
From R.Jesske@telekom.de  Thu Jul  2 05:41:55 2009
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Dear all,
At the moment I'm preparing a new version of =
draft-jesske-sipping-etsi-ngn-reason-04 ( =
http://tools.ietf.org/html/draft-jesske-sipping-etsi-ngn-reason-04 ) to =
be submitted.
There were some comments which were stating that this is the wrong group =
for the draft.
I would like to know what the feeling of the group is to which group =
(BLISS, DISPATCH or SIPCORE) this draft should belong.

Thank you and Best Regards

Roland

Deutsche Telekom Netzproduktion GmbH=20
Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE32-1
Heinrich-Hertz-Str. 3-7, 64295 Darmstadt,=20
Postfach, 64307 Darmstadt (Postanschrift)=20
+496151 628 2766 (Tel)
+491718618445 (Mobile)=20
E-Mail: r.jesske@telekom.de=20
http://www.telekom.de <http://www.telekom.de/> =20



Registerrechtliche Unternehmensangaben:=20
Deutsche Telekom Netzproduktion GmbH=20
Aufsichtsrat: Timotheus H=F6ttges (Vorsitzender)=20
Gesch=E4ftsf=FChrung: Friedrich Fu=DF (Vorsitzender), Albert Matheis, =
Klaus Peren=20
Handelsregister: Amtsgericht Bonn HRB 14190=20
Sitz der Gesellschaft: Bonn=20
USt-IdNr.: DE 814645262



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<P><FONT SIZE=3D2 FACE=3D"Arial">Dear all,</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Arial">At the moment I'm preparing a new =
version of draft-jesske-sipping-etsi-ngn-reason-04 ( </FONT><A =
HREF=3D"http://tools.ietf.org/html/draft-jesske-sipping-etsi-ngn-reason-0=
4"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://tools.ietf.org/html/draft-jesske-sipping-etsi-ngn-r=
eason-04</FONT></U></A><FONT SIZE=3D2 FACE=3D"Arial"> ) to be =
submitted.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">There were some comments which were =
stating that this is the wrong group for the draft.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Arial">I would like to know what the feeling =
of the group is to which group (BLISS, DISPATCH or SIPCORE) this draft =
should belong.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thank you and Best Regards</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Roland</FONT>
</P>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Deutsche Telekom =
Netzproduktion GmbH<BR>
Zentrum Technik Einf=FChrung</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Roland Jesske</FONT><FONT FACE=3D"Times =
New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">Gateways, Protokolle, Konvergenz, =
TE32-1</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Heinrich-Hertz-Str. 3-7, 64295 =
Darmstadt,</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Postfach, 64307 Darmstadt =
(Postanschrift)</FONT><FONT FACE=3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">+496151 628 2766 (Tel)</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">+491718618445 (Mobile)</FONT><FONT =
FACE=3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">E-Mail: </FONT><A =
HREF=3D"mailto:r.jesske@telekom.de"><U><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Arial">r.jesske@telekom.de</FONT></U></A><FONT SIZE=3D2 =
FACE=3D"Arial"></FONT>=20

<BR><A HREF=3D"http://www.telekom.de/"><U></U><U></U><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://www.telekom.de</FONT></U></A><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"></FONT>=20
</P>
<BR>
<BR>

<P><U><FONT SIZE=3D2 FACE=3D"Arial">Registerrechtliche =
Unternehmensangaben:</FONT></U><FONT SIZE=3D2 FACE=3D"Arial"></FONT>=20

<BR><FONT SIZE=3D2 FACE=3D"Arial">Deutsche Telekom Netzproduktion =
GmbH<BR>
Aufsichtsrat: Timotheus H=F6ttges (Vorsitzender)<BR>
Gesch=E4ftsf=FChrung: Friedrich Fu=DF (Vorsitzender), Albert Matheis, =
Klaus Peren<BR>
Handelsregister: Amtsgericht Bonn HRB 14190<BR>
Sitz der Gesellschaft: Bonn<BR>
USt-IdNr.: DE 814645262</FONT>
</P>
<BR>

</BODY>
</HTML>
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From alan@sipstation.com  Thu Jul  2 14:06:18 2009
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Subject: [dispatch] Fwd: New Version Notification for draft-johnston-dispatch-sip-cc-uui-00
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This draft contains the requirements analysis, scenarios, and call flows 
from draft-johnston-sipping-cc-uui-07, as it was Last Called in the 
SIPPING WG.

I will be submitting proposed charter text for a mechanism to meet this 
requirements to the DISPATCH list after July 13 for discussion prior to 
IETF-75.

Discussion is most welcome.

Thanks,
Alan

-------- Original Message --------
Subject: 	New Version Notification for 
draft-johnston-dispatch-sip-cc-uui-00
Date: 	Thu, 2 Jul 2009 13:55:28 -0700 (PDT)
From: 	IETF I-D Submission Tool <idsubmission@ietf.org>
To: 	alan@sipstation.com
CC: 	joanne@avaya.com



A new version of I-D, draft-johnston-dispatch-sip-cc-uui-00.txt has been successfuly submitted by Alan Johnston and posted to the IETF repository.

Filename:	 draft-johnston-dispatch-sip-cc-uui
Revision:	 00
Title:		 Requirements for Transporting User to User Call Control Information in SIP for ISDN Interworking
Creation_date:	 2009-07-02
WG ID:		 Independent Submission
Number_of_pages: 11

Abstract:
Several approaches to transporting the ITU-T Q.931 User to User
Information Element (UU IE) data in SIP have been proposed.  As
networks move to SIP it is important that applications requiring this
data can continue to function in SIP networks as well as the ability
to interwork with this ISDN service for end-to-end transparency.
This document discusses requirements and approaches.  This extension
will also be used for native SIP endpoints implementing similar
services and interworking with ISDN services.  Example use cases
include an exchange between two user agents, retargeting by a proxy,
and redirection.  An example application is an Automatic Call
Distributor (ACD) in a contact center.
                                                                                  


The IETF Secretariat.





From alan@sipstation.com  Thu Jul  2 14:09:53 2009
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This revision of the document references the requirements, use cases, 
and call flows in draft-johnston-dispatch-sip-cc-uui and proposes a new 
SIP header field User-to-User. 

New to this mechanism are the "purpose" and "content" header field 
parameters which help tag and identify the UUI information carried in 
the header field.

Discussion of this mechanism, which the authors feel best meets these 
requirements, is most welcome.

Thanks,
Alan

-------- Original Message --------
Subject: 	New Version Notification for draft-johnston-sipping-cc-uui-08
Date: 	Thu, 2 Jul 2009 14:00:35 -0700 (PDT)
From: 	IETF I-D Submission Tool <idsubmission@ietf.org>
To: 	alan@sipstation.com
CC: 	joanne@avaya.com



A new version of I-D, draft-johnston-sipping-cc-uui-08.txt has been successfuly submitted by Alan Johnston and posted to the IETF repository.

Filename:	 draft-johnston-sipping-cc-uui
Revision:	 08
Title:		 Transporting User to User Call Control Information in SIP for ISDN Interworking
Creation_date:	 2009-07-02
WG ID:		 Independent Submission
Number_of_pages: 12

Abstract:
Several approaches to transporting the ITU-T Q.931 User to User
Information Element (UU IE) data in SIP have been proposed.  As
networks move to SIP it is important that applications requiring this
data can continue to function in SIP networks as well as the ability
to interwork with this ISDN service for end-to- end transparency.
This document discusses three mechanisms to meet the requirements
defined in the Requirements for SIP Call Control UUI document.  A new
SIP header field which bests meets these requirements is proposed.
                                                                                  


The IETF Secretariat.





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Subject: Re: [dispatch] I-D Action:draft-johnston-sipping-cc-uui-08.txt
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Hi Alan,

I took a quick look at this doc and the requirements doc as well. I have 
a few thoughts:

 From section 1:

>    In the future, where both endpoints are intelligent SIP user agents,
>    it may be possible for them to understand and interpret the UUI data.
>    There may be some cases where the UUI information is relevant to SIP.
>    In this case, it might be worthwhile attempting to map UUI data to an
>    appropriate SIP header field or to standardize a new header field.
>    However, the requirements and use cases for this are different enough
>    from those described in this document that these two situations
>    should be examined separately.  This document looks only at the
>    requirements and mechanisms for replicating the existing, widely used
>    and deployed ISDN UUI service.

This *still* troubles me!
Is the assumption here that *both* the UAC and UAS are gateways, so that 
this mechanism is solely for the tunneling of Q.931 and Q.763 UUI 
message over SIP?

I get the impression that an important use case is where at least *one* 
end (probably the UAS) is a SIP device. In that case it will already be 
*necessary* for it to understand and interpret UUI data. And where there 
is redundancy between SIP data and the UUI data, it will need to resolve 
the discrepancy.

Shouldn't that all be in scope here?

> 3.1.  Why INFO is Not Used
...
>    control user-to-user information, INFO can not be used.  As the call
>    flows in the previous section show, the information is related to an

There are no call flows in the previous section. (They are now in 
another document.)

> 3.2.  MIME body Approach
> 
>    One method of transport is to transport the UUI information as a MIME
>    body.  This is in keeping with the SIP-T architecture [RFC3372] in
>    which MIME bodies are used to transport ISUP information.  Since the
>    INVITE will normally have an SDP message body, the resulting INVITE
>    with SDP and UUI will be multipart MIME.  This is not ideal as many
>    SIP UAs do not support multipart MIME INVITEs.

I don't understand this argument, since *no* SIP UAs currently support 
the UUI approach being proposed. The only entities that need to 
understand this are the UAC and the UAS. Both of them will have to 
support this UUI stuff for it to be used. If it were to be transported 
via Mime then that is what they would implement.

But the above is just nit picking, since I do agree with the problem of 
using mime bodies with REFER. I don't have any real problem with using a 
new header for this.

> 5.  Syntax for UUI Header Field
...
>        UUI         = "User-to-User" HCOLON uui-data *(SEMI uui-param)
>        uui-data    = token
>        uui-param   = enc-param | cont-param | purp-param | generic-param
>        enc-param   = "encoding=" ("hex" | token)
>        cont-param  = "content=" ("isdn-uui" | token)
>        purp-param  = "purpose=" ("isdn-interwork" | token)

At the very least, I would request that this be expanded a little bit 
for future flexibility, by permitting the uui-data to be either a token 
or a string. While the encoding you have chosen works as a token, other 
encodings may need the additional flexibility of a string.

        uui-data    = token | quoted-string

> 6.3.  Registration of SIP Option Tag

This section registers the new option tag. But I find almost nothing 
that defines the semantics of the option. (Section 5 does include the 
following:

>    A UA that supports this feature and the "uui" option tag MUST support
>    the call flows in [johnston-dispatch-sip-cc-uui]...

but that is pretty thin from a normative perspective.

Of particular note, does support for the option tag mean support for the 
particular encoding, content, and purpose included in this document? Or 
does it only mean support for the header and params in the abstract? 
(That wouldn't be very useful.)

I don't see how it can mean support for some other yet to be defined 
values for those. Yet if it only means support for the current ones, 
then how will one indicate support for future ones?

The only way out of this that I can see is to have separate options, 
either for particular values of each parameter, or for particular 
combinations of values of all the parameters.

So you might have options uui-purpose-isdn-interwork, uui-content-isdn, 
and uui-encoding-hex. Or you might just have option uui-isdn-interwork 
that means the combination.

I also am not finding much explanation of the semantics of the content 
and purpose parameters. I can only extrapolate from a single example of 
each, which I'm having trouble doing.

I'm guessing that 'content' must refer to the precise syntax of the 
contained data, after decoding. So presumably it must map onto some 
particular spec. For isdn-uui, would that be [Q931] or [Q763]? If so, 
shouldn't it refer to something specific in that document?

'purpose' is even less clear to me. Does it refer to why it is being 
included? Or what is to be done with it? If received by a UA that is not 
an ISDN gateway, how can it decide if this is something it should be 
trying to process? Is it still ISDN interworking if neither the UAC nor 
UAS is connected to an ISDN network?

Suppose I was developing a sophisticated UA that potentially might be 
usable by a call center operator. How should purpose=isdn-interwork 
affect the operation of this device?

	Thanks,
	Paul




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Subject: [dispatch] I-D Action:draft-kuthan-dispatch-diagrevived-00.txt
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Hi Folks,

We have just submitted a draft which proposes a new header field
called Debug.  The purpose of this header field is to convey extra
debugging information that can be used to locate errors in SIP
entities involved in the processing of a SIP transaction.

Abstract:
A major responsibility of Session Initiation Protocol (SIP) servers is
to provide application-layer routing.  SIP routing can be quite
complex and lead to similarly complicated paths that SIP requests
traverse on the way to their actual destinations.  It is therefore
important to be in position to troubleshoot errors that occur along a
SIP path, inside and outside troubleshooters' administrative domains.
Particularly important for the troubleshooters is knowledge of where
an error occurred in a SIP path.  This document introduces a new
header field called Debug.  The purpose of the header field is to
convey extra debugging information that can be used to locate errors
in SIP implementations involved in processing of a SIP transaction.

A temporal URL for this Internet-Draft is:
http://www.ietf.org/proceedings/staging/draft-kuthan-dispatch-diagrevived-0=
0.txt

Comments are welcome!

Regards,
--=20
Victor Pascual =C3=81vila

From xavier.marjou@gmail.com  Fri Jul  3 08:05:51 2009
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Subject: [dispatch] draft-shen-interaction-ind
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Hi,

As it seems there will be time for additional topics, I have a
suggestion:=A0I would be interested by a presentation of the following
draft:=A0http://tools.ietf.org/html/draft-shen-interaction-ind-05

Cheers,
Xavier

On Fri, Jun 26, 2009 at 12:51 AM, Mary Barnes <mary.barnes@nortel.com> wrot=
e:
>
> Hi folks,
>
> We have uploaded a revised (tentative) agenda for IETF-75 for the
> DISPATCH WG session:
> http://www.ietf.org/proceedings/09jul/agenda/dispatch.html
>
> Due to popular demand, we have added a slot for Session Recording. Note,
> that the current agenda has us meeting on Thursday afternoon, but Cullen
> has asked for that to be changed to Monday. As always, the agenda is
> subject to change up until the meeting.
>
> As a reminder any new documents related to the agenda topics must be
> submitted by July 6th and any updates to current docs by July 13th. And,
> most importantly we really should be seeing traffic on the list on these
> topics before the meeting so we can have focused discussions with
> conclusions and a well defined way forward.
>
> Please send any comments on the proposed agenda to the list and/or
> Gonzalo and myself.
>
> Regards,
> Mary
> DISPATCH WG co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From alan@sipstation.com  Fri Jul  3 09:14:14 2009
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Paul,

Thanks for your comments.  See mine below.

Thanks,
Alan

Paul Kyzivat wrote:
> Hi Alan,
>
> I took a quick look at this doc and the requirements doc as well. I 
> have a few thoughts:
>
> From section 1:
>
>>    In the future, where both endpoints are intelligent SIP user agents,
>>    it may be possible for them to understand and interpret the UUI data.
>>    There may be some cases where the UUI information is relevant to SIP.
>>    In this case, it might be worthwhile attempting to map UUI data to an
>>    appropriate SIP header field or to standardize a new header field.
>>    However, the requirements and use cases for this are different enough
>>    from those described in this document that these two situations
>>    should be examined separately.  This document looks only at the
>>    requirements and mechanisms for replicating the existing, widely used
>>    and deployed ISDN UUI service.
>
> This *still* troubles me!
> Is the assumption here that *both* the UAC and UAS are gateways, so 
> that this mechanism is solely for the tunneling of Q.931 and Q.763 UUI 
> message over SIP?
>
> I get the impression that an important use case is where at least 
> *one* end (probably the UAS) is a SIP device. In that case it will 
> already be *necessary* for it to understand and interpret UUI data. 
> And where there is redundancy between SIP data and the UUI data, it 
> will need to resolve the discrepancy.

Not quite.  We are not making the assumption that both are gateways, 
although that is possible.  Most likely, one element is SIP enabled.  
However, it is not a fully intelligent SIP endpoint - the basic software 
and application (probably many years old) is unchanged but a SIP 
interface (SIP trunk) has been added.  As such, the SIP stack does not 
know anything about the UUI as it is implementing exactly the ISDN UUI 
service - the information is just pushed up the stack to another 
application, and this application knows nothing about SIP.  It still 
thinks it is using an ISDN trunk.  So this mechanism allows us to easily 
and quickly get SIP into these applications without having to rewrite 
the entire application to make it SIP aware.  Does that make any sense?  
Honestly don't know how I can explain it more clearly than this.  And 
this is not some corner case or exception - this is a *huge* opportunity 
for SIP if we can get this mechanism standardized.

>
> Shouldn't that all be in scope here?

For the case I described above, where only SIP trunking is being used, 
the best you can do is indicate the purpose (the application using the 
UUI) is the ISDN UUE, and the content is how the ISDN spec defines it - 
an opaque blob of information.  The ISDN does not know if the UUI 
contains an account number, a name, etc - but the application making use 
of the SIP trunking does.

>
>> 3.1.  Why INFO is Not Used
> ...
>>    control user-to-user information, INFO can not be used.  As the call
>>    flows in the previous section show, the information is related to an
>
> There are no call flows in the previous section. (They are now in 
> another document.)

Yes, I missed this in splitting the documents.  I need to edit the text 
a little as well so they go together better but I ran out of time as 
I'll be out of the office all next week.

>
>> 3.2.  MIME body Approach
>>
>>    One method of transport is to transport the UUI information as a MIME
>>    body.  This is in keeping with the SIP-T architecture [RFC3372] in
>>    which MIME bodies are used to transport ISUP information.  Since the
>>    INVITE will normally have an SDP message body, the resulting INVITE
>>    with SDP and UUI will be multipart MIME.  This is not ideal as many
>>    SIP UAs do not support multipart MIME INVITEs.
>
> I don't understand this argument, since *no* SIP UAs currently support 
> the UUI approach being proposed. 

No, Avaya and at least one other vendor have implemented and deployed 
the User-to-User header field in customer networks.  As I said, there is 
a big demand for this, and the marketplace will not wait much longer for 
us to get this finalized.

But in our, and others, experience, adding support for a single new 
header field is easy.  Adding complete support for Multipart MIME is 
much harder and impacts many more things.


> The only entities that need to understand this are the UAC and the 
> UAS. Both of them will have to support this UUI stuff for it to be 
> used. If it were to be transported via Mime then that is what they 
> would implement.
>
> But the above is just nit picking, since I do agree with the problem 
> of using mime bodies with REFER. I don't have any real problem with 
> using a new header for this.

Excellent - I am very glad to hear this.


>
>> 5.  Syntax for UUI Header Field
> ...
>>        UUI         = "User-to-User" HCOLON uui-data *(SEMI uui-param)
>>        uui-data    = token
>>        uui-param   = enc-param | cont-param | purp-param | generic-param
>>        enc-param   = "encoding=" ("hex" | token)
>>        cont-param  = "content=" ("isdn-uui" | token)
>>        purp-param  = "purpose=" ("isdn-interwork" | token)
>
> At the very least, I would request that this be expanded a little bit 
> for future flexibility, by permitting the uui-data to be either a 
> token or a string. While the encoding you have chosen works as a 
> token, other encodings may need the additional flexibility of a string.
>
>        uui-data    = token | quoted-string
>

OK, I guess I'm OK with that.  For the isdn-interwork purpose, we'll 
require the token, but other applications could utilize the quoted string.


>> 6.3.  Registration of SIP Option Tag
>
> This section registers the new option tag. But I find almost nothing 
> that defines the semantics of the option. (Section 5 does include the 
> following:
>
>>    A UA that supports this feature and the "uui" option tag MUST support
>>    the call flows in [johnston-dispatch-sip-cc-uui]...
>
> but that is pretty thin from a normative perspective.
>
> Of particular note, does support for the option tag mean support for 
> the particular encoding, content, and purpose included in this 
> document? Or does it only mean support for the header and params in 
> the abstract? (That wouldn't be very useful.)

Yes, this needs clarification.  I'd suggest we define the option tag to 
be uui-isdn which means it understands the header field and the 
isdn-interwork purpose, and the call flows, including escaping into 
redirection and Refer-To URIs.


>
> I don't see how it can mean support for some other yet to be defined 
> values for those. Yet if it only means support for the current ones, 
> then how will one indicate support for future ones?
>
> The only way out of this that I can see is to have separate options, 
> either for particular values of each parameter, or for particular 
> combinations of values of all the parameters.

New applications using this header field would have the option of 
defining a new SIP option tag which would mean support for the header 
field and their new purpose.


>
> So you might have options uui-purpose-isdn-interwork, 
> uui-content-isdn, and uui-encoding-hex. Or you might just have option 
> uui-isdn-interwork that means the combination.

I think the latter.  I see it as a hierarchy - the application or 
purpose is the top one.  Each purpose has a number of contents that it 
supports.  Each content has a number of encoding schemes it supports.


>
> I also am not finding much explanation of the semantics of the content 
> and purpose parameters. I can only extrapolate from a single example 
> of each, which I'm having trouble doing.
>
> I'm guessing that 'content' must refer to the precise syntax of the 
> contained data, after decoding. So presumably it must map onto some 
> particular spec. For isdn-uui, would that be [Q931] or [Q763]? If so, 
> shouldn't it refer to something specific in that document?

Not really.  In this case, isdn-uui effectively means "opaque data" 
which is how ISDN defines the UUI - it is completely undefined, 
necessarily so  by the strict layering used in the ISDN application.


>
> 'purpose' is even less clear to me. Does it refer to why it is being 
> included? Or what is to be done with it? If received by a UA that is 
> not an ISDN gateway, how can it decide if this is something it should 
> be trying to process? Is it still ISDN interworking if neither the UAC 
> nor UAS is connected to an ISDN network?

The purpose is the application using the UUI.  Others have expressed an 
interest in carrying other data in the header field, in which case a new 
purpose would be defined.  I'll see if I can think of an example one.

Basically, if two UAs both support the header but have different 
applications generating and consuming the UUI, it will not work.  This 
is not a SIP failure, and there is nothing SIP can do about it.  But 
this purpose header field allows the UAs to detect this condition.


>
> Suppose I was developing a sophisticated UA that potentially might be 
> usable by a call center operator. How should purpose=isdn-interwork 
> affect the operation of this device?

To make it work in today's contact center applications, supporting 
isdn-interwork would likely make it work in an application, provided the 
appropriate call center application also ran on it.  Some contact 
centers do not use ISDN UUI, instead they use all kinds of really, 
really ugly CTI protocols running in parallel.  This is a way of moving 
those to the ISDN model, and from there to a more intelligent endpoint 
SIP model.


>
>     Thanks,
>     Paul
>
>
>
>


From pkyzivat@cisco.com  Fri Jul  3 16:40:44 2009
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Subject: Re: [dispatch] I-D Action:draft-johnston-sipping-cc-uui-08.txt
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Hi Alan,

responses inline. I've trimmed out the parts that don't require more 
discussion.

	Thanks,
	Paul

Alan Johnston wrote:
> Paul,
> 
> Thanks for your comments.  See mine below.
> 
> Thanks,
> Alan
> 
> Paul Kyzivat wrote:
>> Hi Alan,
>>
>> I took a quick look at this doc and the requirements doc as well. I 
>> have a few thoughts:
>>
>> From section 1:
>>
>>>    In the future, where both endpoints are intelligent SIP user agents,
>>>    it may be possible for them to understand and interpret the UUI data.
>>>    There may be some cases where the UUI information is relevant to SIP.
>>>    In this case, it might be worthwhile attempting to map UUI data to an
>>>    appropriate SIP header field or to standardize a new header field.
>>>    However, the requirements and use cases for this are different enough
>>>    from those described in this document that these two situations
>>>    should be examined separately.  This document looks only at the
>>>    requirements and mechanisms for replicating the existing, widely used
>>>    and deployed ISDN UUI service.
>>
>> This *still* troubles me!
>> Is the assumption here that *both* the UAC and UAS are gateways, so 
>> that this mechanism is solely for the tunneling of Q.931 and Q.763 UUI 
>> message over SIP?
>>
>> I get the impression that an important use case is where at least 
>> *one* end (probably the UAS) is a SIP device. In that case it will 
>> already be *necessary* for it to understand and interpret UUI data. 
>> And where there is redundancy between SIP data and the UUI data, it 
>> will need to resolve the discrepancy.
> 
> Not quite.  We are not making the assumption that both are gateways, 
> although that is possible.  Most likely, one element is SIP enabled.  
> However, it is not a fully intelligent SIP endpoint - the basic software 
> and application (probably many years old) is unchanged but a SIP 
> interface (SIP trunk) has been added.  As such, the SIP stack does not 
> know anything about the UUI as it is implementing exactly the ISDN UUI 
> service - the information is just pushed up the stack to another 
> application, and this application knows nothing about SIP.  It still 
> thinks it is using an ISDN trunk. 

Then in the way I meant the question, both ends *are* gateways.

But then if someone wishes to build native sip gear to plug into this 
environment, then it will still have to understand this isdn encoding. 
Perhaps that is the cruel truth, unless all the equipment is replaced at 
once.

>>> 5.  Syntax for UUI Header Field
>> ...
>>>        UUI         = "User-to-User" HCOLON uui-data *(SEMI uui-param)
>>>        uui-data    = token
>>>        uui-param   = enc-param | cont-param | purp-param | generic-param
>>>        enc-param   = "encoding=" ("hex" | token)
>>>        cont-param  = "content=" ("isdn-uui" | token)
>>>        purp-param  = "purpose=" ("isdn-interwork" | token)
>>
>> At the very least, I would request that this be expanded a little bit 
>> for future flexibility, by permitting the uui-data to be either a 
>> token or a string. While the encoding you have chosen works as a 
>> token, other encodings may need the additional flexibility of a string.
>>
>>        uui-data    = token | quoted-string
> 
> OK, I guess I'm OK with that.  For the isdn-interwork purpose, we'll 
> require the token, but other applications could utilize the quoted string.

I would hope that when the token encoding is sufficient, then either 
form would be accepted. Or it would perhaps be ok to specify for a 
particular encoding (hex) that the token encoding must be used. I don't 
see how the purpose has anything to do with it.

>>> 6.3.  Registration of SIP Option Tag
>>
>> This section registers the new option tag. But I find almost nothing 
>> that defines the semantics of the option. (Section 5 does include the 
>> following:
>>
>>>    A UA that supports this feature and the "uui" option tag MUST support
>>>    the call flows in [johnston-dispatch-sip-cc-uui]...
>>
>> but that is pretty thin from a normative perspective.
>>
>> Of particular note, does support for the option tag mean support for 
>> the particular encoding, content, and purpose included in this 
>> document? Or does it only mean support for the header and params in 
>> the abstract? (That wouldn't be very useful.)
> 
> Yes, this needs clarification.  I'd suggest we define the option tag to 
> be uui-isdn which means it understands the header field and the 
> isdn-interwork purpose, and the call flows, including escaping into 
> redirection and Refer-To URIs.

I'd like to hear some other opinions on whether to have tag per param 
for tag, or a tag for the combination of param values. I think I'm ok 
with what you propose.

>> I don't see how it can mean support for some other yet to be defined 
>> values for those. Yet if it only means support for the current ones, 
>> then how will one indicate support for future ones?
>>
>> The only way out of this that I can see is to have separate options, 
>> either for particular values of each parameter, or for particular 
>> combinations of values of all the parameters.
> 
> New applications using this header field would have the option of 
> defining a new SIP option tag which would mean support for the header 
> field and their new purpose.
> 
>> So you might have options uui-purpose-isdn-interwork, 
>> uui-content-isdn, and uui-encoding-hex. Or you might just have option 
>> uui-isdn-interwork that means the combination.
> 
> I think the latter.  I see it as a hierarchy - the application or 
> purpose is the top one.  Each purpose has a number of contents that it 
> supports.  Each content has a number of encoding schemes it supports.

I guess I'm ok with that, pending nailing down the definition of "purpose".

>> I also am not finding much explanation of the semantics of the content 
>> and purpose parameters. I can only extrapolate from a single example 
>> of each, which I'm having trouble doing.
>>
>> I'm guessing that 'content' must refer to the precise syntax of the 
>> contained data, after decoding. So presumably it must map onto some 
>> particular spec. For isdn-uui, would that be [Q931] or [Q763]? If so, 
>> shouldn't it refer to something specific in that document?
> 
> Not really.  In this case, isdn-uui effectively means "opaque data" 
> which is how ISDN defines the UUI - it is completely undefined, 
> necessarily so  by the strict layering used in the ISDN application.

I don't believe you!

If that is the case, then the UAC and UAS must have a private side 
agreement about what the isdn-uui content means. Is that really what you 
mean?

You already state that the first byte is a protocol discriminator. There 
must also be something, somewhere, that specifies the mapping from a 
particular value for that byte to a particular protocol/format. If there 
is, then that should be part of the definition of this content.

>> 'purpose' is even less clear to me. Does it refer to why it is being 
>> included? Or what is to be done with it? If received by a UA that is 
>> not an ISDN gateway, how can it decide if this is something it should 
>> be trying to process? Is it still ISDN interworking if neither the UAC 
>> nor UAS is connected to an ISDN network?
> 
> The purpose is the application using the UUI.  Others have expressed an 
> interest in carrying other data in the header field, in which case a new 
> purpose would be defined.  I'll see if I can think of an example one.

So are you saying that a particular call center application might 
constitute a distinct "purpose"?

> Basically, if two UAs both support the header but have different 
> applications generating and consuming the UUI, it will not work.  This 
> is not a SIP failure, and there is nothing SIP can do about it.  But 
> this purpose header field allows the UAs to detect this condition.

In that case, perhaps "isdn-interwork" isn't really a single purpose at 
all, unless any two pieces of equipment that support that purpose can 
then interwork without knowing any more about each other.

>> Suppose I was developing a sophisticated UA that potentially might be 
>> usable by a call center operator. How should purpose=isdn-interwork 
>> affect the operation of this device?
> 
> To make it work in today's contact center applications, supporting 
> isdn-interwork would likely make it work in an application, provided the 
> appropriate call center application also ran on it.  Some contact 
> centers do not use ISDN UUI, instead they use all kinds of really, 
> really ugly CTI protocols running in parallel. 

Yes, I know! :-(

> This is a way of moving 
> those to the ISDN model, and from there to a more intelligent endpoint 
> SIP model.
> 
>>     Thanks,
>>     Paul

From mary.barnes@nortel.com  Sat Jul  4 08:31:43 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Xavier Marjou" <xavier.marjou@orange-ftgroup.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] draft-shen-interaction-ind
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Xavier,

Can you point the WG to past ML discussions (in any WG) for this =
document?  I don't recall nor could I find any email threads that =
indicate this document had ever been on the radar for the SIPPING WG. =
And, as usual, we don't allocate agenda time for documents that have =
received no ML discussion and a fair amount of interest in committing to =
work on the problem.  Per the approach we're taking in DISPATCH, the =
work item needs to have a problem description with a defined scope and =
definition of a deliverable. I can see this document is defining a =
P-header, but the usage of this "service indicator" is not clear to me, =
perhaps because it's based on IMS functionality. The one example (alarm =
call) seems to be related to emergency services, in which case I would =
think the protocol work in ECRIT would be applicable (or if it's =
"authority to citizen" then the proposed ATOCA work might apply). If the =
author would like to provide a chart or two summarizing the problem the =
document is solving, we can present those in the chair charts. =20

Regards,
Mary.
DISPATCH WG co-chair

-----Original Message-----
From: xavier.marjou@gmail.com [mailto:xavier.marjou@gmail.com] On Behalf =
Of Xavier Marjou
Sent: Friday, July 03, 2009 10:06 AM
To: dispatch@ietf.org; Barnes, Mary (RICH2:AR00)
Subject: draft-shen-interaction-ind

Hi,

As it seems there will be time for additional topics, I have a
suggestion:=A0I would be interested by a presentation of the following
draft:=A0
http://tools.ietf.org/html/draft-shen-interaction-ind-05.txt


Cheers,
Xavier

On Fri, Jun 26, 2009 at 12:51 AM, Mary Barnes <mary.barnes@nortel.com> =
wrote:
>
> Hi folks,
>
> We have uploaded a revised (tentative) agenda for IETF-75 for the=20
> DISPATCH WG session:
> http://www.ietf.org/proceedings/09jul/agenda/dispatch.html
>
> Due to popular demand, we have added a slot for Session Recording.=20
> Note, that the current agenda has us meeting on Thursday afternoon,=20
> but Cullen has asked for that to be changed to Monday. As always, the=20
> agenda is subject to change up until the meeting.
>
> As a reminder any new documents related to the agenda topics must be=20
> submitted by July 6th and any updates to current docs by July 13th.=20
> And, most importantly we really should be seeing traffic on the list=20
> on these topics before the meeting so we can have focused discussions=20
> with conclusions and a well defined way forward.
>
> Please send any comments on the proposed agenda to the list and/or=20
> Gonzalo and myself.
>
> Regards,
> Mary
> DISPATCH WG co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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The updates after expert review of the
draft-vanelburg-sipping-private-network-indication-03
has been submitted to dispatch as:
http://tools.ietf.org/html/draft-vanelburg-dispatch-private-network-ind-00

or plain text:
http://tools.ietf.org/id/draft-vanelburg-dispatch-private-network-ind-00.txt

Best Regards,
/Hans Erik van Elburg


---------- Forwarded message ----------
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Mon, Jul 6, 2009 at 1:06 PM
Subject: Re: [Sipping] Fwd: Expert review
ofdraft-vanelburg-sipping-private-network-indication-03
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
Cc: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, IETF Sipping List <
sipping@ietf.org>


Draft has been uploaded as
http://tools.ietf.org/html/draft-vanelburg-dispatch-private-network-ind-00

Best Regards,
/Hans Erik van Elburg


On Mon, Jul 6, 2009 at 11:42 AM, Hans Erik van Elburg <
ietf.hanserik@gmail.com> wrote:

> Hi John,
>
> 1. Done, moved 10 up and inserted as 1.2. Added the following sentence to
> 1.1:
> "3GPP and ETSI TISPAN have identified a set of requirements that can be met
> by defining a new SIP P-header, according to the procedures in RFC 3427
> [RFC3427]."
> 2. Done
> 4a. Done
> 4b. Done
> 4c. Done
> 5. I added the following terminology description:
>
> "3.1. Traffic
>
> In the context of this document the term traffic is understood as all
> communication pertaining to and/or controlled by a SIP transaction or
> dialog."
>
> 8. Done
>
> Will upload this as cutoff day for 00 drafts is today.
>
> /Hans Erik van Elburg
>
>
>
> On Mon, Jul 6, 2009 at 8:52 AM, Elwell, John <
> john.elwell@siemens-enterprise.com> wrote:
>
>> Hans Erik,
>>
>> > -----Original Message-----
>> > From: sipping-bounces@ietf.org
>> > [mailto:sipping-bounces@ietf.org] On Behalf Of Hans Erik van Elburg
>> > Sent: 03 July 2009 15:17
>> > To: Elwell, John; DRAGE, Keith (Keith)
>> > Cc: IETF Sipping List
>> > Subject: [Sipping] Fwd: Expert review
>> > ofdraft-vanelburg-sipping-private-network-indication-03
>> >
>> > Resent and history removed as otherwise rejected by mailing list.
>> >
>> > Hi John,
>> >
>> > 1. Again the required applicability statement is in section
>> > 10 of the main body. (Same pattern has been followed in
>> > RFC5502, 5002 and 4457) which I took as examples. Together
>> > with the additional text in the abstract that should be
>> > sufficient. If the text in section 10 needs to be improved
>> > then please indicate so.
>> [JRE] I must have written the original comment 1 before I got down to
>> section 10. Basically section 10 is far too late in the document, and I
>> would have expected the statement, perhaps as section 1, or perhaps as a
>> section 1.2 before the existing 1.2.
>>
>>
>> >
>> > 2. Took in your nits, the abstract now reads:
>> >       "This document specifies the SIP P-Private-Network-Indication
>> > P-header. The use of this private network indication extension
>> > is only applicable inside an administrative domain with
>> > previously agreed-upon policies for generation, transport and
>> > usage of such information.  A private network indication
>> > allows nodes in such a domain to treat private network traffic
>> > according to a different set of rules then than the set
>> > applicable to public network traffic. The indication also
>> > distinguishes traffic from one private network from another
>> > private network."
>> [JRE]
>> Delete the word "then".
>>
>> >
>> > 4a. Regarding 1.2 item b). Would the following rewrite help?:
>> > OLD:
>> > " b) business trunking application, where the NGN hosts
>> > transit capabilities between NGCN's, break-in capabilities
>> > from NGN to NGCN and break-out capabilities from NGCN to NGN; and"
>> > /OLD
>> >
>> >
>> > NEW:
>> > " b) business trunking application, where the NGN hosts
>> > transit capabilities between NGCN's, break-in capabilities
>> > where the NGN converts public network traffic to private
>> > network traffic for delivery at a served NGCN and break-out
>> > capabilities where the NGN converts private network traffic
>> > from a served NGCN to public network traffic; and"
>> >
>> > /NEW
>> >
>> > If not please suggest an improved sentence that covers your
>> > concern. Please note that in the example that you gave it is
>> > not the business trunking application that does the
>> > conversion, but the hosted enterprise application.
>> [JRE] That would do.
>>
>>
>> >
>> > 4b. "normal rules"
>> > Looking at the definition, would the following rewrite help:
>> > OLD
>> > " Traffic sent to or received from a public telecommunication
>> > network for processing according to the normal rules."
>> > /OLD
>> >
>> > NEW
>> > " Traffic sent to or received from a public telecommunication
>> > network for processing according to the rules for ordinary
>> > subscribers of a public telecommunication network."
>> > /NEW
>> [JRE] That would do.
>>
>> >
>> > 4c. NGN is a public telecommunication network
>> > >       - "To summarize a few example reasons for a public
>> > telecommunication network to make the distinction between the
>> > two types of traffic" Isn't it an NGN that needs to make the
>> > distinction?
>> > > [HE] NGN is a public telecommunication network. But we can
>> > rephrase to: "To summarize a few example reasons for a public
>> > NGN to make the distinction between the two types of traffic" [/HE]
>> [JRE] OK
>>
>> >
>> > [JRE] I think that is the problem I am having. I believe an
>> > NGN is a network infrastructure that supports both public
>> > network traffic and private network traffic, or in other
>> > words it supports both a public network and a number of
>> > private networks.[/JRE]
>> > [HE2]Yes, but its main purpose is to serve as a public
>> > network with all the regulations that apply to such networks
>> > etc. This does not prohibit an NGN to be used as a private
>> > network of course, but still it has been designed from the
>> > perspective of having to serve the needs of public network
>> > operators. That is why the "normal"/default  procedures are
>> > for public network traffic. As we want to introduce the
>> > capability for such a public network (NGN) to also support
>> > private network traffic that has to be processed to a
>> > different set of rules, the Private-Network-Indication is
>> > needed.[/HE2]
>> >
>> >
>> > 5. Traffic --> SIP traffic
>> > Calling traffic SIP traffic suggest that the media is not
>> > part of the traffic, is that what we want??
>> [JRE] The point is, it is not HTTP traffic or FTP traffic or SMTP
>> traffic or whatever.
>>
>> >
>> >
>> > 6. Example private network traffic can also exist between two
>> > different enterprises
>> > Where would you like to see this? Obviously section 1.2 is
>> > not the right place for such an example.
>> > Isn't this too obvious, if you imagine that two companies
>> > have agreed to form a cooperation and communicate under this
>> > agreement?
>> > Would you like to see this as an additional example in section 4?
>> [JRE] I was not necessarily seeking a further example. However, given
>> that the private network indication identifies the particular private
>> network, what form does this indication take when traffic is between two
>> enterprises? Is there some interworking function where the identifier
>> changes from that of the first private network to that of the second
>> private network?
>>
>> >
>> >
>> > 8. calling line identification
>> > Yes we agree fully on that "calling line identification is
>> > not an adequate distinction". I think that is what the
>> > current text tries to explain. Actually what I dont like
>> > about this text is that it starts explaining what the
>> > indication is not about. I propose that we completely remove
>> > the 1st paragraph in section 1.3 and the 1st word "Rather"
>> > from the 2nd paragraph.
>> [JRE] Yes. And in the second paragraph, perhaps we could enhance:
>> "This
>>   indication is not for the end user on the private network."
>> to say:
>> "This indication does not identify an end user on a private network and
>> is not for delivery to an end user on the private network."
>>
>>
>> John
>>
>>
>> >
>> > /Hans Erik van Elburg
>> >
>> >
>> >
>> >
>>
>
>

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<div>The updates after expert review of the d<span class=3D"Apple-style-spa=
n" style=3D"border-collapse: collapse; white-space: pre; -webkit-border-hor=
izontal-spacing: 2px; -webkit-border-vertical-spacing: 2px; ">raft-vanelbur=
g-sipping-private-network-indication-03 has been submitted to dispatch as:<=
/span></div>
<div><span class=3D"Apple-style-span" style=3D"border-collapse: collapse; w=
hite-space: pre; -webkit-border-horizontal-spacing: 2px; -webkit-border-ver=
tical-spacing: 2px;"><a href=3D"http://tools.ietf.org/html/draft-vanelburg-=
dispatch-private-network-ind-00">http://tools.ietf.org/html/draft-vanelburg=
-dispatch-private-network-ind-00</a></span></div>
<div><span class=3D"Apple-style-span" style=3D"border-collapse: collapse; w=
hite-space: pre; -webkit-border-horizontal-spacing: 2px; -webkit-border-ver=
tical-spacing: 2px;"><br></span></div><div><span class=3D"Apple-style-span"=
 style=3D"border-collapse: collapse; white-space: pre; -webkit-border-horiz=
ontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;">or plain text:</=
span></div>
<div><span class=3D"Apple-style-span" style=3D"border-collapse: collapse; w=
hite-space: pre; -webkit-border-horizontal-spacing: 2px; -webkit-border-ver=
tical-spacing: 2px;"><a href=3D"http://tools.ietf.org/id/draft-vanelburg-di=
spatch-private-network-ind-00.txt">http://tools.ietf.org/id/draft-vanelburg=
-dispatch-private-network-ind-00.txt</a></span></div>
<div><span class=3D"Apple-style-span" style=3D"border-collapse: collapse; w=
hite-space: pre; -webkit-border-horizontal-spacing: 2px; -webkit-border-ver=
tical-spacing: 2px;"><br></span></div><div><span class=3D"Apple-style-span"=
 style=3D"border-collapse: collapse; white-space: pre; -webkit-border-horiz=
ontal-spacing: 2px; -webkit-border-vertical-spacing: 2px;">Best Regards,</s=
pan></div>
/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">---------- Forwarded message ----------<=
br>From: <b class=3D"gmail_sendername">Hans Erik van Elburg</b> <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ietf.hanserik@gmail.com">ietf.hanserik@gmail=
.com</a>&gt;</span><br>
Date: Mon, Jul 6, 2009 at 1:06 PM<br>Subject: Re: [Sipping] Fwd: Expert rev=
iew ofdraft-vanelburg-sipping-private-network-indication-03<br>To: &quot;El=
well, John&quot; &lt;<a href=3D"mailto:john.elwell@siemens-enterprise.com">=
john.elwell@siemens-enterprise.com</a>&gt;<br>
Cc: &quot;DRAGE, Keith (Keith)&quot; &lt;<a href=3D"mailto:drage@alcatel-lu=
cent.com">drage@alcatel-lucent.com</a>&gt;, IETF Sipping List &lt;<a href=
=3D"mailto:sipping@ietf.org">sipping@ietf.org</a>&gt;<br><br><br><div>Draft=
 has been uploaded as=A0<a href=3D"http://tools.ietf.org/html/draft-vanelbu=
rg-dispatch-private-network-ind-00" target=3D"_blank">http://tools.ietf.org=
/html/draft-vanelburg-dispatch-private-network-ind-00</a></div>
<div><br></div><div>Best Regards,</div>
/Hans Erik van Elburg<div><div></div><div class=3D"h5"><br>
<br><div class=3D"gmail_quote">On Mon, Jul 6, 2009 at 11:42 AM, Hans Erik v=
an Elburg <span dir=3D"ltr">&lt;<a href=3D"mailto:ietf.hanserik@gmail.com" =
target=3D"_blank">ietf.hanserik@gmail.com</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">

<div>Hi John,</div><div><br></div><div>1. Done, moved 10 up and inserted as=
 1.2. Added the following sentence to 1.1:</div>&quot;3GPP and ETSI TISPAN =
have identified a set of=A0requirements that can be met by defining a new S=
IP P-header, according to the procedures in RFC 3427 [RFC3427].&quot;<br>


<div>2. Done</div><div>4a. Done</div><div>4b. Done</div><div>4c. Done</div>=
<div>5. I added the following terminology description:</div><blockquote sty=
le=3D"margin:0 0 0 40px;border:none;padding:0px">
&quot;3.1. Traffic<br><br>In the context of this document the term traffic =
is understood as all<br>communication pertaining to and/or controlled by a =
SIP transaction or<br>dialog.&quot;</blockquote><div><div>8. Done</div>


