cases are out of scope, including anything related
>> to or requiring server integration, multiparty communication,
>> SIP-based IM and presence, XMPP-based voice and video, file
>> transfer, generalized service discovery and capabilities exchange,
>> full protocol translation in communication gateways, shared
>> credentials across both SIP and XMPP accounts, rich presence
>> extensions for features such as geolocation, etc.  Although such
>> topics are important and interesting, they are out of scope for
>> this group.
>>=20
>> However, in addition to the protocol extensions explicitly
>> mentioned above, the group may also define best practices related
>> to the implementation and deployment of dual-stack SIP/XMPP
>> endpoints, including topics such as user agent configuration.
>>=20
>> Deliverables
>>=20
>> - Use cases and protocol requirements
>>=20
>> - XMPP presence extension for SIP-based availability states
>>=20
>> - Media session correlation extensions for SIP and XMPP
>>=20
>> - Contact address advertisement extensions for SIP and XMPP
>>=20
>> - Best practices for implementation and deployment of dual-stack=20
>> endpoints
>>=20
>> Milestones
>>=20
>> Feb 2011  Submit use cases and protocol requirements document to=20
>> IESG (Informational)
>>=20
>> Oct 2011  Submit XMPP presence extension document to IESG=20
>> (Standards Track)
>>=20
>> Oct 2011  Submit media session correlation extensions document to
>> IESG (Standards Track)
>>=20
>> Oct 2011  Submit contact address advertisement extension document
>> to IESG (Standards Track)
>>=20
>> Oct 2011  Submit best practices document to IESG (Informational)
>>=20
>> ###
>>=20
>> _______________________________________________ dispatch mailing
>> list dispatch@ietf.org=20
>> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________ dispatch mailing
> list dispatch@ietf.org=20
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

--=20
Emil Ivov, Ph.D.                               67000 Strasbourg,
Project Lead                                   France
SIP Communicator
emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
http://sip-communicator.org                    FAX:   +33.1.77.62.47.31


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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
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On Oct 24, 2010, at 11:40 PM, Christer Holmberg wrote:

> Hi,
>=20
>> Uh, same as Paul, sorry, ignore my previous email. Don't know how I =
missed this on first search. Thank you for sending the pointer to this.
>>=20
>> What is the sate of this 3GPP doc and what is going on with the =
registration?
>=20
> The document is stable. A reason the package hasn't been registered =
yet might be that people assume the info-event draft has to become RFC =
first.
>=20
> But, if that is not the case, I can inform people that the package can =
be registered.
>=20
> Regards,
>=20
> Christer

My understanding of the process is that the IANA registration for this =
could be done as soon as IESG had approved info-events. And that has =
already happened so I think this  could start any time.=20

You can see that IANA has already created the registry at=20
http://www.iana.org/assignments/sip-parameters



From christer.holmberg@ericsson.com  Wed Oct 27 22:28:07 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@cisco.com>
Date: Thu, 28 Oct 2010 07:29:53 +0200
Thread-Topic: [dispatch] I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-info-dtmf-package-00.txt
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Hi,=20

>>>Uh, same as Paul, sorry, ignore my previous email. Don't=20
>>>know how I missed this on first search. Thank you for sending=20
>>>the pointer to this.
>>>=20
>>>What is the sate of this 3GPP doc and what is going on=20
>>>with the registration?
>>=20
>>The document is stable. A reason the package hasn't been=20
>>registered yet might be that people assume the info-event=20
>>draft has to become RFC first.
>>=20
>>But, if that is not the case, I can inform people that the=20
>>package can be registered.
>=20
>My understanding of the process is that the IANA registration=20
>for this could be done as soon as IESG had approved=20
>info-events. And that has already happened so I think this =20
>could start any time.=20
>=20
>You can see that IANA has already created the registry at=20
>http://www.iana.org/assignments/sip-parameters

Yes. I expect the registration to take place after the next 3GPP CT1 meetin=
g (which is held the week after IETF).