<div><br></div><div>Will upload this as cutoff day for 00 drafts is today.<=
/div><div><br></div><font color=3D"#888888">/Hans Erik van Elburg</font><di=
v><div></div><div><br>
<br><br><div class=3D"gmail_quote">On Mon, Jul 6, 2009 at 8:52 AM, Elwell, =
John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens-enterprise=
.com" target=3D"_blank">john.elwell@siemens-enterprise.com</a>&gt;</span> w=
rote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hans Erik,<br>
<div><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sipping-bounces@ietf.org" target=3D"_blank">si=
pping-bounces@ietf.org</a><br>
&gt; [mailto:<a href=3D"mailto:sipping-bounces@ietf.org" target=3D"_blank">=
sipping-bounces@ietf.org</a>] On Behalf Of Hans Erik van Elburg<br>
&gt; Sent: 03 July 2009 15:17<br>
&gt; To: Elwell, John; DRAGE, Keith (Keith)<br>
&gt; Cc: IETF Sipping List<br>
&gt; Subject: [Sipping] Fwd: Expert review<br>
&gt; ofdraft-vanelburg-sipping-private-network-indication-03<br>
&gt;<br>
&gt; Resent and history removed as otherwise rejected by mailing list.<br>
&gt;<br>
&gt; Hi John,<br>
&gt;<br>
&gt; 1. Again the required applicability statement is in section<br>
&gt; 10 of the main body. (Same pattern has been followed in<br>
&gt; RFC5502, 5002 and 4457) which I took as examples. Together<br>
&gt; with the additional text in the abstract that should be<br>
&gt; sufficient. If the text in section 10 needs to be improved<br>
&gt; then please indicate so.<br>
</div>[JRE] I must have written the original comment 1 before I got down to=
<br>
section 10. Basically section 10 is far too late in the document, and I<br>
would have expected the statement, perhaps as section 1, or perhaps as a<br=
>
section 1.2 before the existing 1.2.<br>
<div><br>
<br>
&gt;<br>
&gt; 2. Took in your nits, the abstract now reads:<br>
&gt; =A0 =A0 =A0 &quot;This document specifies the SIP P-Private-Network-In=
dication<br>
&gt; P-header. The use of this private network indication extension<br>
&gt; is only applicable inside an administrative domain with<br>
&gt; previously agreed-upon policies for generation, transport and<br>
&gt; usage of such information. =A0A private network indication<br>
&gt; allows nodes in such a domain to treat private network traffic<br>
&gt; according to a different set of rules then than the set<br>
&gt; applicable to public network traffic. The indication also<br>
&gt; distinguishes traffic from one private network from another<br>
&gt; private network.&quot;<br>
</div>[JRE]<br>
Delete the word &quot;then&quot;.<br>
<div><br>
&gt;<br>
&gt; 4a. Regarding 1.2 item b). Would the following rewrite help?:<br>
&gt; OLD:<br>
&gt; &quot; b) business trunking application, where the NGN hosts<br>
&gt; transit capabilities between NGCN&#39;s, break-in capabilities<br>
&gt; from NGN to NGCN and break-out capabilities from NGCN to NGN; and&quot=
;<br>
&gt; /OLD<br>
&gt;<br>
&gt;<br>
&gt; NEW:<br>
&gt; &quot; b) business trunking application, where the NGN hosts<br>
&gt; transit capabilities between NGCN&#39;s, break-in capabilities<br>
&gt; where the NGN converts public network traffic to private<br>
&gt; network traffic for delivery at a served NGCN and break-out<br>
&gt; capabilities where the NGN converts private network traffic<br>
&gt; from a served NGCN to public network traffic; and&quot;<br>
&gt;<br>
&gt; /NEW<br>
&gt;<br>
&gt; If not please suggest an improved sentence that covers your<br>
&gt; concern. Please note that in the example that you gave it is<br>
&gt; not the business trunking application that does the<br>
&gt; conversion, but the hosted enterprise application.<br>
</div>[JRE] That would do.<br>
<div><br>
<br>
&gt;<br>
&gt; 4b. &quot;normal rules&quot;<br>
&gt; Looking at the definition, would the following rewrite help:<br>
&gt; OLD<br>
&gt; &quot; Traffic sent to or received from a public telecommunication<br>
&gt; network for processing according to the normal rules.&quot;<br>
&gt; /OLD<br>
&gt;<br>
&gt; NEW<br>
&gt; &quot; Traffic sent to or received from a public telecommunication<br>
&gt; network for processing according to the rules for ordinary<br>
&gt; subscribers of a public telecommunication network.&quot;<br>
&gt; /NEW<br>
</div>[JRE] That would do.<br>
<div><br>
&gt;<br>
&gt; 4c. NGN is a public telecommunication network<br>
&gt; &gt; =A0 =A0 =A0 - &quot;To summarize a few example reasons for a publ=
ic<br>
&gt; telecommunication network to make the distinction between the<br>
&gt; two types of traffic&quot; Isn&#39;t it an NGN that needs to make the<=
br>
&gt; distinction?<br>
&gt; &gt; [HE] NGN is a public telecommunication network. But we can<br>
&gt; rephrase to: &quot;To summarize a few example reasons for a public<br>
&gt; NGN to make the distinction between the two types of traffic&quot; [/H=
E]<br>
</div>[JRE] OK<br>
<div><br>
&gt;<br>
&gt; [JRE] I think that is the problem I am having. I believe an<br>
&gt; NGN is a network infrastructure that supports both public<br>
&gt; network traffic and private network traffic, or in other<br>
&gt; words it supports both a public network and a number of<br>
&gt; private networks.[/JRE]<br>
&gt; [HE2]Yes, but its main purpose is to serve as a public<br>
&gt; network with all the regulations that apply to such networks<br>
&gt; etc. This does not prohibit an NGN to be used as a private<br>
&gt; network of course, but still it has been designed from the<br>
&gt; perspective of having to serve the needs of public network<br>
&gt; operators. That is why the &quot;normal&quot;/default =A0procedures ar=
e<br>
&gt; for public network traffic. As we want to introduce the<br>
&gt; capability for such a public network (NGN) to also support<br>
&gt; private network traffic that has to be processed to a<br>
&gt; different set of rules, the Private-Network-Indication is<br>
&gt; needed.[/HE2]<br>
&gt;<br>
&gt;<br>
&gt; 5. Traffic --&gt; SIP traffic<br>
&gt; Calling traffic SIP traffic suggest that the media is not<br>
&gt; part of the traffic, is that what we want??<br>
</div>[JRE] The point is, it is not HTTP traffic or FTP traffic or SMTP<br>
traffic or whatever.<br>
<div><br>
&gt;<br>
&gt;<br>
&gt; 6. Example private network traffic can also exist between two<br>
&gt; different enterprises<br>
&gt; Where would you like to see this? Obviously section 1.2 is<br>
&gt; not the right place for such an example.<br>
&gt; Isn&#39;t this too obvious, if you imagine that two companies<br>
&gt; have agreed to form a cooperation and communicate under this<br>
&gt; agreement?<br>
&gt; Would you like to see this as an additional example in section 4?<br>
</div>[JRE] I was not necessarily seeking a further example. However, given=
<br>
that the private network indication identifies the particular private<br>
network, what form does this indication take when traffic is between two<br=
>
enterprises? Is there some interworking function where the identifier<br>
changes from that of the first private network to that of the second<br>
private network?<br>
<div><br>
&gt;<br>
&gt;<br>
&gt; 8. calling line identification<br>
&gt; Yes we agree fully on that &quot;calling line identification is<br>
&gt; not an adequate distinction&quot;. I think that is what the<br>
&gt; current text tries to explain. Actually what I dont like<br>
&gt; about this text is that it starts explaining what the<br>
&gt; indication is not about. I propose that we completely remove<br>
&gt; the 1st paragraph in section 1.3 and the 1st word &quot;Rather&quot;<b=
r>
&gt; from the 2nd paragraph.<br>
</div>[JRE] Yes. And in the second paragraph, perhaps we could enhance:<br>
&quot;This<br>
 =A0 indication is not for the end user on the private network.&quot;<br>
to say:<br>
&quot;This indication does not identify an end user on a private network an=
d<br>
is not for delivery to an end user on the private network.&quot;<br>
<br>
<br>
John<br>
<br>
<br>
&gt;<br>
&gt; /Hans Erik van Elburg<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
</blockquote></div><br></div></div></div>
</blockquote></div><br>
</div></div></div><br>

--0015174be6d20b4185046e078c0c--

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From: <R.Jesske@telekom.de>
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Dear all,
After getting no response to my question from the dispatch. I assume =
that there is no interest on this draft within dispatch.

So now I would like to ask the people from SIPCORE and BILSS where do =
they see the work.

The below mentioned draft describes the use of the Reason Header within =
Resoponses for interoperability with the existing PSTN/ISDN networks.

Best Regards

Roland

> _____________________________________________=20
> Von: 	Jesske, Roland =20
> Gesendet:	Donnerstag, 2. Juli 2009 14:39
> An:	dispatch@ietf.org
> Betreff:	draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Dear all,
> At the moment I'm preparing a new version of =
draft-jesske-sipping-etsi-ngn-reason-04 ( =
http://tools.ietf.org/html/draft-jesske-sipping-etsi-ngn-reason-04 ) to =
be submitted.
> There were some comments which were stating that this is the wrong =
group for the draft.
> I would like to know what the feeling of the group is to which group =
(BLISS, DISPATCH or SIPCORE) this draft should belong.
>=20
> Thank you and Best Regards
>=20
> Roland
>=20
> Deutsche Telekom Netzproduktion GmbH=20
> Zentrum Technik Einf=FChrung=20
> Roland Jesske=20
> Gateways, Protokolle, Konvergenz, TE32-1
> Heinrich-Hertz-Str. 3-7, 64295 Darmstadt,=20
> Postfach, 64307 Darmstadt (Postanschrift)=20
> +496151 628 2766 (Tel)
> +491718618445 (Mobile)=20
> E-Mail: r.jesske@telekom.de=20
> http://www.telekom.de <http://www.telekom.de/> =20
>=20
>=20
>=20
> Registerrechtliche Unternehmensangaben:=20
> Deutsche Telekom Netzproduktion GmbH=20
> Aufsichtsrat: Timotheus H=F6ttges (Vorsitzender)=20
> Gesch=E4ftsf=FChrung: Friedrich Fu=DF (Vorsitzender), Albert Matheis, =
Klaus Peren=20
> Handelsregister: Amtsgericht Bonn HRB 14190=20
> Sitz der Gesellschaft: Bonn=20
> USt-IdNr.: DE 814645262
>=20
>=20

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<TITLE>AW: draft-jesske-sipping-etsi-ngn-reason-04</TITLE>
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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Dear all,</FONT>

<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">After getting no =
response to my question from the dispatch. I assume that there is no =
interest on this draft within dispatch.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">So now I would like =
to ask the people from SIPCORE and BILSS where do they see the =
work.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">The below mentioned =
draft describes the use of the Reason Header within Resoponses for =
interoperability with the existing PSTN/ISDN networks.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Best Regards</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Roland</FONT>
</P>

<P><FONT SIZE=3D1 =
FACE=3D"Tahoma">_____________________________________________ </FONT>

<BR><B><FONT SIZE=3D1 FACE=3D"Tahoma">Von: &nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Tahoma">Jesske, Roland&nbsp; </FONT>

<BR><B><FONT SIZE=3D1 =
FACE=3D"Tahoma">Gesendet:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> =
<FONT SIZE=3D1 FACE=3D"Tahoma">Donnerstag, 2. Juli 2009 14:39</FONT>

<BR><B><FONT SIZE=3D1 =
FACE=3D"Tahoma">An:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Tahoma">dispatch@ietf.org</FONT>

<BR><B><FONT SIZE=3D1 =
FACE=3D"Tahoma">Betreff:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 =
FACE=3D"Tahoma">draft-jesske-sipping-etsi-ngn-reason-04</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Dear all,</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Arial">At the moment I'm preparing a new =
version of draft-jesske-sipping-etsi-ngn-reason-04 ( </FONT><A =
HREF=3D"http://tools.ietf.org/html/draft-jesske-sipping-etsi-ngn-reason-0=
4"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://tools.ietf.org/html/draft-jesske-sipping-etsi-ngn-r=
eason-04</FONT></U></A><FONT SIZE=3D2 FACE=3D"Arial"> ) to be =
submitted.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">There were some comments which were =
stating that this is the wrong group for the draft.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Arial">I would like to know what the feeling =
of the group is to which group (BLISS, DISPATCH or SIPCORE) this draft =
should belong.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thank you and Best Regards</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Roland</FONT>
</P>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Deutsche Telekom =
Netzproduktion GmbH<BR>
Zentrum Technik Einf=FChrung</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Roland Jesske</FONT><FONT FACE=3D"Times =
New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">Gateways, Protokolle, Konvergenz, =
TE32-1</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Heinrich-Hertz-Str. 3-7, 64295 =
Darmstadt,</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Postfach, 64307 Darmstadt =
(Postanschrift)</FONT><FONT FACE=3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">+496151 628 2766 (Tel)</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">+491718618445 (Mobile)</FONT><FONT =
FACE=3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">E-Mail: </FONT><A =
HREF=3D"mailto:r.jesske@telekom.de"><U><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Arial">r.jesske@telekom.de</FONT></U></A><FONT SIZE=3D2 =
FACE=3D"Arial"></FONT>=20

<BR><A HREF=3D"http://www.telekom.de/"><U></U><U></U><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://www.telekom.de</FONT></U></A><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"></FONT>=20
</P>
<BR>
<BR>

<P><U><FONT SIZE=3D2 FACE=3D"Arial">Registerrechtliche =
Unternehmensangaben:</FONT></U><FONT SIZE=3D2 FACE=3D"Arial"></FONT>=20

<BR><FONT SIZE=3D2 FACE=3D"Arial">Deutsche Telekom Netzproduktion =
GmbH<BR>
Aufsichtsrat: Timotheus H=F6ttges (Vorsitzender)<BR>
Gesch=E4ftsf=FChrung: Friedrich Fu=DF (Vorsitzender), Albert Matheis, =
Klaus Peren<BR>
Handelsregister: Amtsgericht Bonn HRB 14190<BR>
Sitz der Gesellschaft: Bonn<BR>
USt-IdNr.: DE 814645262</FONT>
</P>
<BR>

</BODY>
</HTML>
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From gonzalo.camarillo@ericsson.com  Mon Jul  6 06:57:49 2009
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Date: Mon, 06 Jul 2009 16:56:49 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [dispatch] Preconditions and intermediaries
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Folks,

[as individual]

the following draft was presented in SIPPING in the the last IETF meeting.

http://tools.ietf.org/internet-drafts/draft-camarillo-sipping-precons-00.txt

This draft was a spin off from the "rollback" discussions that led to 
the following draft. It was a separate issue we decided to document in a 
separate draft.

http://www.ietf.org/internet-drafts/draft-camarillo-sipcore-reinvite-00.txt

Now, I would like to ask comments on the preconditions draft. If this is 
something people are interested in?

Cheers,

Gonzalo

From dworley@nortel.com  Mon Jul  6 08:37:39 2009
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From: "Dale Worley" <dworley@nortel.com>
To: R.Jesske@telekom.de
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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On Mon, 2009-07-06 at 13:50 +0200, R.Jesske@telekom.de wrote:
> Dear all, 
> After getting no response to my question from the dispatch. I assume
> that there is no interest on this draft within dispatch.
> 
> So now I would like to ask the people from SIPCORE and BILSS where do
> they see the work.
> 
> The below mentioned draft describes the use of the Reason Header
> within Resoponses for interoperability with the existing PSTN/ISDN
> networks.

Is the Abstract right?  As far as I can tell (by reading section 2), the
Abstract doesn't describe the draft at all.  That might be a reason that
you haven't gotten any response -- the Abstract "proposes the use of the
Reason header field in SIP responses", but that is something that is
already defined and allowed.

Dale



From salvatore.loreto@ericsson.com  Mon Jul  6 09:33:33 2009
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: [dispatch] Disaggregated Media in SIP
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Hi there,

I have just submitted a draft that talks of Disaggregated Media in the 
Session Initiation Protocol (SIP).
http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disaggregated-media-00.txt


Abstract:
Disaggregated media refers to the ability for a user to create a 
multimedia session combining different media streams, coming from
different devices under his or her control, so that they are treated by 
the far end of the session as a single media session.
This document lists several use cases that involve disaggregated media 
in SIP.
Additionally, this document analyzes what types of disaggregated media 
can be implemented using existing protocol
mechanisms, and the pros and cons of using each of those mechanisms.
Finally, this document describes scenarios that are not covered by 
current mechanisms
and proposes new IETF work to cover them.


cheers
Sal

From AUDET@nortel.com  Mon Jul  6 10:09:37 2009
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Date: Mon, 6 Jul 2009 12:07:53 -0500
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Thread-Topic: [dispatch] Disaggregated Media in SIP
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References: <4A522751.9010607@ericsson.com>
From: "Francois Audet" <audet@nortel.com>
To: "Salvatore Loreto" <salvatore.loreto@ericsson.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] Disaggregated Media in SIP
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I'm glad to see this topic coming back.

I see that this draft doesn't propose a solution to problem: it list
three options, and describes why they are not adequate. I agree with
the conclusions.

We've looked at various approaches to solve this important problem=20
several times before:=20

- Feature ref (refer to urn: indicating specific features)
  http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00

- Remote control using REFER to requests & responses=20
  http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
  (Also, versions -04, -03,-02, -00)

- Remore control using REFER with XML body describing function
  http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01

- Remote control using MBUS
  http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
  http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01

On top of that there are various proprietary mechanisms, and even some =
legacy
PBX-CTI protocols.

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore Loreto
> Sent: Monday, July 06, 2009 09:33
> To: dispatch@ietf.org
> Subject: [dispatch] Disaggregated Media in SIP
>=20
> Hi there,
>=20
> I have just submitted a draft that talks of Disaggregated=20
> Media in the Session Initiation Protocol (SIP).
> http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
ggregated-media-00.txt
>=20
>=20
> Abstract:
> Disaggregated media refers to the ability for a user to create a=20
> multimedia session combining different media streams, coming from
> different devices under his or her control, so that they are=20
> treated by=20
> the far end of the session as a single media session.
> This document lists several use cases that involve=20
> disaggregated media=20
> in SIP.
> Additionally, this document analyzes what types of=20
> disaggregated media=20
> can be implemented using existing protocol
> mechanisms, and the pros and cons of using each of those mechanisms.
> Finally, this document describes scenarios that are not covered by=20
> current mechanisms
> and proposes new IETF work to cover them.
>=20
>=20
> cheers
> Sal
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From nadia.bishai@ericsson.com  Mon Jul  6 10:18:34 2009
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From: "Nadia Bishai" <nadia.bishai@ericsson.com>
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Hi,

During the presentation of draft-loreto-sipping-3892bis-01.txt at IETF =
74 in SF,
we got a question from Cullen related to the importance for a receiver =
to get all the identities expressions and what it does with multiple =
identity expressions.

The reason why multiple identity experssions are needed by the receiver =
is that some receivers can only handle one format of identity expression =
(e.g. TEL URI) and not other formats (e.g., SIP URI). =20
Some examples:=20
-in a Conference call, when an incoming session invitation carries the =
Referred-By header, the TEL URI is needed to find the corresponding =
contact information in the user's address book when the address book =
only stores telephone numbers.
- When a MESSAGE message is sent to a group where one recipient is an =
SMS user, the MESSAGE is intercepted by an interworking function that =
translates the MESSAGE into an SMS message. The SMS system can only =
handle a MSISDN identifier. The interworking function will use the TEL =
URI received in the Referred-By header to derive an MSISDN value to be =
sent in SMS signalling.


Nadia Bishai=20
E      =20
E-Mail: Nadia.Bishai@ericsson.com=20
Tel:  (+1) 514-345-7900   (X 42234)         8400 Decarie Boulevard=20
Cell: (+1) 514-591-6739                          Montreal, Quebec, =
Canada
 www.ericsson.com                                H4P-2N2=20
ECN 8105 2234=20



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<P><FONT FACE=3D"Times New Roman">Hi,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">During the presentation of =
draft-loreto-sipping-3892bis-01.txt at IETF 74 in SF,</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">we got a question from Cullen =
related to the importance for a receiver to get all the identities =
expressions and what it does with multiple identity =
expressions.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The reason why multiple identity =
experssions are needed by the receiver is that some receivers can only =
handle one format of identity expression (e.g. TEL URI) and not other =
formats (e.g., SIP URI).&nbsp; </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Some examples: </FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">-in a Conference call, when an =
incoming session invitation carries the Referred-By header, the TEL URI =
is needed to find the corresponding contact information in the user's =
address book when the address book only stores telephone =
numbers.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- When a MESSAGE message is sent =
to a group where one recipient is an SMS user, the MESSAGE is =
intercepted by an interworking function that translates the MESSAGE into =
an SMS message. The SMS system can only handle a MSISDN identifier. The =
interworking function will use the TEL URI received in the Referred-By =
header to derive an MSISDN value to be sent in SMS =
signalling.</FONT></P>
<BR>

<P><B><SPAN LANG=3D"en-us"><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">Nadia Bishai</FONT></SPAN></B><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Arial"><BR>
</FONT><FONT COLOR=3D"#3366FF" SIZE=3D6 =
FACE=3D"EriLogo">E</FONT><B></B><B></B><B><FONT FACE=3D"Arial"> =
=A0=A0=A0=A0=A0</FONT></B><FONT FACE=3D"Arial"><BR>
</FONT><FONT SIZE=3D1 FACE=3D"Arial">E-Mail: =
Nadia.Bishai@ericsson.com</FONT><FONT FACE=3D"Arial"><BR>
</FONT><FONT SIZE=3D1 FACE=3D"Arial">Tel:=A0 (+1) 514-345-7900=A0=A0 =
(</FONT><B><FONT SIZE=3D1 FACE=3D"Arial">X 42234</FONT></B><FONT =
SIZE=3D1 FACE=3D"Arial">)=A0=A0=A0=A0=A0=A0=A0=A0 8400 Decarie =
Boulevard</FONT><FONT FACE=3D"Arial"><BR>
</FONT><FONT SIZE=3D1 FACE=3D"Arial">Cell: (+1) =
514-591-6739=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 Montreal, Quebec, Canada</FONT><BR>
<FONT SIZE=3D1 =
FACE=3D"Arial">=A0www.ericsson.com=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
H4P-2N2</FONT><FONT FACE=3D"Arial"><BR>
</FONT><B><FONT SIZE=3D1 FACE=3D"Arial">ECN 8105 2234</FONT></B><FONT =
FACE=3D"Arial"> </FONT></SPAN>
</P>
<BR>

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From: "Francois Audet" <audet@nortel.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, "Salvatore Loreto" <salvatore.loreto@ericsson.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] Disaggregated Media in SIP
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Sure.
=20
That would be functionaly identical to the Feature referal using REFER =
to URL/URN, but using MESSAGE instead.=20
=20
Works for me.
=20
=20


________________________________

	From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
	Sent: Monday, July 06, 2009 10:29
	To: Audet, Francois (SC100:3055); Salvatore Loreto; dispatch@ietf.org
	Cc: Henning Schulzrinne
	Subject: Re: [dispatch] Disaggregated Media in SIP
=09
=09
	>We've looked at various approaches to solve this important=20
	>problem several times before
=09
	Actually there is one more: IM-ing a URI to some resource, mentioned by =
Henning Schulzrinne (I don't recall the document or presentation).
=09
	My two cents is that IM-ing a URL is the most general solution, or is =
it?
=09
	Henry
=09
=09
	On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
=09
=09

		I'm glad to see this topic coming back.
	=09
		I see that this draft doesn't propose a solution to problem: it list
		three options, and describes why they are not adequate. I agree with
		the conclusions.
	=09
		We've looked at various approaches to solve this important problem
		several times before:
	=09
		- Feature ref (refer to urn: indicating specific features)
		  http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
	=09
		- Remote control using REFER to requests & responses
		  http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
		  (Also, versions -04, -03,-02, -00)
	=09
		- Remore control using REFER with XML body describing function
		  http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
	=09
		- Remote control using MBUS
		  http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
		  http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
	=09
		On top of that there are various proprietary mechanisms, and even some =
legacy
		PBX-CTI protocols.
	=09
		> -----Original Message-----
		> From: dispatch-bounces@ietf.org
		> [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore Loreto
		> Sent: Monday, July 06, 2009 09:33
		> To: dispatch@ietf.org
		> Subject: [dispatch] Disaggregated Media in SIP
		>
		> Hi there,
		>
		> I have just submitted a draft that talks of Disaggregated
		> Media in the Session Initiation Protocol (SIP).
		> http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
		ggregated-media-00.txt
		>
		>
		> Abstract:
		> Disaggregated media refers to the ability for a user to create a
		> multimedia session combining different media streams, coming from
		> different devices under his or her control, so that they are
		> treated by
		> the far end of the session as a single media session.
		> This document lists several use cases that involve
		> disaggregated media
		> in SIP.
		> Additionally, this document analyzes what types of
		> disaggregated media
		> can be implemented using existing protocol
		> mechanisms, and the pros and cons of using each of those mechanisms.
		> Finally, this document describes scenarios that are not covered by
		> current mechanisms
		> and proposes new IETF work to cover them.
		>
		>
		> cheers
		> Sal
		> _______________________________________________
		> 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
	=09
	=09


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<BODY>
<DIV><SPAN class=3D942393317-06072009><FONT face=3DArial color=3D#800000 =

size=3D2>Sure.</FONT></SPAN></DIV>
<DIV><SPAN class=3D942393317-06072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D942393317-06072009><FONT face=3DArial color=3D#800000 =
size=3D2>That=20
would be functionaly identical to the Feature referal using REFER to =
URL/URN,=20
but using MESSAGE instead. </FONT></SPAN></DIV>
<DIV><SPAN class=3D942393317-06072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D942393317-06072009><FONT face=3DArial color=3D#800000 =
size=3D2>Works=20
for me.</FONT></SPAN></DIV>
<DIV><SPAN class=3D942393317-06072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D942393317-06072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Henry Sinnreich=20
  [mailto:hsinnrei@adobe.com] <BR><B>Sent:</B> Monday, July 06, 2009=20
  10:29<BR><B>To:</B> Audet, Francois (SC100:3055); Salvatore Loreto;=20
  dispatch@ietf.org<BR><B>Cc:</B> Henning Schulzrinne<BR><B>Subject:</B> =
Re:=20
  [dispatch] Disaggregated Media in SIP<BR></FONT><BR></DIV>
  <DIV></DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
  style=3D"FONT-SIZE: 18pt">&gt;We've looked at various approaches to =
solve this=20
  important <BR>&gt;problem several times before<BR><BR>Actually there =
is one=20
  more: IM-ing a URI to some resource, mentioned by Henning Schulzrinne =
(I don=92t=20
  recall the document or presentation).<BR><BR>My two cents is that =
IM-ing a URL=20
  is the most general solution, or is it?<BR><BR>Henry<BR><BR><BR>On =
7/6/09=20
  12:07 PM, "Francois Audet" &lt;<A=20
  href=3D"audet@nortel.com">audet@nortel.com</A>&gt; =
wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">I'm glad to see this topic coming =
back.<BR><BR>I see=20
    that this draft doesn't propose a solution to problem: it =
list<BR>three=20
    options, and describes why they are not adequate. I agree =
with<BR>the=20
    conclusions.<BR><BR>We've looked at various approaches to solve this =

    important problem<BR>several times before:<BR><BR>- Feature ref =
(refer to=20
    urn: indicating specific features)<BR>&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00">ht=
tp://tools.ietf.org/html/draft-audet-sipping-feature-ref-00</A><BR><BR>- =

    Remote control using REFER to requests &amp; =
responses<BR>&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05">http://to=
ols.ietf.org/html/draft-mahy-sip-remote-cc-05</A><BR>&nbsp;&nbsp;(Also,=20
    versions -04, -03,-02, -00)<BR><BR>- Remore control using REFER with =
XML=20
    body describing function<BR>&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01">http://to=
ols.ietf.org/html/draft-mahy-sip-remote-cc-01</A><BR><BR>-=20
    Remote control using MBUS<BR>&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01">ht=
tp://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01</A>=20
    &amp;<BR>&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01">http://=
tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01</A><BR><BR>On=20
    top of that there are various proprietary mechanisms, and even some=20
    legacy<BR>PBX-CTI protocols.<BR><BR>&gt; -----Original =
Message-----<BR>&gt;=20
    From: <A=20
    =
href=3D"dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A><BR>&gt; =
[<A=20
    =
href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.or=
g</A>]=20
    On Behalf Of Salvatore Loreto<BR>&gt; Sent: Monday, July 06, 2009=20
    09:33<BR>&gt; To: <A =
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
    Subject: [dispatch] Disaggregated Media in SIP<BR>&gt;<BR>&gt; Hi=20
    there,<BR>&gt;<BR>&gt; I have just submitted a draft that talks of=20
    Disaggregated<BR>&gt; Media in the Session Initiation Protocol=20
    (SIP).<BR>&gt; <A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa">h=
ttp://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa</A><BR>ggre=
gated-media-00.txt<BR>&gt;<BR>&gt;<BR>&gt;=20
    Abstract:<BR>&gt; Disaggregated media refers to the ability for a =
user to=20
    create a<BR>&gt; multimedia session combining different media =
streams,=20
    coming from<BR>&gt; different devices under his or her control, so =
that they=20
    are<BR>&gt; treated by<BR>&gt; the far end of the session as a =
single media=20
    session.<BR>&gt; This document lists several use cases that =
involve<BR>&gt;=20
    disaggregated media<BR>&gt; in SIP.<BR>&gt; Additionally, this =
document=20
    analyzes what types of<BR>&gt; disaggregated media<BR>&gt; can be=20
    implemented using existing protocol<BR>&gt; mechanisms, and the pros =
and=20
    cons of using each of those mechanisms.<BR>&gt; Finally, this =
document=20
    describes scenarios that are not covered by<BR>&gt; current=20
    mechanisms<BR>&gt; and proposes new IETF work to cover=20
    them.<BR>&gt;<BR>&gt;<BR>&gt; cheers<BR>&gt; Sal<BR>&gt;=20
    _______________________________________________<BR>&gt; dispatch =
mailing=20
    list<BR>&gt; <A =
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt; <A=20
    =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;<BR>____________________________=
___________________<BR>dispatch=20
    mailing list<BR><A =
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR><A=20
    =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR><BR></SPAN></FONT></BLOCKQUOTE></BLO=
CKQUOTE></BODY></HTML>

------_=_NextPart_001_01C9FE60.20324DE8--

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Subject: Re: [dispatch] Disaggregated Media in SIP
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Sal,

Glad to see more focus on the objectives rather than the solutions.

A few comments:

> 2.6.  Answering a call using Two Separate Devices
> 
>    Laura has a PC with a softclient and a desk phone.  Bob has an
>    integrated advanced multimedia phone with camera.
> 
>    Laura receives on her desk phone an incoming voice-video call from
>    Bob.
> 
>    Laura decides to answer the Bob's session invitation by establishing
>    a voice session with Bob using the desk phone and the video session
>    using her multimedia phone. 

OK

>    Two SIP dialogs need to be established:
>    one between Bob's device and Laura's desk phone and one between Bob's
>    device and Laura's multimedia phone.

The above statement assumes the outcome that this draft is trying to 
make a case for.

>         ----                                 ----
>        | UA |\                             /| UA |
>         ----  \                           /  ----
>                \                         /
>       ----      \----               ----/    ----
>      | UA |-----| UA |-------------| UA |---| UA |
>       ----      /----     SIP       ----\    ----
>                /                         \
>         ----  /                           \  ----
>        | UA |/                             \| UA |
>         ----                                 ----
>         Alice                                 Bob
> 
> 
>                       Figure 2: Centralized scenario
> 
>    Figure 2 shows the generic signaling flow common to all centralized
>    solutions, where a Central Node is able to manage all signaling
>    messages needed to coordinate the different user's devices and hide
>    from the far end of the session all the mechanisms used to distribute
>    the media among different devices.

That is but one of many topologies. Others include:
- Alice has a multimedia device that supports all the media
- Alice has a device that supports *some* of the media and is
   also the controller for the other devices.
- Either of the above for Bob
- all permutations of approaches for Alice and Bob

The key is that there is a single sip session between caller and callee, 
which negotiates potentially many media sessions. The caller and callee 
may then each use whatever technique(s) desired to establish connection 
of the individual media sessions to particular devices. That may involve 
other sip signaling for some or all of the media, or not.

> 3.3.1.  3pcc issues

These are indeed issues to some extent. But its debatable whether they 
are significant issues, or how hard it is to mitigate them. To be fair 
we will have to look at issues for alternative solutions and balance them.

> 4.  Scenarios Not Covered by Existing Mechanisms
> 
>    As discussed previously, all existing mechanisms implement
>    disaggregated media using a centralized approach.  Scenarios not
>    covered by existing mechanisms include those where none of the nodes
>    can act as a controller because it does not support the necessary
>    functionality

Well, its clear that some devices will need to support some new 
functionality to aggregate the disaggregated media. Actually it seems 
that something new will need to be implemented at each end. So we are 
really only discussing *which* new functionality should be implemented 
at each end.

>    or because it will not participate in the whole session
>    (transferring the SIP dialog from a controller to a new one using
>    REFER and Replaces is complex and requires support from the far end).
>    These scenarios are better implemented using a more distributed
>    approach.

If it would be possible to do a transfer at all, it should be possible 
to do a transfer involving this disaggregated media controller.

>       /------------\
>      |     ---- ____|________________
>      |    | UA |    |                \
>      |     ----     |                 \ ------
>      |   ----       |                  |      |
>      |  | UA |-------------------------|  UA  |
>      |   ----       |                  |      |
>      |     ----     |                 / ------
>      |    | UA |____|________________/   Bob
>      |      ----    |
>       \------------/
>           Alice
> 
>    Figure 5 shows the generic signaling flow in a Distributed Scenario,
>    where all the signaling messages go from the different user's devices
>    to the far end of the session.

That shows some of the signaling paths. What it doesn't show is what is 
required to organize Alice's devices and decide which should be involved 
in the call, for which media. That is an important piece, unless it 
handled entirely by Bob.

Every one of the use cases will require some special processing and 
control signaling among the disaggregated devices before a call topology 
such as shown in figure 5 can be established.

Also, its far from clear what order of signaling would be required, and 
in which direction, to use the approach of figure 5 for each use case. 
Its quite possible that one call might be established from Bob to Alice 
and a supporting call from Alice to Bob. This would confuse 
responsibility for charging.

Another problem with figure 5 arises if it becomes necessary to transfer 
the resulting call, say from Bob to Charlie. There are then three calls 
that need to be transferred. Getting all that to happen smoothly will be 
a great challenge. Its hard enough if Charlie, like Bob, can handle all 
the media on a single device. If Charlie needs to aggregate some devices 
to handle all the media coming from Alice, then the problem becomes 
exceedingly complex.

Hence, IMO it remains far from clear that the new mechanism actually 
offers any benefits over the 3pcc mechanism. If you think it does 
(apparently you do) then I think it becomes necessary to compare the two 
in more detail. I think that requires nailing down in more detail how 
the important use cases should behave to an end user, and then how that 
behavior might be achieved using each technique. And then identifying 
gaps that need to be filled in each in order to have sufficient 
capability to achieve an equivalent and satisfactory result.

I think this analysis must also ensure that the resulting calls can work 
with other common features such as transfers, conferencing, "voice"mail, 
etc.

	Thanks,
	Paul

Salvatore Loreto wrote:
> Hi there,
> 
> I have just submitted a draft that talks of Disaggregated Media in the 
> Session Initiation Protocol (SIP).
> http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disaggregated-media-00.txt 
> 
> 
> 
> Abstract:
> Disaggregated media refers to the ability for a user to create a 
> multimedia session combining different media streams, coming from
> different devices under his or her control, so that they are treated by 
> the far end of the session as a single media session.
> This document lists several use cases that involve disaggregated media 
> in SIP.
> Additionally, this document analyzes what types of disaggregated media 
> can be implemented using existing protocol
> mechanisms, and the pros and cons of using each of those mechanisms.
> Finally, this document describes scenarios that are not covered by 
> current mechanisms
> and proposes new IETF work to cover them.
> 
> 
> cheers
> Sal
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From hsinnrei@adobe.com  Mon Jul  6 13:32:11 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Francois Audet <audet@nortel.com>, Salvatore Loreto <salvatore.loreto@ericsson.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 6 Jul 2009 10:28:57 -0700
Thread-Topic: [dispatch] Disaggregated Media in SIP
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--_000_C6779E89487Ehsinnreiadobecom_
Content-Type: text/plain; charset="iso-8859-1"
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>We've looked at various approaches to solve this important
>problem several times before

Actually there is one more: IM-ing a URI to some resource, mentioned by Hen=
ning Schulzrinne (I don't recall the document or presentation).

My two cents is that IM-ing a URL is the most general solution, or is it?

Henry


On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:

I'm glad to see this topic coming back.

I see that this draft doesn't propose a solution to problem: it list
three options, and describes why they are not adequate. I agree with
the conclusions.

We've looked at various approaches to solve this important problem
several times before:

- Feature ref (refer to urn: indicating specific features)
  http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00

- Remote control using REFER to requests & responses
  http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
  (Also, versions -04, -03,-02, -00)

- Remore control using REFER with XML body describing function
  http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01

- Remote control using MBUS
  http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
  http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01

On top of that there are various proprietary mechanisms, and even some lega=
cy
PBX-CTI protocols.

> -----Original Message-----
> From: dispatch-bounces@ietf.org
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore Loreto
> Sent: Monday, July 06, 2009 09:33
> To: dispatch@ietf.org
> Subject: [dispatch] Disaggregated Media in SIP
>
> Hi there,
>
> I have just submitted a draft that talks of Disaggregated
> Media in the Session Initiation Protocol (SIP).
> http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
ggregated-media-00.txt
>
>
> Abstract:
> Disaggregated media refers to the ability for a user to create a
> multimedia session combining different media streams, coming from
> different devices under his or her control, so that they are
> treated by
> the far end of the session as a single media session.
> This document lists several use cases that involve
> disaggregated media
> in SIP.
> Additionally, this document analyzes what types of
> disaggregated media
> can be implemented using existing protocol
> mechanisms, and the pros and cons of using each of those mechanisms.
> Finally, this document describes scenarios that are not covered by
> current mechanisms
> and proposes new IETF work to cover them.
>
>
> cheers
> Sal
> _______________________________________________
> 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


--_000_C6779E89487Ehsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [dispatch] Disaggregated Media in SIP</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>&gt;We've looked at various approaches to solve this important <BR>
&gt;problem several times before<BR>
<BR>
Actually there is one more: IM-ing a URI to some resource, mentioned by Hen=
ning Schulzrinne (I don&#8217;t recall the document or presentation).<BR>
<BR>
My two cents is that IM-ing a URL is the most general solution, or is it?<B=
R>
<BR>
Henry<BR>
<BR>
<BR>
On 7/6/09 12:07 PM, &quot;Francois Audet&quot; &lt;<a href=3D"audet@nortel.=
com">audet@nortel.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>I'm glad to see this topic coming back.<BR>
<BR>
I see that this draft doesn't propose a solution to problem: it list<BR>
three options, and describes why they are not adequate. I agree with<BR>
the conclusions.<BR>
<BR>
We've looked at various approaches to solve this important problem<BR>
several times before:<BR>
<BR>
- Feature ref (refer to urn: indicating specific features)<BR>
&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/html/draft-audet-sipping-featu=
re-ref-00">http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00</a=
><BR>
<BR>
- Remote control using REFER to requests &amp; responses<BR>
&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-=
05">http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05</a><BR>
&nbsp;&nbsp;(Also, versions -04, -03,-02, -00)<BR>
<BR>
- Remore control using REFER with XML body describing function<BR>
&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-=
01">http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01</a><BR>
<BR>
- Remote control using MBUS<BR>
&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-re=
motecc-01">http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01</a=
> &amp;<BR>
&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sd=
p-01">http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01</a><BR>
<BR>
On top of that there are various proprietary mechanisms, and even some lega=
cy<BR>
PBX-CTI protocols.<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: <a href=3D"dispatch-bounces@ietf.org">dispatch-bounces@ietf.org<=
/a><BR>
&gt; [<a href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@=
ietf.org</a>] On Behalf Of Salvatore Loreto<BR>
&gt; Sent: Monday, July 06, 2009 09:33<BR>
&gt; To: <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt; Subject: [dispatch] Disaggregated Media in SIP<BR>
&gt;<BR>
&gt; Hi there,<BR>
&gt;<BR>
&gt; I have just submitted a draft that talks of Disaggregated<BR>
&gt; Media in the Session Initiation Protocol (SIP).<BR>
&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-loreto-dispatch-d=
isa">http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa</a><BR>
ggregated-media-00.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; Abstract:<BR>
&gt; Disaggregated media refers to the ability for a user to create a<BR>
&gt; multimedia session combining different media streams, coming from<BR>
&gt; different devices under his or her control, so that they are<BR>
&gt; treated by<BR>
&gt; the far end of the session as a single media session.<BR>
&gt; This document lists several use cases that involve<BR>
&gt; disaggregated media<BR>
&gt; in SIP.<BR>
&gt; Additionally, this document analyzes what types of<BR>
&gt; disaggregated media<BR>
&gt; can be implemented using existing protocol<BR>
&gt; mechanisms, and the pros and cons of using each of those mechanisms.<B=
R>
&gt; Finally, this document describes scenarios that are not covered by<BR>
&gt; current mechanisms<BR>
&gt; and proposes new IETF work to cover them.<BR>
&gt;<BR>
&gt;<BR>
&gt; cheers<BR>
&gt; Sal<BR>
&gt; _______________________________________________<BR>
&gt; dispatch mailing list<BR>
&gt; <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www=
.ietf.org/mailman/listinfo/dispatch</a><BR>
&gt;<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf=
.org/mailman/listinfo/dispatch</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
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Subject: Re: [dispatch] Disaggregated Media in SIP
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Henry Sinnreich wrote:
>>We've looked at various approaches to solve this important
>>problem several times before
> 
> Actually there is one more: IM-ing a URI to some resource, mentioned by 
> Henning Schulzrinne (I don’t recall the document or presentation).
> 
> My two cents is that IM-ing a URL is the most general solution, or is it?

Past suggestions by various people to send control signals (intended to 
be acted upon by automata rather than people) via IM have generally been 
rejected as inappropriate. (The exception so far has been file transfer, 
which has some control behavior and some expected human interaction.)

Now if you just want to say "Bob, please make a video call to 
sip:alice_camera@alice.com in order to see me" then I guess IM is ok. 
But IMO its not otherwise good. Its just a hack.

	Thanks,
	Paul

> Henry
> 
> 
> On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
> 
>     I'm glad to see this topic coming back.
> 
>     I see that this draft doesn't propose a solution to problem: it list
>     three options, and describes why they are not adequate. I agree with
>     the conclusions.
> 
>     We've looked at various approaches to solve this important problem
>     several times before:
> 
>     - Feature ref (refer to urn: indicating specific features)
>       http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
> 
>     - Remote control using REFER to requests & responses
>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
>       (Also, versions -04, -03,-02, -00)
> 
>     - Remore control using REFER with XML body describing function
>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
> 
>     - Remote control using MBUS
>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
> 
>     On top of that there are various proprietary mechanisms, and even
>     some legacy
>     PBX-CTI protocols.
> 
>     >  -----Original Message-----
>     >  From: dispatch-bounces@ietf.org
>     >  [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore Loreto
>     >  Sent: Monday, July 06, 2009 09:33
>     >  To: dispatch@ietf.org
>     >  Subject: [dispatch] Disaggregated Media in SIP
>     >
>     >  Hi there,
>     >
>     >  I have just submitted a draft that talks of Disaggregated
>     >  Media in the Session Initiation Protocol (SIP).
>     >  http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
>     ggregated-media-00.txt
>     >
>     >
>     >  Abstract:
>     >  Disaggregated media refers to the ability for a user to create a
>     >  multimedia session combining different media streams, coming from
>     >  different devices under his or her control, so that they are
>     >  treated by
>     >  the far end of the session as a single media session.
>     >  This document lists several use cases that involve
>     >  disaggregated media
>     >  in SIP.
>     >  Additionally, this document analyzes what types of
>     >  disaggregated media
>     >  can be implemented using existing protocol
>     >  mechanisms, and the pros and cons of using each of those mechanisms.
>     >  Finally, this document describes scenarios that are not covered by
>     >  current mechanisms
>     >  and proposes new IETF work to cover them.
>     >
>     >
>     >  cheers
>     >  Sal
>     >  _______________________________________________
>     >  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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To: Vijay Gurbani <vkg@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 6 Jul 2009 18:48:04 -0400
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Subject: Re: [dispatch] Session Recording in SIP
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All: The following draft on SIP session recording requirements was submitte=
d today:

http://www.ietf.org/internet-drafts/draft-jain-dispatch-session-recording-p=
rotocol-req-00.txt

Title:           Requirements for Session Recording Protocol (SRP)

Abstract:
Session recording is a critical requirement in many business
communications environments such as call centers and financial
trading floors.  In some of these environments, all calls must be
recorded for regulatory and compliance reasons.  In others, calls may
be recorded for quality control or business analytics.  Recording is
typically done by sending a copy of the media to the recording
devices.  This document specifies requirements for a protocol that
will manage delivery of media from an end-point that originates media
or that has access to it to a recording device.  This protocol is
being referred to as Session Recording Protocol and will most likely
be based on SIP.




> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf
> Of Vijay Gurbani
> Sent: Monday, June 22, 2009 12:09 AM
> To: dispatch@ietf.org
> Subject: [dispatch] Summary on Session Recording in SIP
>
> All: Here is the summary of the Session Recording thread from June 9
> to June 19, 2009.  If I misrepresented anything, please let me know.
>
> There was a fair amount of discussion on where to conduct this
> work.  In the end, it was decided to continue discussions on
> dispatch, further decisions what to do -- new working group or
> put this work in an existing working group -- will be decided
> later.  Regardless of where the work is done, opinions were
> expressed that the work is important that it should commence
> in the IETF.
>
> Need to be crisp about the purpose of this work: is it that users
> record sessions to hear later, or sessions need to be recorded
> because of some business needs?  Or both?  Is SRTP covered or
> only RTP?
>
> Solutions should cover:
>   Always on recording
>   Recording on demand
>   Required recording
>   Pause and resume recording
>   Playback facilities and how they interact with recording
>   Recording formats and protocols for controlling the stored recording.
>
> A lot of discussion ensued on the specific architecture or
> architectures that would be possible.  No decision was reached on
> which specific architecture is good or bad, although various opinions
> were expressed in support for each.
>
> There are essentially four architectures to realize recording
> services: two that discuss where the media stream is forked
> out to the recorder -- at the UA or in a middlebox of some
> sort are outlined here:
>    http://www.ietf.org/mail-archive/web/dispatch/current/msg00226.html
>
> The third architecture has both UAs sending media to a
> recorder.  See:
>      http://www.ietf.org/mail-archive/web/dispatch/current/msg00236.html
>
> A fourth architecture that is a superset of the third one is at:
>      http://www.ietf.org/mail-archive/web/dispatch/current/msg00256.html
>
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> WWW:   http://ect.bell-labs.com/who/vkg
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Preconditions and intermediaries
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> 2.  Starting Using the Media Parameters Subject to Preconditions
...
>    However, the fact that the UAs can start using the new media
>    parameters does not mean that they need to start using them
>    immediately.  When preconditions are used, the UAS decides when to
>    start using them.  

Why do you say that?

For one thing offerer and answerer have more significance than UAC and 
UAS in any such decision.

For another, QoS needs to be obtained by both ends before it can be used 
by either end. The order in which it is obtained, and signaled that it 
has been obtained, is indeterminate.

When one end has been told that the other end has obtained it, and knows 
that it has obtained it, then it can start using it.

But that doesn't change the point about signaling the intent to start using.

	Thanks,
	Paul

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Thread-Topic: [dispatch] Disaggregated Media in SIP
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References: <4A5261E2.4050506@cisco.com> <C677FFD1.48EB%hsinnrei@adobe.com>
From: "Francois Audet" <audet@nortel.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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I think what Paul calls automata is the application on the IM client, so =
that would undermine what this spec and all of us in the Enterprise =
space have been trying to do for years.
=20
I will note that the "istyping" indication is already done today with =
MESSAGE. And the istyping indicator is certainly an automata. And that =
is an RFC today, and is widly deployed.
=20
I personally don't really care if its a MESSAGE, a REFER, or an INFO =
(although we certainly can rule out MBUS). Or a new message.
=20
I don't think "other protocols" is a good answer: it has to be routable =
just like SIP.


________________________________

	From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
	Sent: Monday, July 06, 2009 17:24
	To: Paul Kyzivat
	Cc: Audet, Francois (SC100:3055); Salvatore Loreto; dispatch@ietf.org
	Subject: Re: [dispatch] Disaggregated Media in SIP
=09
=09
	Paul Kyzivat wrote:
	>Past suggestions by various people to send control signals (intended =
tobe acted upon=20
	>by automata rather than >people) via IM have generally been
	>rejected as inappropriate.=20
=09
	I am not sure how many people expect a usage scenario for IM with an =
automata in the middle or=20
	what the deployment statistics are for such automata (I have never =
encountered one). =20
=09
	All SIP (or other protocol ) Communicator packages have IM and the URI =
works there very nicely.
=09
	Do you have any usage statistics that justifies the assertion automata =
are the=20
	key usage scenario and "plain person to person" IM does not count?
=09
	Henry
=09
=09
	On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
=09
=09

	=09
	=09
	=09
		Henry Sinnreich wrote:
		>>We've looked at various approaches to solve this important
		>>problem several times before
		>
		> Actually there is one more: IM-ing a URI to some resource, mentioned =
by
		> Henning Schulzrinne (I don't recall the document or presentation).
		>
		> My two cents is that IM-ing a URL is the most general solution, or =
is it?
	=09
		Past suggestions by various people to send control signals (intended =
to
		be acted upon by automata rather than people) via IM have generally =
been
		rejected as inappropriate. (The exception so far has been file =
transfer,
		which has some control behavior and some expected human interaction.)
	=09
		Now if you just want to say "Bob, please make a video call to
		sip:alice_camera@alice.com in order to see me" then I guess IM is ok.
		But IMO its not otherwise good. Its just a hack.
	=09
		        Thanks,
		        Paul
	=09
		> Henry
		>
		>
		> On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
		>
		>     I'm glad to see this topic coming back.
		>
		>     I see that this draft doesn't propose a solution to problem: it =
list
		>     three options, and describes why they are not adequate. I agree =
with
		>     the conclusions.
		>
		>     We've looked at various approaches to solve this important =
problem
		>     several times before:
		>
		>     - Feature ref (refer to urn: indicating specific features)
		>       http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
		>
		>     - Remote control using REFER to requests & responses
		>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
		>       (Also, versions -04, -03,-02, -00)
		>
		>     - Remore control using REFER with XML body describing function
		>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
		>
		>     - Remote control using MBUS
		>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 =
&
		>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
		>
		>     On top of that there are various proprietary mechanisms, and =
even
		>     some legacy
		>     PBX-CTI protocols.
		>
		>     >  -----Original Message-----
		>     >  From: dispatch-bounces@ietf.org
		>     >  [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore =
Loreto
		>     >  Sent: Monday, July 06, 2009 09:33
		>     >  To: dispatch@ietf.org
		>     >  Subject: [dispatch] Disaggregated Media in SIP
		>     >
		>     >  Hi there,
		>     >
		>     >  I have just submitted a draft that talks of Disaggregated
		>     >  Media in the Session Initiation Protocol (SIP).
		>     >  =
http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
		>     ggregated-media-00.txt
		>     >
		>     >
		>     >  Abstract:
		>     >  Disaggregated media refers to the ability for a user to =
create a
		>     >  multimedia session combining different media streams, coming =
from
		>     >  different devices under his or her control, so that they are
		>     >  treated by
		>     >  the far end of the session as a single media session.
		>     >  This document lists several use cases that involve
		>     >  disaggregated media
		>     >  in SIP.
		>     >  Additionally, this document analyzes what types of
		>     >  disaggregated media
		>     >  can be implemented using existing protocol
		>     >  mechanisms, and the pros and cons of using each of those =
mechanisms.
		>     >  Finally, this document describes scenarios that are not =
covered by
		>     >  current mechanisms
		>     >  and proposes new IETF work to cover them.
		>     >
		>     >
		>     >  cheers
		>     >  Sal
		>     >  _______________________________________________
		>     >  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
	=09
	=09


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<HTML><HEAD><TITLE>Re: [dispatch] Disaggregated Media in SIP</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
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<META content=3D"MSHTML 6.00.2900.3562" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D970023600-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
think what Paul calls automata is the application on the IM client, so =
that=20
would undermine what this spec and all of us in the Enterprise space =
have been=20
trying to do for years.</FONT></SPAN></DIV>
<DIV><SPAN class=3D970023600-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D970023600-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>I will=20
note that the "istyping" indication is already done today with MESSAGE. =
And the=20
istyping indicator is certainly an automata. And that is an RFC today, =
and is=20
widly deployed.</FONT></SPAN></DIV>
<DIV><SPAN class=3D970023600-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D970023600-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
personally don't really care if its a MESSAGE, a REFER, or an INFO =
(although we=20
certainly can rule out MBUS). Or a new message.</FONT></SPAN></DIV>
<DIV><SPAN class=3D970023600-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D970023600-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
don't think "other protocols" is a good answer: it has to be routable =
just like=20
SIP.</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Henry Sinnreich=20
  [mailto:hsinnrei@adobe.com] <BR><B>Sent:</B> Monday, July 06, 2009=20
  17:24<BR><B>To:</B> Paul Kyzivat<BR><B>Cc:</B> Audet, Francois =
(SC100:3055);=20
  Salvatore Loreto; dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch]=20
  Disaggregated Media in SIP<BR></FONT><BR></DIV>
  <DIV></DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
  style=3D"FONT-SIZE: 18pt">Paul Kyzivat wrote:<BR>&gt;Past suggestions =
by various=20
  people to send control signals (intended tobe acted upon <BR>&gt;by =
automata=20
  rather than &gt;people) via IM have generally been<BR>&gt;rejected as=20
  inappropriate. <BR><BR>I am not sure how many people expect a usage =
scenario=20
  for IM with an automata in the middle or <BR>what the deployment =
statistics=20
  are for such automata (I have never encountered one). =
&nbsp;<BR><BR>All SIP=20
  (or other protocol ) Communicator packages have IM and the URI works =
there=20
  very nicely.<BR><BR>Do you have any usage statistics that justifies =
the=20
  assertion automata are the <BR>key usage scenario and =93plain person =
to person=94=20
  IM does not count?<BR><BR>Henry<BR><BR><BR>On 7/6/09 3:43 PM, "Paul =
Kyzivat"=20
  &lt;<A href=3D"pkyzivat@cisco.com">pkyzivat@cisco.com</A>&gt;=20
  wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt"><BR><BR><BR>Henry Sinnreich =
wrote:<BR>&gt;&gt;We've=20
    looked at various approaches to solve this =
important<BR>&gt;&gt;problem=20
    several times before<BR>&gt;<BR>&gt; Actually there is one more: =
IM-ing a=20
    URI to some resource, mentioned by<BR>&gt; Henning Schulzrinne (I =
don=92t=20
    recall the document or presentation).<BR>&gt;<BR>&gt; My two cents =
is that=20
    IM-ing a URL is the most general solution, or is it?<BR><BR>Past =
suggestions=20
    by various people to send control signals (intended to<BR>be acted =
upon by=20
    automata rather than people) via IM have generally been<BR>rejected =
as=20
    inappropriate. (The exception so far has been file =
transfer,<BR>which has=20
    some control behavior and some expected human =
interaction.)<BR><BR>Now if=20
    you just want to say "Bob, please make a video call to<BR>sip:<A=20
    href=3D"alice_camera@alice.com">alice_camera@alice.com</A> in order =
to see me"=20
    then I guess IM is ok.<BR>But IMO its not otherwise good. Its just a =

    =
hack.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Thanks,<BR>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Paul<BR><BR>&gt;=20
    Henry<BR>&gt;<BR>&gt;<BR>&gt; On 7/6/09 12:07 PM, "Francois Audet" =
&lt;<A=20
    href=3D"audet@nortel.com">audet@nortel.com</A>&gt; =
wrote:<BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;I'm glad to see this topic coming=20
    back.<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;I see that this draft =
doesn't=20
    propose a solution to problem: it list<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;three=20
    options, and describes why they are not adequate. I agree =
with<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;the conclusions.<BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;We've looked at various approaches to solve =
this=20
    important problem<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;several times=20
    before:<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Feature ref (refer =
to urn:=20
    indicating specific features)<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00">ht=
tp://tools.ietf.org/html/draft-audet-sipping-feature-ref-00</A><BR>&gt;<B=
R>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;- Remote control using REFER to requests =
&amp;=20
    responses<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05">http://to=
ols.ietf.org/html/draft-mahy-sip-remote-cc-05</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Also, versions -04, -03,-02,=20
    -00)<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Remore control using =
REFER=20
    with XML body describing function<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01">http://to=
ols.ietf.org/html/draft-mahy-sip-remote-cc-01</A><BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;- Remote control using MBUS<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01">ht=
tp://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01</A>=20
    &amp;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01">http://=
tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01</A><BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;On top of that there are various proprietary =

    mechanisms, and even<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;some =
legacy<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;PBX-CTI protocols.<BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;-----Original =
Message-----<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;From: <A=20
    =
href=3D"dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A><BR>&gt; =

    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;[<A=20
    =
href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.or=
g</A>]=20
    On Behalf Of Salvatore Loreto<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Sent: Monday, July 06, 2009 09:33<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;To: <A =
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Subject: [dispatch] Disaggregated =
Media=20
    in SIP<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Hi there,<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;I=20
    have just submitted a draft that talks of Disaggregated<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Media in the Session Initiation =
Protocol=20
    (SIP).<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa">h=
ttp://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa</A><BR>&gt;=
=20
    &nbsp;&nbsp;&nbsp;&nbsp;ggregated-media-00.txt<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Abstract:<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Disaggregated media refers to the =
ability=20
    for a user to create a<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;multimedia=20
    session combining different media streams, coming from<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;different devices under his or =
her=20
    control, so that they are<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;treated=20
    by<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;the far end of the =
session as=20
    a single media session.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;This=20
    document lists several use cases that involve<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;disaggregated media<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;in SIP.<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Additionally, this document =
analyzes what=20
    types of<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;disaggregated=20
    media<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;can be implemented =
using=20
    existing protocol<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;mechanisms, and=20
    the pros and cons of using each of those mechanisms.<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Finally, this document describes=20
    scenarios that are not covered by<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;current mechanisms<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;and=20
    proposes new IETF work to cover them.<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;cheers<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Sal<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;_______________________________________________<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;dispatch mailing list<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
    href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
    =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    =
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R>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;dispatch mailing list<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;<A=20
    href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
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    _______________________________________________<BR>&gt; dispatch =
mailing=20
    list<BR>&gt; <A =
href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt; <A=20
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Francois Audet <audet@nortel.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 6 Jul 2009 17:54:47 -0700
Thread-Topic: [dispatch] Disaggregated Media in SIP
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Francois Audet wrote:
>spec and all of us in the Enterprise space have been trying to do for year=
s.

Now this makes sense.
But still, can you share some data of its real life usage, as compared to w=
hat is out today in various SIP Communicators? That would rule our the pers=
on-to-person IM scenarios?

>I don't think "other protocols" is a good answer: it has to be routable ju=
st like SIP.
This again makes sense to me.
(neglecting the facts about Jabber IM [such as in the IETF], Skype,  etc.)

Henry


On 7/6/09 7:41 PM, "Francois Audet" <audet@nortel.com> wrote:

I think what Paul calls automata is the application on the IM client, so th=
at would undermine what this spec and all of us in the Enterprise space hav=
e been trying to do for years.

I will note that the "istyping" indication is already done today with MESSA=
GE. And the istyping indicator is certainly an automata. And that is an RFC=
 today, and is widly deployed.

I personally don't really care if its a MESSAGE, a REFER, or an INFO (altho=
ugh we certainly can rule out MBUS). Or a new message.

I don't think "other protocols" is a good answer: it has to be routable jus=
t like SIP.



________________________________
From: Henry Sinnreich  [mailto:hsinnrei@adobe.com]
Sent: Monday, July 06, 2009  17:24
To: Paul Kyzivat
Cc: Audet, Francois (SC100:3055);  Salvatore Loreto; dispatch@ietf.org
Subject: Re: [dispatch]  Disaggregated Media in SIP


Paul Kyzivat wrote:
>Past suggestions by various  people to send control signals (intended tobe=
 acted upon
>by automata  rather than >people) via IM have generally been
>rejected as  inappropriate.

I am not sure how many people expect a usage scenario  for IM with an autom=
ata in the middle or
what the deployment statistics  are for such automata (I have never encount=
ered one).

All SIP  (or other protocol ) Communicator packages have IM and the URI wor=
ks there  very nicely.

Do you have any usage statistics that justifies the  assertion automata are=
 the
key usage scenario and "plain person to person"  IM does not count?

Henry


On 7/6/09 3:43 PM, "Paul Kyzivat"  <pkyzivat@cisco.com>  wrote:





Henry Sinnreich wrote:
>>We've  looked at various approaches to solve this important
>>problem  several times before
>
> Actually there is one more: IM-ing a  URI to some resource, mentioned by
> Henning Schulzrinne (I don't  recall the document or presentation).
>
> My two cents is that  IM-ing a URL is the most general solution, or is it=
?

Past suggestions  by various people to send control signals (intended to
be acted upon by  automata rather than people) via IM have generally been
rejected as  inappropriate. (The exception so far has been file transfer,
which has  some control behavior and some expected human interaction.)

Now if  you just want to say "Bob, please make a video call to
sip:alice_camera@alice.com in order to see me"  then I guess IM is ok.
But IMO its not otherwise good. Its just a  hack.

        Thanks,
        Paul

>  Henry
>
>
> On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
>
>      I'm glad to see this topic coming  back.
>
>     I see that this draft doesn't  propose a solution to problem: it list
>     three  options, and describes why they are not adequate. I agree with
>      the conclusions.
>
>      We've looked at various approaches to solve this  important problem
>     several times  before:
>
>     - Feature ref (refer to urn:  indicating specific features)
>       http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
>
>      - Remote control using REFER to requests &  responses
>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
>        (Also, versions -04, -03,-02,  -00)
>
>     - Remore control using REFER  with XML body describing function
>        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
>
>      - Remote control using MBUS
>        http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01  &
>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
>
>      On top of that there are various proprietary  mechanisms, and even
>     some legacy
>      PBX-CTI protocols.
>
>      >  -----Original Message-----
>      >  From: dispatch-bounces@ietf.org
>      >  [mailto:dispatch-bounces@ietf.org]  On Behalf Of Salvatore Loreto
>     >   Sent: Monday, July 06, 2009 09:33
>     >   To: dispatch@ietf.org
>      >  Subject: [dispatch] Disaggregated Media  in SIP
>     >
>      >  Hi there,
>      >
>     >  I  have just submitted a draft that talks of Disaggregated
>      >  Media in the Session Initiation Protocol  (SIP).
>     >  http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
>      ggregated-media-00.txt
>      >
>     >
>      >  Abstract:
>      >  Disaggregated media refers to the ability  for a user to create a
>     >  multimedia  session combining different media streams, coming from
>      >  different devices under his or her  control, so that they are
>     >  treated  by
>     >  the far end of the session as  a single media session.
>     >  This  document lists several use cases that involve
>      >  disaggregated media
>      >  in SIP.
>      >  Additionally, this document analyzes what  types of
>     >  disaggregated  media
>     >  can be implemented using  existing protocol
>     >  mechanisms, and  the pros and cons of using each of those mechanis=
ms.
>      >  Finally, this document describes  scenarios that are not covered =
by
>     >   current mechanisms
>     >  and  proposes new IETF work to cover them.
>      >
>     >
>      >  cheers
>      >  Sal
>     >   _______________________________________________
>      >  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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<HTML>
<HEAD>
<TITLE>Re: [dispatch] Disaggregated Media in SIP</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Francois Audet wrote:<BR>
&gt;</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#7F0000"><F=
ONT FACE=3D"Arial">spec and all of us in the Enterprise space have been try=
ing to do for years.<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
Now this makes sense. <BR>
But still, can you share some data of its real life usage, as compared to w=
hat is out today in various SIP Communicators? That would rule our the pers=
on-to-person IM scenarios?<BR>
<BR>
&gt;</FONT><FONT COLOR=3D"#7F0000"><FONT FACE=3D"Arial">I don't think &quot=
;other protocols&quot; is a good answer: it has to be routable just like SI=
P.<BR>
</FONT></FONT><FONT FACE=3D"Arial">This again makes sense to me.<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">(neglecting the fa=
cts about Jabber IM [such as in the IETF], Skype, &nbsp;etc.)<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 7/6/09 7:41 PM, &quot;Francois Audet&quot; &lt;<a href=3D"audet@nortel.c=
om">audet@nortel.com</a>&gt; wrote:<BR>
<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#80=
0000"><FONT FACE=3D"Arial">I think what Paul calls automata is the applicat=
ion on the IM client, so that would undermine what this spec and all of us =
in the Enterprise space have been trying to do for years.<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT><FONT COLOR=3D"#800000"><FONT FACE=3D"Arial">I will note that the &q=
uot;istyping&quot; indication is already done today with MESSAGE. And the i=
styping indicator is certainly an automata. And that is an RFC today, and i=
s widly deployed.<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT><FONT COLOR=3D"#800000"><FONT FACE=3D"Arial">I personally don't real=
ly care if its a MESSAGE, a REFER, or an INFO (although we certainly can ru=
le out MBUS). Or a new message.<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT><FONT COLOR=3D"#800000"><FONT FACE=3D"Arial">I don't think &quot;oth=
er protocols&quot; is a good answer: it has to be routable just like SIP.<B=
R>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"> <BR>
&nbsp;<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"100%"> </FONT><FONT FACE=3D"Tahoma, =
Verdana, Helvetica, Arial"><B>From:</B> Henry Sinnreich &nbsp;[<a href=3D"m=
ailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com</a>] <BR>
<B>Sent:</B> Monday, July 06, 2009 &nbsp;17:24<BR>
<B>To:</B> Paul Kyzivat<BR>
<B>Cc:</B> Audet, Francois (SC100:3055); &nbsp;Salvatore Loreto; <a href=3D=
"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<B>Subject:</B> Re: [dispatch] &nbsp;Disaggregated Media in SIP<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
&nbsp;<BR>
Paul Kyzivat wrote:<BR>
&gt;Past suggestions by various &nbsp;people to send control signals (inten=
ded tobe acted upon <BR>
&gt;by automata &nbsp;rather than &gt;people) via IM have generally been<BR=
>
&gt;rejected as &nbsp;inappropriate. <BR>
<BR>
I am not sure how many people expect a usage scenario &nbsp;for IM with an =
automata in the middle or <BR>
what the deployment statistics &nbsp;are for such automata (I have never en=
countered one). &nbsp;<BR>
<BR>
All SIP &nbsp;(or other protocol ) Communicator packages have IM and the UR=
I works there &nbsp;very nicely.<BR>
<BR>
Do you have any usage statistics that justifies the &nbsp;assertion automat=
a are the <BR>
key usage scenario and &#8220;plain person to person&#8221; &nbsp;IM does n=
ot count?<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 7/6/09 3:43 PM, &quot;Paul Kyzivat&quot; &nbsp;&lt;<a href=3D"pkyzivat@c=
isco.com">pkyzivat@cisco.com</a>&gt; &nbsp;wrote:<BR>
<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"><BR>
<BR>
<BR>
Henry Sinnreich wrote:<BR>
&gt;&gt;We've &nbsp;looked at various approaches to solve this important<BR=
>
&gt;&gt;problem &nbsp;several times before<BR>
&gt;<BR>
&gt; Actually there is one more: IM-ing a &nbsp;URI to some resource, menti=
oned by<BR>
&gt; Henning Schulzrinne (I don&#8217;t &nbsp;recall the document or presen=
tation).<BR>
&gt;<BR>
&gt; My two cents is that &nbsp;IM-ing a URL is the most general solution, =
or is it?<BR>
<BR>
Past suggestions &nbsp;by various people to send control signals (intended =
to<BR>
be acted upon by &nbsp;automata rather than people) via IM have generally b=
een<BR>
rejected as &nbsp;inappropriate. (The exception so far has been file transf=
er,<BR>
which has &nbsp;some control behavior and some expected human interaction.)=
<BR>
<BR>
Now if &nbsp;you just want to say &quot;Bob, please make a video call to<BR=
>
sip:<a href=3D"alice_camera@alice.com">alice_camera@alice.com</a> in order =
to see me&quot; &nbsp;then I guess IM is ok.<BR>
But IMO its not otherwise good. Its just a &nbsp;hack.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Thanks,<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Paul<BR>
<BR>
&gt; &nbsp;Henry<BR>
&gt;<BR>
&gt;<BR>
&gt; On 7/6/09 12:07 PM, &quot;Francois Audet&quot; &lt;<a href=3D"audet@no=
rtel.com">audet@nortel.com</a>&gt; wrote:<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;I'm glad to see this topic coming &nbsp;=
back.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;I see that this draft doesn't &nbsp;propose a =
solution to problem: it list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;three &nbsp;options, and describes why they ar=
e not adequate. I agree with<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;the conclusions.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We've looked at various approaches to so=
lve this &nbsp;important problem<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;several times &nbsp;before:<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Feature ref (refer to urn: &nbsp;indicating =
specific features)<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-audet-sipping-feature-ref-00">http://tools.ietf.org/html/draft-au=
det-sipping-feature-ref-00</a><BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remote control using REFER to requests=
 &amp; &nbsp;responses<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-mahy-sip-remote-cc-05">http://tools.ietf.org/html/draft-mahy-sip-=
remote-cc-05</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Also, versions -04, -03,-02=
, &nbsp;-00)<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Remore control using REFER &nbsp;with XML bo=
dy describing function<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf=
.org/html/draft-mahy-sip-remote-cc-01">http://tools.ietf.org/html/draft-mah=
y-sip-remote-cc-01</a><BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remote control using MBUS<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf=
.org/html/draft-mahy-mmusic-mbus-remotecc-01">http://tools.ietf.org/html/dr=
aft-mahy-mmusic-mbus-remotecc-01</a> &nbsp;&amp;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-mahy-mmusic-mbus-sdp-01">http://tools.ietf.org/html/draft-mahy-mm=
usic-mbus-sdp-01</a><BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;On top of that there are various proprie=
tary &nbsp;mechanisms, and even<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;some legacy<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;PBX-CTI protocols.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;-----Original Message-----<BR=
>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;From: <a href=3D"dispatch-bou=
nces@ietf.org">dispatch-bounces@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;[<a href=3D"mailto:dispatch-b=
ounces@ietf.org">mailto:dispatch-bounces@ietf.org</a>] &nbsp;On Behalf Of S=
alvatore Loreto<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;Sent: Monday, July 06, 2009 0=
9:33<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;To: <a href=3D"dispatch@ietf.=
org">dispatch@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Subject: [dispatch] Disaggreg=
ated Media &nbsp;in SIP<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Hi there,<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;I &nbsp;have just submitted a draft=
 that talks of Disaggregated<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Media in the Session Initiati=
on Protocol &nbsp;(SIP).<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<a href=3D"http://www.ietf.org/inte=
rnet-drafts/draft-loreto-dispatch-disa">http://www.ietf.org/internet-drafts=
/draft-loreto-dispatch-disa</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;ggregated-media-00.txt<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Abstract:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Disaggregated media refers to=
 the ability &nbsp;for a user to create a<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;multimedia &nbsp;session combining =
different media streams, coming from<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;different devices under his o=
r her &nbsp;control, so that they are<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;treated &nbsp;by<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;the far end of the session as &nbsp=
;a single media session.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;This &nbsp;document lists several u=
se cases that involve<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;disaggregated media<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;in SIP.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Additionally, this document a=
nalyzes what &nbsp;types of<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;disaggregated &nbsp;media<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;can be implemented using &nbsp;exis=
ting protocol<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;mechanisms, and &nbsp;the pros and =
cons of using each of those mechanisms.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Finally, this document descri=
bes &nbsp;scenarios that are not covered by<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;current mechanisms<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;and &nbsp;proposes new IETF work to=
 cover them.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;cheers<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Sal<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;_____________________________=
__________________<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;dispatch mailing list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<a href=3D"dispatch@ietf.org"=
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 6 Jul 2009 17:24:01 -0700
Thread-Topic: [dispatch] Disaggregated Media in SIP
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Paul Kyzivat wrote:
>Past suggestions by various people to send control signals (intended tobe =
acted upon
>by automata rather than >people) via IM have generally been
>rejected as inappropriate.

I am not sure how many people expect a usage scenario for IM with an automa=
ta in the middle or
what the deployment statistics are for such automata (I have never encounte=
red one).

All SIP (or other protocol ) Communicator packages have IM and the URI work=
s there very nicely.

Do you have any usage statistics that justifies the assertion automata are =
the
key usage scenario and "plain person to person" IM does not count?

Henry


On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:




Henry Sinnreich wrote:
>>We've looked at various approaches to solve this important
>>problem several times before
>
> Actually there is one more: IM-ing a URI to some resource, mentioned by
> Henning Schulzrinne (I don't recall the document or presentation).
>
> My two cents is that IM-ing a URL is the most general solution, or is it?

Past suggestions by various people to send control signals (intended to
be acted upon by automata rather than people) via IM have generally been
rejected as inappropriate. (The exception so far has been file transfer,
which has some control behavior and some expected human interaction.)

Now if you just want to say "Bob, please make a video call to
sip:alice_camera@alice.com in order to see me" then I guess IM is ok.
But IMO its not otherwise good. Its just a hack.

        Thanks,
        Paul

> Henry
>
>
> On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
>
>     I'm glad to see this topic coming back.
>
>     I see that this draft doesn't propose a solution to problem: it list
>     three options, and describes why they are not adequate. I agree with
>     the conclusions.
>
>     We've looked at various approaches to solve this important problem
>     several times before:
>
>     - Feature ref (refer to urn: indicating specific features)
>       http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
>
>     - Remote control using REFER to requests & responses
>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
>       (Also, versions -04, -03,-02, -00)
>
>     - Remore control using REFER with XML body describing function
>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
>
>     - Remote control using MBUS
>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
>
>     On top of that there are various proprietary mechanisms, and even
>     some legacy
>     PBX-CTI protocols.
>
>     >  -----Original Message-----
>     >  From: dispatch-bounces@ietf.org
>     >  [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore Loreto
>     >  Sent: Monday, July 06, 2009 09:33
>     >  To: dispatch@ietf.org
>     >  Subject: [dispatch] Disaggregated Media in SIP
>     >
>     >  Hi there,
>     >
>     >  I have just submitted a draft that talks of Disaggregated
>     >  Media in the Session Initiation Protocol (SIP).
>     >  http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
>     ggregated-media-00.txt
>     >
>     >
>     >  Abstract:
>     >  Disaggregated media refers to the ability for a user to create a
>     >  multimedia session combining different media streams, coming from
>     >  different devices under his or her control, so that they are
>     >  treated by
>     >  the far end of the session as a single media session.
>     >  This document lists several use cases that involve
>     >  disaggregated media
>     >  in SIP.
>     >  Additionally, this document analyzes what types of
>     >  disaggregated media
>     >  can be implemented using existing protocol
>     >  mechanisms, and the pros and cons of using each of those mechanism=
s.
>     >  Finally, this document describes scenarios that are not covered by
>     >  current mechanisms
>     >  and proposes new IETF work to cover them.
>     >
>     >
>     >  cheers
>     >  Sal
>     >  _______________________________________________
>     >  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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<HTML>
<HEAD>
<TITLE>Re: [dispatch] Disaggregated Media in SIP</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Paul Kyzivat wrote:<BR>
&gt;Past suggestions by various people to send control signals (intended to=
be acted upon <BR>
&gt;by automata rather than &gt;people) via IM have generally been<BR>
&gt;rejected as inappropriate. <BR>
<BR>
I am not sure how many people expect a usage scenario for IM with an automa=
ta in the middle or <BR>
what the deployment statistics are for such automata (I have never encounte=
red one). &nbsp;<BR>
<BR>
All SIP (or other protocol ) Communicator packages have IM and the URI work=
s there very nicely.<BR>
<BR>
Do you have any usage statistics that justifies the assertion automata are =
the <BR>
key usage scenario and &#8220;plain person to person&#8221; IM does not cou=
nt?<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 7/6/09 3:43 PM, &quot;Paul Kyzivat&quot; &lt;<a href=3D"pkyzivat@cisco.c=
om">pkyzivat@cisco.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'><BR>
<BR>
<BR>
Henry Sinnreich wrote:<BR>
&gt;&gt;We've looked at various approaches to solve this important<BR>
&gt;&gt;problem several times before<BR>
&gt;<BR>
&gt; Actually there is one more: IM-ing a URI to some resource, mentioned b=
y<BR>
&gt; Henning Schulzrinne (I don&#8217;t recall the document or presentation=
).<BR>
&gt;<BR>
&gt; My two cents is that IM-ing a URL is the most general solution, or is =
it?<BR>
<BR>
Past suggestions by various people to send control signals (intended to<BR>
be acted upon by automata rather than people) via IM have generally been<BR=
>
rejected as inappropriate. (The exception so far has been file transfer,<BR=
>
which has some control behavior and some expected human interaction.)<BR>
<BR>
Now if you just want to say &quot;Bob, please make a video call to<BR>
sip:<a href=3D"alice_camera@alice.com">alice_camera@alice.com</a> in order =
to see me&quot; then I guess IM is ok.<BR>
But IMO its not otherwise good. Its just a hack.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Thanks,<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Paul<BR>
<BR>
&gt; Henry<BR>
&gt;<BR>
&gt;<BR>
&gt; On 7/6/09 12:07 PM, &quot;Francois Audet&quot; &lt;<a href=3D"audet@no=
rtel.com">audet@nortel.com</a>&gt; wrote:<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;I'm glad to see this topic coming back.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;I see that this draft doesn't propose a soluti=
on to problem: it list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;three options, and describes why they are not =
adequate. I agree with<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;the conclusions.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;We've looked at various approaches to solve th=
is important problem<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;several times before:<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Feature ref (refer to urn: indicating specif=
ic features)<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-audet-sipping-feature-ref-00">http://tools.ietf.org/html/draft-au=
det-sipping-feature-ref-00</a><BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Remote control using REFER to requests &amp;=
 responses<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-mahy-sip-remote-cc-05">http://tools.ietf.org/html/draft-mahy-sip-=
remote-cc-05</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Also, versions -04, -03,-02, -00)=
<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Remore control using REFER with XML body des=
cribing function<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-mahy-sip-remote-cc-01">http://tools.ietf.org/html/draft-mahy-sip-=
remote-cc-01</a><BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;- Remote control using MBUS<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-mahy-mmusic-mbus-remotecc-01">http://tools.ietf.org/html/draft-ma=
hy-mmusic-mbus-remotecc-01</a> &amp;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools.ietf.org/h=
tml/draft-mahy-mmusic-mbus-sdp-01">http://tools.ietf.org/html/draft-mahy-mm=
usic-mbus-sdp-01</a><BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;On top of that there are various proprietary m=
echanisms, and even<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;some legacy<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;PBX-CTI protocols.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;-----Original Message-----<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;From: <a href=3D"dispatch-bounces@i=
etf.org">dispatch-bounces@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;[<a href=3D"mailto:dispatch-bounces=
@ietf.org">mailto:dispatch-bounces@ietf.org</a>] On Behalf Of Salvatore Lor=
eto<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Sent: Monday, July 06, 2009 09:33<B=
R>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;To: <a href=3D"dispatch@ietf.org">d=
ispatch@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Subject: [dispatch] Disaggregated M=
edia in SIP<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Hi there,<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;I have just submitted a draft that =
talks of Disaggregated<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Media in the Session Initiation Pro=
tocol (SIP).<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<a href=3D"http://www.ietf.org/inte=
rnet-drafts/draft-loreto-dispatch-disa">http://www.ietf.org/internet-drafts=
/draft-loreto-dispatch-disa</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;ggregated-media-00.txt<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Abstract:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Disaggregated media refers to the a=
bility for a user to create a<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;multimedia session combining differ=
ent media streams, coming from<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;different devices under his or her =
control, so that they are<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;treated by<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;the far end of the session as a sin=
gle media session.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;This document lists several use cas=
es that involve<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;disaggregated media<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;in SIP.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Additionally, this document analyze=
s what types of<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;disaggregated media<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;can be implemented using existing p=
rotocol<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;mechanisms, and the pros and cons o=
f using each of those mechanisms.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Finally, this document describes sc=
enarios that are not covered by<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;current mechanisms<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;and proposes new IETF work to cover=
 them.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;cheers<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Sal<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;___________________________________=
____________<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;dispatch mailing list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<a href=3D"dispatch@ietf.org">dispa=
tch@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<a href=3D"https://www.ietf.org/mai=
lman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a><=
BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;______________________________________________=
_<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;dispatch mailing list<BR>
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&gt; ----------------------------------------------------------------------=
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Date: Tue, 7 Jul 2009 01:06:16 -0400
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Cc: "sipping@ietf.org" <sipping@ietf.org>, "dispatch@ietf.org" <dispatch@ietf.org>, "mohamed.boucadair@orange-ftgroup.com" <mohamed.boucadair@orange-ftgroup.com>
Subject: [dispatch] A replacement for ANAT
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BTW, this draft was submitted in response to the discussion around the need=
 for an ANAT-type model that was backwards-compatible, and for a v4/v6 stac=
k option-tag indicator, which was a topic of discussion on SIPPING back in =
February and March.  That thread started with:
http://www.ietf.org/mail-archive/web/sipping/current/msg16718.html


-hadriel
p.s. sorry about the cross-posting, but SIPPING is effectively closed (righ=
t?) so it's hard to know where to post such notices which are tightly relat=
ed to SIP.


> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of mohamed.boucadair@orange-ftgroup.com
> Sent: Monday, July 06, 2009 7:44 AM
>=20
> Dear all,
>=20
> This draft has been submitted.
>=20
> Comments and suggestions are more than welcome.
>=20
>=20
> Cheers,
> Med
>=20
>=20
> -----Message d'origine-----
> De : i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
> De la part de Internet-Drafts@ietf.org
> Envoy=E9 : lundi 6 juillet 2009 11:45
> =C0 : i-d-announce@ietf.org
> Objet : I-D Action:draft-boucadair-mmusic-ccap-00.txt
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>=20
> 	Title           : Session Description Protocol (SDP) Connectivity
> Capability (CCAP) Attribute
> 	Author(s)       : M. Boucadair, H. Kaplan
> 	Filename        : draft-boucadair-mmusic-ccap-00.txt
> 	Pages           : 14
> 	Date            : 2009-07-06
>=20
> This memo proposes a mechanism which allows to carry multiple IP
> addresses, of different address families (e.g., IPv4, IPv6), in the same
> SDP offer/answer.  The proposed attribute solves the backward
> compatibility problem which plagued ANAT, due to its syntax.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-boucadair-mmusic-ccap-00.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.

From R.Jesske@telekom.de  Mon Jul  6 22:17:26 2009
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Hi Dale,
Within the SIPPING group I had already discussions on that issue.
The assumption was that the use of the Reason Header within Responses is =
conditionally allowed.=20

The RFC 3326 says:
   Initially, the Reason header field defined here appears to be most
   useful for BYE and CANCEL requests, but it can appear in any request
   within a dialog, in any CANCEL request and in any response whose
   status code explicitly allows the presence of this header field.


So we need a draft where it is stated that a reason header can be =
included within the SIP Response. That was also the recommendation of =
that discussion. That is why I wrote a draft.
Nevertheless some standards already include this behaviour like the =
ITU-T Q.1912.5 specification.
Also 3GPP needs the reason header within Responses.

Now due to the fact that the draft is expired and with the =
reorganisation of SIP and SIPPING the questioned appeared where such =
draft would like to fit best.
 SIPCORE, DISPATCH or BLISS.

=20
Best Regards,

Roland =20

-----Urspr=FCngliche Nachricht-----
Von: Dale Worley [mailto:dworley@nortel.com]=20
Gesendet: Montag, 6. Juli 2009 17:37
An: Jesske, Roland
Cc: DISPATCH
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

On Mon, 2009-07-06 at 13:50 +0200, R.Jesske@telekom.de wrote:
> Dear all,=20
> After getting no response to my question from the dispatch. I assume
> that there is no interest on this draft within dispatch.
>=20
> So now I would like to ask the people from SIPCORE and BILSS where do
> they see the work.
>=20
> The below mentioned draft describes the use of the Reason Header
> within Resoponses for interoperability with the existing PSTN/ISDN
> networks.

Is the Abstract right?  As far as I can tell (by reading section 2), the
Abstract doesn't describe the draft at all.  That might be a reason that
you haven't gotten any response -- the Abstract "proposes the use of the
Reason header field in SIP responses", but that is something that is
already defined and allowed.

Dale



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Subject: Re: [dispatch] Disaggregated Media in SIP
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Henry Sinnreich wrote:
> Paul Kyzivat wrote:
>>Past suggestions by various people to send control signals (intended 
> tobe acted upon
>>by automata rather than >people) via IM have generally been
>>rejected as inappropriate.
> 
> I am not sure how many people expect a usage scenario for IM with an 
> automata in the middle or
> what the deployment statistics are for such automata (I have never 
> encountered one).  
> 
> All SIP (or other protocol ) Communicator packages have IM and the URI 
> works there very nicely.
> 
> Do you have any usage statistics that justifies the assertion automata 
> are the
> key usage scenario and “plain person to person” IM does not count?
> 
> Henry

I don't know of one either.

But my observation is about using the IM message as a solution to the 
kinds of use cases that were presented in the draft, at least as I 
imagine them playing out.

My mental model is of an environment where a significant fraction of 
people have multimedia UAs, and an expectation that those various media 
will be available in calls that they make. Now add to that people who 
have support for various media, but not in a single device. They then 
want to play in that same multimedia environment.

In such a case, using IMs between people, so that they may establish 
separate calls for various media is clearly the poor man's choice - 
better than nothing but not ideal. And its especially poor if the 
"initial" call is not an IM call. Then just establishing an IM channel 
between the same two people is non-trivial.

	Thanks,
	Paul


> On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
> 
> 
> 
> 
>     Henry Sinnreich wrote:
>     > >We've looked at various approaches to solve this important
>     > >problem several times before
>     >
>     >  Actually there is one more: IM-ing a URI to some resource,
>     mentioned by
>     >  Henning Schulzrinne (I don’t recall the document or presentation).
>     >
>     >  My two cents is that IM-ing a URL is the most general solution, or
>     is it?
> 
>     Past suggestions by various people to send control signals (intended to
>     be acted upon by automata rather than people) via IM have generally been
>     rejected as inappropriate. (The exception so far has been file transfer,
>     which has some control behavior and some expected human interaction.)
> 
>     Now if you just want to say "Bob, please make a video call to
>     sip:alice_camera@alice.com in order to see me" then I guess IM is ok.
>     But IMO its not otherwise good. Its just a hack.
> 
>             Thanks,
>             Paul
> 
>     >  Henry
>     >
>     >
>     >  On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
>     >
>     >      I'm glad to see this topic coming back.
>     >
>     >      I see that this draft doesn't propose a solution to problem:
>     it list
>     >      three options, and describes why they are not adequate. I
>     agree with
>     >      the conclusions.
>     >
>     >      We've looked at various approaches to solve this important problem
>     >      several times before:
>     >
>     >      - Feature ref (refer to urn: indicating specific features)
>     >        http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
>     >
>     >      - Remote control using REFER to requests & responses
>     >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
>     >        (Also, versions -04, -03,-02, -00)
>     >
>     >      - Remore control using REFER with XML body describing function
>     >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
>     >
>     >      - Remote control using MBUS
>     >        http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
>     >        http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
>     >
>     >      On top of that there are various proprietary mechanisms, and even
>     >      some legacy
>     >      PBX-CTI protocols.
>     >
>     >      >  -----Original Message-----
>     >      >  From: dispatch-bounces@ietf.org
>     >      >  [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore
>     Loreto
>     >      >  Sent: Monday, July 06, 2009 09:33
>     >      >  To: dispatch@ietf.org
>     >      >  Subject: [dispatch] Disaggregated Media in SIP
>     >      >
>     >      >  Hi there,
>     >      >
>     >      >  I have just submitted a draft that talks of Disaggregated
>     >      >  Media in the Session Initiation Protocol (SIP).
>     >      >  http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
>     >      ggregated-media-00.txt
>     >      >
>     >      >
>     >      >  Abstract:
>     >      >  Disaggregated media refers to the ability for a user to
>     create a
>     >      >  multimedia session combining different media streams,
>     coming from
>     >      >  different devices under his or her control, so that they are
>     >      >  treated by
>     >      >  the far end of the session as a single media session.
>     >      >  This document lists several use cases that involve
>     >      >  disaggregated media
>     >      >  in SIP.
>     >      >  Additionally, this document analyzes what types of
>     >      >  disaggregated media
>     >      >  can be implemented using existing protocol
>     >      >  mechanisms, and the pros and cons of using each of those
>     mechanisms.
>     >      >  Finally, this document describes scenarios that are not
>     covered by
>     >      >  current mechanisms
>     >      >  and proposes new IETF work to cover them.
>     >      >
>     >      >
>     >      >  cheers
>     >      >  Sal
>     >      >  _______________________________________________
>     >      >  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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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Francois Audet wrote:
> I think what Paul calls automata is the application on the IM client, so 
> that would undermine what this spec and all of us in the Enterprise 
> space have been trying to do for years.
>  
> I will note that the "istyping" indication is already done today with 
> MESSAGE. And the istyping indicator is certainly an automata. And that 
> is an RFC today, and is widly deployed.

It is not something that is clear cut. I can make a case for istyping - 
it is indeed a message intended for human eyes, it is just encoded 
differently from plain text so that the receiving application has more 
options in rendering it. In that respect it isn't so different from html.

> I personally don't really care if its a MESSAGE, a REFER, or an INFO 
> (although we certainly can rule out MBUS). Or a new message.

I still remain unconvinced that it is either necessary or appropriate 
for one end of a call to be involved in how the other end aggregates 
media resources for the call.