Regards,

Christer

From lili.yang@huawei.com  Thu Oct 28 00:42:08 2010
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From melodysong@huawei.com  Thu Oct 28 22:58:42 2010
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Date: Fri, 29 Oct 2010 14:00:18 +0800
From: Song Haibin <melodysong@huawei.com>
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Subject: [dispatch] Is there a BOF on telepresence at Beijing meeting?
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Thanks,
Haibin



From stephen.botzko@gmail.com  Fri Oct 29 04:30:48 2010
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From: Stephen Botzko <stephen.botzko@gmail.com>
To: Song Haibin <melodysong@huawei.com>
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Subject: Re: [dispatch] Is there a BOF on telepresence at Beijing meeting?
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There is no BOF, but it is on the agenda for the Dispatch session.

Stephen Botzko

On Fri, Oct 29, 2010 at 2:00 AM, Song Haibin <melodysong@huawei.com> wrote:

> Thanks,
> Haibin
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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There is no BOF, but it is on the agenda for the Dispatch session. <br><br>=
Stephen Botzko<br><br><div class=3D"gmail_quote">On Fri, Oct 29, 2010 at 2:=
00 AM, Song Haibin <span dir=3D"ltr">&lt;<a href=3D"mailto:melodysong@huawe=
i.com">melodysong@huawei.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Thanks,<br>
Haibin<br>
<br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div><br>

--00504502e44cfb344b0493bfd095--

From mary.ietf.barnes@gmail.com  Fri Oct 29 06:46:50 2010
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Subject: [dispatch] IETF-79 Agenda for DISPATCH
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Hi folks,

FYI, an updated version of the DISPATCH WG agenda (links updated to
reflect most recent charters) has been uploaded to the meeting
materials:
http://www.ietf.org/proceedings/79/agenda/dispatch.html

As a reminder, the wiki also contains a summary of the work items that
are the focus for IETF-79 (as well as a summary of all previous
meetings):
http://trac.tools.ietf.org/wg/dispatch/trac/wiki

Regards,
Mary.

From HKaplan@acmepacket.com  Fri Oct 29 09:50:35 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Date: Fri, 29 Oct 2010 12:52:06 -0400
Thread-Topic: [dispatch] IETF-79 Agenda for DISPATCH
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I don't think I'll need 45 min for DTMF - maybe 30 mins tops.  So teleprese=
nce can have another 15.
And I'd put mine after telepresence since it's a charter discussion, so the=
y can run over.

I expect the result of mine will be to go the individual submission path an=
yway, since DTMF is too contentious a topic for most folks.  But at least t=
he presentation debate should be lively and entertaining. :)

-hadriel

On Oct 29, 2010, at 9:48 AM, Mary Barnes wrote:

> Hi folks,
>=20
> FYI, an updated version of the DISPATCH WG agenda (links updated to
> reflect most recent charters) has been uploaded to the meeting
> materials:
> http://www.ietf.org/proceedings/79/agenda/dispatch.html
>=20
> As a reminder, the wiki also contains a summary of the work items that
> are the focus for IETF-79 (as well as a summary of all previous
> meetings):
> http://trac.tools.ietf.org/wg/dispatch/trac/wiki
>=20
> Regards,
> Mary.
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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A revised version of draft-allen-dispatch-imei-urn-as-instanceid was
uploaded earlier this week and can be found at

 

http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-01.tx
t

 

Please provide any feedback.

 

Thanks

Andrew Allen


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<p class=3DMsoNormal>A revised version of draft-allen-dispatch-imei-urn-as-i=
nstanceid
was uploaded earlier this week and can be found at<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><a
href=3D"http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-0=
1.txt">http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-01=
.txt</a><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Please provide any feedback.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Andrew Allen<o:p></o:p></p>

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On 10/29/2010 12:52 PM, Hadriel Kaplan wrote:

> ...
> since DTMF is too contentious a topic for most folks.
> But at least the presentation debate should be lively and entertaining. :)

Yes, no doubt!