	Thanks,
	Paul

> I don't think "other protocols" is a good answer: it has to be routable 
> just like SIP.
> 
>     ------------------------------------------------------------------------
>     *From:* Henry Sinnreich [mailto:hsinnrei@adobe.com]
>     *Sent:* Monday, July 06, 2009 17:24
>     *To:* Paul Kyzivat
>     *Cc:* Audet, Francois (SC100:3055); Salvatore Loreto; dispatch@ietf.org
>     *Subject:* Re: [dispatch] Disaggregated Media in SIP
> 
>     Paul Kyzivat wrote:
>     >Past suggestions by various people to send control signals (intended
>     tobe acted upon
>     >by automata rather than >people) via IM have generally been
>     >rejected as inappropriate.
> 
>     I am not sure how many people expect a usage scenario for IM with an
>     automata in the middle or
>     what the deployment statistics are for such automata (I have never
>     encountered one).  
> 
>     All SIP (or other protocol ) Communicator packages have IM and the
>     URI works there very nicely.
> 
>     Do you have any usage statistics that justifies the assertion
>     automata are the
>     key usage scenario and “plain person to person” IM does not count?
> 
>     Henry
> 
> 
>     On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
> 
> 
> 
> 
>         Henry Sinnreich wrote:
>         > >We've looked at various approaches to solve this important
>         > >problem several times before
>         >
>         >  Actually there is one more: IM-ing a URI to some resource,
>         mentioned by
>         >  Henning Schulzrinne (I don’t recall the document or presentation).
>         >
>         >  My two cents is that IM-ing a URL is the most general
>         solution, or is it?
> 
>         Past suggestions by various people to send control signals
>         (intended to
>         be acted upon by automata rather than people) via IM have
>         generally been
>         rejected as inappropriate. (The exception so far has been file
>         transfer,
>         which has some control behavior and some expected human
>         interaction.)
> 
>         Now if you just want to say "Bob, please make a video call to
>         sip:alice_camera@alice.com in order to see me" then I guess IM
>         is ok.
>         But IMO its not otherwise good. Its just a hack.
> 
>                 Thanks,
>                 Paul
> 
>         >  Henry
>         >
>         >
>         >  On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
>         >
>         >      I'm glad to see this topic coming back.
>         >
>         >      I see that this draft doesn't propose a solution to
>         problem: it list
>         >      three options, and describes why they are not adequate. I
>         agree with
>         >      the conclusions.
>         >
>         >      We've looked at various approaches to solve this important
>         problem
>         >      several times before:
>         >
>         >      - Feature ref (refer to urn: indicating specific features)
>         >
>               http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
>         >
>         >      - Remote control using REFER to requests & responses
>         >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
>         >        (Also, versions -04, -03,-02, -00)
>         >
>         >      - Remore control using REFER with XML body describing function
>         >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
>         >
>         >      - Remote control using MBUS
>         >
>               http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01
>         &
>         >        http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
>         >
>         >      On top of that there are various proprietary mechanisms,
>         and even
>         >      some legacy
>         >      PBX-CTI protocols.
>         >
>         >      >  -----Original Message-----
>         >      >  From: dispatch-bounces@ietf.org
>         >      >  [mailto:dispatch-bounces@ietf.org] On Behalf Of
>         Salvatore Loreto
>         >      >  Sent: Monday, July 06, 2009 09:33
>         >      >  To: dispatch@ietf.org
>         >      >  Subject: [dispatch] Disaggregated Media in SIP
>         >      >
>         >      >  Hi there,
>         >      >
>         >      >  I have just submitted a draft that talks of Disaggregated
>         >      >  Media in the Session Initiation Protocol (SIP).
>         >      >
>          http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
>         >      ggregated-media-00.txt
>         >      >
>         >      >
>         >      >  Abstract:
>         >      >  Disaggregated media refers to the ability for a user to
>         create a
>         >      >  multimedia session combining different media streams,
>         coming from
>         >      >  different devices under his or her control, so that
>         they are
>         >      >  treated by
>         >      >  the far end of the session as a single media session.
>         >      >  This document lists several use cases that involve
>         >      >  disaggregated media
>         >      >  in SIP.
>         >      >  Additionally, this document analyzes what types of
>         >      >  disaggregated media
>         >      >  can be implemented using existing protocol
>         >      >  mechanisms, and the pros and cons of using each of
>         those mechanisms.
>         >      >  Finally, this document describes scenarios that are not
>         covered by
>         >      >  current mechanisms
>         >      >  and proposes new IETF work to cover them.
>         >      >
>         >      >
>         >      >  cheers
>         >      >  Sal
>         >      >  _______________________________________________
>         >      >  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 drage@alcatel-lucent.com  Tue Jul  7 07:48:28 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Francois Audet <audet@nortel.com>
Date: Tue, 7 Jul 2009 16:47:29 +0200
Thread-Topic: [dispatch] Disaggregated Media in SIP
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Disaggregated Media in SIP
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I understood the key distinction for MESSAGE method contents was not what g=
enerated it, but how it was treated on reception. The prime use of the MESS=
AGE method was where the contents were rendered to the user (even if they a=
re generated by something that is an automata). If the contents were to be =
understood by some automata, then PUBLISH or SUBSCRIBE / NOTIFY may be more=
 appropriate, or if there is an INVITE dialog around that pertains to the c=
ommunication, then something else.

Hopever the multiple methods for similar purposes discussion in SIP never r=
eally resolved itself so I would say there is no definitive answer to this.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Tuesday, July 07, 2009 2:09 PM
> To: Francois Audet
> Cc: Henry Sinnreich; dispatch@ietf.org
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
>=20
>=20
> Francois Audet wrote:
> > I think what Paul calls automata is the application on the=20
> IM client,=20
> > so that would undermine what this spec and all of us in the=20
> Enterprise=20
> > space have been trying to do for years.
> > =20
> > I will note that the "istyping" indication is already done=20
> today with=20
> > MESSAGE. And the istyping indicator is certainly an=20
> automata. And that=20
> > is an RFC today, and is widly deployed.
>=20
> It is not something that is clear cut. I can make a case for=20
> istyping - it is indeed a message intended for human eyes, it=20
> is just encoded differently from plain text so that the=20
> receiving application has more options in rendering it. In=20
> that respect it isn't so different from html.
>=20
> > I personally don't really care if its a MESSAGE, a REFER,=20
> or an INFO=20
> > (although we certainly can rule out MBUS). Or a new message.
>=20
> I still remain unconvinced that it is either necessary or=20
> appropriate for one end of a call to be involved in how the=20
> other end aggregates media resources for the call.
>=20
> 	Thanks,
> 	Paul
>=20
> > I don't think "other protocols" is a good answer: it has to be=20
> > routable just like SIP.
> >=20
> >    =20
> --------------------------------------------------------------
> ----------
> >     *From:* Henry Sinnreich [mailto:hsinnrei@adobe.com]
> >     *Sent:* Monday, July 06, 2009 17:24
> >     *To:* Paul Kyzivat
> >     *Cc:* Audet, Francois (SC100:3055); Salvatore Loreto;=20
> dispatch@ietf.org
> >     *Subject:* Re: [dispatch] Disaggregated Media in SIP
> >=20
> >     Paul Kyzivat wrote:
> >     >Past suggestions by various people to send control=20
> signals (intended
> >     tobe acted upon
> >     >by automata rather than >people) via IM have generally been
> >     >rejected as inappropriate.
> >=20
> >     I am not sure how many people expect a usage scenario=20
> for IM with an
> >     automata in the middle or
> >     what the deployment statistics are for such automata (I=20
> have never
> >     encountered one). =20
> >=20
> >     All SIP (or other protocol ) Communicator packages have=20
> IM and the
> >     URI works there very nicely.
> >=20
> >     Do you have any usage statistics that justifies the assertion
> >     automata are the
> >     key usage scenario and =93plain person to person=94 IM does=20
> not count?
> >=20
> >     Henry
> >=20
> >=20
> >     On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
> >=20
> >=20
> >=20
> >=20
> >         Henry Sinnreich wrote:
> >         > >We've looked at various approaches to solve this=20
> important
> >         > >problem several times before
> >         >
> >         >  Actually there is one more: IM-ing a URI to some=20
> resource,
> >         mentioned by
> >         >  Henning Schulzrinne (I don=92t recall the document=20
> or presentation).
> >         >
> >         >  My two cents is that IM-ing a URL is the most general
> >         solution, or is it?
> >=20
> >         Past suggestions by various people to send control signals
> >         (intended to
> >         be acted upon by automata rather than people) via IM have
> >         generally been
> >         rejected as inappropriate. (The exception so far=20
> has been file
> >         transfer,
> >         which has some control behavior and some expected human
> >         interaction.)
> >=20
> >         Now if you just want to say "Bob, please make a=20
> video call to
> >         sip:alice_camera@alice.com in order to see me" then=20
> I guess IM
> >         is ok.
> >         But IMO its not otherwise good. Its just a hack.
> >=20
> >                 Thanks,
> >                 Paul
> >=20
> >         >  Henry
> >         >
> >         >
> >         >  On 7/6/09 12:07 PM, "Francois Audet"=20
> <audet@nortel.com> wrote:
> >         >
> >         >      I'm glad to see this topic coming back.
> >         >
> >         >      I see that this draft doesn't propose a solution to
> >         problem: it list
> >         >      three options, and describes why they are=20
> not adequate. I
> >         agree with
> >         >      the conclusions.
> >         >
> >         >      We've looked at various approaches to solve=20
> this important
> >         problem
> >         >      several times before:
> >         >
> >         >      - Feature ref (refer to urn: indicating=20
> specific features)
> >         >
> >              =20
> http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
> >         >
> >         >      - Remote control using REFER to requests & responses
> >         >       =20
> http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
> >         >        (Also, versions -04, -03,-02, -00)
> >         >
> >         >      - Remore control using REFER with XML body=20
> describing function
> >         >       =20
> http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
> >         >
> >         >      - Remote control using MBUS
> >         >
> >              =20
> http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01
> >         &
> >         >       =20
> http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
> >         >
> >         >      On top of that there are various proprietary=20
> mechanisms,
> >         and even
> >         >      some legacy
> >         >      PBX-CTI protocols.
> >         >
> >         >      >  -----Original Message-----
> >         >      >  From: dispatch-bounces@ietf.org
> >         >      >  [mailto:dispatch-bounces@ietf.org] On Behalf Of
> >         Salvatore Loreto
> >         >      >  Sent: Monday, July 06, 2009 09:33
> >         >      >  To: dispatch@ietf.org
> >         >      >  Subject: [dispatch] Disaggregated Media in SIP
> >         >      >
> >         >      >  Hi there,
> >         >      >
> >         >      >  I have just submitted a draft that talks=20
> of Disaggregated
> >         >      >  Media in the Session Initiation Protocol (SIP).
> >         >      >
> >         =20
> http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
> >         >      ggregated-media-00.txt
> >         >      >
> >         >      >
> >         >      >  Abstract:
> >         >      >  Disaggregated media refers to the ability=20
> for a user to
> >         create a
> >         >      >  multimedia session combining different=20
> media streams,
> >         coming from
> >         >      >  different devices under his or her=20
> control, so that
> >         they are
> >         >      >  treated by
> >         >      >  the far end of the session as a single=20
> media session.
> >         >      >  This document lists several use cases that involve
> >         >      >  disaggregated media
> >         >      >  in SIP.
> >         >      >  Additionally, this document analyzes what types of
> >         >      >  disaggregated media
> >         >      >  can be implemented using existing protocol
> >         >      >  mechanisms, and the pros and cons of using each of
> >         those mechanisms.
> >         >      >  Finally, this document describes=20
> scenarios that are not
> >         covered by
> >         >      >  current mechanisms
> >         >      >  and proposes new IETF work to cover them.
> >         >      >
> >         >      >
> >         >      >  cheers
> >         >      >  Sal
> >         >      >  _______________________________________________
> >         >      >  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
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From AUDET@nortel.com  Tue Jul  7 08:23:01 2009
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References: <4A5261E2.4050506@cisco.com> <C677FFD1.48EB%hsinnrei@adobe.com> <1ECE0EB50388174790F9694F77522CCF1ED91794@zrc2hxm0.corp.nortel.com> <4A5348F5.5030200@cisco.com>
From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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> I still remain unconvinced that it is either necessary or=20
> appropriate for one end of a call to be involved in how the=20
> other end aggregates media resources for the call.

I don't understand what this statement means.

Everybody now has multiple devices these days.

With the amount of implementations of Remote Call Control=20
using various proprietary mechanisms out there, it is pretty
clear that there is a need for this type of functionality.

That does not necessarily means that one end is necessarily involved
in how "the other end aggregates media resources". In my mind, it's
more of how do you share the session, and partition the media streams.

From AUDET@nortel.com  Tue Jul  7 08:44:42 2009
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References: <4A53479B.70908@cisco.com> <C678D29E.492B%hsinnrei@adobe.com>
From: "Francois Audet" <audet@nortel.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Correct.
=20
I agree with Henry. The bootstrap is typically the IM/Presence session, =
not the voice session. That would mean XMPP or SIMPLE.
=20
To me this is a different than the "multiple device" issue, but they are =
quite intertwinned.
=20
A scenario where you contact somebody using presence on a PC client or =
Web Page, or smart phone/PDA, perhaps IM, and then initiate a voice call =
using a desktop phone is a classic example.
=20
In a "loosely coupled" model, all that you'd need from the PC Client/Web =
Page/Smart Phone/PDA is:

*	The ability to initiate the call using the alternate device
*	The ability to do basic control functions for the call using the =
primary (rich UI) interface, for that call on the alternate device: =
answer/terminate/hold/etc.

=20


________________________________

	From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
	Sent: Tuesday, July 07, 2009 08:23
	To: Paul Kyzivat
	Cc: Audet, Francois (SC100:3055); Salvatore Loreto; dispatch@ietf.org
	Subject: Re: [dispatch] Disaggregated Media in SIP
=09
=09
	Paul,
=09
	This is getting interesting, so let's see...
=09
	>In such a case, using IMs between people, so that they may establish
	>separate calls for various media is clearly the poor man's choice -
	>better than nothing but not ideal. And its especially poor if the =
"initial" call is not an IM call.=20
	>Then just establishing an IM channel
	>between the same two people is non-trivial.
=09
	No matter what type of communicator we use: SIP, Skype, Jabber, iChat, =
etc., most courteous people will first IM to ask if the other party can =
speak right now or later. While talking, they may also chose to add =
video. And if planned, also add desktop sharing.
=09
	So the natural order of communications seems to be mostly:=20
	Presence (or the web conference URI), IM, voice, video, desktop =
sharing, conferencing, but certainly it all starts with a URI for the =
rendezvous function.
=09
	It seems we differ (or do we?) on three points:
=09

	1.	Communicator "calls" are not the poor mans choice,=20
	2.	It most often starts with presence and IM,=20
	3.	The URI (or Skype name) of the other party in the address book is =
all that's required. Or the social network as an address book.
	=09

=09
	Now leaving for a moment telephony aside, let's look at some new usage =
scenarios:
=09

	*	People in social networks (Twitter, Facebook, etc.)=20
	*	Telepresence (business)=20
	*	Digital living room (consumer)
	=09

	It is a pretty safe guess that SIP PBX-style voice communications are =
not the right model for these emerging scenarios. I guess even contact =
centers for customer care may benefit from looking at the emerging Web =
applications for communications and especially for sharing of documents =
(having a URI).=20
	Don't we often prefer to IM or email so as not to be tortured by IVR =
Hell?
	And even for voice, just to be called back by an agent?
=09
	Now if we could only outlaw "automata"   :-)
=09
	Henry
=09
=09
	On 7/7/09 8:03 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
=09
=09

		Henry Sinnreich wrote:
		> Paul Kyzivat wrote:
		>>Past suggestions by various people to send control signals (intended
		> tobe acted upon
		>>by automata rather than >people) via IM have generally been
		>>rejected as inappropriate.
		>
		> I am not sure how many people expect a usage scenario for IM with an
		> automata in the middle or
		> what the deployment statistics are for such automata (I have never
		> encountered one).=20
		>
		> All SIP (or other protocol ) Communicator packages have IM and the =
URI
		> works there very nicely.
		>
		> Do you have any usage statistics that justifies the assertion =
automata
		> are the
		> key usage scenario and "plain person to person" IM does not count?
		>
		> Henry
	=09
		I don't know of one either.
	=09
		But my observation is about using the IM message as a solution to the
		kinds of use cases that were presented in the draft, at least as I
		imagine them playing out.
	=09
		My mental model is of an environment where a significant fraction of
		people have multimedia UAs, and an expectation that those various =
media
		will be available in calls that they make. Now add to that people who
		have support for various media, but not in a single device. They then
		want to play in that same multimedia environment.
	=09
		In such a case, using IMs between people, so that they may establish
		separate calls for various media is clearly the poor man's choice -
		better than nothing but not ideal. And its especially poor if the
		"initial" call is not an IM call. Then just establishing an IM channel
		between the same two people is non-trivial.
	=09
		        Thanks,
		        Paul
	=09
	=09
		> On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
		>
		>
		>
		>
		>     Henry Sinnreich wrote:
		>     > >We've looked at various approaches to solve this important
		>     > >problem several times before
		>     >
		>     >  Actually there is one more: IM-ing a URI to some resource,
		>     mentioned by
		>     >  Henning Schulzrinne (I don't recall the document or =
presentation).
		>     >
		>     >  My two cents is that IM-ing a URL is the most general =
solution, or
		>     is it?
		>
		>     Past suggestions by various people to send control signals =
(intended to
		>     be acted upon by automata rather than people) via IM have =
generally been
		>     rejected as inappropriate. (The exception so far has been file =
transfer,
		>     which has some control behavior and some expected human =
interaction.)
		>
		>     Now if you just want to say "Bob, please make a video call to
		>     sip:alice_camera@alice.com in order to see me" then I guess IM =
is ok.
		>     But IMO its not otherwise good. Its just a hack.
		>
		>             Thanks,
		>             Paul
		>
		>     >  Henry
		>     >
		>     >
		>     >  On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> =
wrote:
		>     >
		>     >      I'm glad to see this topic coming back.
		>     >
		>     >      I see that this draft doesn't propose a solution to =
problem:
		>     it list
		>     >      three options, and describes why they are not adequate. I
		>     agree with
		>     >      the conclusions.
		>     >
		>     >      We've looked at various approaches to solve this =
important problem
		>     >      several times before:
		>     >
		>     >      - Feature ref (refer to urn: indicating specific =
features)
		>     >        =
http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
		>     >
		>     >      - Remote control using REFER to requests & responses
		>     >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
		>     >        (Also, versions -04, -03,-02, -00)
		>     >
		>     >      - Remore control using REFER with XML body describing =
function
		>     >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
		>     >
		>     >      - Remote control using MBUS
		>     >        =
http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
		>     >        =
http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
		>     >
		>     >      On top of that there are various proprietary mechanisms, =
and even
		>     >      some legacy
		>     >      PBX-CTI protocols.
		>     >
		>     >      >  -----Original Message-----
		>     >      >  From: dispatch-bounces@ietf.org
		>     >      >  [mailto:dispatch-bounces@ietf.org] On Behalf Of =
Salvatore
		>     Loreto
		>     >      >  Sent: Monday, July 06, 2009 09:33
		>     >      >  To: dispatch@ietf.org
		>     >      >  Subject: [dispatch] Disaggregated Media in SIP
		>     >      >
		>     >      >  Hi there,
		>     >      >
		>     >      >  I have just submitted a draft that talks of =
Disaggregated
		>     >      >  Media in the Session Initiation Protocol (SIP).
		>     >      >  =
http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
		>     >      ggregated-media-00.txt
		>     >      >
		>     >      >
		>     >      >  Abstract:
		>     >      >  Disaggregated media refers to the ability for a user =
to
		>     create a
		>     >      >  multimedia session combining different media streams,
		>     coming from
		>     >      >  different devices under his or her control, so that =
they are
		>     >      >  treated by
		>     >      >  the far end of the session as a single media session.
		>     >      >  This document lists several use cases that involve
		>     >      >  disaggregated media
		>     >      >  in SIP.
		>     >      >  Additionally, this document analyzes what types of
		>     >      >  disaggregated media
		>     >      >  can be implemented using existing protocol
		>     >      >  mechanisms, and the pros and cons of using each of =
those
		>     mechanisms.
		>     >      >  Finally, this document describes scenarios that are =
not
		>     covered by
		>     >      >  current mechanisms
		>     >      >  and proposes new IETF work to cover them.
		>     >      >
		>     >      >
		>     >      >  cheers
		>     >      >  Sal
		>     >      >  _______________________________________________
		>     >      >  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
		>
	=09
	=09


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<HTML><HEAD><TITLE>Re: [dispatch] Disaggregated Media in SIP</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<BODY>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =

size=3D2>Correct.</FONT></SPAN></DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>I=20
agree with Henry. The bootstrap is typically the IM/Presence session, =
not the=20
voice session. That would mean XMPP or SIMPLE.</FONT></SPAN></DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>To me=20
this is a different&nbsp;than the "multiple device" issue, but they are =
quite=20
intertwinned.</FONT></SPAN></DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>A=20
scenario where you contact somebody using presence on a PC client or Web =
Page,=20
or smart phone/PDA, perhaps IM, and then initiate a voice call using a =
desktop=20
phone is a classic example.</FONT></SPAN></DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =
size=3D2>In a=20
"loosely coupled" model, all that you'd need from the PC Client/Web =
Page/Smart=20
Phone/PDA is:</FONT></SPAN></DIV>
<UL>
  <LI><SPAN class=3D520072915-07072009><FONT face=3DArial =
color=3D#800000 size=3D2>The=20
  ability to initiate the call using the alternate =
device</FONT></SPAN></LI>
  <LI><SPAN class=3D520072915-07072009><FONT face=3DArial =
color=3D#800000 size=3D2>The=20
  ability to do basic control functions for the call using the primary =
(rich UI)=20
  interface, for that call on the alternate device:=20
  answer/terminate/hold/etc.</FONT></SPAN></LI></UL>
<DIV><SPAN class=3D520072915-07072009><FONT face=3DArial color=3D#800000 =

size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Henry Sinnreich=20
  [mailto:hsinnrei@adobe.com] <BR><B>Sent:</B> Tuesday, July 07, 2009=20
  08:23<BR><B>To:</B> Paul Kyzivat<BR><B>Cc:</B> Audet, Francois =
(SC100:3055);=20
  Salvatore Loreto; dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch]=20
  Disaggregated Media in SIP<BR></FONT><BR></DIV>
  <DIV></DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
  style=3D"FONT-SIZE: 18pt">Paul,<BR><BR>This is getting interesting, so =
let=92s=20
  see...<BR><BR>&gt;In such a case, using IMs between people, so that =
they may=20
  establish<BR>&gt;separate calls for various media is clearly the poor =
man's=20
  choice -<BR>&gt;better than nothing but not ideal. And its especially =
poor if=20
  the "initial" call is not an IM call. <BR>&gt;Then just establishing =
an IM=20
  channel<BR>&gt;between the same two people is non-trivial.<BR><BR>No =
matter=20
  what type of communicator we use: SIP, Skype, Jabber, iChat, etc., =
most=20
  courteous people will first IM to ask if the other party can speak =
right now=20
  or later. While talking, they may also chose to add video. And if =
planned,=20
  also add desktop sharing.<BR><BR>So the natural order of =
communications seems=20
  to be mostly: <BR>Presence (or the web conference URI), IM, voice, =
video,=20
  desktop sharing, conferencing, but certainly it all starts with a URI =
for the=20
  rendezvous function.<BR><BR>It seems we differ (or do we?) on three=20
  points:<BR></SPAN></FONT>
  <OL>
    <LI><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">Communicator =93calls=94 are not the poor =
mans choice,=20
    </SPAN></FONT>
    <LI><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">It most often starts with presence and IM, =

    </SPAN></FONT>
    <LI><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">The URI (or Skype name) of the other party =
in the=20
    address book is all that=92s required. Or the social network as an =
address=20
    book.<BR></SPAN></FONT></LI></OL><FONT=20
  face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
  style=3D"FONT-SIZE: 18pt"><BR>Now leaving for a moment telephony =
aside, let=92s=20
  look at some new usage scenarios:<BR></SPAN></FONT>
  <UL>
    <LI><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">People in social networks (Twitter, =
Facebook, etc.)=20
    </SPAN></FONT>
    <LI><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">Telepresence (business) </SPAN></FONT>
    <LI><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">Digital living room=20
  (consumer)<BR></SPAN></FONT></LI></UL><FONT=20
  face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN style=3D"FONT-SIZE: =
18pt">It is=20
  a pretty safe guess that SIP PBX-style voice communications are not =
the right=20
  model for these emerging scenarios. I guess even contact centers for =
customer=20
  care may benefit from looking at the emerging Web applications for=20
  communications and especially for sharing of documents (having a URI). =

  <BR>Don=92t we often prefer to IM or email so as not to be tortured by =
IVR=20
  Hell?<BR>And even for voice, just to be called back by an =
agent?<BR><BR>Now if=20
  we could only outlaw =93automata=94 =
&nbsp;&nbsp;:-)<BR><BR>Henry<BR><BR><BR>On=20
  7/7/09 8:03 AM, "Paul Kyzivat" &lt;<A=20
  href=3D"pkyzivat@cisco.com">pkyzivat@cisco.com</A>&gt;=20
  wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">Henry Sinnreich wrote:<BR>&gt; Paul =
Kyzivat=20
    wrote:<BR>&gt;&gt;Past suggestions by various people to send control =
signals=20
    (intended<BR>&gt; tobe acted upon<BR>&gt;&gt;by automata rather than =

    &gt;people) via IM have generally been<BR>&gt;&gt;rejected as=20
    inappropriate.<BR>&gt;<BR>&gt; I am not sure how many people expect =
a usage=20
    scenario for IM with an<BR>&gt; automata in the middle or<BR>&gt; =
what the=20
    deployment statistics are for such automata (I have never<BR>&gt;=20
    encountered one). <BR>&gt;<BR>&gt; All SIP (or other protocol ) =
Communicator=20
    packages have IM and the URI<BR>&gt; works there very=20
    nicely.<BR>&gt;<BR>&gt; Do you have any usage statistics that =
justifies the=20
    assertion automata<BR>&gt; are the<BR>&gt; key usage scenario and =
=93plain=20
    person to person=94 IM does not count?<BR>&gt;<BR>&gt; =
Henry<BR><BR>I don't=20
    know of one either.<BR><BR>But my observation is about using the IM =
message=20
    as a solution to the<BR>kinds of use cases that were presented in =
the draft,=20
    at least as I<BR>imagine them playing out.<BR><BR>My mental model is =
of an=20
    environment where a significant fraction of<BR>people have =
multimedia UAs,=20
    and an expectation that those various media<BR>will be available in =
calls=20
    that they make. Now add to that people who<BR>have support for =
various=20
    media, but not in a single device. They then<BR>want to play in that =
same=20
    multimedia environment.<BR><BR>In such a case, using IMs between =
people, so=20
    that they may establish<BR>separate calls for various media is =
clearly the=20
    poor man's choice -<BR>better than nothing but not ideal. And its =
especially=20
    poor if the<BR>"initial" call is not an IM call. Then just =
establishing an=20
    IM channel<BR>between the same two people is=20
    =
non-trivial.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Thank=
s,<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Paul<BR><BR><BR>&gt=
;=20
    On 7/6/09 3:43 PM, "Paul Kyzivat" &lt;<A=20
    href=3D"pkyzivat@cisco.com">pkyzivat@cisco.com</A>&gt;=20
    wrote:<BR>&gt;<BR>&gt;<BR>&gt;<BR>&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;Henry=20
    Sinnreich wrote:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &gt;We've =
looked at=20
    various approaches to solve this important<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &gt;problem several times =
before<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Actually there is one more: IM-ing a URI to some =
resource,<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;mentioned by<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Henning Schulzrinne (I don=92t recall the document or=20
    presentation).<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;My two cents is that IM-ing a URL =
is the=20
    most general solution, or<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;is=20
    it?<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;Past suggestions by =
various=20
    people to send control signals (intended to<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;be acted upon by automata rather than =
people) via IM=20
    have generally been<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;rejected as=20
    inappropriate. (The exception so far has been file transfer,<BR>&gt; =

    &nbsp;&nbsp;&nbsp;&nbsp;which has some control behavior and some =
expected=20
    human interaction.)<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;Now if =
you just=20
    want to say "Bob, please make a video call to<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;sip:<A=20
    href=3D"alice_camera@alice.com">alice_camera@alice.com</A> in order =
to see me"=20
    then I guess IM is ok.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;But IMO its =
not=20
    otherwise good. Its just a hack.<BR>&gt;<BR>&gt;=20
    =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;T=
hanks,<BR>&gt;=20
    =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;P=
aul<BR>&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Henry<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;On 7/6/09 12:07 PM, "Francois =
Audet"=20
    &lt;<A href=3D"audet@nortel.com">audet@nortel.com</A>&gt; =
wrote:<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;I'm glad to see this topic coming=20
    back.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;I see =
that this=20
    draft doesn't propose a solution to problem:<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;it list<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =

    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;three options, and describes why they =
are not=20
    adequate. I<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;agree with<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;the=20
    conclusions.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We've =
looked at=20
    various approaches to solve this important problem<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;several =
times=20
    before:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Feature =
ref=20
    (refer to urn: indicating specific features)<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00">ht=
tp://tools.ietf.org/html/draft-audet-sipping-feature-ref-00</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remote control using REFER to =
requests &amp;=20
    responses<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05">http://to=
ols.ietf.org/html/draft-mahy-sip-remote-cc-05</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Also, versions -04, =
-03,-02,=20
    -00)<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remore =
control=20
    using REFER with XML body describing function<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01">http://to=
ols.ietf.org/html/draft-mahy-sip-remote-cc-01</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remote control using MBUS<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01">ht=
tp://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01</A>=20
    &amp;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01">http://=
tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;On top of that there are various =
proprietary=20
    mechanisms, and even<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;some legacy<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;PBX-CTI=20
    protocols.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;-----Original Message-----<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;From: <A=20
    =
href=3D"dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A><BR>&gt; =

    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;[<A=20
    =
href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.or=
g</A>]=20
    On Behalf Of Salvatore<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;Loreto<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;Sent:=20
    Monday, July 06, 2009 09:33<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;To: <A=20
    href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Subject: [dispatch] Disaggregated Media in SIP<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;Hi=20
    there,<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;I have just submitted a =
draft that=20
    talks of Disaggregated<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Media in the Session =
Initiation=20
    Protocol (SIP).<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa">h=
ttp://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa</A><BR>&gt;=
=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;ggregated-media-00.txt<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Abstract:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Disaggregated media refers =
to the=20
    ability for a user to<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;create =
a<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;multimedia session combining different media streams,<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;coming from<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;different devices under his =
or her=20
    control, so that they are<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;treated by<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;the=20
    far end of the session as a single media session.<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;This=20
    document lists several use cases that involve<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;disaggregated media<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;in SIP.<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Additionally, this document analyzes what types of<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;disaggregated media<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;can be implemented using =
existing=20
    protocol<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;mechanisms, and the pros =
and cons=20
    of using each of those<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;mechanisms.<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Finally, this document describes scenarios that are =
not<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;covered by<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;current mechanisms<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;and=20
    proposes new IETF work to cover them.<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;cheers<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;Sal<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;_______________________________________________<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;dispatch mailing list<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
    href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;<A=20
    =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;___________________________________________=
____<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;dispatch =
mailing=20
    list<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<A=20
    =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt;=20
    =
&nbsp;&nbsp;&nbsp;&nbsp;-------------------------------------------------=
-----------------------<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;=20
    &nbsp;_______________________________________________<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;dispatch mailing list<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
    href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<A=20
    =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>&gt;<BR><BR></SPAN></FONT></BLOCKQUO=
TE></BLOCKQUOTE></BODY></HTML>

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Date: Tue, 07 Jul 2009 11:47:05 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Francois Audet <audet@nortel.com>
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Francois Audet wrote:
>> I still remain unconvinced that it is either necessary or 
>> appropriate for one end of a call to be involved in how the 
>> other end aggregates media resources for the call.
> 
> I don't understand what this statement means.
> 
> Everybody now has multiple devices these days.
> 
> With the amount of implementations of Remote Call Control 
> using various proprietary mechanisms out there, it is pretty
> clear that there is a need for this type of functionality.
> 
> That does not necessarily means that one end is necessarily involved
> in how "the other end aggregates media resources". In my mind, it's
> more of how do you share the session, and partition the media streams.

My point is that between caller and callee there should be one sip 
session, with as many media streams as the two ends want to share. If 
you stick to that principle, then features such as various forms of 
transfer, conferencing, answering automata, etc. can all work is the 
same manner as they do when there is really only a single device.

If one, or both, ends needs to aggregate multiple devices to assemble 
its end of those media streams, then that is its problem, not a problem 
for the other end.

IMO, to a first level of approximation, 3pcc is sufficient to do that on 
one end when the devices contributing media support are sip devices. 
Certainly there is more that can be done to make that nicer, easier to 
use, etc. I'm uncertain if there is a need for standardization of that.

There may be some cases when I want to aggregate multiple devices of 
yours in order to provide you with a virtual multimedia device that I 
can then communicate with. But even then I would just view that as my 
setting up something separate for you in lieu of your doing it for 
yourself. I wouldn't view it as the normal course of events. Not one of 
Sal's use cases called for that.

	Thanks,
	Paul

	Thanks,
	Paul

From AUDET@nortel.com  Tue Jul  7 08:58:36 2009
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Thread-Topic: [dispatch] Disaggregated Media in SIP
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From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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=20
> My point is that between caller and callee there should be=20
> one sip session, with as many media streams as the two ends=20
> want to share. If you stick to that principle, then features=20
> such as various forms of transfer, conferencing, answering=20
> automata, etc. can all work is the same manner as they do=20
> when there is really only a single device.

I think that's OK for some scenarios (as per my previous email),
and low-level remote speaker/microphone/camera techniques may work.

But I don't believe it is sufficient.

In some cases, the complete opposite may be more appropriate, i.e.,
where both devices are bona fides SIP UAs with very "loosely coupled"
interactions between them.

That allows you to do things like take IM on one device and
voice on the other. Or initiate a call on one on behalf of the
other, and so forth.

But I agree with you that this must be done in a way that is
transparent to the other side.=20

> If one, or both, ends needs to aggregate multiple devices to=20
> assemble its end of those media streams, then that is its=20
> problem, not a problem for the other end.
>=20
> IMO, to a first level of approximation, 3pcc is sufficient to=20
> do that on one end when the devices contributing media=20
> support are sip devices.=20
> Certainly there is more that can be done to make that nicer,=20
> easier to use, etc. I'm uncertain if there is a need for=20
> standardization of that.
>=20
> There may be some cases when I want to aggregate multiple=20
> devices of yours in order to provide you with a virtual=20
> multimedia device that I can then communicate with. But even=20
> then I would just view that as my setting up something=20
> separate for you in lieu of your doing it for yourself. I=20
> wouldn't view it as the normal course of events. Not one of=20
> Sal's use cases called for that.
>=20
> 	Thanks,
> 	Paul
>=20
> 	Thanks,
> 	Paul
>=20

From spromano@unina.it  Tue Jul  7 09:04:45 2009
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Vijay Gurbani <vkg@alcatel-lucent.com>
Subject: Re: [dispatch] Session Recording in SIP
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Dear all,

please also have a look at the following draft, which has been =20
sumbitted yesterday:

Filename:         draft-romano-dcon-recording
Revision:         00
Title:                 Session Recording for Conferences using SMIL
Creation_date:         2009-07-06
WG ID:                 Independent Submission
Number_of_pages: 24

Abstract:
This document deals with session recording, specifically for what
concerns recording of multimedia conferences, both centralized and
distributed.  Each involved media is recorded separately, and is then
properly tagged.  A SMIL [W3C.CR-SMIL3-20080115] metadata is used to
put all the separate recordings together and handle their
synchronization, as well as the possibly asynchronous opening and
closure of media within the context of a conference.  This SMIL
metadata can subsequently be used by an interested user by means of a
compliant player in order to passively receive a playout of the whole
multimedia conference session.  The motivation for this document
comes from our experience with our conferencing framework, Meetecho,
for which we implemented a recording functionality.

Cheers,

Simon


Quoting "Jain, Rajnish" <Rajnish.Jain@ipc.com>:

> All: The following draft on SIP session recording requirements was  =20
> submitted today:
>
> http://www.ietf.org/internet-drafts/draft-jain-dispatch-session-recording-=
protocol-req-00.txt
>
> Title:           Requirements for Session Recording Protocol (SRP)
>
> Abstract:
> Session recording is a critical requirement in many business
> communications environments such as call centers and financial
> trading floors.  In some of these environments, all calls must be
> recorded for regulatory and compliance reasons.  In others, calls may
> be recorded for quality control or business analytics.  Recording is
> typically done by sending a copy of the media to the recording
> devices.  This document specifies requirements for a protocol that
> will manage delivery of media from an end-point that originates media
> or that has access to it to a recording device.  This protocol is
> being referred to as Session Recording Protocol and will most likely
> be based on SIP.
>
>
>
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf
>> Of Vijay Gurbani
>> Sent: Monday, June 22, 2009 12:09 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Summary on Session Recording in SIP
>>
>> All: Here is the summary of the Session Recording thread from June 9
>> to June 19, 2009.  If I misrepresented anything, please let me know.
>>
>> There was a fair amount of discussion on where to conduct this
>> work.  In the end, it was decided to continue discussions on
>> dispatch, further decisions what to do -- new working group or
>> put this work in an existing working group -- will be decided
>> later.  Regardless of where the work is done, opinions were
>> expressed that the work is important that it should commence
>> in the IETF.
>>
>> Need to be crisp about the purpose of this work: is it that users
>> record sessions to hear later, or sessions need to be recorded
>> because of some business needs?  Or both?  Is SRTP covered or
>> only RTP?
>>
>> Solutions should cover:
>>   Always on recording
>>   Recording on demand
>>   Required recording
>>   Pause and resume recording
>>   Playback facilities and how they interact with recording
>>   Recording formats and protocols for controlling the stored recording.
>>
>> A lot of discussion ensued on the specific architecture or
>> architectures that would be possible.  No decision was reached on
>> which specific architecture is good or bad, although various opinions
>> were expressed in support for each.
>>
>> There are essentially four architectures to realize recording
>> services: two that discuss where the media stream is forked
>> out to the recorder -- at the UA or in a middlebox of some
>> sort are outlined here:
>>    http://www.ietf.org/mail-archive/web/dispatch/current/msg00226.html
>>
>> The third architecture has both UAs sending media to a
>> recorder.  See:
>>      http://www.ietf.org/mail-archive/web/dispatch/current/msg00236.html
>>
>> A fourth architecture that is a superset of the third one is at:
>>      http://www.ietf.org/mail-archive/web/dispatch/current/msg00256.html
>>
>> - vijay
>> --
>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
>> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>> WWW:   http://ect.bell-labs.com/who/vkg
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>
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--=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

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


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Subject: Re: [dispatch] Disaggregated Media in SIP
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Henry Sinnreich wrote:
> Paul,
> 
> This is getting interesting, so let’s see...
> 
>>In such a case, using IMs between people, so that they may establish
>>separate calls for various media is clearly the poor man's choice -
>>better than nothing but not ideal. And its especially poor if the 
> "initial" call is not an IM call.
>>Then just establishing an IM channel
>>between the same two people is non-trivial.
> 
> No matter what type of communicator we use: SIP, Skype, Jabber, iChat, 
> etc., most courteous people will first IM to ask if the other party can 
> speak right now or later. While talking, they may also chose to add 
> video. And if planned, also add desktop sharing.

I don't think that most people consider that to be required of courteous 
people. AFAIK most people with cell phones don't send an SMS first to 
ask "can I call you?"

I think this is more a matter of differences in opinion about what the 
preferred medium of communication is. For some people it is definitely 
voice, for others IM or email. So people who consider IM their preferred 
mode of conversation may well IM first and then "escalate to voice" if 
that seems justified. I do that. And currently it often involves IMing a 
phone number. But I would much rather simply have an "add voice" button 
on my IM client.

I think we are just starting to get to the point where multimedia 
communication is common enough that we need to develop an etiquette for 
using it:
- how do we indicate and enforce our preferences of media?
- is it polite to request to receive video without offering
   to send it?
- is it ok to initiate addition of another medium to a call
   without first negotiating its use person to person via the
   existing media?

> So the natural order of communications seems to be mostly:
> Presence (or the web conference URI), IM, voice, video, desktop sharing, 
> conferencing, but certainly it all starts with a URI for the rendezvous 
> function.

I think that is just *your* natural order. (And probably mine too.) But 
not for the world at large. For many the natural order is: voice, then 
SMS if the voice doesn't work.

As long as people endeavor to have one URI that works for all the media 
they support, then that is indeed the starting point.

> It seems we differ (or do we?) on three points:
> 
>    1. Communicator “calls” are not the poor mans choice,
>    2. It most often starts with presence and IM,
>    3. The URI (or Skype name) of the other party in the address book is
>       all that’s required. Or the social network as an address book.

I've answered above. IMO there is no one answer to this.

If everyone had presence on every communication device they use, then it 
would likely be a starting point when it provided useful info. But note 
that I am unlikely to ever have presence data about all the people I 
want to communicate with.

> Now leaving for a moment telephony aside, let’s look at some new usage 
> scenarios:
> 
>     * People in social networks (Twitter, Facebook, etc.)
>     * Telepresence (business)
>     * Digital living room (consumer)
> 
> It is a pretty safe guess that SIP PBX-style voice communications are 
> not the right model for these emerging scenarios. I guess even contact 
> centers for customer care may benefit from looking at the emerging Web 
> applications for communications and especially for sharing of documents 
> (having a URI).
> Don’t we often prefer to IM or email so as not to be tortured by IVR Hell?
> And even for voice, just to be called back by an agent?

As I noted earlier, we are in the very early stages of this because of 
the diversity of devices with varying capabilities. There currently is 
no least common denominator that is shared by all. The closest to one is 
probably still voice. And we still haven't converged the address spaces 
for these things. So its all very cumbersome.

I agree with you that there is a lot of room for progress in this space.

> Now if we could only outlaw “automata”   :-)

You can outlaw them for yourself. Personally I *like* automata that do 
useful services for me. Of note I very much like the automaton that 
answers voice calls, takes voice messages, and sends me an email with a 
text translation of the voice as well as the recorded audio.

The key is that it is an automaton that I have chosen to act as an agent 
on my behalf.

Maybe you don't want to talk to my automaton. That is fine. If you'd 
rather send me an email in the first place, that's fine too.

	Thanks,
	Paul

> Henry
> 
> 
> On 7/7/09 8:03 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
> 
>     Henry Sinnreich wrote:
>     >  Paul Kyzivat wrote:
>     > >Past suggestions by various people to send control signals (intended
>     >  tobe acted upon
>     > >by automata rather than >people) via IM have generally been
>     > >rejected as inappropriate.
>     >
>     >  I am not sure how many people expect a usage scenario for IM with an
>     >  automata in the middle or
>     >  what the deployment statistics are for such automata (I have never
>     >  encountered one).
>     >
>     >  All SIP (or other protocol ) Communicator packages have IM and the URI
>     >  works there very nicely.
>     >
>     >  Do you have any usage statistics that justifies the assertion automata
>     >  are the
>     >  key usage scenario and “plain person to person” IM does not count?
>     >
>     >  Henry
> 
>     I don't know of one either.
> 
>     But my observation is about using the IM message as a solution to the
>     kinds of use cases that were presented in the draft, at least as I
>     imagine them playing out.
> 
>     My mental model is of an environment where a significant fraction of
>     people have multimedia UAs, and an expectation that those various media
>     will be available in calls that they make. Now add to that people who
>     have support for various media, but not in a single device. They then
>     want to play in that same multimedia environment.
> 
>     In such a case, using IMs between people, so that they may establish
>     separate calls for various media is clearly the poor man's choice -
>     better than nothing but not ideal. And its especially poor if the
>     "initial" call is not an IM call. Then just establishing an IM channel
>     between the same two people is non-trivial.
> 
>             Thanks,
>             Paul
> 
> 
>     >  On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
>     >
>     >
>     >
>     >
>     >      Henry Sinnreich wrote:
>     >      > >We've looked at various approaches to solve this important
>     >      > >problem several times before
>     >      >
>     >      >  Actually there is one more: IM-ing a URI to some resource,
>     >      mentioned by
>     >      >  Henning Schulzrinne (I don’t recall the document or
>     presentation).
>     >      >
>     >      >  My two cents is that IM-ing a URL is the most general
>     solution, or
>     >      is it?
>     >
>     >      Past suggestions by various people to send control signals
>     (intended to
>     >      be acted upon by automata rather than people) via IM have
>     generally been
>     >      rejected as inappropriate. (The exception so far has been file
>     transfer,
>     >      which has some control behavior and some expected human
>     interaction.)
>     >
>     >      Now if you just want to say "Bob, please make a video call to
>     >      sip:alice_camera@alice.com in order to see me" then I guess IM
>     is ok.
>     >      But IMO its not otherwise good. Its just a hack.
>     >
>     >              Thanks,
>     >              Paul
>     >
>     >      >  Henry
>     >      >
>     >      >
>     >      >  On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
>     >      >
>     >      >      I'm glad to see this topic coming back.
>     >      >
>     >      >      I see that this draft doesn't propose a solution to
>     problem:
>     >      it list
>     >      >      three options, and describes why they are not adequate. I
>     >      agree with
>     >      >      the conclusions.
>     >      >
>     >      >      We've looked at various approaches to solve this
>     important problem
>     >      >      several times before:
>     >      >
>     >      >      - Feature ref (refer to urn: indicating specific features)
>     >      >
>            http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
>     >      >
>     >      >      - Remote control using REFER to requests & responses
>     >      >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
>     >      >        (Also, versions -04, -03,-02, -00)
>     >      >
>     >      >      - Remore control using REFER with XML body describing
>     function
>     >      >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
>     >      >
>     >      >      - Remote control using MBUS
>     >      >
>            http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01 &
>     >      >        http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
>     >      >
>     >      >      On top of that there are various proprietary
>     mechanisms, and even
>     >      >      some legacy
>     >      >      PBX-CTI protocols.
>     >      >
>     >      >      >  -----Original Message-----
>     >      >      >  From: dispatch-bounces@ietf.org
>     >      >      >  [mailto:dispatch-bounces@ietf.org] On Behalf Of
>     Salvatore
>     >      Loreto
>     >      >      >  Sent: Monday, July 06, 2009 09:33
>     >      >      >  To: dispatch@ietf.org
>     >      >      >  Subject: [dispatch] Disaggregated Media in SIP
>     >      >      >
>     >      >      >  Hi there,
>     >      >      >
>     >      >      >  I have just submitted a draft that talks of
>     Disaggregated
>     >      >      >  Media in the Session Initiation Protocol (SIP).
>     >      >      >
>      http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
>     >      >      ggregated-media-00.txt
>     >      >      >
>     >      >      >
>     >      >      >  Abstract:
>     >      >      >  Disaggregated media refers to the ability for a user to
>     >      create a
>     >      >      >  multimedia session combining different media streams,
>     >      coming from
>     >      >      >  different devices under his or her control, so that
>     they are
>     >      >      >  treated by
>     >      >      >  the far end of the session as a single media session.
>     >      >      >  This document lists several use cases that involve
>     >      >      >  disaggregated media
>     >      >      >  in SIP.
>     >      >      >  Additionally, this document analyzes what types of
>     >      >      >  disaggregated media
>     >      >      >  can be implemented using existing protocol
>     >      >      >  mechanisms, and the pros and cons of using each of those
>     >      mechanisms.
>     >      >      >  Finally, this document describes scenarios that are not
>     >      covered by
>     >      >      >  current mechanisms
>     >      >      >  and proposes new IETF work to cover them.
>     >      >      >
>     >      >      >
>     >      >      >  cheers
>     >      >      >  Sal
>     >      >      >  _______________________________________________
>     >      >      >  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 AUDET@nortel.com  Tue Jul  7 09:31:46 2009
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From: "Francois Audet" <audet@nortel.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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You asked for real-life data.

I've seen many ways to achieve this type of "remote call control" =
functionality:

1 - Proprietary Master/Slave protocols

	This is similar to H.248, but proprietary. Specifically, when in=20
      "remote control mode", the device (e.g.., a phone) sees the UA=20
	(e.g. a PC client) as it's "PBX". In this mode, the Phone is not=20
	a SIP UA anymore in control mode. When control is relinquished,=20
	Phone re-SIPifies itself.

2 - CTI-like protocols

	Things like ECMA TR-87 (CSTA tunnelled over SIP). The two devices =
remain
	SIP UAs at all time, and attempt to "coordinate" their state.

3 - Layer 1/2 integration

	The device on the LAN is seen as a peripheral (i.e., remote speaker/
	microphone/camera) to the device (e.g., PC) controlling it, when
	in control mode. Typically, this is USB or Bluetooth, or some emulation
	over Ethernet. Like number 1, the Phone is not a SIP UA anymore in=20
	control mode. When control is relinquished, Phone re-SIPifies itself.=20

4 - SIP extensions/B2BUA

	You can do a lot directly with SIP, if you have a B2BUA, but it is
	non-trivial.

Number 3, and we don't need to standardized anything at IETF for this.

I do not think that it is adequate for all applications, and a much more =

loosely coupled mechanism is what we would need for those applications.
And by "loosely coupled", I really mean it: not media control. More like
session sharing, where each UA ultimately retains control over their own =
destiny.

Something akin to Feature Referal. It could be based on presence or =
other=20
mechanisms.

________________________________

	From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
	Sent: Monday, July 06, 2009 17:55
	To: Audet, Francois (SC100:3055); Paul Kyzivat
	Cc: Salvatore Loreto; dispatch@ietf.org
	Subject: Re: [dispatch] Disaggregated Media in SIP
=09
=09
	Francois Audet wrote:
	>spec and all of us in the Enterprise space have been trying to do for =
years.
=09
	Now this makes sense.=20
	But still, can you share some data of its real life usage, as compared =
to what is out today in various SIP Communicators? That would rule our =
the person-to-person IM scenarios?
=09
	>I don't think "other protocols" is a good answer: it has to be =
routable just like SIP.
	This again makes sense to me.
	(neglecting the facts about Jabber IM [such as in the IETF], Skype,  =
etc.)
=09
	Henry
=09
=09
	On 7/6/09 7:41 PM, "Francois Audet" <audet@nortel.com> wrote:
=09
=09

		I think what Paul calls automata is the application on the IM client, =
so that would undermine what this spec and all of us in the Enterprise =
space have been trying to do for years.
	=09
		I will note that the "istyping" indication is already done today with =
MESSAGE. And the istyping indicator is certainly an automata. And that =
is an RFC today, and is widly deployed.
	=09
		I personally don't really care if its a MESSAGE, a REFER, or an INFO =
(although we certainly can rule out MBUS). Or a new message.
	=09
		I don't think "other protocols" is a good answer: it has to be =
routable just like SIP.
	=09
	=09

		=09
			=20
		=09
________________________________

			From: Henry Sinnreich  [mailto:hsinnrei@adobe.com]=20
			Sent: Monday, July 06, 2009  17:24
			To: Paul Kyzivat
			Cc: Audet, Francois (SC100:3055);  Salvatore Loreto; =
dispatch@ietf.org
			Subject: Re: [dispatch]  Disaggregated Media in SIP
		=09
			=20
			Paul Kyzivat wrote:
			>Past suggestions by various  people to send control signals =
(intended tobe acted upon=20
			>by automata  rather than >people) via IM have generally been
			>rejected as  inappropriate.=20
		=09
			I am not sure how many people expect a usage scenario  for IM with an =
automata in the middle or=20
			what the deployment statistics  are for such automata (I have never =
encountered one). =20
		=09
			All SIP  (or other protocol ) Communicator packages have IM and the =
URI works there  very nicely.
		=09
			Do you have any usage statistics that justifies the  assertion =
automata are the=20
			key usage scenario and "plain person to person"  IM does not count?
		=09
			Henry
		=09
		=09
			On 7/6/09 3:43 PM, "Paul Kyzivat"  <pkyzivat@cisco.com>  wrote:
		=09
			=20
		=09

			=09
			=09
			=09
				Henry Sinnreich wrote:
				>>We've  looked at various approaches to solve this important
				>>problem  several times before
				>
				> Actually there is one more: IM-ing a  URI to some resource, =
mentioned by
				> Henning Schulzrinne (I don't  recall the document or =
presentation).
				>
				> My two cents is that  IM-ing a URL is the most general solution, =
or is it?
			=09
				Past suggestions  by various people to send control signals =
(intended to
				be acted upon by  automata rather than people) via IM have generally =
been
				rejected as  inappropriate. (The exception so far has been file =
transfer,
				which has  some control behavior and some expected human =
interaction.)
			=09
				Now if  you just want to say "Bob, please make a video call to
				sip:alice_camera@alice.com in order to see me"  then I guess IM is =
ok.
				But IMO its not otherwise good. Its just a  hack.
			=09
				        Thanks,
				        Paul
			=09
				>  Henry
				>
				>
				> On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
				>
				>      I'm glad to see this topic coming  back.
				>
				>     I see that this draft doesn't  propose a solution to problem: =
it list
				>     three  options, and describes why they are not adequate. I =
agree with
				>      the conclusions.
				>
				>      We've looked at various approaches to solve this  important =
problem
				>     several times  before:
				>
				>     - Feature ref (refer to urn:  indicating specific features)
				>       =
http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00
				>
				>      - Remote control using REFER to requests &  responses
				>       http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
				>        (Also, versions -04, -03,-02,  -00)
				>
				>     - Remore control using REFER  with XML body describing =
function
				>        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
				>
				>      - Remote control using MBUS
				>        =
http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01  &
				>       http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
				>
				>      On top of that there are various proprietary  mechanisms, and =
even
				>     some legacy
				>      PBX-CTI protocols.
				>
				>      >  -----Original Message-----
				>      >  From: dispatch-bounces@ietf.org
				>      >  [mailto:dispatch-bounces@ietf.org]  On Behalf Of Salvatore =
Loreto
				>     >   Sent: Monday, July 06, 2009 09:33
				>     >   To: dispatch@ietf.org
				>      >  Subject: [dispatch] Disaggregated Media  in SIP
				>     >
				>      >  Hi there,
				>      >
				>     >  I  have just submitted a draft that talks of Disaggregated
				>      >  Media in the Session Initiation Protocol  (SIP).
				>     >  =
http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
				>      ggregated-media-00.txt
				>      >
				>     >
				>      >  Abstract:
				>      >  Disaggregated media refers to the ability  for a user to =
create a
				>     >  multimedia  session combining different media streams, =
coming from
				>      >  different devices under his or her  control, so that they =
are
				>     >  treated  by
				>     >  the far end of the session as  a single media session.
				>     >  This  document lists several use cases that involve
				>      >  disaggregated media
				>      >  in SIP.
				>      >  Additionally, this document analyzes what  types of
				>     >  disaggregated  media
				>     >  can be implemented using  existing protocol
				>     >  mechanisms, and  the pros and cons of using each of those =
mechanisms.
				>      >  Finally, this document describes  scenarios that are not =
covered by
				>     >   current mechanisms
				>     >  and  proposes new IETF work to cover them.
				>      >
				>     >
				>      >  cheers
				>      >  Sal
				>     >   _______________________________________________
				>      >  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
			=09
			=09

	=09
	=09


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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Francois Audet wrote:
>  
>> My point is that between caller and callee there should be 
>> one sip session, with as many media streams as the two ends 
>> want to share. If you stick to that principle, then features 
>> such as various forms of transfer, conferencing, answering 
>> automata, etc. can all work is the same manner as they do 
>> when there is really only a single device.
> 
> I think that's OK for some scenarios (as per my previous email),
> and low-level remote speaker/microphone/camera techniques may work.
> 
> But I don't believe it is sufficient.
> 
> In some cases, the complete opposite may be more appropriate, i.e.,
> where both devices are bona fides SIP UAs with very "loosely coupled"
> interactions between them.
> 
> That allows you to do things like take IM on one device and
> voice on the other. Or initiate a call on one on behalf of the
> other, and so forth.
> 
> But I agree with you that this must be done in a way that is
> transparent to the other side. 

We aren't communicating very well.
I can't piece together the things you say above into a coherent whole.

But apparently we agree on the key point - that this should be invisible 
to the other "side". Maybe we should just start there, since I don't 
think Sal agrees with that. At least his draft, and its predecessors, don't.

If we can agree that one communication session between two parties has 
one sip session *between them*, then we can work on what the 
architecture should be like at each end.

Of course you are still free to make multiple concurrent "calls" between 
the same two parties. But if you do, they are unrelated, except perhaps 
in your head.

	Thanks,
	Paul

>> If one, or both, ends needs to aggregate multiple devices to 
>> assemble its end of those media streams, then that is its 
>> problem, not a problem for the other end.
>>
>> IMO, to a first level of approximation, 3pcc is sufficient to 
>> do that on one end when the devices contributing media 
>> support are sip devices. 
>> Certainly there is more that can be done to make that nicer, 
>> easier to use, etc. I'm uncertain if there is a need for 
>> standardization of that.
>>
>> There may be some cases when I want to aggregate multiple 
>> devices of yours in order to provide you with a virtual 
>> multimedia device that I can then communicate with. But even 
>> then I would just view that as my setting up something 
>> separate for you in lieu of your doing it for yourself. I 
>> wouldn't view it as the normal course of events. Not one of 
>> Sal's use cases called for that.
>>
>> 	Thanks,
>> 	Paul
>>
>> 	Thanks,
>> 	Paul
>>
> 

From hsinnrei@adobe.com  Tue Jul  7 09:45:11 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 7 Jul 2009 08:23:26 -0700
Thread-Topic: [dispatch] Disaggregated Media in SIP
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Paul,

This is getting interesting, so let's see...

>In such a case, using IMs between people, so that they may establish
>separate calls for various media is clearly the poor man's choice -
>better than nothing but not ideal. And its especially poor if the "initial=
" call is not an IM call.
>Then just establishing an IM channel
>between the same two people is non-trivial.

No matter what type of communicator we use: SIP, Skype, Jabber, iChat, etc.=
, most courteous people will first IM to ask if the other party can speak r=
ight now or later. While talking, they may also chose to add video. And if =
planned, also add desktop sharing.

So the natural order of communications seems to be mostly:
Presence (or the web conference URI), IM, voice, video, desktop sharing, co=
nferencing, but certainly it all starts with a URI for the rendezvous funct=
ion.

It seems we differ (or do we?) on three points:

 1.  Communicator "calls" are not the poor mans choice,
 2.  It most often starts with presence and IM,
 3.  The URI (or Skype name) of the other party in the address book is all =
that's required. Or the social network as an address book.

Now leaving for a moment telephony aside, let's look at some new usage scen=
arios:

 *   People in social networks (Twitter, Facebook, etc.)
 *   Telepresence (business)
 *   Digital living room (consumer)

It is a pretty safe guess that SIP PBX-style voice communications are not t=
he right model for these emerging scenarios. I guess even contact centers f=
or customer care may benefit from looking at the emerging Web applications =
for communications and especially for sharing of documents (having a URI).
Don't we often prefer to IM or email so as not to be tortured by IVR Hell?
And even for voice, just to be called back by an agent?

Now if we could only outlaw "automata"   :-)

Henry


On 7/7/09 8:03 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

Henry Sinnreich wrote:
> Paul Kyzivat wrote:
>>Past suggestions by various people to send control signals (intended
> tobe acted upon
>>by automata rather than >people) via IM have generally been
>>rejected as inappropriate.
>
> I am not sure how many people expect a usage scenario for IM with an
> automata in the middle or
> what the deployment statistics are for such automata (I have never
> encountered one).
>
> All SIP (or other protocol ) Communicator packages have IM and the URI
> works there very nicely.
>
> Do you have any usage statistics that justifies the assertion automata
> are the
> key usage scenario and "plain person to person" IM does not count?
>
> Henry

I don't know of one either.

But my observation is about using the IM message as a solution to the
kinds of use cases that were presented in the draft, at least as I
imagine them playing out.

My mental model is of an environment where a significant fraction of
people have multimedia UAs, and an expectation that those various media
will be available in calls that they make. Now add to that people who
have support for various media, but not in a single device. They then
want to play in that same multimedia environment.

In such a case, using IMs between people, so that they may establish
separate calls for various media is clearly the poor man's choice -
better than nothing but not ideal. And its especially poor if the
"initial" call is not an IM call. Then just establishing an IM channel
between the same two people is non-trivial.

        Thanks,
        Paul


> On 7/6/09 3:43 PM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
>
>
>
>
>     Henry Sinnreich wrote:
>     > >We've looked at various approaches to solve this important
>     > >problem several times before
>     >
>     >  Actually there is one more: IM-ing a URI to some resource,
>     mentioned by
>     >  Henning Schulzrinne (I don't recall the document or presentation).
>     >
>     >  My two cents is that IM-ing a URL is the most general solution, or
>     is it?
>
>     Past suggestions by various people to send control signals (intended =
to
>     be acted upon by automata rather than people) via IM have generally b=
een
>     rejected as inappropriate. (The exception so far has been file transf=
er,
>     which has some control behavior and some expected human interaction.)
>
>     Now if you just want to say "Bob, please make a video call to
>     sip:alice_camera@alice.com in order to see me" then I guess IM is ok.
>     But IMO its not otherwise good. Its just a hack.
>
>             Thanks,
>             Paul
>
>     >  Henry
>     >
>     >
>     >  On 7/6/09 12:07 PM, "Francois Audet" <audet@nortel.com> wrote:
>     >
>     >      I'm glad to see this topic coming back.
>     >
>     >      I see that this draft doesn't propose a solution to problem:
>     it list
>     >      three options, and describes why they are not adequate. I
>     agree with
>     >      the conclusions.
>     >
>     >      We've looked at various approaches to solve this important pro=
blem
>     >      several times before:
>     >
>     >      - Feature ref (refer to urn: indicating specific features)
>     >        http://tools.ietf.org/html/draft-audet-sipping-feature-ref-0=
0
>     >
>     >      - Remote control using REFER to requests & responses
>     >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05
>     >        (Also, versions -04, -03,-02, -00)
>     >
>     >      - Remore control using REFER with XML body describing function
>     >        http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01
>     >
>     >      - Remote control using MBUS
>     >        http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-0=
1 &
>     >        http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01
>     >
>     >      On top of that there are various proprietary mechanisms, and e=
ven
>     >      some legacy
>     >      PBX-CTI protocols.
>     >
>     >      >  -----Original Message-----
>     >      >  From: dispatch-bounces@ietf.org
>     >      >  [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore
>     Loreto
>     >      >  Sent: Monday, July 06, 2009 09:33
>     >      >  To: dispatch@ietf.org
>     >      >  Subject: [dispatch] Disaggregated Media in SIP
>     >      >
>     >      >  Hi there,
>     >      >
>     >      >  I have just submitted a draft that talks of Disaggregated
>     >      >  Media in the Session Initiation Protocol (SIP).
>     >      >  http://www.ietf.org/internet-drafts/draft-loreto-dispatch-d=
isa
>     >      ggregated-media-00.txt
>     >      >
>     >      >
>     >      >  Abstract:
>     >      >  Disaggregated media refers to the ability for a user to
>     create a
>     >      >  multimedia session combining different media streams,
>     coming from
>     >      >  different devices under his or her control, so that they ar=
e
>     >      >  treated by
>     >      >  the far end of the session as a single media session.
>     >      >  This document lists several use cases that involve
>     >      >  disaggregated media
>     >      >  in SIP.
>     >      >  Additionally, this document analyzes what types of
>     >      >  disaggregated media
>     >      >  can be implemented using existing protocol
>     >      >  mechanisms, and the pros and cons of using each of those
>     mechanisms.
>     >      >  Finally, this document describes scenarios that are not
>     covered by
>     >      >  current mechanisms
>     >      >  and proposes new IETF work to cover them.
>     >      >
>     >      >
>     >      >  cheers
>     >      >  Sal
>     >      >  _______________________________________________
>     >      >  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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<HTML>
<HEAD>
<TITLE>Re: [dispatch] Disaggregated Media in SIP</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Paul,<BR>
<BR>
This is getting interesting, so let&#8217;s see...<BR>
<BR>
&gt;In such a case, using IMs between people, so that they may establish<BR=
>
&gt;separate calls for various media is clearly the poor man's choice -<BR>
&gt;better than nothing but not ideal. And its especially poor if the &quot=
;initial&quot; call is not an IM call. <BR>
&gt;Then just establishing an IM channel<BR>
&gt;between the same two people is non-trivial.<BR>
<BR>
No matter what type of communicator we use: SIP, Skype, Jabber, iChat, etc.=
, most courteous people will first IM to ask if the other party can speak r=
ight now or later. While talking, they may also chose to add video. And if =
planned, also add desktop sharing.<BR>
<BR>
So the natural order of communications seems to be mostly: <BR>
Presence (or the web conference URI), IM, voice, video, desktop sharing, co=
nferencing, but certainly it all starts with a URI for the rendezvous funct=
ion.<BR>
<BR>
It seems we differ (or do we?) on three points:<BR>
</SPAN></FONT><OL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN STYLE=3D'font-size:18pt'>Communicator &#8220;calls&#8221; are not the po=
or mans choice,
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN S=
TYLE=3D'font-size:18pt'>It most often starts with presence and IM,
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN S=
TYLE=3D'font-size:18pt'>The URI (or Skype name) of the other party in the a=
ddress book is all that&#8217;s required. Or the social network as an addre=
ss book.<BR>
</SPAN></FONT></OL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:18pt'><BR>
Now leaving for a moment telephony aside, let&#8217;s look at some new usag=
e scenarios:<BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN STYLE=3D'font-size:18pt'>People in social networks (Twitter, Facebook, e=
tc.)
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN S=
TYLE=3D'font-size:18pt'>Telepresence (business)
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN S=
TYLE=3D'font-size:18pt'>Digital living room (consumer)<BR>
</SPAN></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:18pt'>It is a pretty safe guess that SIP PBX-style voice=
 communications are not the right model for these emerging scenarios. I gue=
ss even contact centers for customer care may benefit from looking at the e=
merging Web applications for communications and especially for sharing of d=
ocuments (having a URI). <BR>
Don&#8217;t we often prefer to IM or email so as not to be tortured by IVR =
Hell?<BR>
And even for voice, just to be called back by an agent?<BR>
<BR>
Now if we could only outlaw &#8220;automata&#8221; &nbsp;&nbsp;:-)<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 7/7/09 8:03 AM, &quot;Paul Kyzivat&quot; &lt;<a href=3D"pkyzivat@cisco.c=
om">pkyzivat@cisco.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>Henry Sinnreich wrote:<BR>
&gt; Paul Kyzivat wrote:<BR>
&gt;&gt;Past suggestions by various people to send control signals (intende=
d<BR>
&gt; tobe acted upon<BR>
&gt;&gt;by automata rather than &gt;people) via IM have generally been<BR>
&gt;&gt;rejected as inappropriate.<BR>
&gt;<BR>
&gt; I am not sure how many people expect a usage scenario for IM with an<B=
R>
&gt; automata in the middle or<BR>
&gt; what the deployment statistics are for such automata (I have never<BR>
&gt; encountered one). <BR>
&gt;<BR>
&gt; All SIP (or other protocol ) Communicator packages have IM and the URI=
<BR>
&gt; works there very nicely.<BR>
&gt;<BR>
&gt; Do you have any usage statistics that justifies the assertion automata=
<BR>
&gt; are the<BR>
&gt; key usage scenario and &#8220;plain person to person&#8221; IM does no=
t count?<BR>
&gt;<BR>
&gt; Henry<BR>
<BR>
I don't know of one either.<BR>
<BR>
But my observation is about using the IM message as a solution to the<BR>
kinds of use cases that were presented in the draft, at least as I<BR>
imagine them playing out.<BR>
<BR>
My mental model is of an environment where a significant fraction of<BR>
people have multimedia UAs, and an expectation that those various media<BR>
will be available in calls that they make. Now add to that people who<BR>
have support for various media, but not in a single device. They then<BR>
want to play in that same multimedia environment.<BR>
<BR>
In such a case, using IMs between people, so that they may establish<BR>
separate calls for various media is clearly the poor man's choice -<BR>
better than nothing but not ideal. And its especially poor if the<BR>
&quot;initial&quot; call is not an IM call. Then just establishing an IM ch=
annel<BR>
between the same two people is non-trivial.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Thanks,<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Paul<BR>
<BR>
<BR>
&gt; On 7/6/09 3:43 PM, &quot;Paul Kyzivat&quot; &lt;<a href=3D"pkyzivat@ci=
sco.com">pkyzivat@cisco.com</a>&gt; wrote:<BR>
&gt;<BR>
&gt;<BR>
&gt;<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;Henry Sinnreich wrote:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &gt;We've looked at various approaches to=
 solve this important<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &gt;problem several times before<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Actually there is one more: IM-ing =
a URI to some resource,<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;mentioned by<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Henning Schulzrinne (I don&#8217;t =
recall the document or presentation).<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;My two cents is that IM-ing a URL i=
s the most general solution, or<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;is it?<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;Past suggestions by various people to send con=
trol signals (intended to<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;be acted upon by automata rather than people) =
via IM have generally been<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;rejected as inappropriate. (The exception so f=
ar has been file transfer,<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;which has some control behavior and some expec=
ted human interaction.)<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;Now if you just want to say &quot;Bob, please =
make a video call to<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;sip:<a href=3D"alice_camera@alice.com">alice_c=
amera@alice.com</a> in order to see me&quot; then I guess IM is ok.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;But IMO its not otherwise good. Its just a hac=
k.<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;Thanks,<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;Paul<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;Henry<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;On 7/6/09 12:07 PM, &quot;Francois =
Audet&quot; &lt;<a href=3D"audet@nortel.com">audet@nortel.com</a>&gt; wrote=
:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;I'm glad to=
 see this topic coming back.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;I see that =
this draft doesn't propose a solution to problem:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;it list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;three optio=
ns, and describes why they are not adequate. I<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;agree with<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;the conclus=
ions.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We've looke=
d at various approaches to solve this important problem<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;several tim=
es before:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Feature r=
ef (refer to urn: indicating specific features)<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;<a href=3D"http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00">=
http://tools.ietf.org/html/draft-audet-sipping-feature-ref-00</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remote co=
ntrol using REFER to requests &amp; responses<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;<a href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-05">http://=
tools.ietf.org/html/draft-mahy-sip-remote-cc-05</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;(Also, versions -04, -03,-02, -00)<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remore co=
ntrol using REFER with XML body describing function<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;<a href=3D"http://tools.ietf.org/html/draft-mahy-sip-remote-cc-01">http://=
tools.ietf.org/html/draft-mahy-sip-remote-cc-01</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- Remote co=
ntrol using MBUS<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;<a href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01">=
http://tools.ietf.org/html/draft-mahy-mmusic-mbus-remotecc-01</a> &amp;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;<a href=3D"http://tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01">http:=
//tools.ietf.org/html/draft-mahy-mmusic-mbus-sdp-01</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;On top of t=
hat there are various proprietary mechanisms, and even<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;some legacy=
<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;PBX-CTI pro=
tocols.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
-----Original Message-----<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
From: <a href=3D"dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a><B=
R>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
[<a href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.=
org</a>] On Behalf Of Salvatore<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;Loreto<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Sent: Monday, July 06, 2009 09:33<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
To: <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Subject: [dispatch] Disaggregated Media in SIP<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Hi there,<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
I have just submitted a draft that talks of Disaggregated<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Media in the Session Initiation Protocol (SIP).<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
<a href=3D"http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa">=
http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;ggregated-m=
edia-00.txt<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Abstract:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Disaggregated media refers to the ability for a user to<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;create a<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
multimedia session combining different media streams,<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;coming from<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
different devices under his or her control, so that they are<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
treated by<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
the far end of the session as a single media session.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
This document lists several use cases that involve<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
disaggregated media<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
in SIP.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Additionally, this document analyzes what types of<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
disaggregated media<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
can be implemented using existing protocol<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
mechanisms, and the pros and cons of using each of those<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;mechanisms.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Finally, this document describes scenarios that are not<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;covered by<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
current mechanisms<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
and proposes new IETF work to cover them.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
cheers<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
Sal<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
_______________________________________________<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
dispatch mailing list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;=
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf=
.org/mailman/listinfo/dispatch</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;___________=
____________________________________<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;dispatch ma=
iling list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"=
dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"=
https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.org/mailma=
n/listinfo/dispatch</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;----------------------------------------------=
--------------------------<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;___________________________________=
____________<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;dispatch mailing list<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<a href=3D"dispatch@ietf.org">dispa=
tch@ietf.org</a><BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&gt; &nbsp;<a href=3D"https://www.ietf.org/mai=
lman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a><=
BR>
&gt;<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


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From AUDET@nortel.com  Tue Jul  7 10:49:15 2009
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References: <4A5261E2.4050506@cisco.com> <C677FFD1.48EB%hsinnrei@adobe.com> <1ECE0EB50388174790F9694F77522CCF1ED91794@zrc2hxm0.corp.nortel.com> <4A5348F5.5030200@cisco.com> <1ECE0EB50388174790F9694F77522CCF1ED91D73@zrc2hxm0.corp.nortel.com> <4A536DF9.4020906@cisco.com> <1ECE0EB50388174790F9694F77522CCF1ED91E95@zrc2hxm0.corp.nortel.com> <4A537B66.4000608@cisco.com>
From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Tuesday, July 07, 2009 09:44
> To: Audet, Francois (SC100:3055)
> Cc: Henry Sinnreich; Salvatore Loreto; dispatch@ietf.org
> Subject: Re: [dispatch] Disaggregated Media in SIP
>
> Of course you are still free to make multiple concurrent=20
> "calls" between the same two parties. But if you do, they are=20
> unrelated, except perhaps in your head.

Actually, the "in your head" part is important.

If it is "loosely coupled", then you don't need to do complex state =
synchronization,
because, as you say, it is "done in your head".

If I have my PC client, and I click "Make call" or "Answer" or something =
similar,
it just needs to tell my Phone to do do so.

That's what I mean by "loose coupling".

The other guy on the other end sees your Phone in this case.

For "Make call", it's just a SIP INVITE coming from the Phone.

For "Answer", the INVITE from the guy gets forked, the PC client tells =
the
phone to send a 200.=20

We don't have to make this complicated.


From pkyzivat@cisco.com  Tue Jul  7 11:17:05 2009
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References: <4A5261E2.4050506@cisco.com> <C677FFD1.48EB%hsinnrei@adobe.com> <1ECE0EB50388174790F9694F77522CCF1ED91794@zrc2hxm0.corp.nortel.com> <4A5348F5.5030200@cisco.com> <1ECE0EB50388174790F9694F77522CCF1ED91D73@zrc2hxm0.corp.nortel.com> <4A536DF9.4020906@cisco.com> <1ECE0EB50388174790F9694F77522CCF1ED91E95@zrc2hxm0.corp.nortel.com> <4A537B66.4000608@cisco.com> <1ECE0EB50388174790F9694F77522CCF1EDE4E4E@zrc2hxm0.corp.nortel.com>
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Inline

Francois Audet wrote:
>  
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Tuesday, July 07, 2009 09:44
>> To: Audet, Francois (SC100:3055)
>> Cc: Henry Sinnreich; Salvatore Loreto; dispatch@ietf.org
>> Subject: Re: [dispatch] Disaggregated Media in SIP
>>
>> Of course you are still free to make multiple concurrent 
>> "calls" between the same two parties. But if you do, they are 
>> unrelated, except perhaps in your head.
> 
> Actually, the "in your head" part is important.
> 
> If it is "loosely coupled", then you don't need to do complex state synchronization,
> because, as you say, it is "done in your head".
> 
> If I have my PC client, and I click "Make call" or "Answer" or something similar,
> it just needs to tell my Phone to do do so.
> 
> That's what I mean by "loose coupling".
> 
> The other guy on the other end sees your Phone in this case.
> 
> For "Make call", it's just a SIP INVITE coming from the Phone.
> 
> For "Answer", the INVITE from the guy gets forked, the PC client tells the
> phone to send a 200. 
> 
> We don't have to make this complicated.

Most of those examples aren't of multimedia at all - they are just 
remote control of a pt-pt call.

And the "in your head" part also means that call control actions, such 
ans transfer, don't work on the collection - just on an individual one. 
And recordings won't know the association between them either.

Nevertheless, I'm not opposed to such things. They all can have a place.

To get anywhere on this I think we need to concentrate on some use 
cases. The ones in Sal's draft are a reasonable starting point. For 
instance:

> 2.6.  Answering a call using Two Separate Devices
> 
>    Laura has a PC with a softclient and a desk phone.  Bob has an
>    integrated advanced multimedia phone with camera.
> 
>    Laura receives on her desk phone an incoming voice-video call from
>    Bob.
> 
>    Laura decides to answer the Bob's session invitation by establishing
>    a voice session with Bob using the desk phone and the video session
>    using her multimedia phone.  Two SIP dialogs need to be established:
>    one between Bob's device and Laura's desk phone and one between Bob's
>    device and Laura's multimedia phone.
> 
>    Laura wants Bob to treat the audio stream from her deskphone and the
>    video stream from her softclient as part of the same communication
>    space.  That is, Laura wants Bob to treat both streams as belonging
>    to the same multimedia session.

To address this without bothering Bob, *something* on Laura's end must 
have the intelligence to coordinate this. And then there must be some 
communications to set it all up.

- we might assume the desk phone, while media poor, is intelligent
   and can handle all the setup. This might be realistic in some cases,
   but probably not often.

- or we might assume that the softphone on her desk is smart enough
   and has a rich enough UI, to handle this. I find that more likely.

Assuming the latter, I can image it popping up a user dialog when it 
receives the INVITE, offering some options of how to handle the call.
(Of course the desk phone is probably ringing too.)
One of the options might be to recruit the desk phone for audio. So 
Laura might just need to click something to select the call setup that 
the use case describes. Then the softphone would answer the call. There 
would be some sort of signaling to incorporate the desk phone, but that 
would all be hidden from Bob. We can discuss what sorts of signaling 
would be good to accomplish this. (I think it sounds like an excellent 
task for BLISS.)

	Thanks,
	Paul

From adam@nostrum.com  Tue Jul  7 12:47:33 2009
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This is a multi-part message in MIME format.
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I've just submitted a new version of draft-roach-sip-http-subscribe, 
which addresses a couple of minor comments made regarding -01 [1]. I 
know there are use cases coming out of the BLISS working group that 
would benefit from such a mechanism. I suspect it has a potential 
community of interest beyond that single working group, which is why I'm 
bringing it to DISPATCH.

I'm happy to keep working on the document as long as there is enough 
interest to make the work worthwhile. If we were to form a mailing list 
with the goal of making progress on this work, who would be interested 
in joining?

/a


[1] The only substantive change was in the third paragraph of section 
2.2, which now requires the server to indicate which URIs are causing it 
to forbid a subscription.

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A new version of I-D, draft-roach-sip-http-subscribe-02.txt has been successfuly submitted by Adam Roach and posted to the IETF repository.

Filename:	 draft-roach-sip-http-subscribe
Revision:	 02
Title:		 A SIP Event Package for Subscribing to Changes to an HTTP Resource
Creation_date:	 2009-07-07
WG ID:		 Independent Submission
Number_of_pages: 12

Abstract:
The Session Initiation Protocol (SIP) is increasingly being used in
systems that are tightly coupled with Hypertext Transport Protocol
(HTTP) servers for a variety of reasons.  In many of these cases,
applications can benefit from being able to discover, in near-real-
time, when a specific HTTP resource is created, changed, or deleted.
This document proposes a mechanism, based on the SIP events
framework, for doing so.

This document further proposes that the HTTP work necessary to make
such a mechanism work be extensible to support protocols other than
SIP for monitoring HTTP resources.
                                                                                  


The IETF Secretariat.



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From michael@voip.co.uk  Tue Jul  7 14:33:12 2009
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2009/7/7 Adam Roach <adam@nostrum.com>:
> I'm happy to keep working on the document as long as there is enough
> interest to make the work worthwhile. If we were to form a mailing list with
> the goal of making progress on this work, who would be interested in
> joining?
>

I'd be interested in joining.  I think there are a number of
interesting things that can be done with a package like this, and I
would be happy to help it progress.  This draft seems to be a
fundamentally useful building block for making advanced applications,
yet remains small enough in scope and application-agnostic to mean we
should be able to achieve consensus relatively quickly.

Michael

From theo@crazygreek.co.uk  Tue Jul  7 16:15:32 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
Date: Wed, 8 Jul 2009 00:08:36 +0100
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On Tue, Jul 7, 2009 at 8:47 PM, Adam Roach<adam@nostrum.com> wrote:

> I'm happy to keep working on the document as long as there is enough
> interest to make the work worthwhile. If we were to form a mailing list with
> the goal of making progress on this work, who would be interested in
> joining?

I would absolutely be interested, and really would like to see this
work progress.

 ~ Theo

From john.elwell@siemens-enterprise.com  Wed Jul  8 02:11:51 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Sal,

Thanks for writing this. The centralised approach is what I and some
others have been suggesting over the past months. It is good to see a
comparison between the centralised approach (in its various flavours)
and the distributed approach that you have been advocating for some
time.

It seems the biggest issue for deciding whether or not to work on the
distributed approach is whether there is a need for the "master" device
(the one operating SIP) to drop out of the call, while allowing the call
and certain media to continue at other devices.=20

I still have a concern that with the distributed approach the remote end
of the call needs to exhibit special behaviour in order to handle the
multiple dialogs arising from the distributed approach (as Paul also
notes). So in practice, to make it work with any degree of reliability,
calls will need to go through a B2BUA that provides the necessary
aggregation. So it effectively ends up as a centralised approach anyway.

The distributed approach (section 4) is not really described in enough
detail. There still has to be some communication between Alice's UAs,
e.g., to convey information to a UA to be added to the call to allow it
to make a call to Bob and allow Bob to associate the new call with the
existing call. Also if Alice wants to clear all calls (i.e., all
components of the same call) with one button push, there needs to be
some communication between the UAs.

John

John

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore Loreto
> Sent: 06 July 2009 17:33
> To: dispatch@ietf.org
> Subject: [dispatch] Disaggregated Media in SIP
>=20
> Hi there,
>=20
> I have just submitted a draft that talks of Disaggregated=20
> Media in the=20
> Session Initiation Protocol (SIP).
> http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
> ggregated-media-00.txt
>=20
>=20
> Abstract:
> Disaggregated media refers to the ability for a user to create a=20
> multimedia session combining different media streams, coming from
> different devices under his or her control, so that they are=20
> treated by=20
> the far end of the session as a single media session.
> This document lists several use cases that involve=20
> disaggregated media=20
> in SIP.
> Additionally, this document analyzes what types of=20
> disaggregated media=20
> can be implemented using existing protocol
> mechanisms, and the pros and cons of using each of those mechanisms.
> Finally, this document describes scenarios that are not covered by=20
> current mechanisms
> and proposes new IETF work to cover them.
>=20
>=20
> cheers
> Sal
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From john.elwell@siemens-enterprise.com  Thu Jul  9 00:14:33 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Adam,

I am interested because of the existing BLISS use cases, but I am sure
there will be other uses. It seems that having this event package
available could save the effort of writing more specific SIP event
packages in certain cases.

John=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: 07 July 2009 20:48
> To: IETF Dispatch
> Subject: [dispatch] SIP HTTP Event Package [Fwd: New Version=20
> Notificationfor draft-roach-sip-http-subscribe-02]
>=20
> I've just submitted a new version of draft-roach-sip-http-subscribe,=20
> which addresses a couple of minor comments made regarding -01 [1]. I=20
> know there are use cases coming out of the BLISS working group that=20
> would benefit from such a mechanism. I suspect it has a potential=20
> community of interest beyond that single working group, which=20
> is why I'm=20
> bringing it to DISPATCH.
>=20
> I'm happy to keep working on the document as long as there is enough=20
> interest to make the work worthwhile. If we were to form a=20
> mailing list=20
> with the goal of making progress on this work, who would be=20
> interested=20
> in joining?
>=20
> /a
>=20
>=20
> [1] The only substantive change was in the third paragraph of section=20
> 2.2, which now requires the server to indicate which URIs are=20
> causing it=20
> to forbid a subscription.
>=20

From john.elwell@siemens-enterprise.com  Thu Jul  9 09:24:15 2009
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Subject: Re: [dispatch] Fwd: [Sipping] Fwd: Expert reviewofdraft-vanelburg-sipping-private-network-indication-03
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My review comments have been resolved with the exception of the
following:

- I failed to find an example showing how it would work for private
network traffic between two enterprises (as claimed in 1.3 last bullet).
Which enterprise is identified, or is there an interworking function
that changes the indicator?

- "There may be cases where treating the call as a public network
       call although both participants are from the same enterprise is
       advantageous to the enterprise."
I requested examples, but I didn't find any in the new draft.


- "The business
   trunking application providing routeing capabilities for the
   enterprise traffic, and supports the identification of calls to and
   from public network users, break-in and break out of that traffic."
This does not parse. I think "providing" should be "provides".

- I didn't have time to check the document for correction of the many
nits that I mentioned in general terms during my original review.