But then *all* of Hadriel's presentations seem to be lively and 
entertaining.

	Thanks,
	Paul

From pkyzivat@cisco.com  Fri Oct 29 10:31:13 2010
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Subject: Re: [dispatch] Revised draft draft-allen-dispatch-imei-urn-as-instanceid
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Andrew,

This revision resolves my concerns.
I'm happy with this draft.

	Thanks,
	Paul

On 10/29/2010 1:00 PM, Andrew Allen wrote:
> A revised version of draft-allen-dispatch-imei-urn-as-instanceid was
> uploaded earlier this week and can be found at
>
> http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-01.txt
>
> Please provide any feedback.
>
> Thanks
>
> Andrew Allen
>
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From HKaplan@acmepacket.com  Fri Oct 29 12:49:10 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Andrew Allen <aallen@rim.com>
Date: Fri, 29 Oct 2010 15:50:49 -0400
Thread-Topic: [dispatch] Revised draft draft-allen-dispatch-imei-urn-as-instanceid
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I've read the draft and think it's both clear and useful.  The only substan=
tive question I have is why is it a separate document from the montemurro d=
raft?  Why don't you just incorporate both the format and the instance-id u=
sage in one draft?  The single draft can state the URN format is for genera=
l use, while also specifying the instance-id usage, no?

Regardless of whether it's one or two drafts, I believe it/they should be D=
ISPATCHed as an A-D sponsored document.
I don't think it falls within the scope of any current WG, nor does it need=
 a min-WG created for it.

-hadriel

On Oct 29, 2010, at 1:00 PM, Andrew Allen wrote:

A revised version of draft-allen-dispatch-imei-urn-as-instanceid was upload=
ed earlier this week and can be found at

http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-01.txt

Please provide any feedback.

Thanks
Andrew Allen
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<html><head><base href=3D"x-msg://1022/"></head><body style=3D"word-wrap: b=
reak-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;=
 "><div><br></div><div>I've read the draft and think it's both clear and us=
eful. &nbsp;The only substantive question I have is why is it a separate do=
cument from the montemurro draft? &nbsp;Why don't you just incorporate both=
 the format and the instance-id usage in one draft? &nbsp;The single draft =
can state the URN format is for general use, while also specifying the inst=
ance-id usage, no?</div><div><br></div><div>Regardless of whether it's one =
or two drafts, I believe it/they should be DISPATCHed as an A-D sponsored d=
ocument. &nbsp;</div><div>I don't think it falls within the scope of any cu=
rrent WG, nor does it need a min-WG created for it.</div><div><br></div><di=
v>-hadriel</div><br><div><div>On Oct 29, 2010, at 1:00 PM, Andrew Allen wro=
te:</div><br class=3D"Apple-interchange-newline"><blockquote type=3D"cite">=
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; font-f=
amily: Helvetica; font-style: normal; font-variant: normal; font-weight: no=
rmal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent:=
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px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing=
: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div lang=3D"EN-=
US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" style=3D"pag=
e: WordSection1; "><div style=3D"margin-top: 0in; margin-right: 0in; margin=
-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri,=
 sans-serif; ">A revised version of draft-allen-dispatch-imei-urn-as-instan=
ceid was uploaded earlier this week and can be found at<o:p></o:p></div><di=
v style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; mar=
gin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p>&n=
bsp;</o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; margin-bo=
ttom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sa=
ns-serif; "><a href=3D"http://www.ietf.org/id/draft-allen-dispatch-imei-urn=
-as-instanceid-01.txt" style=3D"color: blue; text-decoration: underline; ">=
http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-01.txt</=
a><o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; margin=
-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri,=
 sans-serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0in; margin=
-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; fo=
nt-family: Calibri, sans-serif; ">Please provide any feedback.<o:p></o:p></=
div><div style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001=
pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">=
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rgin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Cali=
bri, sans-serif; ">Thanks<o:p></o:p></div><div style=3D"margin-top: 0in; ma=
rgin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt=
; font-family: Calibri, sans-serif; ">Andrew Allen<o:p></o:p></div></div>--=
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authorized and may be unlawful.<span class=3D"Apple-converted-space">&nbsp;=
</span><span>&lt;ATT00001..c&gt;</span></div></span></blockquote></div><br>=
</body></html>=