John
=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Hans Erik van Elburg
> Sent: 06 July 2009 12:10
> To: dispatch@ietf.org
> Subject: [dispatch] Fwd: [Sipping] Fwd: Expert=20
> reviewofdraft-vanelburg-sipping-private-network-indication-03
>=20
> The updates after expert review of the=20
> draft-vanelburg-sipping-private-network-indication-03 has=20
> been submitted to dispatch as:
> http://tools.ietf.org/html/draft-vanelburg-dispatch-private-ne
> twork-ind-00
>=20
>=20
> or plain text:
> http://tools.ietf.org/id/draft-vanelburg-dispatch-private-netw
> ork-ind-00.txt
>=20
>=20
> Best Regards,
> /Hans Erik van Elburg
>=20
>=20
>=20
> ---------- Forwarded message ----------
> From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
> Date: Mon, Jul 6, 2009 at 1:06 PM
> Subject: Re: [Sipping] Fwd: Expert review=20
> ofdraft-vanelburg-sipping-private-network-indication-03
> To: "Elwell, John" <john.elwell@siemens-enterprise.com>
> Cc: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, IETF=20
> Sipping List <sipping@ietf.org>
>=20
>=20
>=20
> Draft has been uploaded as=20
> http://tools.ietf.org/html/draft-vanelburg-dispatch-private-ne
> twork-ind-00
>=20
> Best Regards,
> /Hans Erik van Elburg=20
>=20
>=20
> On Mon, Jul 6, 2009 at 11:42 AM, Hans Erik van Elburg=20
> <ietf.hanserik@gmail.com> wrote:
>=20
>=20
> 	Hi John,
>=20
> 	1. Done, moved 10 up and inserted as 1.2. Added the=20
> following sentence to 1.1:
> 	"3GPP and ETSI TISPAN have identified a set of=20
> requirements that can be met by defining a new SIP P-header,=20
> according to the procedures in RFC 3427 [RFC3427]."
> =09
> 	2. Done
> 	4a. Done
> 	4b. Done
> 	4c. Done
> 	5. I added the following terminology description:
>=20
> 		"3.1. Traffic
> 	=09
> 		In the context of this document the term=20
> traffic is understood as all
> 		communication pertaining to and/or controlled=20
> by a SIP transaction or
> 		dialog."
>=20
> 	8. Done
>=20
> 	Will upload this as cutoff day for 00 drafts is today.
>=20
> 	/Hans Erik van Elburg=20
>=20
>=20
>=20
> 	On Mon, Jul 6, 2009 at 8:52 AM, Elwell, John=20
> <john.elwell@siemens-enterprise.com> wrote:
> =09
>=20
> 		Hans Erik,
> 	=09
>=20
> 		> -----Original Message-----
> 		> From: sipping-bounces@ietf.org
> 		> [mailto:sipping-bounces@ietf.org] On Behalf=20
> Of Hans Erik van Elburg
> 		> Sent: 03 July 2009 15:17
> 		> To: Elwell, John; DRAGE, Keith (Keith)
> 		> Cc: IETF Sipping List
> 		> Subject: [Sipping] Fwd: Expert review
> 		>=20
> ofdraft-vanelburg-sipping-private-network-indication-03
> 		>
> 		> Resent and history removed as otherwise=20
> rejected by mailing list.
> 		>
> 		> Hi John,
> 		>
> 		> 1. Again the required applicability statement=20
> is in section
> 		> 10 of the main body. (Same pattern has been=20
> followed in
> 		> RFC5502, 5002 and 4457) which I took as=20
> examples. Together
> 		> with the additional text in the abstract that=20
> should be
> 		> sufficient. If the text in section 10 needs=20
> to be improved
> 		> then please indicate so.
> 	=09
> 		[JRE] I must have written the original comment=20
> 1 before I got down to
> 		section 10. Basically section 10 is far too=20
> late in the document, and I
> 		would have expected the statement, perhaps as=20
> section 1, or perhaps as a
> 		section 1.2 before the existing 1.2.
> 	=09
>=20
>=20
> 		>
> 		> 2. Took in your nits, the abstract now reads:
> 		>       "This document specifies the SIP=20
> P-Private-Network-Indication
> 		> P-header. The use of this private network=20
> indication extension
> 		> is only applicable inside an administrative=20
> domain with
> 		> previously agreed-upon policies for=20
> generation, transport and
> 		> usage of such information.  A private network=20
> indication
> 		> allows nodes in such a domain to treat=20
> private network traffic
> 		> according to a different set of rules then=20
> than the set
> 		> applicable to public network traffic. The=20
> indication also
> 		> distinguishes traffic from one private=20
> network from another
> 		> private network."
> 	=09
> 		[JRE]
> 		Delete the word "then".
> 	=09
>=20
> 		>
> 		> 4a. Regarding 1.2 item b). Would the=20
> following rewrite help?:
> 		> OLD:
> 		> " b) business trunking application, where the=20
> NGN hosts
> 		> transit capabilities between NGCN's, break-in=20
> capabilities
> 		> from NGN to NGCN and break-out capabilities=20
> from NGCN to NGN; and"
> 		> /OLD
> 		>
> 		>
> 		> NEW:
> 		> " b) business trunking application, where the=20
> NGN hosts
> 		> transit capabilities between NGCN's, break-in=20
> capabilities
> 		> where the NGN converts public network traffic=20
> to private
> 		> network traffic for delivery at a served NGCN=20
> and break-out
> 		> capabilities where the NGN converts private=20
> network traffic
> 		> from a served NGCN to public network traffic; and"
> 		>
> 		> /NEW
> 		>
> 		> If not please suggest an improved sentence=20
> that covers your
> 		> concern. Please note that in the example that=20
> you gave it is
> 		> not the business trunking application that does the
> 		> conversion, but the hosted enterprise application.
> 	=09
> 		[JRE] That would do.
> 	=09
>=20
>=20
> 		>
> 		> 4b. "normal rules"
> 		> Looking at the definition, would the=20
> following rewrite help:
> 		> OLD
> 		> " Traffic sent to or received from a public=20
> telecommunication
> 		> network for processing according to the normal rules."
> 		> /OLD
> 		>
> 		> NEW
> 		> " Traffic sent to or received from a public=20
> telecommunication
> 		> network for processing according to the rules=20
> for ordinary
> 		> subscribers of a public telecommunication network."
> 		> /NEW
> 	=09
> 		[JRE] That would do.
> 	=09
>=20
> 		>
> 		> 4c. NGN is a public telecommunication network
> 		> >       - "To summarize a few example reasons=20
> for a public
> 		> telecommunication network to make the=20
> distinction between the
> 		> two types of traffic" Isn't it an NGN that=20
> needs to make the
> 		> distinction?
> 		> > [HE] NGN is a public telecommunication=20
> network. But we can
> 		> rephrase to: "To summarize a few example=20
> reasons for a public
> 		> NGN to make the distinction between the two=20
> types of traffic" [/HE]
> 	=09
> 		[JRE] OK
> 	=09
>=20
> 		>
> 		> [JRE] I think that is the problem I am=20
> having. I believe an
> 		> NGN is a network infrastructure that supports=20
> both public
> 		> network traffic and private network traffic,=20
> or in other
> 		> words it supports both a public network and a=20
> number of
> 		> private networks.[/JRE]
> 		> [HE2]Yes, but its main purpose is to serve as a public
> 		> network with all the regulations that apply=20
> to such networks
> 		> etc. This does not prohibit an NGN to be used=20
> as a private
> 		> network of course, but still it has been=20
> designed from the
> 		> perspective of having to serve the needs of=20
> public network
> 		> operators. That is why the "normal"/default =20
> procedures are
> 		> for public network traffic. As we want to=20
> introduce the
> 		> capability for such a public network (NGN) to=20
> also support
> 		> private network traffic that has to be processed to a
> 		> different set of rules, the=20
> Private-Network-Indication is
> 		> needed.[/HE2]
> 		>
> 		>
> 		> 5. Traffic --> SIP traffic
> 		> Calling traffic SIP traffic suggest that the=20
> media is not
> 		> part of the traffic, is that what we want??
> 	=09
> 		[JRE] The point is, it is not HTTP traffic or=20
> FTP traffic or SMTP
> 		traffic or whatever.
> 	=09
>=20
> 		>
> 		>
> 		> 6. Example private network traffic can also=20
> exist between two
> 		> different enterprises
> 		> Where would you like to see this? Obviously=20
> section 1.2 is
> 		> not the right place for such an example.
> 		> Isn't this too obvious, if you imagine that=20
> two companies
> 		> have agreed to form a cooperation and=20
> communicate under this
> 		> agreement?
> 		> Would you like to see this as an additional=20
> example in section 4?
> 	=09
> 		[JRE] I was not necessarily seeking a further=20
> example. However, given
> 		that the private network indication identifies=20
> the particular private
> 		network, what form does this indication take=20
> when traffic is between two
> 		enterprises? Is there some interworking=20
> function where the identifier
> 		changes from that of the first private network=20
> to that of the second
> 		private network?
> 	=09
>=20
> 		>
> 		>
> 		> 8. calling line identification
> 		> Yes we agree fully on that "calling line=20
> identification is
> 		> not an adequate distinction". I think that is what the
> 		> current text tries to explain. Actually what=20
> I dont like
> 		> about this text is that it starts explaining what the
> 		> indication is not about. I propose that we=20
> completely remove
> 		> the 1st paragraph in section 1.3 and the 1st=20
> word "Rather"
> 		> from the 2nd paragraph.
> 	=09
> 		[JRE] Yes. And in the second paragraph, perhaps=20
> we could enhance:
> 		"This
> 		  indication is not for the end user on the=20
> private network."
> 		to say:
> 		"This indication does not identify an end user=20
> on a private network and
> 		is not for delivery to an end user on the=20
> private network."
> 	=09
> 	=09
> 		John
> 	=09
> 	=09
> 		>
> 		> /Hans Erik van Elburg
> 		>
> 		>
> 		>
> 		>
> 	=09
>=20
>=20
>=20
>=20
>=20

From HKaplan@acmepacket.com  Thu Jul  9 10:36:40 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 9 Jul 2009 13:37:06 -0400
Thread-Topic: Fwd: I-D Action:draft-kaplan-dispatch-sip-implicit-registrations-00.txt 
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Howdy,
As part of a discussion in the SIP Forum regarding a model for Registration=
 we are using for IP-PBX's, I was asked by someone to submit an I-D describ=
ing it, in case it would be of any value.  Since I hadn't seen an I-D befor=
e on how 3GPP/ETSI did their model, and since their model is different from=
 SIP-Forum's, I submitted one which describes the models for both SIP Forum=
's SIP-Connect v1.1 and 3GPP/ETSI.

NOTE: this draft was not sanctioned nor reviewed by the SIP-Forum nor 3GPP,=
 and I am still talking with colleagues in 3GPP to make sure I got that par=
t right.  This is purely an individual submission in all respects.

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

	Title           : Session Initiation Protocol (SIP) Implicit Registrations
	Author(s)       : H. Kaplan
	Filename        : draft-kaplan-dispatch-sip-implicit-registrations-00.txt
	Pages           : 12
	Date            : 2009-07-05

This document identifies several approaches to provide reachability informa=
tion for a domain or multiple AoR's using a single SIP REGISTER method tran=
saction, in ways not originally envisioned or documented by RFC 3261.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-kaplan-dispatch-sip-implicit-regi=
strations-00.txt

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

Below is the data which will enable a MIME compliant mail reader implementa=
tion to automatically retrieve the ASCII version of the Internet-Draft.


From gonzalo.camarillo@ericsson.com  Thu Jul  9 13:48:18 2009
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Date: Thu, 09 Jul 2009 23:48:42 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [dispatch] Input requested: abandoning draft-ietf-sipping-profile-datasets-03?
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Folks,

the SIPPING WG agreed to work on the following draft a long time ago:

http://tools.ietf.org/html/draft-ietf-sipping-profile-datasets-03

As you can see in the following slide, the media-policy draft depends on 
the draft above:

http://www.ietf.org/proceedings/07jul/slides/sipping-4/sld11.htm

While there are people interested in the media-policy draft, we have not 
found anyone interested in the profile-datasets draft.

One of the things we agreed to do when restructured RAI and chartered 
DISPATCH was to drop items people were no longer interested in. 
Therefore, at this point, we are considering abandoning the 
profile-dataset draft. If we do this, we would need to import a few 
things from that draft to the media-policy draft (e.g., the merging 
rules) but the idea would be to only progress the media-policy draft.

Does anyone have any objection to this way forward?

Note that in order to progress a draft we need people interested in it 
and people willing to put cycles on it. Therefore, if you are interested 
in keeping the draft alive, you will be automatically volunteering to 
put cycles on it in order to progress it ;o)

Thanks,

Gonzalo
DISPATCH co-chair

From AUDET@nortel.com  Thu Jul  9 13:52:17 2009
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Subject: Re: [dispatch] Input requested: abandoningdraft-ietf-sipping-profile-datasets-03?
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...and is anybody interested in the media-policy draft?=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> Sent: Thursday, July 09, 2009 13:49
> To: DISPATCH
> Subject: [dispatch] Input requested:=20
> abandoningdraft-ietf-sipping-profile-datasets-03?
>=20
> Folks,
>=20
> the SIPPING WG agreed to work on the following draft a long time ago:
>=20
> http://tools.ietf.org/html/draft-ietf-sipping-profile-datasets-03
>=20
> As you can see in the following slide, the media-policy draft=20
> depends on the draft above:
>=20
> http://www.ietf.org/proceedings/07jul/slides/sipping-4/sld11.htm
>=20
> While there are people interested in the media-policy draft,=20
> we have not found anyone interested in the profile-datasets draft.
>=20
> One of the things we agreed to do when restructured RAI and=20
> chartered DISPATCH was to drop items people were no longer=20
> interested in.=20
> Therefore, at this point, we are considering abandoning the=20
> profile-dataset draft. If we do this, we would need to import=20
> a few things from that draft to the media-policy draft (e.g.,=20
> the merging
> rules) but the idea would be to only progress the media-policy draft.
>=20
> Does anyone have any objection to this way forward?
>=20
> Note that in order to progress a draft we need people=20
> interested in it and people willing to put cycles on it.=20
> Therefore, if you are interested in keeping the draft alive,=20
> you will be automatically volunteering to put cycles on it in=20
> order to progress it ;o)
>=20
> Thanks,
>=20
> Gonzalo
> DISPATCH co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

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From gonzalo.camarillo@ericsson.com  Thu Jul  9 21:40:33 2009
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Subject: Re: [dispatch] Input requested: abandoningdraft-ietf-sipping-profile-datasets-03?
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Hi,

yes, as I stated in my email below, there are people interested in the 
functionality that draft provides. However, we have not found anybody 
interested in defining new profiles (besides the one defined in the 
media-policy draft).

Cheers,

Gonzalo


Francois Audet wrote:
> ...and is anybody interested in the media-policy draft? 
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>> Sent: Thursday, July 09, 2009 13:49
>> To: DISPATCH
>> Subject: [dispatch] Input requested: 
>> abandoningdraft-ietf-sipping-profile-datasets-03?
>>
>> Folks,
>>
>> the SIPPING WG agreed to work on the following draft a long time ago:
>>
>> http://tools.ietf.org/html/draft-ietf-sipping-profile-datasets-03
>>
>> As you can see in the following slide, the media-policy draft 
>> depends on the draft above:
>>
>> http://www.ietf.org/proceedings/07jul/slides/sipping-4/sld11.htm
>>
>> While there are people interested in the media-policy draft, 
>> we have not found anyone interested in the profile-datasets draft.
>>
>> One of the things we agreed to do when restructured RAI and 
>> chartered DISPATCH was to drop items people were no longer 
>> interested in. 
>> Therefore, at this point, we are considering abandoning the 
>> profile-dataset draft. If we do this, we would need to import 
>> a few things from that draft to the media-policy draft (e.g., 
>> the merging
>> rules) but the idea would be to only progress the media-policy draft.
>>
>> Does anyone have any objection to this way forward?
>>
>> Note that in order to progress a draft we need people 
>> interested in it and people willing to put cycles on it. 
>> Therefore, if you are interested in keeping the draft alive, 
>> you will be automatically volunteering to put cycles on it in 
>> order to progress it ;o)
>>
>> Thanks,
>>
>> Gonzalo
>> DISPATCH co-chair
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>


From salvatore.loreto@ericsson.com  Fri Jul 10 01:41:03 2009
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [dispatch] SIP HTTP Event Package [Fwd: New Version Notification for draft-roach-sip-http-subscribe-02]
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Hi Adam,

I'd be interested in joining, but to discuss and decide about the real 
need of this package.

I would understand better:
- the advantage of having a SIP Event Package to monitor the changes in 
an HTTP resource ;
  (by the way the draft doesn't explain how it would be possible monitor 
the "creation" or the "deletion" of
   an HTTP resource!)

versus

-using directly the HTTP long-polling techniques that can perform at 
same time the monitoring
and the content transfer of HTTP resources.
(for an overview of the HTTP long-polling techniques check:   
http://www.ietf.org/internet-drafts/draft-loreto-http-bidirectional-00.txt )



thanks
Sal

Adam Roach wrote:
> I've just submitted a new version of draft-roach-sip-http-subscribe, 
> which addresses a couple of minor comments made regarding -01 [1]. I 
> know there are use cases coming out of the BLISS working group that 
> would benefit from such a mechanism. I suspect it has a potential 
> community of interest beyond that single working group, which is why 
> I'm bringing it to DISPATCH.
>
> I'm happy to keep working on the document as long as there is enough 
> interest to make the work worthwhile. If we were to form a mailing 
> list with the goal of making progress on this work, who would be 
> interested in joining?
>
> /a
>
>
> [1] The only substantive change was in the third paragraph of section 
> 2.2, which now requires the server to indicate which URIs are causing 
> it to forbid a subscription.
>
> ------------------------------------------------------------------------
>
> Subject:
> New Version Notification for draft-roach-sip-http-subscribe-02
> From:
> IETF I-D Submission Tool <idsubmission@ietf.org>
> Date:
> Tue, 7 Jul 2009 12:26:36 -0700 (PDT)
> To:
> adam@nostrum.com
>
> To:
> adam@nostrum.com
>
>
> A new version of I-D, draft-roach-sip-http-subscribe-02.txt has been successfuly submitted by Adam Roach and posted to the IETF repository.
>
> Filename:	 draft-roach-sip-http-subscribe
> Revision:	 02
> Title:		 A SIP Event Package for Subscribing to Changes to an HTTP Resource
> Creation_date:	 2009-07-07
> WG ID:		 Independent Submission
> Number_of_pages: 12
>
> Abstract:
> The Session Initiation Protocol (SIP) is increasingly being used in
> systems that are tightly coupled with Hypertext Transport Protocol
> (HTTP) servers for a variety of reasons.  In many of these cases,
> applications can benefit from being able to discover, in near-real-
> time, when a specific HTTP resource is created, changed, or deleted.
> This document proposes a mechanism, based on the SIP events
> framework, for doing so.
>
> This document further proposes that the HTTP work necessary to make
> such a mechanism work be extensible to support protocols other than
> SIP for monitoring HTTP resources.
>                                                                                   
>
>
> The IETF Secretariat.
>
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>   


From theo@crazygreek.co.uk  Fri Jul 10 06:17:03 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
Date: Fri, 10 Jul 2009 14:17:09 +0100
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Subject: Re: [dispatch] SIP HTTP Event Package [Fwd: New Version Notification for draft-roach-sip-http-subscribe-02]
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On Fri, Jul 10, 2009 at 9:41 AM, Salvatore
Loreto<salvatore.loreto@ericsson.com> wrote:

> - the advantage of having a SIP Event Package to monitor the changes in a=
n
> HTTP resource ;
> =A0(by the way the draft doesn't explain how it would be possible monitor=
 the
> "creation" or the "deletion" of
> =A0an HTTP resource!)

see the last paragraph in 3.5.1 for deletion...

> -using directly the HTTP long-polling techniques that can perform at same
> time the monitoring
> and the content transfer of HTTP resources.
> (for an overview of the HTTP long-polling techniques check:
> http://www.ietf.org/internet-drafts/draft-loreto-http-bidirectional-00.tx=
t )

This would require the connection between the notifier and the server
to stay open.  When a single client is monitoring tens of thousands of
resources potentially over as many servers, this becomes very
unfriendly.  Consider that the period between a SUBSCRIBE and being
notified of changes could potentially be days.  Additionally,
monitoring can be done by millions of clients to a single server -
each of which would need an open connection to the server(s).

 ~ Theo

From salvatore.loreto@ericsson.com  Fri Jul 10 06:40:46 2009
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Subject: Re: [dispatch] SIP HTTP Event Package [Fwd: New Version Notification for draft-roach-sip-http-subscribe-02]
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don't take me wrong, I am just thinking about the pro and cons...
in line!

Theo Zourzouvillys wrote:
> On Fri, Jul 10, 2009 at 9:41 AM, Salvatore
> Loreto<salvatore.loreto@ericsson.com> wrote:
>
>   
>> - the advantage of having a SIP Event Package to monitor the changes in an
>> HTTP resource ;
>>  (by the way the draft doesn't explain how it would be possible monitor the
>> "creation" or the "deletion" of
>>  an HTTP resource!)
>>     
>
> see the last paragraph in 3.5.1 for deletion...
>   
right!
but then what about "creation" ?
>   
>> -using directly the HTTP long-polling techniques that can perform at same
>> time the monitoring
>> and the content transfer of HTTP resources.
>> (for an overview of the HTTP long-polling techniques check:
>> http://www.ietf.org/internet-drafts/draft-loreto-http-bidirectional-00.txt )
>>     
>
> This would require the connection between the notifier and the server
> to stay open.
right a point in favour of the Event package is that SIP goes over UDP
>   When a single client is monitoring tens of thousands of
> resources potentially over as many servers, this becomes very
> unfriendly.  
How to monitoring more resources on the same server is something that 
people are starting to think
also for web-application using HTTP.
> Consider that the period between a SUBSCRIBE and being
> notified of changes could potentially be days.
right, but then you have refresh the SUBSCRIBE... etc etc.
>   Additionally,
> monitoring can be done by millions of clients to a single server -
> each of which would need an open connection to the server(s).
>   
that is not a real problem, with HTML5 and the growing number of web 
applications a Server (a Web Server in this case)
will end up in exactly the same scenario.
>  ~ Theo

From theo@crazygreek.co.uk  Fri Jul 10 07:04:44 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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On Fri, Jul 10, 2009 at 2:41 PM, Salvatore
Loreto<salvatore.loreto@ericsson.com> wrote:

> right!
> but then what about "creation" ?

Sorry, i missed that bit :-)

My understanding (and how we implement it) is server could return a
Link header in a 404 response in the same way as a 200, which can then
be monitored.  Of course, servers may elect not to return a Link
header for the 404, or could reject the SUBSCRIBE request as received
if they don't support monitoring of non-existent resources.

>> This would require the connection between the notifier and the server
>> to stay open.
>
> right a point in favour of the Event package is that SIP goes over UDP

or one connection for many clients on the same proxy.

>> Consider that the period between a SUBSCRIBE and being
>> notified of changes could potentially be days.
>
> right, but then you have refresh the SUBSCRIBE... etc etc.

only for every <expiry period> interval!

>> =A0Additionally,
>> monitoring can be done by millions of clients to a single server -
>> each of which would need an open connection to the server(s).
>>
>
> that is not a real problem, with HTML5 and the growing number of web
> applications a Server (a Web Server in this case)
> will end up in exactly the same scenario.

agreed.

Note that this is just one protocol that can be used for monitoring
(using the Link 'monitor' header).  There could be others, such as
over bi-direcitonal HTTP, XMPP, etc :)

 ~ Theo

From adam@nostrum.com  Fri Jul 10 08:40:50 2009
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Theo Zourzouvillys wrote:
> On Fri, Jul 10, 2009 at 2:41 PM, Salvatore
> Loreto<salvatore.loreto@ericsson.com> wrote:
>
>   
>> right!
>> but then what about "creation" ?
>>     
>
> Sorry, i missed that bit :-)
>
> My understanding (and how we implement it) is server could return a
> Link header in a 404 response in the same way as a 200, which can then
> be monitored.  Of course, servers may elect not to return a Link
> header for the 404, or could reject the SUBSCRIBE request as received
> if they don't support monitoring of non-existent resources.
>   

That seems like a really good idea. I'll put it on the list for the next 
version.

/a

>>> Consider that the period between a SUBSCRIBE and being
>>> notified of changes could potentially be days.
>>>       
>> right, but then you have refresh the SUBSCRIBE... etc etc.
>>     
>
> only for every <expiry period> interval!
>   

And the draft explicitly addresses this point:

   Subscriptions to slower-changing resources lack this property, and
   the need to periodically refresh subscriptions render short
   subscriptions wasteful.  For these type of subscriptions, expirations
   as long as 604800 (one week) or even longer may well make sense.


/a

From dworley@nortel.com  Fri Jul 10 12:42:07 2009
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From: "Dale Worley" <dworley@nortel.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [dispatch] Disaggregated Media in SIP
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On Tue, 2009-07-07 at 09:09 -0400, Paul Kyzivat wrote:
> I still remain unconvinced that it is either necessary or appropriate 
> for one end of a call to be involved in how the other end aggregates 
> media resources for the call.

To support this, I would say that a *primary* requirement for a solution
should be that the "other end" does not have to be aware how "this end"
is arranging for multiple media source/sinks.  Otherwise, there is no
interoperability with UAs that do not understand the aggregation scheme.

Reading some of the discussion, I realized there is already a *very*
common instance of disaggregated media:  music-on-hold.  UA X puts a
call on hold, and then arranges for the music-on-hold source to send RTP
to UA Y.  The entire design problem for music-on-hold is arranging for
UA Y *not* to have to understand what is going on.

A typical solution is
http://tools.ietf.org/html/draft-worley-service-example-03.  We've had
luck getting phone manufacturers to implement this, and there are a
number of similar schemes that other phones use.

Dale



From dworley@nortel.com  Fri Jul 10 12:51:14 2009
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On Tue, 2009-07-07 at 07:13 +0200, R.Jesske@telekom.de wrote:
> Hi Dale,
> Within the SIPPING group I had already discussions on that issue.
> The assumption was that the use of the Reason Header within Responses is conditionally allowed. 
> 
> The RFC 3326 says:
>    Initially, the Reason header field defined here appears to be most
>    useful for BYE and CANCEL requests, but it can appear in any request
>    within a dialog, in any CANCEL request and in any response whose
>    status code explicitly allows the presence of this header field.
> 
> 
> So we need a draft where it is stated that a reason header can be included within the SIP Response. That was also the recommendation of that discussion. That is why I wrote a draft.
> Nevertheless some standards already include this behaviour like the ITU-T Q.1912.5 specification.
> Also 3GPP needs the reason header within Responses.

It is still unclear to me what is being changed.  Is the change the
replacement of "conditionally allowed" to "allowed"?  In any case, in my
mind the purpose of the Reason header is to carry Q.850 cause-codes in
failure responses.

The point is that the Abstract is not at all clear about the change from
the status quo (at least to someone who isn't already thoroughly
familiar with how Reason is currently specified).

And the secondary point is that if the Abstract was clearer about what
it is being changed, more people might read and discuss the draft.

Dale



From volkerh@bell-labs.com  Sat Jul 11 09:33:11 2009
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Subject: Re: [dispatch] SIP Overload Control Design: Loop Control
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Tim,

the case you are describing is discussed in the current draft in the 
Topologies section:

    If A can reliably determine that D, E and F are its only downstream
    neighbors and all of them are in overload, it may choose to report
    overload upstream on behalf of D, E and F.

The section you were referring to discusses end-to-end overload control, 
however, the example is a hop-by-hop control.

Thanks,

Volker

Moran, Timothy L. wrote:
> Volker,
> 
> 
> 
> A comment on the control loop paragraph in section 2 of the design 
> considerations for OLC:
> 
> “A control loop spanning multiple hops can be used if the sending
> entity has full knowledge about the SIP servers on the path of a SIP
> message.”
> 
> 
> 
> I read this as the sending entity is required to have full knowledge
> of the entire path.  I offer that that may not be the case.  It may
> be possible to only require knowledge of one _level_ downstream.
> This sounds pretty much like your example except that the example
> limits the case to “one” downstream server.  Referring back to the
> requirements case of two load balancing servers and the upstream
> server “PA”, server PA can declare overload upstream by knowing that
> servers P1 _and_ P2 are overloaded.
> 
> 
> 
> Perhaps what I am trying to say is that an upstream server in a
> managed network would declare overload (upstream) based on knowledge
> that all next-hop servers are overloaded.
> 
> 
> 
> Br,
> 
> Tim M.
> 
> 
> 
> 
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> 
> _______________________________________________ dispatch mailing list
>  dispatch@ietf.org https://www.ietf.org/mailman/listinfo/dispatch


From R.Jesske@telekom.de  Sun Jul 12 23:16:39 2009
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Hi Dale,
The main probblem is that reasons are currently not allowed in =
responses. Neither conditionally nor allowed

Please see: =
http://www.ietf.org/mail-archive/web/sipping/current/msg09682.html

The draft tries to close this gap and allows conditionally Q.850 =
resasons within Resopnses. So it would be a service provider decision to =
include Reasons in Responses when mapping from ISUP to SIP.

The following call flow shows an example:

         A                Gateway             Gateway              B=20
         |        IAM        |                  |                  |=20
         |------------------>|     INVITE       |                  |=20
         |                   |----------------->|      IAM         |=20
         |                   |     100 Trying   |----------------->|=20
         |                   |<-----------------|    100 Trying    |=20
         |                   |   403 Decline    |                  |
         |                   |  Reason Q850 #87 |  REL Cause #87   |
         |   REL Cause #87   |                  |<-----------------|=20
         |                   |<-----------------|                  |=20
         |<----------------- |                  |                  |=20
         |                   |                  |                  |=20
         |                   |                  |                  |=20
         |                   |                  |                  |=20

Due to RFC 3398 a 403 will be mapped back to a ISUP Cause 21. With the =
Reason in responses the Cause Value 87 will be secured and transferred =
back to ISUP.

Mheanwile I got some indications from some persons who see this issue =
within the BLISS group due to the fact that this draft solves =
interoperability issues within SIP and PSTN.

Views?

BR,

Roland
=20

-----Urspr=FCngliche Nachricht-----
Von: Dale Worley [mailto:dworley@nortel.com]=20
Gesendet: Freitag, 10. Juli 2009 21:52
An: Jesske, Roland
Cc: dispatch@ietf.org
Betreff: Re: AW: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

On Tue, 2009-07-07 at 07:13 +0200, R.Jesske@telekom.de wrote:
> Hi Dale,
> Within the SIPPING group I had already discussions on that issue.
> The assumption was that the use of the Reason Header within Responses =
is conditionally allowed.=20
>=20
> The RFC 3326 says:
>    Initially, the Reason header field defined here appears to be most
>    useful for BYE and CANCEL requests, but it can appear in any =
request
>    within a dialog, in any CANCEL request and in any response whose
>    status code explicitly allows the presence of this header field.
>=20
>=20
> So we need a draft where it is stated that a reason header can be =
included within the SIP Response. That was also the recommendation of =
that discussion. That is why I wrote a draft.
> Nevertheless some standards already include this behaviour like the =
ITU-T Q.1912.5 specification.
> Also 3GPP needs the reason header within Responses.

It is still unclear to me what is being changed.  Is the change the
replacement of "conditionally allowed" to "allowed"?  In any case, in my
mind the purpose of the Reason header is to carry Q.850 cause-codes in
failure responses.

The point is that the Abstract is not at all clear about the change from
the status quo (at least to someone who isn't already thoroughly
familiar with how Reason is currently specified).

And the secondary point is that if the Abstract was clearer about what
it is being changed, more people might read and discuss the draft.

Dale



From salvatore.loreto@ericsson.com  Mon Jul 13 01:32:42 2009
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Subject: Re: [dispatch] SIP HTTP Event Package [Fwd: New Version Notification for draft-roach-sip-http-subscribe-02]
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Theo Zourzouvillys wrote:
> On Fri, Jul 10, 2009 at 2:41 PM, Salvatore
> Loreto<salvatore.loreto@ericsson.com> wrote:
>
>   
>> right!
>> but then what about "creation" ?
>>     
>
> Sorry, i missed that bit :-)
>
> My understanding (and how we implement it) is server could return a
> Link header in a 404 response in the same way as a 200, which can then
> be monitored.  Of course, servers may elect not to return a Link
> header for the 404, or could reject the SUBSCRIBE request as received
> if they don't support monitoring of non-existent resources.
>   
this could be a possibility,
but it should be explicitly insert the SUBSCRIBE the willingness to 
monitor also *non-existent resource*
moreover  if a subscriber has several subscriptions to the same server, 
it should be avoided that the server sends
this kind of information to each single subscription.
for example the server should accept only one SUBSCRIBE willing to 
monitor *non-existent resource* per each UA.

/Sal

From RIFATYU@nortel.com  Mon Jul 13 12:47:55 2009
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From: "Rifaat Shekh-Yusef" <rifatyu@nortel.com>
To: "Adam Roach" <adam@nostrum.com>, "IETF Dispatch" <dispatch@ietf.org>
Subject: Re: [dispatch] SIP HTTP Event Package [Fwd: New Version Notificationfor draft-roach-sip-http-subscribe-02]
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I am interested and willing to help this work progress.

Regards,
 Rifaat

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Adam Roach
Sent: Tuesday, July 07, 2009 3:48 PM
To: IETF Dispatch
Subject: [dispatch] SIP HTTP Event Package [Fwd: New Version
Notificationfor draft-roach-sip-http-subscribe-02]

I've just submitted a new version of draft-roach-sip-http-subscribe,
which addresses a couple of minor comments made regarding -01 [1]. I
know there are use cases coming out of the BLISS working group that
would benefit from such a mechanism. I suspect it has a potential
community of interest beyond that single working group, which is why I'm
bringing it to DISPATCH.

I'm happy to keep working on the document as long as there is enough
interest to make the work worthwhile. If we were to form a mailing list
with the goal of making progress on this work, who would be interested
in joining?

/a


[1] The only substantive change was in the third paragraph of section=20
2.2, which now requires the server to indicate which URIs are causing it

to forbid a subscription.

From scott.lawrence@nortel.com  Tue Jul 14 07:34:45 2009
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From: "Scott Lawrence" <scott.lawrence@nortel.com>
To: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Adam wrote:


> I've just submitted a new version of draft-roach-sip-http-subscribe, 
> which addresses a couple of minor comments made regarding -01 [1]. I 
> know there are use cases coming out of the BLISS working group that 
> would benefit from such a mechanism. I suspect it has a potential 
> community of interest beyond that single working group, which is why
> I'm bringing it to DISPATCH.

I think this draft is a great idea - I've got a few comments, but will
send those separately

On Fri, 2009-07-10 at 11:41 +0300, Salvatore Loreto wrote:

> I'd be interested in joining, but to discuss and decide about the real 
> need of this package.
> 
> I would understand better:
> - the advantage of having a SIP Event Package to monitor the changes in 
> an HTTP resource ;
>   (by the way the draft doesn't explain how it would be possible monitor 
> the "creation" or the "deletion" of
>    an HTTP resource!)

I don't think that the mechanisms used to implement the service need to
be a part of the discussion at all.

The draft should concern itself with the on-the-wire behavior of the
event package, not how the service providing it gets the information it
needs.  There are any number of ways it could be done, including
integrating a SIP UA into the web server, the file store behind the web
server, some application providing data via the web server, etc.  None
of that needs to be (or should be) in the draft defining the event
package.



From dworley@nortel.com  Thu Jul 16 10:02:09 2009
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From: "Dale Worley" <dworley@nortel.com>
To: R.Jesske@telekom.de
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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On Mon, 2009-07-13 at 08:16 +0200, R.Jesske@telekom.de wrote:

> The main probblem is that reasons are currently not allowed in
> responses. Neither conditionally nor allowed
> 
> Please see: http://www.ietf.org/mail-archive/web/sipping/current/msg09682.html

That clarifies things.

Let me rephrase my complaint:  Although the text of the RFC states "The
Reason header field MAY appear in [...] any response whose status code
explicitly allows the presence of this header field.", that text is
stunningly unclear that the consequence is that the Reason header is not
allowed in *any* defined response, and a casual reader might not realize
that Reason is thus effectively forbidden in all responses.

Expanding the Abstract along these lines would make the significance and
importance of the draft much clearer:

        Although the use of the Reason header in responses is considered
        in RFC 3326, doing so is not specified for any existing response
        code.  This document specifies the use of the Reason header
        field in SIP responses to carry Q.850 reason codes for the
        failure of an INVITE.
        
Dale



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On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois (SC100:3055) wrote:
> Hi Roland,
> 
> You use case is very common.  
> 
> I believe you are incorrect in saying that "reasons are currently 
> not allowed in responses. Neither conditionally nor allowed".
> 
> RFC 3326 says in 1.0:
>   "[...] it can appear in any request
>    within a dialog, in any CANCEL request and in any response whose
>    status code explicitly allows the presence of this header field."
> 
> To be honest, I believe Q.850 codes are much more common in 
> Responses than in requests.

Googling

     "sip/2.0" reason "q.850"

turns up numerous examples of SIP responses using the Reason header in
the forbidden manner.

I'd say that your draft formally allows what people are already doing.

Dale



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From: "Francois Audet" <audet@nortel.com>
To: "Dale Worley" <dworley@nortel.com>
Cc: dispatch@ietf.org, R.Jesske@telekom.de
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Again, the spec is very clear that it IS allowed.

I believe the wishy-washy text about "status code explicitly
allowing it" was meant to exclude responses that were not appropriate,
and restricing it to effectively error responses.

At the time this was written, I believe we were not clear on which
codes were supposed to be appropriate or not.

Seems to me:
- Any Error response code should be allowed.
- I don't think 2XX makes sense.
- 3XX is controversial (as per the email quoted by Roland): seems to me =
it
  would be quite useful
- Provisional is interesting... Sounds like 199 error response to me...

> -----Original Message-----
> From: Worley, Dale (BL60:9D30)=20
> Sent: Thursday, July 16, 2009 10:09
> To: Audet, Francois (SC100:3055)
> Cc: R.Jesske@telekom.de; dispatch@ietf.org
> Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois (SC100:3055) wrote:
> > Hi Roland,
> >=20
> > You use case is very common. =20
> >=20
> > I believe you are incorrect in saying that "reasons are=20
> currently not=20
> > allowed in responses. Neither conditionally nor allowed".
> >=20
> > RFC 3326 says in 1.0:
> >   "[...] it can appear in any request
> >    within a dialog, in any CANCEL request and in any response whose
> >    status code explicitly allows the presence of this header field."
> >=20
> > To be honest, I believe Q.850 codes are much more common in=20
> Responses=20
> > than in requests.
>=20
> Googling
>=20
>      "sip/2.0" reason "q.850"
>=20
> turns up numerous examples of SIP responses using the Reason=20
> header in the forbidden manner.
>=20
> I'd say that your draft formally allows what people are already doing.
>=20
> Dale
>=20
>=20
>=20

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From: "Francois Audet" <audet@nortel.com>
To: <R.Jesske@telekom.de>, "Dale Worley" <dworley@nortel.com>
Cc: dispatch@ietf.org
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi Roland,

You use case is very common. =20

I believe you are incorrect in saying that "reasons are currently=20
not allowed in responses. Neither conditionally nor allowed".

RFC 3326 says in 1.0:
  "[...] it can appear in any request
   within a dialog, in any CANCEL request and in any response whose
   status code explicitly allows the presence of this header field."

To be honest, I believe Q.850 codes are much more common in=20
Responses than in requests.

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of R.Jesske@telekom.de
> Sent: Sunday, July 12, 2009 23:17
> To: Worley, Dale (BL60:9D30)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Hi Dale,
> The main probblem is that reasons are currently not allowed=20
> in responses. Neither conditionally nor allowed
>=20
> Please see:=20
> http://www.ietf.org/mail-archive/web/sipping/current/msg09682.html
>=20
> The draft tries to close this gap and allows conditionally=20
> Q.850 resasons within Resopnses. So it would be a service=20
> provider decision to include Reasons in Responses when=20
> mapping from ISUP to SIP.
>=20
> The following call flow shows an example:
>=20
>          A                Gateway             Gateway              B=20
>          |        IAM        |                  |                  |=20
>          |------------------>|     INVITE       |                  |=20
>          |                   |----------------->|      IAM         |=20
>          |                   |     100 Trying   |----------------->|=20
>          |                   |<-----------------|    100 Trying    |=20
>          |                   |   403 Decline    |                  |
>          |                   |  Reason Q850 #87 |  REL Cause #87   |
>          |   REL Cause #87   |                  |<-----------------|=20
>          |                   |<-----------------|                  |=20
>          |<----------------- |                  |                  |=20
>          |                   |                  |                  |=20
>          |                   |                  |                  |=20
>          |                   |                  |                  |=20
>=20
> Due to RFC 3398 a 403 will be mapped back to a ISUP Cause 21.=20
> With the Reason in responses the Cause Value 87 will be=20
> secured and transferred back to ISUP.
>=20
> Mheanwile I got some indications from some persons who see=20
> this issue within the BLISS group due to the fact that this=20
> draft solves interoperability issues within SIP and PSTN.
>=20
> Views?
>=20
> BR,
>=20
> Roland
> =20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Dale Worley [mailto:dworley@nortel.com]
> Gesendet: Freitag, 10. Juli 2009 21:52
> An: Jesske, Roland
> Cc: dispatch@ietf.org
> Betreff: Re: AW: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> On Tue, 2009-07-07 at 07:13 +0200, R.Jesske@telekom.de wrote:
> > Hi Dale,
> > Within the SIPPING group I had already discussions on that issue.
> > The assumption was that the use of the Reason Header within=20
> Responses is conditionally allowed.=20
> >=20
> > The RFC 3326 says:
> >    Initially, the Reason header field defined here appears=20
> to be most
> >    useful for BYE and CANCEL requests, but it can appear in=20
> any request
> >    within a dialog, in any CANCEL request and in any response whose
> >    status code explicitly allows the presence of this header field.
> >=20
> >=20
> > So we need a draft where it is stated that a reason header=20
> can be included within the SIP Response. That was also the=20
> recommendation of that discussion. That is why I wrote a draft.
> > Nevertheless some standards already include this behaviour=20
> like the ITU-T Q.1912.5 specification.
> > Also 3GPP needs the reason header within Responses.
>=20
> It is still unclear to me what is being changed.  Is the=20
> change the replacement of "conditionally allowed" to=20
> "allowed"?  In any case, in my mind the purpose of the Reason=20
> header is to carry Q.850 cause-codes in failure responses.
>=20
> The point is that the Abstract is not at all clear about the=20
> change from the status quo (at least to someone who isn't=20
> already thoroughly familiar with how Reason is currently specified).
>=20
> And the secondary point is that if the Abstract was clearer=20
> about what it is being changed, more people might read and=20
> discuss the draft.
>=20
> Dale
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From drage@alcatel-lucent.com  Thu Jul 16 11:29:47 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Francois Audet <audet@nortel.com>, Dale Worley <dworley@nortel.com>
Date: Thu, 16 Jul 2009 20:29:44 +0200
Thread-Topic: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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If by "the spec" you are referring to the original Reason header RFC 3326, =
then yes, it did not preclude it, but specifically indicated that any usage=
 had to be documented further.

   Initially, the Reason header field defined here appears to be most
   useful for BYE and CANCEL requests, but it can appear in any request
   within a dialog, in any CANCEL request and in any response whose
   status code explicitly allows the presence of this header field.

There are no RFCs currently specifying the use with any status code.

Note also that the rest of the text in the document is specifically about i=
ts use in requests.

If you go back, you will remember that Reason header was being addressed or=
iginally at two problems. One was the issue of transferring clearing inform=
ation from elsewhere in a BYE request, the other was the HERFP problem. The=
 latter was the use case that required things in a response, but there was =
no agreement on the way forward on this part of the problem. Therefore the =
Reason header draft was cut down only to address the first problem.

I would note that the argument has been made in the past, and indeed is hin=
ted at in RFC 3326, is that in responses, you should use the response code.=
 This is what SIP UAs are designed to understand and deal with on reception=
. Therefore it has been stated in the past that if one wishes to deliver in=
formation to end UAs that is not covered by the existing response codes, th=
en a new status code should be defined. Therefore no general use of Reason =
headers in responses was given in RFC 3326 for that reason. I did not origi=
nate that argument, and maybe the people who gave that argument should spea=
k up, but it is certainly the argument that shaped RFC 3326 the way it is.

There is a case where you need to get information from a SIP UA acting as a=
 gateway to another SIP UA acting as a gateway and in that case the Reason =
header in a response would be well justified in that case, and that is cert=
ainly one of the cases covered by the draft.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Francois Audet
> Sent: Thursday, July 16, 2009 6:16 PM
> To: Dale Worley
> Cc: dispatch@ietf.org; R.Jesske@telekom.de
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Again, the spec is very clear that it IS allowed.
>=20
> I believe the wishy-washy text about "status code explicitly=20
> allowing it" was meant to exclude responses that were not=20
> appropriate, and restricing it to effectively error responses.
>=20
> At the time this was written, I believe we were not clear on=20
> which codes were supposed to be appropriate or not.
>=20
> Seems to me:
> - Any Error response code should be allowed.
> - I don't think 2XX makes sense.
> - 3XX is controversial (as per the email quoted by Roland):=20
> seems to me it
>   would be quite useful
> - Provisional is interesting... Sounds like 199 error=20
> response to me...
>=20
> > -----Original Message-----
> > From: Worley, Dale (BL60:9D30)
> > Sent: Thursday, July 16, 2009 10:09
> > To: Audet, Francois (SC100:3055)
> > Cc: R.Jesske@telekom.de; dispatch@ietf.org
> > Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >=20
> > On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois=20
> (SC100:3055) wrote:
> > > Hi Roland,
> > >=20
> > > You use case is very common. =20
> > >=20
> > > I believe you are incorrect in saying that "reasons are
> > currently not
> > > allowed in responses. Neither conditionally nor allowed".
> > >=20
> > > RFC 3326 says in 1.0:
> > >   "[...] it can appear in any request
> > >    within a dialog, in any CANCEL request and in any=20
> response whose
> > >    status code explicitly allows the presence of this=20
> header field."
> > >=20
> > > To be honest, I believe Q.850 codes are much more common in
> > Responses
> > > than in requests.
> >=20
> > Googling
> >=20
> >      "sip/2.0" reason "q.850"
> >=20
> > turns up numerous examples of SIP responses using the=20
> Reason header in=20
> > the forbidden manner.
> >=20
> > I'd say that your draft formally allows what people are=20
> already doing.
> >=20
> > Dale
> >=20
> >=20
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From R.Jesske@telekom.de  Thu Jul 16 22:08:19 2009
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From: <R.Jesske@telekom.de>
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Hi Dale,
You are right. The Q.850 causes will be used within ITU-T Q.1912.5 for =
interworking SIP-ISUP and SIP-I.
Within ETSI ES 383 001 for interworking SIP-ISUP and SIP-I.
Within 3GPP TS 29.163 also interworking SIP with ISUP but there it is =
referred to the regarding draft we are discussing.

Also it is widely deployed also within our network.

But nevertheless within 3GPP meetings I have got requests to progress =
the ietf draft because people thinks that this is the way to ratify the =
use of reason within responses.

Best regards

Roland

-----Urspr=FCngliche Nachricht-----
Von: Dale Worley [mailto:dworley@nortel.com]=20
Gesendet: Donnerstag, 16. Juli 2009 19:09
An: Francois Audet
Cc: Jesske, Roland; dispatch@ietf.org
Betreff: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois (SC100:3055) wrote:
> Hi Roland,
>=20
> You use case is very common. =20
>=20
> I believe you are incorrect in saying that "reasons are currently=20
> not allowed in responses. Neither conditionally nor allowed".
>=20
> RFC 3326 says in 1.0:
>   "[...] it can appear in any request
>    within a dialog, in any CANCEL request and in any response whose
>    status code explicitly allows the presence of this header field."
>=20
> To be honest, I believe Q.850 codes are much more common in=20
> Responses than in requests.

Googling

     "sip/2.0" reason "q.850"

turns up numerous examples of SIP responses using the Reason header in
the forbidden manner.

I'd say that your draft formally allows what people are already doing.

Dale



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Thank you Dale for your input.
I have changed the Abstract.

BR,

Roland=20

-----Urspr=FCngliche Nachricht-----
Von: Dale Worley [mailto:dworley@nortel.com]=20
Gesendet: Donnerstag, 16. Juli 2009 19:01
An: Jesske, Roland
Cc: dispatch@ietf.org; esasaki@shui-usjapan.com
Betreff: Re: AW: AW: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

On Mon, 2009-07-13 at 08:16 +0200, R.Jesske@telekom.de wrote:

> The main probblem is that reasons are currently not allowed in
> responses. Neither conditionally nor allowed
>=20
> Please see: =
http://www.ietf.org/mail-archive/web/sipping/current/msg09682.html

That clarifies things.

Let me rephrase my complaint:  Although the text of the RFC states "The
Reason header field MAY appear in [...] any response whose status code
explicitly allows the presence of this header field.", that text is
stunningly unclear that the consequence is that the Reason header is not
allowed in *any* defined response, and a casual reader might not realize
that Reason is thus effectively forbidden in all responses.

Expanding the Abstract along these lines would make the significance and
importance of the draft much clearer:

        Although the use of the Reason header in responses is considered
        in RFC 3326, doing so is not specified for any existing response
        code.  This document specifies the use of the Reason header
        field in SIP responses to carry Q.850 reason codes for the
        failure of an INVITE.
       =20
Dale



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From: <R.Jesske@telekom.de>
To: <audet@nortel.com>, <dworley@nortel.com>
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi Francois,
The problem is that this is not the opinion I've got in the past. I =
would have been happy if somebody had the same view on that issue as you =
have now.

And I have not seen any RFC that explicitly allows the use of Q.850 =
causes within responses.

Of course within ISUP the Release can be sent in both directions also =
before and after Answer Message (equivalent to 200 OK). So a Release =
Message within ISUP has the same functionality as a Response, CANCEL and =
BYE. And that was not really taken into consideration when the Reason =
Header was developed.

BR,

Roland

-----Urspr=FCngliche Nachricht-----
Von: Francois Audet [mailto:audet@nortel.com]=20
Gesendet: Donnerstag, 16. Juli 2009 18:48
An: Jesske, Roland; Dale Worley
Cc: dispatch@ietf.org
Betreff: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

Hi Roland,

You use case is very common. =20

I believe you are incorrect in saying that "reasons are currently=20
not allowed in responses. Neither conditionally nor allowed".

RFC 3326 says in 1.0:
  "[...] it can appear in any request
   within a dialog, in any CANCEL request and in any response whose
   status code explicitly allows the presence of this header field."

To be honest, I believe Q.850 codes are much more common in=20
Responses than in requests.

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of R.Jesske@telekom.de
> Sent: Sunday, July 12, 2009 23:17
> To: Worley, Dale (BL60:9D30)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Hi Dale,
> The main probblem is that reasons are currently not allowed=20
> in responses. Neither conditionally nor allowed
>=20
> Please see:=20
> http://www.ietf.org/mail-archive/web/sipping/current/msg09682.html
>=20
> The draft tries to close this gap and allows conditionally=20
> Q.850 resasons within Resopnses. So it would be a service=20
> provider decision to include Reasons in Responses when=20
> mapping from ISUP to SIP.
>=20
> The following call flow shows an example:
>=20
>          A                Gateway             Gateway              B=20
>          |        IAM        |                  |                  |=20
>          |------------------>|     INVITE       |                  |=20
>          |                   |----------------->|      IAM         |=20
>          |                   |     100 Trying   |----------------->|=20
>          |                   |<-----------------|    100 Trying    |=20
>          |                   |   403 Decline    |                  |
>          |                   |  Reason Q850 #87 |  REL Cause #87   |
>          |   REL Cause #87   |                  |<-----------------|=20
>          |                   |<-----------------|                  |=20
>          |<----------------- |                  |                  |=20
>          |                   |                  |                  |=20
>          |                   |                  |                  |=20
>          |                   |                  |                  |=20
>=20
> Due to RFC 3398 a 403 will be mapped back to a ISUP Cause 21.=20
> With the Reason in responses the Cause Value 87 will be=20
> secured and transferred back to ISUP.
>=20
> Mheanwile I got some indications from some persons who see=20
> this issue within the BLISS group due to the fact that this=20
> draft solves interoperability issues within SIP and PSTN.
>=20
> Views?
>=20
> BR,
>=20
> Roland
> =20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Dale Worley [mailto:dworley@nortel.com]
> Gesendet: Freitag, 10. Juli 2009 21:52
> An: Jesske, Roland
> Cc: dispatch@ietf.org
> Betreff: Re: AW: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> On Tue, 2009-07-07 at 07:13 +0200, R.Jesske@telekom.de wrote:
> > Hi Dale,
> > Within the SIPPING group I had already discussions on that issue.
> > The assumption was that the use of the Reason Header within=20
> Responses is conditionally allowed.=20
> >=20
> > The RFC 3326 says:
> >    Initially, the Reason header field defined here appears=20
> to be most
> >    useful for BYE and CANCEL requests, but it can appear in=20
> any request
> >    within a dialog, in any CANCEL request and in any response whose
> >    status code explicitly allows the presence of this header field.
> >=20
> >=20
> > So we need a draft where it is stated that a reason header=20
> can be included within the SIP Response. That was also the=20
> recommendation of that discussion. That is why I wrote a draft.
> > Nevertheless some standards already include this behaviour=20
> like the ITU-T Q.1912.5 specification.
> > Also 3GPP needs the reason header within Responses.
>=20
> It is still unclear to me what is being changed.  Is the=20
> change the replacement of "conditionally allowed" to=20
> "allowed"?  In any case, in my mind the purpose of the Reason=20
> header is to carry Q.850 cause-codes in failure responses.
>=20
> The point is that the Abstract is not at all clear about the=20
> change from the status quo (at least to someone who isn't=20
> already thoroughly familiar with how Reason is currently specified).
>=20
> And the secondary point is that if the Abstract was clearer=20
> about what it is being changed, more people might read and=20
> discuss the draft.
>=20
> Dale
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From R.Jesske@telekom.de  Thu Jul 16 23:31:08 2009
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From: <R.Jesske@telekom.de>
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi,
With regard to Keith's answer I see that the draft is needed.
But where should this be done?

BLISS, DISPATCH or SIPCORE?

We have now the discussion within DISPATCH due to the fact that the =
draft was issued for the sipping group.

The issue shall solve interoperability problems with the PSTN where =
Q.850 causes shall be transferred through SIP.
An other used case is where applications within SIP networks could =
interpret the Reason header to put an announcement towards a user. This =
is important within Hybrid networks where SIP and ISUP runs in parallel.

Seen from this point of view the work should be done within BLISS to =
solve this interoperability problem.

RFC3326 was done within SIP and the draft could be seen as an extension =
to this RFC so that could be a reason to do this within SIPCORE.

Or the discussion takes place within DISPATCH and the draft was brought =
to SIPPING.

So how to proceed?

BR

Roland

-----Urspr=FCngliche Nachricht-----
Von: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
Gesendet: Donnerstag, 16. Juli 2009 20:30
An: Francois Audet; Dale Worley
Cc: dispatch@ietf.org; Jesske, Roland
Betreff: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

If by "the spec" you are referring to the original Reason header RFC =
3326, then yes, it did not preclude it, but specifically indicated that =
any usage had to be documented further.

   Initially, the Reason header field defined here appears to be most
   useful for BYE and CANCEL requests, but it can appear in any request
   within a dialog, in any CANCEL request and in any response whose
   status code explicitly allows the presence of this header field.

There are no RFCs currently specifying the use with any status code.

Note also that the rest of the text in the document is specifically =
about its use in requests.

If you go back, you will remember that Reason header was being addressed =
originally at two problems. One was the issue of transferring clearing =
information from elsewhere in a BYE request, the other was the HERFP =
problem. The latter was the use case that required things in a response, =
but there was no agreement on the way forward on this part of the =
problem. Therefore the Reason header draft was cut down only to address =
the first problem.

I would note that the argument has been made in the past, and indeed is =
hinted at in RFC 3326, is that in responses, you should use the response =
code. This is what SIP UAs are designed to understand and deal with on =
reception. Therefore it has been stated in the past that if one wishes =
to deliver information to end UAs that is not covered by the existing =
response codes, then a new status code should be defined. Therefore no =
general use of Reason headers in responses was given in RFC 3326 for =
that reason. I did not originate that argument, and maybe the people who =
gave that argument should speak up, but it is certainly the argument =
that shaped RFC 3326 the way it is.

There is a case where you need to get information from a SIP UA acting =
as a gateway to another SIP UA acting as a gateway and in that case the =
Reason header in a response would be well justified in that case, and =
that is certainly one of the cases covered by the draft.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Francois Audet
> Sent: Thursday, July 16, 2009 6:16 PM
> To: Dale Worley
> Cc: dispatch@ietf.org; R.Jesske@telekom.de
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Again, the spec is very clear that it IS allowed.
>=20
> I believe the wishy-washy text about "status code explicitly=20
> allowing it" was meant to exclude responses that were not=20
> appropriate, and restricing it to effectively error responses.
>=20
> At the time this was written, I believe we were not clear on=20
> which codes were supposed to be appropriate or not.
>=20
> Seems to me:
> - Any Error response code should be allowed.
> - I don't think 2XX makes sense.
> - 3XX is controversial (as per the email quoted by Roland):=20
> seems to me it
>   would be quite useful
> - Provisional is interesting... Sounds like 199 error=20
> response to me...
>=20
> > -----Original Message-----
> > From: Worley, Dale (BL60:9D30)
> > Sent: Thursday, July 16, 2009 10:09
> > To: Audet, Francois (SC100:3055)
> > Cc: R.Jesske@telekom.de; dispatch@ietf.org
> > Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >=20
> > On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois=20
> (SC100:3055) wrote:
> > > Hi Roland,
> > >=20
> > > You use case is very common. =20
> > >=20
> > > I believe you are incorrect in saying that "reasons are
> > currently not
> > > allowed in responses. Neither conditionally nor allowed".
> > >=20
> > > RFC 3326 says in 1.0:
> > >   "[...] it can appear in any request
> > >    within a dialog, in any CANCEL request and in any=20
> response whose
> > >    status code explicitly allows the presence of this=20
> header field."
> > >=20
> > > To be honest, I believe Q.850 codes are much more common in
> > Responses
> > > than in requests.
> >=20
> > Googling
> >=20
> >      "sip/2.0" reason "q.850"
> >=20
> > turns up numerous examples of SIP responses using the=20
> Reason header in=20
> > the forbidden manner.
> >=20
> > I'd say that your draft formally allows what people are=20
> already doing.
> >=20
> > Dale
> >=20
> >=20
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Vijay Gurbani <vkg@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Fri, 17 Jul 2009 00:56:03 -0700
Thread-Topic: [dispatch] Session Recording in SIP
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Subject: Re: [dispatch] Session Recording in SIP
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I read this draft and following are my comments :

Section 3 Definitions

Perhaps Recording Server(RS) should simply be called Recorder. I see the no=
te on the last sentence however as long as definitions refer to Client and =
Server, it will always imply to the reader to think that RS is the server. =
The statement that SRP may be SIP is an understatement since the definition=
 for RC already referred to as a SIP UA/B2BUA.

Dynamic Recording, if the recording session is started and ended at random =
points on demand, its length will not be the same as actual media session.

Persistent Recording, it is not clear what is meant by "system start up". I=
s "system" referring to RC or RS ?=20

Depending on the answer, I am curious how are these persistent recording se=
ssions are correlated to many active media sessions and what would be mecha=
nism to keep them alive ? What is the number of persistent recording sessio=
ns per RC/RS ?=20

Perhaps implications of this should be covered somewhere else, maybe in the=
 requirements section ?

Section 5 Requirements

REQ-006 Not a good idea to talk about the solution here and how some RC imp=
lementations handle the problem.

Perhaps REQ-010 and 011 can be combined into one by just saying "over a sin=
gle or separate".

REQ-018 can be extended to say that the mechanism MUST support the ability =
to find out the associated recorded session(s) given that one(for instance =
RS) is only aware of recording session(s). I'd rather prefer this to be a s=
eparate requirement.=20

I have another requirement in mind : The mechanism MUST support the ability=
 to setup recording sessions that will record different conversation direct=
ions of the Recorded session(s) from RC(s) and they too should be correlate=
d accordingly. For instance, while an agent is talking to a customer, there=
 should be a way to setup two recording sessions, one carrying the media fr=
om agent towards customer and vice versa for the other. Furthermore, these =
recording sessions should be marked as such in SRP.

REQ-021 talks about a mode where RC just forks media to RS. However, there =
is value in studying transcoding, media incompatibility use cases and see i=
f there are requirements they bring given that RS/RCs are/will most likely =
have different media capabilities.

If recorded session media is stopped/temporarily interrupted due to problem=
s or features invoked on it(recording session is put on hold or started a c=
onferencing or transfer operation), how does that affect the already setup =
recording session(s) ?

Nit :

Section2, sentence that starts with "that draft focuses on the mechanism to=
 provide the SRTP...." is odd. It is clear that srtp-key draft is not for s=
ession recording, so no need to state the obvious.

Regards,
Serhad Doken

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Jain, Rajnish
> Sent: Monday, July 06, 2009 3:48 PM
> To: Vijay Gurbani; dispatch@ietf.org
> Subject: Re: [dispatch] Session Recording in SIP
>=20
> All: The following draft on SIP session recording requirements was
> submitted today:
>=20
> http://www.ietf.org/internet-drafts/draft-jain-dispatch-session-
> recording-protocol-req-00.txt
>=20
> Title:           Requirements for Session Recording Protocol (SRP)
>=20
> Abstract:
> Session recording is a critical requirement in many business
> communications environments such as call centers and financial
> trading floors.  In some of these environments, all calls must be
> recorded for regulatory and compliance reasons.  In others, calls may
> be recorded for quality control or business analytics.  Recording is
> typically done by sending a copy of the media to the recording
> devices.  This document specifies requirements for a protocol that
> will manage delivery of media from an end-point that originates media
> or that has access to it to a recording device.  This protocol is
> being referred to as Session Recording Protocol and will most likely
> be based on SIP.
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
> > Of Vijay Gurbani
> > Sent: Monday, June 22, 2009 12:09 AM
> > To: dispatch@ietf.org
> > Subject: [dispatch] Summary on Session Recording in SIP
> >
> > All: Here is the summary of the Session Recording thread from June 9
> > to June 19, 2009.  If I misrepresented anything, please let me know.
> >
> > There was a fair amount of discussion on where to conduct this
> > work.  In the end, it was decided to continue discussions on
> > dispatch, further decisions what to do -- new working group or
> > put this work in an existing working group -- will be decided
> > later.  Regardless of where the work is done, opinions were
> > expressed that the work is important that it should commence
> > in the IETF.
> >
> > Need to be crisp about the purpose of this work: is it that users
> > record sessions to hear later, or sessions need to be recorded
> > because of some business needs?  Or both?  Is SRTP covered or
> > only RTP?
> >
> > Solutions should cover:
> >   Always on recording
> >   Recording on demand
> >   Required recording
> >   Pause and resume recording
> >   Playback facilities and how they interact with recording
> >   Recording formats and protocols for controlling the stored
> recording.
> >
> > A lot of discussion ensued on the specific architecture or
> > architectures that would be possible.  No decision was reached on
> > which specific architecture is good or bad, although various opinions
> > were expressed in support for each.
> >
> > There are essentially four architectures to realize recording
> > services: two that discuss where the media stream is forked
> > out to the recorder -- at the UA or in a middlebox of some
> > sort are outlined here:
> >    http://www.ietf.org/mail-
> archive/web/dispatch/current/msg00226.html
> >
> > The third architecture has both UAs sending media to a
> > recorder.  See:
> >      http://www.ietf.org/mail-
> archive/web/dispatch/current/msg00236.html
> >
> > A fourth architecture that is a superset of the third one is at:
> >      http://www.ietf.org/mail-
> archive/web/dispatch/current/msg00256.html
> >
> > - vijay
> > --
> > Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> > 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> > Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> > WWW:   http://ect.bell-labs.com/who/vkg
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>=20
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From mary.barnes@nortel.com  Fri Jul 17 08:06:26 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: <R.Jesske@telekom.de>, <drage@alcatel-lucent.com>, "Francois Audet" <audet@nortel.com>, "Dale Worley" <dworley@nortel.com>
Cc: ericksasaki@sbcglobal.net, dispatch@ietf.org
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi Roland,

I think DISPATCH remains the appropriate place for these discussions.  =
If we did agree specific changes to RFC 3326, that work would not fall =
under the charter of SIPCORE (as it's currently scoped as I understand =
it). =20

It's not clear to me that this fits the scope of BLISS in that the focus =
of this document is on the clarifications to RFC 3326 to support certain =
services/environments.  It's not under the purview of BLISS to make SIP =
protocol changes.  [I will note that the BLISS charter needs to be =
updated to reflect how it does interface with SIPCORE and DISPATCH, =
rather than SIP and SIPPING as it's currently stated.]  I do see that =
more details of services that would need this reason header might fall =
into the BLISS WG charter.

My suggestion is that you update the document to incorporate the =
conclusions of this discussion ( the doc is currently expired) - you'll =
have to wait for the document submissions to be opened again on July =
27th.  In my view, the crux of a work item is whether you need a true =
update to RFC 3326 or whether you are documenting clarifications  - =
similar to the sip-offeranswer document in SIPPING. =20

Regards,
Mary.=20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of R.Jesske@telekom.de
Sent: Friday, July 17, 2009 1:33 AM
To: drage@alcatel-lucent.com; Audet, Francois (SC100:3055); Worley, Dale =
(BL60:9D30)
Cc: ericksasaki@sbcglobal.net; dispatch@ietf.org
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

Hi,
With regard to Keith's answer I see that the draft is needed.
But where should this be done?

BLISS, DISPATCH or SIPCORE?

We have now the discussion within DISPATCH due to the fact that the =
draft was issued for the sipping group.

The issue shall solve interoperability problems with the PSTN where =
Q.850 causes shall be transferred through SIP.
An other used case is where applications within SIP networks could =
interpret the Reason header to put an announcement towards a user. This =
is important within Hybrid networks where SIP and ISUP runs in parallel.

Seen from this point of view the work should be done within BLISS to =
solve this interoperability problem.

RFC3326 was done within SIP and the draft could be seen as an extension =
to this RFC so that could be a reason to do this within SIPCORE.

Or the discussion takes place within DISPATCH and the draft was brought =
to SIPPING.

So how to proceed?

BR

Roland

-----Urspr=FCngliche Nachricht-----
Von: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
Gesendet: Donnerstag, 16. Juli 2009 20:30
An: Francois Audet; Dale Worley
Cc: dispatch@ietf.org; Jesske, Roland
Betreff: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

If by "the spec" you are referring to the original Reason header RFC =
3326, then yes, it did not preclude it, but specifically indicated that =
any usage had to be documented further.

   Initially, the Reason header field defined here appears to be most
   useful for BYE and CANCEL requests, but it can appear in any request
   within a dialog, in any CANCEL request and in any response whose
   status code explicitly allows the presence of this header field.

There are no RFCs currently specifying the use with any status code.

Note also that the rest of the text in the document is specifically =
about its use in requests.

If you go back, you will remember that Reason header was being addressed =
originally at two problems. One was the issue of transferring clearing =
information from elsewhere in a BYE request, the other was the HERFP =
problem. The latter was the use case that required things in a response, =
but there was no agreement on the way forward on this part of the =
problem. Therefore the Reason header draft was cut down only to address =
the first problem.

I would note that the argument has been made in the past, and indeed is =
hinted at in RFC 3326, is that in responses, you should use the response =
code. This is what SIP UAs are designed to understand and deal with on =
reception. Therefore it has been stated in the past that if one wishes =
to deliver information to end UAs that is not covered by the existing =
response codes, then a new status code should be defined. Therefore no =
general use of Reason headers in responses was given in RFC 3326 for =
that reason. I did not originate that argument, and maybe the people who =
gave that argument should speak up, but it is certainly the argument =
that shaped RFC 3326 the way it is.

There is a case where you need to get information from a SIP UA acting =
as a gateway to another SIP UA acting as a gateway and in that case the =
Reason header in a response would be well justified in that case, and =
that is certainly one of the cases covered by the draft.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Francois Audet
> Sent: Thursday, July 16, 2009 6:16 PM
> To: Dale Worley
> Cc: dispatch@ietf.org; R.Jesske@telekom.de
> Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Again, the spec is very clear that it IS allowed.
>=20
> I believe the wishy-washy text about "status code explicitly allowing=20
> it" was meant to exclude responses that were not appropriate, and=20
> restricing it to effectively error responses.
>=20
> At the time this was written, I believe we were not clear on which=20
> codes were supposed to be appropriate or not.
>=20
> Seems to me:
> - Any Error response code should be allowed.
> - I don't think 2XX makes sense.
> - 3XX is controversial (as per the email quoted by Roland):=20
> seems to me it
>   would be quite useful
> - Provisional is interesting... Sounds like 199 error response to=20
> me...
>=20
> > -----Original Message-----
> > From: Worley, Dale (BL60:9D30)
> > Sent: Thursday, July 16, 2009 10:09
> > To: Audet, Francois (SC100:3055)
> > Cc: R.Jesske@telekom.de; dispatch@ietf.org
> > Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >=20
> > On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois
> (SC100:3055) wrote:
> > > Hi Roland,
> > >=20
> > > You use case is very common. =20
> > >=20
> > > I believe you are incorrect in saying that "reasons are
> > currently not
> > > allowed in responses. Neither conditionally nor allowed".
> > >=20
> > > RFC 3326 says in 1.0:
> > >   "[...] it can appear in any request
> > >    within a dialog, in any CANCEL request and in any
> response whose
> > >    status code explicitly allows the presence of this
> header field."
> > >=20
> > > To be honest, I believe Q.850 codes are much more common in
> > Responses
> > > than in requests.
> >=20
> > Googling
> >=20
> >      "sip/2.0" reason "q.850"
> >=20
> > turns up numerous examples of SIP responses using the
> Reason header in
> > the forbidden manner.
> >=20
> > I'd say that your draft formally allows what people are
> already doing.
> >=20
> > Dale
> >=20
> >=20
> >=20
> _______________________________________________
> 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

From dworley@nortel.com  Fri Jul 17 11:50:59 2009
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From: "Dale Worley" <dworley@nortel.com>
To: R.Jesske@telekom.de
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Thinking about this discussion, I would add a sentence to the abstract
telling the reader that the Reason header in responses is widely used,
even though it is not officially specified.  Something along these
lines:

        Although the use of the Reason header field in responses is
        considered in RFC 3326, doing so is not specified for any
        existing response code.  Nonetheless, the Reason header field
        has been widely used in responses to carry Q.850 reason codes
        for failure responses to INVITEs that have been gatewayed to
        PSTN systems.  This document specifies and formally permits the
        use of the Reason header field in SIP responses to carry Q.850
        reason codes for this and other purposes.

I think that makes the current situation clear to the uninformed reader
(e.g., me), and it only requires 3 sentences.

Dale



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Subject: [dispatch] Questions regarding draft-kaplan-sipping-interop-bcp-02
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Section 2:

There are two broad areas of SIP interoperation.  One is "phone to
PBX", which is assumed to be between a UA and a proxy for its domain.
The other is "SIP trunking", that is, between two proxies, often in
different administrative domains.  The recommendations in this draft
seem to cover both areas, and it's likely that the two areas cannot be
clearly separated.  Nonetheless, a naive reader (me) would likely
infer from the author's employer that the draft only covers "SIP
trunking" problems.  An expansion of the Applicability section would
dispel this misinterpretation earlier in the text, and would prevent
some readers from leaving off finishing reading the document.

Section 4.3:

Just after the regexp is the statement "Note this is after removing
visual-separators, and any user parameters."  Given he need for
accuracy here, this statement is problematic because
"visual-separator" is not a defined term in regard to SIP URIs
(although it is for tel: URIs), and "user parameter" is not defined at
all in the grammar (although people generally know what the term
means).

The initial part of a SIP URI is officially called "user".

Isn't the maximum length of an E.164 number 15 digits?  If so, why are
16-digit numbers also restricted?

A more clearly defined specification is:

   It is RECOMMENDED that SIP domains not allocate SIP URI AoR 
   user parts which match the following regular expression syntax, 
   unless the URI is intended to have the same significance as
   the corresponding E.164 number:

   ^+?([-.()]*[0-9]){11,16}[-.()]*(;.*)?$ 

   This regular expression is to be compared to the user part after
   removing any escaping.  The characters "*" and "#", although
   allowed in tel: URIs, are not allowed in E.164 numbers, and so user
   parts containing them are not included in this prohibition.

In regard to the statement

   It is RECOMMENDED, that the SIP Working Group address the 
   fundamental issue of E.164 or telephone number scope in SIP URI's.  

I don't see what issue needs to be addressed, as the working group's
opinion is well-defined and well-known, and nobody has advanced any
argument to change it.

   The TEL URI also has specific syntactic and usage issues beyond 
   simply lack of broad industry support, which may be a reason for not 
   supporting it. 

This is more interesting, since the question to be answered is "Why
don't implementers follow the clear intention of the specification,
and what might be done about that?"

In regard to the URI schemes that are used in various locations, we
need a general statement that elements should tolerate sip:, sips:,
and tel: URIs in all situations, excepting when the element must
understand the URI to process the message.  E.g., there is no excuse
for rejecting a syntactically correct From URI, because an element
never needs to understand the semantics of the From URI.

Similarly, there are situations where a default action is specified to
be taken when the element does not understand a URI which might
otherwise cause it to take specific action.  In such cases, an element
should take the default action for such URIs rather than rejecting the
message outright.

Section 4.4:

   Although the Tel URI has not seen much usage itself using the "tel" 
   scheme, it is quite common to see SIP URI's containing Tel-URI 
   parameters - i.e., Tel URI's in the form of SIP URI's.  There are 
   many URI parameters defined for Tel-URI that are popular, and thus 
   frequently appear in SIP URI's, for example in the request-URI. 
    
What is not made clear here is that these parameters show up as
"user-part parameters", which is a syntax feature not defined in RFC
3261.  Naively, the above paragraph suggests that these parameters
show up as URI-parameters.  This could be fixed by amending the first
sentence to say "... quite common to see SIP URI's user-parts
containing Tel-URI parameters ...", and adding an example:  "E.g.,
sip:+12125551212;phone-context=example.com@example.net"

   The most common interoperability problem in this area is the 
   placement of the Tel URI parameters, for example the "tgrp" and 
   "trunk-context" parameters from [RFC4904], and the "rn", "npdi", and 
   "cic" parameters from [RFC4694].  RFC 3261 section 19.1.6 is very 
   clear that the entire telephone-subscriber portion, including 
   parameters, is placed in the userinfo portion of a SIP URI.  Thus 
   the Tel-URI parameters become user parameters for SIP, not SIP URI 
   parameters. 
    
"userinfo portion" should be "user portion", as the SIP URI syntax
allows phone numbers to be followed by a password part, and presumably
such an inserted telephone-subscriber portion can be followed by a
password part (yuck).

Section 5.1:

In regard to failure responses to REGISTER requests:

   [What can we say here that UA's will actually do?] 

"In any circumstance the UA MUST take care that it never generates
retries of requests without using a suitable back-off strategy.  If
the UA receives a response which suggests that a particular
modification of the request is likely to be successful, it MAY send
the modified request immediately.  The UA MAY NOT retry any failing
request with a long-term average frequency of more than one request
sent (new client transaction initiated) per 5 minutes regardless of
what responses it receives."

Making sure that UAs do not flood systems is a critical need.

Section 5.2:

Let's add "SIP elements MUST process provided Contact URIs opaquely."

Section 5.5:

Shouldn't this section refer to sections 2.4 to 2.6 (various forms of
transfer) in RFC 5359 (the service examples RFC)?

Section 6.1:

I suggest softening the first recommendation to require SDP only in
initial INVITEs, but permitting offer-less re-INVITEs.  The latter is
very useful in implementing MOH, and since the dialog path is already
established, it is unambiguous what media/codecs the UAS and the
middleboxes support/permit *in that dialog*, even if call routing is
affected by media/codecs.

Section 6.4:

   [Note: one trick some vendors use is to pretend the unacceptable 
   response was not received and send a CANCEL - should we recommend 
   that?] 

That's an interesting idea, since it seems to permit the INVITE to
fail-over to another destination which might generate an acceptable
answer.  But thinking more about it, I don't think that works -- the
CANCEL gets a 200 response but doesn't do anything (since the UAS has
already progressed the dialog to confirmed), and the UAS will
retransmit the 200.  What is needed is to eat the 200 response with
the unacceptable answer so that all upstream proxies have a chance to
fork the request to additional destinations.  But that means that the
proxy that eats the 200 must be well downstream, near the UAS that
generated the 200, but must know what the UAC considers unacceptable.
It's hard to make that work unless the proxy that eats the 200 is also
the one that generates further forks, and also services the UAC.

The only workable solution I can think of is for the INVITE to carry
"Request-Disposition: no-cancel" so that the intermediate proxies
continue to generate forks even though one UAS responds 200.  Then the
UAC will receive all the 200s as they are generated.  If it doesn't
like the answer in one 200, it can BYE that dialog.  If it receives a
200 with an answer it likes, it can then CANCEL the INVITE to suppress
further forking.

That's messy, but it might make sense for a UAC to use it as a
fallback strategy.  Assuming that the intermediate proxies support
Request-Disposition, it is even standardized.

Section 7:

The Call-Info header is used in the call-completion architecture
(draft-bliss-ietf-call-completion).  I suspect that many other "rare"
headers are used sporadically in other situations.

A better recommendation is to remind implementers that an
implementation MUST ignore header fields, URI-parameters, and
header-parameters that it does not understand.  Conversely,
implementations MUST reject requests with unknown option-tags in
Require header fields.  If implementations follow these rules,
extraneous information should not cause interoperation problems, since
SIP features are designed (remarkably carefully!) to be
upward-compatible under these rules.

Dale



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Subject: Re: [dispatch] [Sipping] I-D Action:draft-kaplan-sipping-interop-bcp-02.txt
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On Mon, 2009-07-13 at 02:00 -0400, Hadriel Kaplan wrote:
> p.s.  My guess is that it's not because they can't generate a 604
> physically, but rather that the systems can't consistently and
> programmatically know when it would be correct to do so.

That's exactly the problem -- a 604 doesn't mean "the request-URI cannot
be reached", it means "the request-URI of the *original* request cannot
be reached".  But the UAS cannot know the original request-URI, so it
can never validly generate any 6xx-class response.

See draft-worley-6xx-considered-harmful for a whine about this...

Dale



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Please follow and participate in this discussion on the sip-clf list  
if you are interested.

RjS

Begin forwarded message:

> From: Robert Sparks <rjsparks@nostrum.com>
> Date: July 17, 2009 4:13:58 PM CDT
> To: sip-clf@ietf.org
> Subject: draft CLF charter
>
> All -
>
> We are working on forming a CLF working group based on DISPATCH's  
> decision.
>
> Below is a proposed charter for this working group. Please review  
> and comment
> on this list. Depending on the feedback we receive, we will target  
> forming this
> group shortly after the Stockholm meeting.
>
> We'll also be discussing this in Thursday's opsarea meeting.
>
> Thanks,
>
> RjS
>
>
>> The SIP Common Log File (CLF) working group is chartered to define
>> a standard logging format for systems processing SIP messages.
>>
>> Well-known web servers such as Apache and web proxies like Squid
>> support event logging using a common log format.  The logs produced
>> using these de-facto standard formats are invaluable to system
>> administrators for trouble-shooting a server and tool writers to
>> craft tools that mine the log files to produce reports and trends
>> and to search for a certain SIP message or messages, a transaction
>> or a related set of transactions.  Furthermore, these log records
>> can also be used to train anomaly detection systems and feed events
>> into a security event management system.
>>
>> The Session Initiation Protocol does not have a common log
>> format. Diverse element provide distinct log formats making
>> it complex to produce tools to analyze them.
>>
>> The CLF working group will produce a format suitable for logging
>> from any SIP element. The format will anticipate the need to
>> search, merge, and summarize the log records from diverse elements.
>> The format will anticipate the need to correlate messages from
>> multiple elements related to a given request (that may fork)
>> or a given dialog. The format will take SIP's extensibility into
>> consideration, providing a way to represent SIP message components
>> that are defined in the future.  The format will anticipate being
>> used both for off-line analysis and on-line real-time processing
>> applications. The working group will consider the need for
>> efficient processing in its design of this format.
>>
>> The working group is not pre-constrained to producing either a
>> bit-field oriented or text-oriented format, and may choose to
>> provide both. If the group chooses to specify both, it must be
>> possible to mechanically translate between the formats without
>> loss of information.
>>
>> Specifying the mechanics of exchanging, transporting, and storing
>> SIP Common Log Format records is explicitly out of scope. Specifying
>> a real-time transfer mechanism for heuristic analysis is explicitly
>> out of scope.
>>
>> The group will generate:
>>
>> - A problem statement enunciating the motivation,
>> and use cases for a SIP Common Log Format. This analysis
>> will identify the required minimal information that must
>> appear in any record.
>>
>> - A specification of the SIP Common Log Format record.
>>
>> The group will consider providing one or more reference
>> implementations for decoding a CLF record.
>>
>> Goals and Milestones
>> ===========================
>>
>> Nov 09 - Problem statement, motivation, and use cases to IESG  
>> (Informational)
>> Feb 10 - SIP Common Log Format specification to IESG (PS)
>>
>


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">Please follow and participate =
in this discussion on the sip-clf list if you are =
interested.<div><br></div><div>RjS<br><div><div><div><br><div>Begin =
forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>From: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica">Robert Sparks &lt;<a =
href=3D"mailto:rjsparks@nostrum.com">rjsparks@nostrum.com</a>&gt;</font></=
div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Date: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">July 17, 2009 4:13:58 PM =
CDT</font></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><font face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>To: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:sip-clf@ietf.org">sip-clf@ietf.org</a></font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Subject: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><b>draft CLF charter</b></font></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><br></div> </div><div>All -<br><br>We are working on =
forming a CLF working group based on DISPATCH's decision.<br><br>Below =
is a proposed charter for this working group. Please review and =
comment<br>on this list. Depending on the feedback we receive, we will =
target forming this<br>group shortly after the Stockholm =
meeting.<br><br>We'll also be discussing this in Thursday's opsarea =
meeting.<br><br>Thanks,<br><br>RjS<br><br><br><blockquote =
type=3D"cite">The SIP Common Log File (CLF) working group is chartered =
to define<br></blockquote><blockquote type=3D"cite">a standard logging =
format for systems processing SIP messages.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Well-known web =
servers such as Apache and web proxies like =
Squid<br></blockquote><blockquote type=3D"cite">support event logging =
using a common log format. &nbsp;The logs =
produced<br></blockquote><blockquote type=3D"cite">using these de-facto =
standard formats are invaluable to system<br></blockquote><blockquote =
type=3D"cite">administrators for trouble-shooting a server and tool =
writers to<br></blockquote><blockquote type=3D"cite">craft tools that =
mine the log files to produce reports and =
trends<br></blockquote><blockquote type=3D"cite">and to search for a =
certain SIP message or messages, a =
transaction<br></blockquote><blockquote type=3D"cite">or a related set =
of transactions. &nbsp;Furthermore, these log =
records<br></blockquote><blockquote type=3D"cite">can also be used to =
train anomaly detection systems and feed =
events<br></blockquote><blockquote type=3D"cite">into a security event =
management system.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The Session =
Initiation Protocol does not have a common =
log<br></blockquote><blockquote type=3D"cite">format. Diverse element =
provide distinct log formats making<br></blockquote><blockquote =
type=3D"cite">it complex to produce tools to analyze =
them.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The CLF working =
group will produce a format suitable for =
logging<br></blockquote><blockquote type=3D"cite">from any SIP element. =
The format will anticipate the need to<br></blockquote><blockquote =
type=3D"cite">search, merge, and summarize the log records from diverse =
elements.<br></blockquote><blockquote type=3D"cite">The format will =
anticipate the need to correlate messages =
from<br></blockquote><blockquote type=3D"cite">multiple elements related =
to a given request (that may fork)<br></blockquote><blockquote =
type=3D"cite">or a given dialog. The format will take SIP's =
extensibility into<br></blockquote><blockquote =
type=3D"cite">consideration, providing a way to represent SIP message =
components<br></blockquote><blockquote type=3D"cite">that are defined in =
the future. &nbsp;The format will anticipate =
being<br></blockquote><blockquote type=3D"cite">used both for off-line =
analysis and on-line real-time processing<br></blockquote><blockquote =
type=3D"cite">applications. The working group will consider the need =
for<br></blockquote><blockquote type=3D"cite">efficient processing in =
its design of this format.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The working =
group is not pre-constrained to producing either =
a<br></blockquote><blockquote type=3D"cite">bit-field oriented or =
text-oriented format, and may choose to<br></blockquote><blockquote =
type=3D"cite">provide both. If the group chooses to specify both, it =
must be<br></blockquote><blockquote type=3D"cite">possible to =
mechanically translate between the formats =
without<br></blockquote><blockquote type=3D"cite">loss of =
information.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Specifying the =
mechanics of exchanging, transporting, and =
storing<br></blockquote><blockquote type=3D"cite">SIP Common Log Format =
records is explicitly out of scope. =
Specifying<br></blockquote><blockquote type=3D"cite">a real-time =
transfer mechanism for heuristic analysis is =
explicitly<br></blockquote><blockquote type=3D"cite">out of =
scope.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The group will =
generate:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">- A problem =
statement enunciating the motivation,<br></blockquote><blockquote =
type=3D"cite"> and use cases for a SIP Common Log Format. This =
analysis<br></blockquote><blockquote type=3D"cite"> will identify the =
required minimal information that must<br></blockquote><blockquote =
type=3D"cite"> appear in any record.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">- A =
specification of the SIP Common Log Format =
record.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The group will =
consider providing one or more reference<br></blockquote><blockquote =
type=3D"cite">implementations for decoding a CLF =
record.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Goals and =
Milestones<br></blockquote><blockquote =
type=3D"cite">=3D=3D=3D=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></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Nov 09 - =
Problem statement, motivation, and use cases to IESG =
(Informational)<br></blockquote><blockquote type=3D"cite">Feb 10 - SIP =
Common Log Format specification to IESG (PS)<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><br></div></blockquote></div><br></div></di=
v></div></body></html>=

--Apple-Mail-140--1024798114--

From HKaplan@acmepacket.com  Fri Jul 17 14:37:00 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dale Worley <dworley@nortel.com>, DISPATCH <dispatch@ietf.org>
Date: Fri, 17 Jul 2009 17:37:30 -0400
Thread-Topic: [dispatch] Questions regarding draft-kaplan-sipping-interop-bcp-02
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Dale,
Thanks for the detailed review!
I will take your comments for the next rev of the doc. (I thought I had fix=
ed one of them already but realized not in the rev I submitted - d'oh)

-hadriel


> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Dale Worley
> Sent: Friday, July 17, 2009 3:34 PM
> To: DISPATCH
> Subject: [dispatch] Questions regarding draft-kaplan-sipping-interop-bcp-
> 02
>=20
> Section 2:
>=20
> There are two broad areas of SIP interoperation.  One is "phone to
> PBX", which is assumed to be between a UA and a proxy for its domain.
> The other is "SIP trunking", that is, between two proxies, often in
> different administrative domains.  The recommendations in this draft
> seem to cover both areas, and it's likely that the two areas cannot be
> clearly separated.  Nonetheless, a naive reader (me) would likely
> infer from the author's employer that the draft only covers "SIP
> trunking" problems.  An expansion of the Applicability section would
> dispel this misinterpretation earlier in the text, and would prevent
> some readers from leaving off finishing reading the document.
>=20
> Section 4.3:
>=20
> Just after the regexp is the statement "Note this is after removing
> visual-separators, and any user parameters."  Given he need for
> accuracy here, this statement is problematic because
> "visual-separator" is not a defined term in regard to SIP URIs
> (although it is for tel: URIs), and "user parameter" is not defined at
> all in the grammar (although people generally know what the term
> means).
>=20
> The initial part of a SIP URI is officially called "user".
>=20
> Isn't the maximum length of an E.164 number 15 digits?  If so, why are
> 16-digit numbers also restricted?
>=20
> A more clearly defined specification is:
>=20
>    It is RECOMMENDED that SIP domains not allocate SIP URI AoR
>    user parts which match the following regular expression syntax,
>    unless the URI is intended to have the same significance as
>    the corresponding E.164 number:
>=20
>    ^+?([-.()]*[0-9]){11,16}[-.()]*(;.*)?$
>=20
>    This regular expression is to be compared to the user part after
>    removing any escaping.  The characters "*" and "#", although
>    allowed in tel: URIs, are not allowed in E.164 numbers, and so user
>    parts containing them are not included in this prohibition.
>=20
> In regard to the statement
>=20
>    It is RECOMMENDED, that the SIP Working Group address the
>    fundamental issue of E.164 or telephone number scope in SIP URI's.
>=20
> I don't see what issue needs to be addressed, as the working group's
> opinion is well-defined and well-known, and nobody has advanced any
> argument to change it.
>=20
>    The TEL URI also has specific syntactic and usage issues beyond
>    simply lack of broad industry support, which may be a reason for not
>    supporting it.
>=20
> This is more interesting, since the question to be answered is "Why
> don't implementers follow the clear intention of the specification,
> and what might be done about that?"
>=20
> In regard to the URI schemes that are used in various locations, we
> need a general statement that elements should tolerate sip:, sips:,
> and tel: URIs in all situations, excepting when the element must
> understand the URI to process the message.  E.g., there is no excuse
> for rejecting a syntactically correct From URI, because an element
> never needs to understand the semantics of the From URI.
>=20
> Similarly, there are situations where a default action is specified to
> be taken when the element does not understand a URI which might
> otherwise cause it to take specific action.  In such cases, an element
> should take the default action for such URIs rather than rejecting the
> message outright.
>=20
> Section 4.4:
>=20
>    Although the Tel URI has not seen much usage itself using the "tel"
>    scheme, it is quite common to see SIP URI's containing Tel-URI
>    parameters - i.e., Tel URI's in the form of SIP URI's.  There are
>    many URI parameters defined for Tel-URI that are popular, and thus
>    frequently appear in SIP URI's, for example in the request-URI.
>=20
> What is not made clear here is that these parameters show up as
> "user-part parameters", which is a syntax feature not defined in RFC
> 3261.  Naively, the above paragraph suggests that these parameters
> show up as URI-parameters.  This could be fixed by amending the first
> sentence to say "... quite common to see SIP URI's user-parts
> containing Tel-URI parameters ...", and adding an example:  "E.g.,
> sip:+12125551212;phone-context=3Dexample.com@example.net"
>=20
>    The most common interoperability problem in this area is the
>    placement of the Tel URI parameters, for example the "tgrp" and
>    "trunk-context" parameters from [RFC4904], and the "rn", "npdi", and
>    "cic" parameters from [RFC4694].  RFC 3261 section 19.1.6 is very
>    clear that the entire telephone-subscriber portion, including
>    parameters, is placed in the userinfo portion of a SIP URI.  Thus
>    the Tel-URI parameters become user parameters for SIP, not SIP URI
>    parameters.
>=20
> "userinfo portion" should be "user portion", as the SIP URI syntax
> allows phone numbers to be followed by a password part, and presumably
> such an inserted telephone-subscriber portion can be followed by a
> password part (yuck).
>=20
> Section 5.1:
>=20
> In regard to failure responses to REGISTER requests:
>=20
>    [What can we say here that UA's will actually do?]
>=20
> "In any circumstance the UA MUST take care that it never generates
> retries of requests without using a suitable back-off strategy.  If
> the UA receives a response which suggests that a particular
> modification of the request is likely to be successful, it MAY send
> the modified request immediately.  The UA MAY NOT retry any failing
> request with a long-term average frequency of more than one request
> sent (new client transaction initiated) per 5 minutes regardless of
> what responses it receives."
>=20
> Making sure that UAs do not flood systems is a critical need.
>=20
> Section 5.2:
>=20
> Let's add "SIP elements MUST process provided Contact URIs opaquely."
>=20
> Section 5.5:
>=20
> Shouldn't this section refer to sections 2.4 to 2.6 (various forms of
> transfer) in RFC 5359 (the service examples RFC)?
>=20
> Section 6.1:
>=20
> I suggest softening the first recommendation to require SDP only in
> initial INVITEs, but permitting offer-less re-INVITEs.  The latter is
> very useful in implementing MOH, and since the dialog path is already
> established, it is unambiguous what media/codecs the UAS and the
> middleboxes support/permit *in that dialog*, even if call routing is
> affected by media/codecs.
>=20
> Section 6.4:
>=20
>    [Note: one trick some vendors use is to pretend the unacceptable
>    response was not received and send a CANCEL - should we recommend
>    that?]
>=20
> That's an interesting idea, since it seems to permit the INVITE to
> fail-over to another destination which might generate an acceptable
> answer.  But thinking more about it, I don't think that works -- the
> CANCEL gets a 200 response but doesn't do anything (since the UAS has
> already progressed the dialog to confirmed), and the UAS will
> retransmit the 200.  What is needed is to eat the 200 response with
> the unacceptable answer so that all upstream proxies have a chance to
> fork the request to additional destinations.  But that means that the
> proxy that eats the 200 must be well downstream, near the UAS that
> generated the 200, but must know what the UAC considers unacceptable.
> It's hard to make that work unless the proxy that eats the 200 is also
> the one that generates further forks, and also services the UAC.
>=20
> The only workable solution I can think of is for the INVITE to carry
> "Request-Disposition: no-cancel" so that the intermediate proxies
> continue to generate forks even though one UAS responds 200.  Then the
> UAC will receive all the 200s as they are generated.  If it doesn't
> like the answer in one 200, it can BYE that dialog.  If it receives a
> 200 with an answer it likes, it can then CANCEL the INVITE to suppress
> further forking.
>=20
> That's messy, but it might make sense for a UAC to use it as a
> fallback strategy.  Assuming that the intermediate proxies support
> Request-Disposition, it is even standardized.
>=20
> Section 7:
>=20
> The Call-Info header is used in the call-completion architecture
> (draft-bliss-ietf-call-completion).  I suspect that many other "rare"
> headers are used sporadically in other situations.
>=20
> A better recommendation is to remind implementers that an
> implementation MUST ignore header fields, URI-parameters, and
> header-parameters that it does not understand.  Conversely,
> implementations MUST reject requests with unknown option-tags in
> Require header fields.  If implementations follow these rules,
> extraneous information should not cause interoperation problems, since
> SIP features are designed (remarkably carefully!) to be
> upward-compatible under these rules.
>=20
> Dale
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From gonzalo.camarillo@ericsson.com  Mon Jul 20 02:59:16 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [dispatch] Input requested: abandoning	draft-ietf-sipping-profile-datasets-03?
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Hi,

since nobody has expressed interested in this draft, we will be 
abandoning it.