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Hadriel

 

It was previously included in the 04 version of the Montemurro draft but
the RAI AD thought that this ought to be in an RAI draft and not in the
URN definition draft. 

 

I am in favor of this being an AD sponsored document.

 

Andrew

 

From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
Sent: Friday, October 29, 2010 2:51 PM
To: Andrew Allen
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Revised draft
draft-allen-dispatch-imei-urn-as-instanceid

 

 

I've read the draft and think it's both clear and useful.  The only
substantive question I have is why is it a separate document from the
montemurro draft?  Why don't you just incorporate both the format and
the instance-id usage in one draft?  The single draft can state the URN
format is for general use, while also specifying the instance-id usage,
no?

 

Regardless of whether it's one or two drafts, I believe it/they should
be DISPATCHed as an A-D sponsored document.  

I don't think it falls within the scope of any current WG, nor does it
need a min-WG created for it.

 

-hadriel

 

On Oct 29, 2010, at 1:00 PM, Andrew Allen wrote:





A revised version of draft-allen-dispatch-imei-urn-as-instanceid was
uploaded earlier this week and can be found at

 

http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-01.tx
t

 

Please provide any feedback.

 

Thanks

Andrew Allen

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sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif";
color:#1F497D'>It was previously included in the 04 version of the Montemurr=
o draft
but the RAI AD thought that this ought to be in an RAI draft and not in the=
 URN
definition draft. <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif";
color:#1F497D'>I am in favor of this being an AD sponsored document.<o:p></o=
:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif";
color:#1F497D'>Andrew<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0=
in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma"=
,"sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Hadriel Kaplan
[mailto:HKaplan@acmepacket.com] <br>
<b>Sent:</b> Friday, October 29, 2010 2:51 PM<br>
<b>To:</b> Andrew Allen<br>
<b>Cc:</b> dispatch@ietf.org<br>
<b>Subject:</b> Re: [dispatch] Revised draft
draft-allen-dispatch-imei-urn-as-instanceid<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>I've read the draft and think it's both clear and usefu=
l.
&nbsp;The only substantive question I have is why is it a separate document
from the montemurro draft? &nbsp;Why don't you just incorporate both the for=
mat
and the instance-id usage in one draft? &nbsp;The single draft can state the
URN format is for general use, while also specifying the instance-id usage,=
 no?<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Regardless of whether it's one or two drafts, I believe
it/they should be DISPATCHed as an A-D sponsored document. &nbsp;<o:p></o:p>=
</p>

</div>

<div>

<p class=3DMsoNormal>I don't think it falls within the scope of any current=
 WG,
nor does it need a min-WG created for it.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>-hadriel<o:p></o:p></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<div>

<p class=3DMsoNormal>On Oct 29, 2010, at 1:00 PM, Andrew Allen wrote:<o:p></=
o:p></p>

</div>

<p class=3DMsoNormal><br>
<br>
<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'>A
revised version of draft-allen-dispatch-imei-urn-as-instanceid was uploaded
earlier this week and can be found at<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'><a
href=3D"http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-0=
1.txt">http://www.ietf.org/id/draft-allen-dispatch-imei-urn-as-instanceid-01=
.txt</a><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'>Please
provide any feedback.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'>Thanks<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri","=
sans-serif"'>Andrew
Allen<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span style=3D'font-size:13.5pt;font-family:"Helvetica"=
,"sans-serif"'>-------------------------------------------------------------=
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