Cheers,

Gonzalo
DISPATCH co-chair

Gonzalo Camarillo wrote:
> Folks,
> 
> the SIPPING WG agreed to work on the following draft a long time ago:
> 
> http://tools.ietf.org/html/draft-ietf-sipping-profile-datasets-03
> 
> As you can see in the following slide, the media-policy draft depends on 
> the draft above:
> 
> http://www.ietf.org/proceedings/07jul/slides/sipping-4/sld11.htm
> 
> While there are people interested in the media-policy draft, we have not 
> found anyone interested in the profile-datasets draft.
> 
> One of the things we agreed to do when restructured RAI and chartered 
> DISPATCH was to drop items people were no longer interested in. 
> Therefore, at this point, we are considering abandoning the 
> profile-dataset draft. If we do this, we would need to import a few 
> things from that draft to the media-policy draft (e.g., the merging 
> rules) but the idea would be to only progress the media-policy draft.
> 
> Does anyone have any objection to this way forward?
> 
> Note that in order to progress a draft we need people interested in it 
> and people willing to put cycles on it. Therefore, if you are interested 
> in keeping the draft alive, you will be automatically volunteering to 
> put cycles on it in order to progress it ;o)
> 
> Thanks,
> 
> Gonzalo
> DISPATCH co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


From gonzalo.camarillo@ericsson.com  Mon Jul 20 03:32:38 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi,

as Francois suggests, a document specifying the use of Reason header 
fields in responses needs to specify those things (see Francois' list 
below). Additionally, you should think of whether or not Reason header 
fields in responses can carry SIP status codes and what happens if they 
are different to the status code of the response.

In short, the document cannot simply say that now it is OK to use Reason 
in responses. It needs to address the different situations a typical 
implementation may face.

Cheers,

Gonzalo


Francois Audet wrote:
> Again, the spec is very clear that it IS allowed.
> 
> I believe the wishy-washy text about "status code explicitly
> allowing it" was meant to exclude responses that were not appropriate,
> and restricing it to effectively error responses.
> 
> At the time this was written, I believe we were not clear on which
> codes were supposed to be appropriate or not.
> 
> Seems to me:
> - Any Error response code should be allowed.
> - I don't think 2XX makes sense.
> - 3XX is controversial (as per the email quoted by Roland): seems to me it
>   would be quite useful
> - Provisional is interesting... Sounds like 199 error response to me...
> 
>> -----Original Message-----
>> From: Worley, Dale (BL60:9D30) 
>> Sent: Thursday, July 16, 2009 10:09
>> To: Audet, Francois (SC100:3055)
>> Cc: R.Jesske@telekom.de; dispatch@ietf.org
>> Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois (SC100:3055) wrote:
>>> Hi Roland,
>>>
>>> You use case is very common.  
>>>
>>> I believe you are incorrect in saying that "reasons are 
>> currently not 
>>> allowed in responses. Neither conditionally nor allowed".
>>>
>>> RFC 3326 says in 1.0:
>>>   "[...] it can appear in any request
>>>    within a dialog, in any CANCEL request and in any response whose
>>>    status code explicitly allows the presence of this header field."
>>>
>>> To be honest, I believe Q.850 codes are much more common in 
>> Responses 
>>> than in requests.
>> Googling
>>
>>      "sip/2.0" reason "q.850"
>>
>> turns up numerous examples of SIP responses using the Reason 
>> header in the forbidden manner.
>>
>> I'd say that your draft formally allows what people are already doing.
>>
>> Dale
>>
>>
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


From gonzalo.camarillo@ericsson.com  Mon Jul 20 03:43:48 2009
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Hi Paul,

> that this should be invisible 
> to the other "side".

while this is of course a good thing when it can be achieved, it cannot 
be an absolute requirement for solutions to a given problem. Otherwise, 
SIP would not have mechanisms such as the Require header field. That is, 
while in many deployment scenarios you want to be able to deploy things 
incrementally taking advantage of extensions even if only one UA support 
them, in others you can actually update all your UAs to support a new 
extension.

Salvatore has described deployment scenarios where UAs can be upgraded 
to support a new extension. I think we should consider them if they can 
take advantage of extra functionality when compared with other scenarios 
where this cannot be done... that is why we included Require header 
fields in SIP's original design.

Cheers,

Gonzalo

From salvatore.loreto@ericsson.com  Mon Jul 20 04:13:19 2009
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Hi Paul,

Paul Kyzivat wrote:
>
> If we can agree that one communication session between two parties has 
> one sip session *between them*, then we can work on what the 
> architecture should be like at each end.
that would be a work to improve the centralized scenario and perhaps 
making it possible without using 3pcc
or solving the current 3pcc limitations and them to make it nicer and 
easier to use.

However I still think it would be useful to work on a decentralized 
scenario, moreover there could be scenario where knowing
how one end aggregates media resource for the call will help the other 
end to better decide how to manage (i.e. answer) the call .


thanks
Sal

From salvatore.loreto@ericsson.com  Mon Jul 20 04:25:53 2009
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Hi John,

see my answers in line

thanks
Sal

Elwell, John wrote:
> Sal,
>
> Thanks for writing this. The centralised approach is what I and some
> others have been suggesting over the past months. It is good to see a
> comparison between the centralised approach (in its various flavours)
> and the distributed approach that you have been advocating for some
> time.
>
> It seems the biggest issue for deciding whether or not to work on the
> distributed approach is whether there is a need for the "master" device
> (the one operating SIP) to drop out of the call, while allowing the call
> and certain media to continue at other devices. 
>   
exactly! this is one of the main 3pcc issue, but not the only one;

> I still have a concern that with the distributed approach the remote end
> of the call needs to exhibit special behaviour in order to handle the
> multiple dialogs arising from the distributed approach (as Paul also
> notes). So in practice, to make it work with any degree of reliability,
> calls will need to go through a B2BUA that provides the necessary
> aggregation. So it effectively ends up as a centralised approach anyway.
>   
well, if the other end does not support a distributed approach then you 
can not use it.
> The distributed approach (section 4) is not really described in enough
> detail. There still has to be some communication between Alice's UAs,
> e.g., to convey information to a UA to be added to the call to allow it
> to make a call to Bob and allow Bob to associate the new call with the
> existing call. Also if Alice wants to clear all calls (i.e., all
> components of the same call) with one button push, there needs to be
> some communication between the UAs.
>   
right, how the different user devices of one end coordinate and convey 
information
each other in a distributed scenario is something that needs to need to 
be analysed and describe in more
details.

> John
>
> John
>
>   
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Salvatore Loreto
>> Sent: 06 July 2009 17:33
>> To: dispatch@ietf.org
>> Subject: [dispatch] Disaggregated Media in SIP
>>
>> Hi there,
>>
>> I have just submitted a draft that talks of Disaggregated 
>> Media in the 
>> Session Initiation Protocol (SIP).
>> http://www.ietf.org/internet-drafts/draft-loreto-dispatch-disa
>> ggregated-media-00.txt
>>
>>
>> Abstract:
>> Disaggregated media refers to the ability for a user to create a 
>> multimedia session combining different media streams, coming from
>> different devices under his or her control, so that they are 
>> treated by 
>> the far end of the session as a single media session.
>> This document lists several use cases that involve 
>> disaggregated media 
>> in SIP.
>> Additionally, this document analyzes what types of 
>> disaggregated media 
>> can be implemented using existing protocol
>> mechanisms, and the pros and cons of using each of those mechanisms.
>> Finally, this document describes scenarios that are not covered by 
>> current mechanisms
>> and proposes new IETF work to cover them.
>>
>>
>> cheers
>> Sal
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>     


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To: "Doken, Serhad" <sdoken@qualcomm.com>, Vijay Gurbani <vkg@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>
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Subject: Re: [dispatch] Session Recording in SIP
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Hi,

Thanks for your detailed comments. Please see inline:

> -----Original Message-----
> From: Doken, Serhad [mailto:sdoken@qualcomm.com]
> Sent: Friday, July 17, 2009 3:56 AM
> To: Jain, Rajnish; Vijay Gurbani; dispatch@ietf.org
> Subject: RE: [dispatch] Session Recording in SIP
>
> I read this draft and following are my comments :
>
> Section 3 Definitions
>
> Perhaps Recording Server(RS) should simply be called Recorder. I see the
> note on the last sentence however as long as definitions refer to Client =
and
> Server, it will always imply to the reader to think that RS is the server=
.

Naming these things is always a bit tricky. If we called the RS simply the =
Recorder given your reasoning, then the RC should also be named differently=
 (because the RC can act as the "server side" of the SRP signaling). The RS=
 naming is a bit akin to the term "Media Server". If there is enough confus=
ion on RS then, we can consider renaming it.

> The statement that SRP may be SIP is an understatement since the definiti=
on
> for RC already referred to as a SIP UA/B2BUA.

This will be clarified in subsequent revisions when we'll say that SRP is i=
ndeed SIP.

> Dynamic Recording, if the recording session is started and ended at rando=
m
> points on demand, its length will not be the same as actual media session=
.

Ok. We can clarify this.

> Persistent Recording, it is not clear what is meant by "system start up".=
 Is
> "system" referring to RC or RS ?

It meant that when both devices become functional. We can clarify this.

> Depending on the answer, I am curious how are these persistent recording
> sessions are correlated to many active media sessions and what would be
> mechanism to keep them alive ?

The data over the call metadata interface will offer the correlation. A sim=
ple way to keep SRP persistent sessions alive is through the SIP session-ti=
mers.

What is the number of persistent recording
> sessions per RC/RS ?

That's an implementation-dependent choice.

> Perhaps implications of this should be covered somewhere else, maybe in t=
he
> requirements section ?
>
> Section 5 Requirements
>
> REQ-006 Not a good idea to talk about the solution here and how some RC
> implementations handle the problem.

That note is only informative text. I guess we can take it out.

> Perhaps REQ-010 and 011 can be combined into one by just saying "over a
> single or separate".

Fine. Will change this.

> REQ-018 can be extended to say that the mechanism MUST support the abilit=
y
> to find out the associated recorded session(s) given that one(for instanc=
e
> RS) is only aware of recording session(s). I'd rather prefer this to be a
> separate requirement.

The RS is aware of the recorded session metadata. REQ-018 is a bit broad in=
 the sense that it talks about correlation in either direction (i.e. from r=
ecorded sessions to recording sessions and vice-versa). I think your commen=
t is about splitting the two directions, right?

> I have another requirement in mind : The mechanism MUST support the abili=
ty
> to setup recording sessions that will record different conversation
> directions of the Recorded session(s) from RC(s) and they too should be
> correlated accordingly. For instance, while an agent is talking to a
> customer, there should be a way to setup two recording sessions, one
> carrying the media from agent towards customer and vice versa for the oth=
er.
> Furthermore, these recording sessions should be marked as such in SRP.

Yes, this is needed. This requirement was actually in there and somehow got=
 removed in one of the edits. We'll add this.

> REQ-021 talks about a mode where RC just forks media to RS. However, ther=
e
> is value in studying transcoding, media incompatibility use cases and see=
 if
> there are requirements they bring given that RS/RCs are/will most likely
> have different media capabilities.

REQ-021 is primarily about a middle-box type of RC that forks the media. Th=
e RC vendors in this case want their boxes to be optimal and scalable by do=
ing as little media processing as possible. REQ-021 isn't precluding the us=
e cases that you mentioned, but I think it'll make sense to address such us=
e cases in this document.

> If recorded session media is stopped/temporarily interrupted due to probl=
ems
> or features invoked on it(recording session is put on hold or started a
> conferencing or transfer operation), how does that affect the already set=
up
> recording session(s) ?

I think that'll largely depend on the RC/RS implementations and the nature =
of the SRP sessions. For instance, if the recording sessions are persistent=
 and the RC puts recorded session media on hold, then it RC MAY reINVITE th=
e RS and put the recording session media on hold or it MAY just temporarily=
 stop sending such media (silence suppression) to the RS.

> Nit :
>
> Section2, sentence that starts with "that draft focuses on the mechanism =
to
> provide the SRTP...." is odd. It is clear that srtp-key draft is not for
> session recording, so no need to state the obvious.

Ok. We can change some wording here.

Thanks,
Raj

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Gonzalo Camarillo wrote:
> Hi Paul,
> 
>> that this should be invisible to the other "side".
> 
> while this is of course a good thing when it can be achieved, it cannot 
> be an absolute requirement for solutions to a given problem. Otherwise, 
> SIP would not have mechanisms such as the Require header field.

I'm not trying to assert that this is a criterion that is appropriate 
for all problems - just that it is appropriate for *this* problem.

Some problems cannot be solved without the collaboration of both ends, 
and in that case, if the feature is not one that all UAs must have then 
some sort of negotiation to determine whether the two ends support it is 
needed.

But in this case it can be shown that the capability *can* be managed on 
one end without the assistance of the other end. And doing it that way 
vastly increases the interoperability over an approach that requires the 
support of both ends.

> That is, 
> while in many deployment scenarios you want to be able to deploy things 
> incrementally taking advantage of extensions even if only one UA support 
> them, in others you can actually update all your UAs to support a new 
> extension.
> 
> Salvatore has described deployment scenarios where UAs can be upgraded 
> to support a new extension. I think we should consider them if they can 
> take advantage of extra functionality when compared with other scenarios 
> where this cannot be done... that is why we included Require header 
> fields in SIP's original design.

I still haven't seen anything showing why this approach ends up being 
"better" than the "centralized" approach. (Note that I don't agree with 
the term "centralized", because it isn't really more centralized than 
the "distributed" approach is. This would become clearer if we looked at 
all the communications necessary to establish a call with disaggregated 
media, and the various topologies of devices that might play in such 
scenarios.)

Perhaps we should back off from the signaling mechanisms for the moment 
and consider the use cases of interest in more detail, and the various 
sorts of communications between them that are necessary to establish a 
call that involves them.

	Thanks,
	Paul

> Cheers,
> 
> Gonzalo
> 

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Salvatore Loreto wrote:
> Hi Paul,
> 
> Paul Kyzivat wrote:
>>
>> If we can agree that one communication session between two parties has 
>> one sip session *between them*, then we can work on what the 
>> architecture should be like at each end.
> that would be a work to improve the centralized scenario and perhaps 
> making it possible without using 3pcc
> or solving the current 3pcc limitations and them to make it nicer and 
> easier to use.
> 
> However I still think it would be useful to work on a decentralized 
> scenario, moreover there could be scenario where knowing
> how one end aggregates media resource for the call will help the other 
> end to better decide how to manage (i.e. answer) the call .

I will try to keep an open mind, but I suspect we have a fundamental 
disagreement here. IMO there is a fundamental architectural property at 
stake here. IMO the point is as much as possible to add capabilities 
without increasing the complexity of the fundamentals - use what we have 
as building blocks when we can. Only when that is found to be fatally 
flawed should we be adding more stuff for everybody to implement.

IMO there will be many devices that want to support multimedia than 
those that have need to support disaggregated media. Its a bad thing if 
those can't participate in multimedia calls with devices that have need 
support from multiple devices to achieve multimedia capability.

I guess the one way I could buy into your "distributed" approach is to 
simply model it as a conference, where the focus is at one "end". But I 
think we discussed that once.

	Thanks,
	Paul

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Date: Mon, 20 Jul 2009 16:38:35 +0200
Message-ID: <9ae56b1e0907200738n6685284l17d1282fd0653b9@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Fwd: I-D Action:draft-kaplan-dispatch-sip-implicit-registrations-00.txt
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Hi Hadriel,
Thanks for putting this together, this is useful.

- General
I think it would be good to add some informative references to the ETSI
TISPAN specification where the NGN/IMS way of performing business trunking
(SIP trunking) has been specified using the concept of implicitly registered
wildcarded public user identities (AORs)

The specification numbers are:
*ETSI TS 181 019 *V2.0.0 (2007-11)
Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN);*Business Communication Requirements*
*ETSI TS 182 023 V2.1.1* (2009-01)
Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN);*Core and enterprise NGN interaction
scenarios;Architecture and functional description*

*ETSI TS 182 025 V2.1.1 *(2008-09)
Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN);*Business trunking;Architecture and functional
description*

I think it would also be good to add informative references to the SIP
Connect specifications.

Further it might be good to say something about the different viewpoints
taken by the different bodies:
- ETSI TISPAN/3GPP Business Trunking are providing the viewpoint of the
service provider connecting IP-PBX's to its IMS/NGN infrastructure
- SIP Connect is more looking at the interconnect specifciation between the
an SSP and a IP-PBX.

Ideally both should match at some point to get an interoperable corporate
communication space.

- Section 6.2
The 3GPP/ETSI TISPAN mechanism does not add the notion of wildcarded
P-Associated-URI, but it adds the notion of wildcarded public user identity,
which generates so to speak the set of public user identioties to be
registered. This is not only visible on the P-Associated-URI header but also
makes configuration of such sets of public user identities easier.

3GPP and ETSI have extended the mechanism described here with the following
additions, for connected PBX's a subscription parameter can be set that
allows delivery of the target URI is kept in the request URI and the contact
address is added to the Route header after the stored path is inserted,
which means that in this case the P-Called-Party-ID is not strictly needed
for delivery of the target address.

Further this mechanism is extended to allow PBX/corporate proxies to use
P-Asserted-Identity instead of P-Preferred-Identity, which is a more natural
thing to do for corporate proxies.

Regarding the side note: This seems misleading as an SSP may very well serve
the domain name of  the enterprise, even if only for SIP traffic. This can
all be arranged with normal DNS infrastructure.

Best Regards,
/Hans Erik van Elburg

On Thu, Jul 9, 2009 at 7:37 PM, Hadriel Kaplan <HKaplan@acmepacket.com>wrote:

> Howdy,
> As part of a discussion in the SIP Forum regarding a model for Registration
> we are using for IP-PBX's, I was asked by someone to submit an I-D
> describing it, in case it would be of any value.  Since I hadn't seen an I-D
> before on how 3GPP/ETSI did their model, and since their model is different
> from SIP-Forum's, I submitted one which describes the models for both SIP
> Forum's SIP-Connect v1.1 and 3GPP/ETSI.
>
> NOTE: this draft was not sanctioned nor reviewed by the SIP-Forum nor 3GPP,
> and I am still talking with colleagues in 3GPP to make sure I got that part
> right.  This is purely an individual submission in all respects.
>
> Forwarded message:
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>        Title           : Session Initiation Protocol (SIP) Implicit
> Registrations
>        Author(s)       : H. Kaplan
>        Filename        :
> draft-kaplan-dispatch-sip-implicit-registrations-00.txt
>        Pages           : 12
>        Date            : 2009-07-05
>
> This document identifies several approaches to provide reachability
> information for a domain or multiple AoR's using a single SIP REGISTER
> method transaction, in ways not originally envisioned or documented by RFC
> 3261.
>
> A URL for this Internet-Draft is:
>
> http://www.ietf.org/internet-drafts/draft-kaplan-dispatch-sip-implicit-registrations-00.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

--0015174be630ca308c046f2417be
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Hadriel,<div><br></div><div>Thanks for putting this together, this is=A0=
useful.</div><div><br></div><div><span class=3D"Apple-style-span" style=3D"=
text-decoration: underline; ">- General</span></div><div>I think it would b=
e good to add some informative references to the ETSI TISPAN specification =
where the NGN/IMS way of performing business trunking (SIP trunking) has be=
en specified using the concept of implicitly registered wildcarded public u=
ser identities (AORs)</div>
<div><br></div><div>The specification numbers are:</div><b>ETSI TS 181 019 =
</b>V2.0.0 (2007-11)<br>Telecommunications and Internet converged Services =
and Protocols for Advanced Networking (TISPAN);<b>Business Communication Re=
quirements</b><div>
<br><b>ETSI TS 182 023 V2.1.1</b> (2009-01)<br>Telecommunications and Inter=
net converged Services and Protocols for Advanced Networking (TISPAN);<b>Co=
re and enterprise NGN interaction scenarios;Architecture and functional des=
cription</b><br>
<br></div><div><b>ETSI TS 182 025 V2.1.1 </b>(2008-09)<br>Telecommunication=
s and Internet converged Services and Protocols for Advanced Networking (TI=
SPAN);<b>Business trunking;Architecture and functional description</b><br>
<div><span class=3D"Apple-style-span" style=3D"font-weight: 900; "><br></sp=
an></div>I think it would also be good to add informative references to the=
 SIP Connect specifications.<br><div><span class=3D"Apple-style-span" style=
=3D"font-weight: 900; "><br>
</span></div>Further it might be good to say something about the different =
viewpoints taken by the different bodies:</div><div>- ETSI TISPAN/3GPP Busi=
ness Trunking are providing the viewpoint of the service provider connectin=
g IP-PBX&#39;s to its IMS/NGN infrastructure</div>
<div>- SIP Connect is more looking at the interconnect specifciation betwee=
n the an SSP and a IP-PBX.</div><div><br></div><div>Ideally both should mat=
ch at some point to get an interoperable corporate communication space.<br>
<div><span class=3D"Apple-style-span" style=3D"font-weight: 900; "><br></sp=
an></div><span class=3D"Apple-style-span" style=3D"text-decoration: underli=
ne;">- Section 6.2</span><br><div>The 3GPP/ETSI TISPAN mechanism does not a=
dd the notion of wildcarded P-Associated-URI, but it adds the notion of wil=
dcarded public user identity, which generates so to speak the set of public=
 user identioties to be registered. This is not only visible on the P-Assoc=
iated-URI header but also makes configuration of such sets of public user i=
dentities easier.</div>
<div><br></div><div>3GPP and ETSI have extended the mechanism described her=
e with the following additions, for connected PBX&#39;s a subscription para=
meter can be set that allows delivery of the target URI is kept in the requ=
est URI and the contact address is added to the Route header after the stor=
ed path is inserted, which means that in this case the P-Called-Party-ID is=
 not strictly needed for delivery of the target address.</div>
<div><br></div><div>Further this mechanism is extended to allow PBX/corpora=
te proxies to use P-Asserted-Identity instead of P-Preferred-Identity, whic=
h is a more natural thing to do for corporate proxies.</div><div><br></div>
<div>Regarding the side note: This seems misleading as an SSP may very well=
 serve the domain name of =A0the enterprise, even if only for SIP traffic. =
This can all be arranged with normal DNS infrastructure.=A0</div><div><br><=
/div>
<div>Best Regards,</div><div>/Hans Erik van Elburg</div><div><br><div class=
=3D"gmail_quote">On Thu, Jul 9, 2009 at 7:37 PM, Hadriel Kaplan <span dir=
=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepacket.com">HKaplan@acmepacket.c=
om</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;">Howdy,<br>
As part of a discussion in the SIP Forum regarding a model for Registration=
 we are using for IP-PBX&#39;s, I was asked by someone to submit an I-D des=
cribing it, in case it would be of any value. =A0Since I hadn&#39;t seen an=
 I-D before on how 3GPP/ETSI did their model, and since their model is diff=
erent from SIP-Forum&#39;s, I submitted one which describes the models for =
both SIP Forum&#39;s SIP-Connect v1.1 and 3GPP/ETSI.<br>

<br>
NOTE: this draft was not sanctioned nor reviewed by the SIP-Forum nor 3GPP,=
 and I am still talking with colleagues in 3GPP to make sure I got that par=
t right. =A0This is purely an individual submission in all respects.<br>

<br>
Forwarded message:<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
<br>
 =A0 =A0 =A0 =A0Title =A0 =A0 =A0 =A0 =A0 : Session Initiation Protocol (SI=
P) Implicit Registrations<br>
 =A0 =A0 =A0 =A0Author(s) =A0 =A0 =A0 : H. Kaplan<br>
 =A0 =A0 =A0 =A0Filename =A0 =A0 =A0 =A0: draft-kaplan-dispatch-sip-implici=
t-registrations-00.txt<br>
 =A0 =A0 =A0 =A0Pages =A0 =A0 =A0 =A0 =A0 : 12<br>
 =A0 =A0 =A0 =A0Date =A0 =A0 =A0 =A0 =A0 =A0: 2009-07-05<br>
<br>
This document identifies several approaches to provide reachability informa=
tion for a domain or multiple AoR&#39;s using a single SIP REGISTER method =
transaction, in ways not originally envisioned or documented by RFC 3261.<b=
r>

<br>
A URL for this Internet-Draft is:<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-kaplan-dispatch-sip-im=
plicit-registrations-00.txt" target=3D"_blank">http://www.ietf.org/internet=
-drafts/draft-kaplan-dispatch-sip-implicit-registrations-00.txt</a><br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
Below is the data which will enable a MIME compliant mail reader implementa=
tion to automatically retrieve the ASCII version of the Internet-Draft.<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></div></div>

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Thread-Topic: [AVT] [dispatch] Proposal to form Internet Wideband AudioCodec	WG
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, "Dan York" <dyork@voxeo.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] [AVT] Proposal to form Internet Wideband AudioCodec	WG
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Keith,

Comments in-line w/MAR:

Regards,

Michael A. Ramalho, Ph.D.

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
DRAGE, Keith (Keith)
Sent: Thursday, June 04, 2009 6:01 AM
To: Dan York; dispatch@ietf.org
Cc: avt@ietf.org
Subject: Re: [AVT] [dispatch] Proposal to form Internet Wideband
AudioCodec WG

I would argue that G.711 is implemented, not because anyone can
implement it, but because everyone else has implemented it. It
essentially forms the lowest common denominator for interoperability. If
fancy codec x doesn't work, then assuming you have the bandwidth
availabile, G.711 probably will. And I suspect it is royalty free not
because it was always so, but because any IPR that existed has pretty
much expired by now.

G.711 is also the one codec that is probably the most neutral to
transcoding. I can do to codec A to G.711 and back to codec A with less
impact that with any other intermediate codec.

MAR: Let's look at that statement. G.711 is a simple *waveform* codec
that achieves a (single frequency) SNR of approximately 25 dB over an
approximate 25 dB dynamic range. The reason why you can "go to G.711 and
back" is simply that the other (narrowband) codecs you use have less
fidelity (i.e., on average much more distortion - or equivalently lesser
SNR). The distortion introduced by G.711 is less than the distortion
already introduced by the other codecs.

MAR: An equivalent statement is that for the 0 - 4kHz band (narrowband),
25 dB SNR is good enough for telephony.

For wideband there probably is not any one codec that has achieved that
position.

MAR: Well, the game changes with VOICE when it comes to wideband. Speech
has a spectral tilt to it - lesser power per Hz with increasing
frequency (as it comes from a finite energy acoustic source - this MUST
be the case in the limit). The end result is that a PSD for an ensemble
of speech signals has approximately 6dB less energy per octave (i.e.,
over all phonetic content).

MAR: If you use G.711 for wideband (16k sampling) - the higher
frequencies will have more distortion (i.e., much less than 25 dB). Thus
G.711 sounds somewhat bad in wideband application due to this.

MAR: What I believe you are asking for is: 1) a simple "waveform codec",
2) but one that is "pre-equalized" to whiten the spectral tilt in speech
signals. For example, virtually all filter-band-based codecs (e.g.,
MLT/MCT-based) do this via the "bit allocation" process (sometimes in
conjunction with pre-emphasis).

MAR: I believe most of the codecs proposed for this proposed "codec"
working group are speech-model based. If you want a "waveform based"
solution that you can "transform into and out of" (something which I
call a "do-no-harm" codec property) - you simply need pre-equalization
plus a relatively simple waveform codec behind it (G.711 or something
nearly as simple).

MAR: Note these codecs will be less bandwidth efficient than the usual
(model-based) suspects - as the model-based codecs generally only
parameterize the spectrum (NOT the waveform) above 4kHz. But give me
60~90 kbps ... and such a codec becomes a relatively easy task for any
signal processing graduate student (at least one that would have me on
their thesis committee).

MAR: Lastly, waveform based codecs are generally better for speech
recognition applications - a quality that we might also desire to
consider.

I don't believe building a better, even royalty free codec, will
guarantee a position in the market place. The world is littered with
better solutions that never made it. You need a market where everyone
chooses to implement it so that you can use that codec without having to
go to transcoding. Pasting IETF on the front cover does not achieve
that.

I don't really have a problem with people trying to sit down and design
a new codec and IETF then publishing it. Without some other selling
point however it just becomes yet another codec competing for market
place. As such, put it somewhere where it does not interfere with other
work. Sticking it on a separate mailing list, and not letting it compete
for valuable IETF face to face slots as a working group is fine.

regards

Keith
=20

From HKaplan@acmepacket.com  Mon Jul 20 14:02:41 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Mon, 20 Jul 2009 16:49:22 -0400
Thread-Topic: [dispatch] Fwd: I-D Action:draft-kaplan-dispatch-sip-implicit-registrations-00.txt
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Hi,
Thanks for the feedback!
Yes I agree I need to add the references explicitly into the reference sect=
ion.
I also got some similar comments from other folks, so I'll update it with t=
he consolidated set once the submission tool is open (and once we finalize =
this in SIP-Forum as well - it's still under debate).
-hadriel


________________________________
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
Sent: Monday, July 20, 2009 10:39 AM
To: Hadriel Kaplan
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Fwd: I-D Action:draft-kaplan-dispatch-sip-implicit-=
registrations-00.txt

Hi Hadriel,

Thanks for putting this together, this is useful.

- General
I think it would be good to add some informative references to the ETSI TIS=
PAN specification where the NGN/IMS way of performing business trunking (SI=
P trunking) has been specified using the concept of implicitly registered w=
ildcarded public user identities (AORs)

The specification numbers are:
ETSI TS 181 019 V2.0.0 (2007-11)
Telecommunications and Internet converged Services and Protocols for Advanc=
ed Networking (TISPAN);Business Communication Requirements

ETSI TS 182 023 V2.1.1 (2009-01)
Telecommunications and Internet converged Services and Protocols for Advanc=
ed Networking (TISPAN);Core and enterprise NGN interaction scenarios;Archit=
ecture and functional description
ETSI TS 182 025 V2.1.1 (2008-09)
Telecommunications and Internet converged Services and Protocols for Advanc=
ed Networking (TISPAN);Business trunking;Architecture and functional descri=
ption

I think it would also be good to add informative references to the SIP Conn=
ect specifications.

Further it might be good to say something about the different viewpoints ta=
ken by the different bodies:
- ETSI TISPAN/3GPP Business Trunking are providing the viewpoint of the ser=
vice provider connecting IP-PBX's to its IMS/NGN infrastructure
- SIP Connect is more looking at the interconnect specifciation between the=
 an SSP and a IP-PBX.

Ideally both should match at some point to get an interoperable corporate c=
ommunication space.

- Section 6.2
The 3GPP/ETSI TISPAN mechanism does not add the notion of wildcarded P-Asso=
ciated-URI, but it adds the notion of wildcarded public user identity, whic=
h generates so to speak the set of public user identioties to be registered=
. This is not only visible on the P-Associated-URI header but also makes co=
nfiguration of such sets of public user identities easier.

3GPP and ETSI have extended the mechanism described here with the following=
 additions, for connected PBX's a subscription parameter can be set that al=
lows delivery of the target URI is kept in the request URI and the contact =
address is added to the Route header after the stored path is inserted, whi=
ch means that in this case the P-Called-Party-ID is not strictly needed for=
 delivery of the target address.

Further this mechanism is extended to allow PBX/corporate proxies to use P-=
Asserted-Identity instead of P-Preferred-Identity, which is a more natural =
thing to do for corporate proxies.

Regarding the side note: This seems misleading as an SSP may very well serv=
e the domain name of  the enterprise, even if only for SIP traffic. This ca=
n all be arranged with normal DNS infrastructure.

Best Regards,
/Hans Erik van Elburg

On Thu, Jul 9, 2009 at 7:37 PM, Hadriel Kaplan <HKaplan@acmepacket.com<mail=
to:HKaplan@acmepacket.com>> wrote:
Howdy,
As part of a discussion in the SIP Forum regarding a model for Registration=
 we are using for IP-PBX's, I was asked by someone to submit an I-D describ=
ing it, in case it would be of any value.  Since I hadn't seen an I-D befor=
e on how 3GPP/ETSI did their model, and since their model is different from=
 SIP-Forum's, I submitted one which describes the models for both SIP Forum=
's SIP-Connect v1.1 and 3GPP/ETSI.

NOTE: this draft was not sanctioned nor reviewed by the SIP-Forum nor 3GPP,=
 and I am still talking with colleagues in 3GPP to make sure I got that par=
t right.  This is purely an individual submission in all respects.

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

       Title           : Session Initiation Protocol (SIP) Implicit Registr=
ations
       Author(s)       : H. Kaplan
       Filename        : draft-kaplan-dispatch-sip-implicit-registrations-0=
0.txt
       Pages           : 12
       Date            : 2009-07-05

This document identifies several approaches to provide reachability informa=
tion for a domain or multiple AoR's using a single SIP REGISTER method tran=
saction, in ways not originally envisioned or documented by RFC 3261.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-kaplan-dispatch-sip-implicit-regi=
strations-00.txt

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

Below is the data which will enable a MIME compliant mail reader implementa=
tion to automatically retrieve the ASCII version of the Internet-Draft.

_______________________________________________
dispatch mailing list
dispatch@ietf.org<mailto:dispatch@ietf.org>
https://www.ietf.org/mailman/listinfo/dispatch


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Thanks for the feedback!<o:p></o:p></s=
pan></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Yes I agree I need to add the referenc=
es
explicitly into the reference section.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I also got some similar comments from
other folks, so I&#8217;ll update it with the consolidated set once the
submission tool is open (and once we finalize this in SIP-Forum as well &#8=
211;
it&#8217;s still under debate).<o:p></o:p></span></font></p>

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

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

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

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> <st1:Per=
sonName
w:st=3D"on">Hans Erik van Elburg</st1:PersonName>
[mailto:ietf.hanserik@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, July 20, 2009 =
10:39
AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Hadriel Kaplan<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <st1:PersonName w:st=3D"=
on">dispatch@ietf.org</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [dispatch] Fwd:=
 I-D
Action:draft-kaplan-dispatch-sip-implicit-registrations-00.txt</span></font=
><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Hi Hadriel,<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Thanks for putting this together, this is&nbsp;useful.<o:p></o:p></=
span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><span class=3Dapple-style-span><u><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>- General</span><=
/font></u></span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>I think it would be good to add some informative references to the =
ETSI
TISPAN specification where the NGN/IMS way of performing business trunking =
(SIP
trunking) has been specified using the concept of implicitly registered
wildcarded public user identities (AORs)<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>The specification numbers are:<o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><b><font size=3D3 face=3D"Times New Roman"><span
style=3D'font-size:12.0pt;font-weight:bold'>ETSI TS 181 019 </span></font><=
/b>V2.0.0
(2007-11)<br>
Telecommunications and Internet converged Services and Protocols for Advanc=
ed
Networking (TISPAN);<b><span style=3D'font-weight:bold'>Business Communicat=
ion
Requirements</span></b><o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br>
<b><span style=3D'font-weight:bold'>ETSI TS 182 023 V2.1.1</span></b> (2009=
-01)<br>
Telecommunications and Internet converged Services and Protocols for Advanc=
ed
Networking (TISPAN);<b><span style=3D'font-weight:bold'>Core and enterprise=
 NGN
interaction scenarios;Architecture and functional description</span></b><o:=
p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><b><font size=3D3 face=3D"Times New Roman"><span
style=3D'font-size:12.0pt;font-weight:bold'>ETSI TS 182 025 V2.1.1 </span><=
/font></b>(2008-09)<br>
Telecommunications and Internet converged Services and Protocols for Advanc=
ed
Networking (TISPAN);<b><span style=3D'font-weight:bold'>Business
trunking;Architecture and functional description</span></b><o:p></o:p></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>I think it would also be good to add informative references to the =
SIP
Connect specifications.<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Further it might be good to say something about the different
viewpoints taken by the different bodies:<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>- ETSI TISPAN/3GPP Business Trunking are providing the viewpoint of=
 the
service provider connecting IP-PBX's to its IMS/NGN infrastructure<o:p></o:=
p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>- SIP Connect is more looking at the interconnect specifciation bet=
ween
the an SSP and a IP-PBX.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Ideally both should match at some point to get an interoperable
corporate communication space.<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<p class=3DMsoNormal><span class=3Dapple-style-span><u><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>- Section 6.2</sp=
an></font></u></span><o:p></o:p></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>The 3GPP/ETSI TISPAN mechanism does not add the notion of wildcarde=
d
P-Associated-URI, but it adds the notion of wildcarded public user identity=
,
which generates so to speak the set of public user identioties to be
registered. This is not only visible on the P-Associated-URI header but als=
o
makes configuration of such sets of public user identities easier.<o:p></o:=
p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>3GPP and ETSI have extended the mechanism described here with the
following additions, for connected PBX's a subscription parameter can be se=
t
that allows delivery of the target URI is kept in the request URI and the
contact address is added to the Route header after the stored path is inser=
ted,
which means that in this case the P-Called-Party-ID is not strictly needed =
for
delivery of the target address.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Further this mechanism is extended to allow PBX/corporate proxies t=
o
use P-Asserted-Identity instead of P-Preferred-Identity, which is a more
natural thing to do for corporate proxies.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Regarding the side note: This seems misleading as an SSP may very w=
ell
serve the domain name of &nbsp;the enterprise, even if only for SIP traffic=
.
This can all be arranged with normal DNS infrastructure.&nbsp;<o:p></o:p></=
span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Best Regards,<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>/<st1:PersonName w:st=3D"on">Hans Erik van Elburg</st1:PersonName><=
o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On Thu, Jul 9, 2009 at 7:37 PM, Hadriel Kaplan &lt;<a
href=3D"mailto:HKaplan@acmepacket.com">HKaplan@acmepacket.com</a>&gt; wrote=
:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Howdy,<br>
As part of a discussion in the SIP Forum regarding a model for Registration=
 we
are using for IP-PBX's, I was asked by someone to submit an I-D describing =
it,
in case it would be of any value. &nbsp;Since I hadn't seen an I-D before o=
n
how 3GPP/ETSI did their model, and since their model is different from SIP-=
Forum's,
I submitted one which describes the models for both SIP Forum's SIP-Connect
v1.1 and 3GPP/ETSI.<br>
<br>
NOTE: this draft was not sanctioned nor reviewed by the SIP-Forum nor 3GPP,=
 and
I am still talking with colleagues in 3GPP to make sure I got that part rig=
ht.
&nbsp;This is purely an individual submission in all respects.<br>
<br>
Forwarded message:<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp;Title &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : Sessi=
on
Initiation Protocol (SIP) Implicit Registrations<br>
&nbsp; &nbsp; &nbsp; &nbsp;Author(s) &nbsp; &nbsp; &nbsp; : H. Kaplan<br>
&nbsp; &nbsp; &nbsp; &nbsp;Filename &nbsp; &nbsp; &nbsp; &nbsp;:
draft-kaplan-dispatch-sip-implicit-registrations-00.txt<br>
&nbsp; &nbsp; &nbsp; &nbsp;Pages &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : 12<br=
>
&nbsp; &nbsp; &nbsp; &nbsp;Date &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;:
2009-07-05<br>
<br>
This document identifies several approaches to provide reachability informa=
tion
for a domain or multiple AoR's using a single SIP REGISTER method transacti=
on,
in ways not originally envisioned or documented by RFC 3261.<br>
<br>
A URL for this Internet-Draft is:<br>
<a
href=3D"http://www.ietf.org/internet-drafts/draft-kaplan-dispatch-sip-impli=
cit-registrations-00.txt"
target=3D"_blank">http://www.ietf.org/internet-drafts/draft-kaplan-dispatch=
-sip-implicit-registrations-00.txt</a><br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
Below is the data which will enable a MIME compliant mail reader implementa=
tion
to automatically retrieve the ASCII version of the Internet-Draft.<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><o:p></o:p></span></fon=
t></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

</div>

</div>

</div>

</body>

</html>

--_000_E6C2E8958BA59A4FB960963D475F7AC3196CEBD111mail_--

From R.Jesske@telekom.de  Mon Jul 20 22:03:29 2009
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From: <R.Jesske@telekom.de>
To: <Gonzalo.Camarillo@ericsson.com>, <audet@nortel.com>
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi Gonzalo,
Thank you for your comments.
You are correct. The used cases within the document shows where ISUP =
causes will be included.
I think in such cases we should clearly state that SIP Reason should be =
excluded within SIP responses, to avoid contradictions.=20

Then I will include that only within 4xx/5xx/6xx Responses the Reason =
header with an Q.850 Cause makes sense.

There are requirements and three used cases described within the draft =
so I hope that fits.=20

Best Regards

Roland
-----Urspr=FCngliche Nachricht-----
Von: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
Gesendet: Montag, 20. Juli 2009 11:41
An: Francois Audet
Cc: Dale Worley; dispatch@ietf.org; Jesske, Roland
Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04

Hi,

as Francois suggests, a document specifying the use of Reason header=20
fields in responses needs to specify those things (see Francois' list=20
below). Additionally, you should think of whether or not Reason header=20
fields in responses can carry SIP status codes and what happens if they=20
are different to the status code of the response.

In short, the document cannot simply say that now it is OK to use Reason =

in responses. It needs to address the different situations a typical=20
implementation may face.

Cheers,

Gonzalo


Francois Audet wrote:
> Again, the spec is very clear that it IS allowed.
>=20
> I believe the wishy-washy text about "status code explicitly
> allowing it" was meant to exclude responses that were not appropriate,
> and restricing it to effectively error responses.
>=20
> At the time this was written, I believe we were not clear on which
> codes were supposed to be appropriate or not.
>=20
> Seems to me:
> - Any Error response code should be allowed.
> - I don't think 2XX makes sense.
> - 3XX is controversial (as per the email quoted by Roland): seems to =
me it
>   would be quite useful
> - Provisional is interesting... Sounds like 199 error response to =
me...
>=20
>> -----Original Message-----
>> From: Worley, Dale (BL60:9D30)=20
>> Sent: Thursday, July 16, 2009 10:09
>> To: Audet, Francois (SC100:3055)
>> Cc: R.Jesske@telekom.de; dispatch@ietf.org
>> Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>
>> On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois (SC100:3055) =
wrote:
>>> Hi Roland,
>>>
>>> You use case is very common. =20
>>>
>>> I believe you are incorrect in saying that "reasons are=20
>> currently not=20
>>> allowed in responses. Neither conditionally nor allowed".
>>>
>>> RFC 3326 says in 1.0:
>>>   "[...] it can appear in any request
>>>    within a dialog, in any CANCEL request and in any response whose
>>>    status code explicitly allows the presence of this header field."
>>>
>>> To be honest, I believe Q.850 codes are much more common in=20
>> Responses=20
>>> than in requests.
>> Googling
>>
>>      "sip/2.0" reason "q.850"
>>
>> turns up numerous examples of SIP responses using the Reason=20
>> header in the forbidden manner.
>>
>> I'd say that your draft formally allows what people are already =
doing.
>>
>> Dale
>>
>>
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20


From gonzalo.camarillo@ericsson.com  Tue Jul 21 00:53:58 2009
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Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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Hi,

yes, it is normal that the draft has focused on requirements and use 
cases so far. Once (or if) folks agree with those, we have to make sure 
that the normative behavior we define is correct and covers most typical 
cases.

Thanks,

Gonzalo


R.Jesske@telekom.de wrote:
> Hi Gonzalo,
> Thank you for your comments.
> You are correct. The used cases within the document shows where ISUP causes will be included.
> I think in such cases we should clearly state that SIP Reason should be excluded within SIP responses, to avoid contradictions. 
> 
> Then I will include that only within 4xx/5xx/6xx Responses the Reason header with an Q.850 Cause makes sense.
> 
> There are requirements and three used cases described within the draft so I hope that fits. 
> 
> Best Regards
> 
> Roland
> -----Ursprüngliche Nachricht-----
> Von: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
> Gesendet: Montag, 20. Juli 2009 11:41
> An: Francois Audet
> Cc: Dale Worley; dispatch@ietf.org; Jesske, Roland
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> 
> Hi,
> 
> as Francois suggests, a document specifying the use of Reason header 
> fields in responses needs to specify those things (see Francois' list 
> below). Additionally, you should think of whether or not Reason header 
> fields in responses can carry SIP status codes and what happens if they 
> are different to the status code of the response.
> 
> In short, the document cannot simply say that now it is OK to use Reason 
> in responses. It needs to address the different situations a typical 
> implementation may face.
> 
> Cheers,
> 
> Gonzalo
> 
> 
> Francois Audet wrote:
>> Again, the spec is very clear that it IS allowed.
>>
>> I believe the wishy-washy text about "status code explicitly
>> allowing it" was meant to exclude responses that were not appropriate,
>> and restricing it to effectively error responses.
>>
>> At the time this was written, I believe we were not clear on which
>> codes were supposed to be appropriate or not.
>>
>> Seems to me:
>> - Any Error response code should be allowed.
>> - I don't think 2XX makes sense.
>> - 3XX is controversial (as per the email quoted by Roland): seems to me it
>>   would be quite useful
>> - Provisional is interesting... Sounds like 199 error response to me...
>>
>>> -----Original Message-----
>>> From: Worley, Dale (BL60:9D30) 
>>> Sent: Thursday, July 16, 2009 10:09
>>> To: Audet, Francois (SC100:3055)
>>> Cc: R.Jesske@telekom.de; dispatch@ietf.org
>>> Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>>>
>>> On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois (SC100:3055) wrote:
>>>> Hi Roland,
>>>>
>>>> You use case is very common.  
>>>>
>>>> I believe you are incorrect in saying that "reasons are 
>>> currently not 
>>>> allowed in responses. Neither conditionally nor allowed".
>>>>
>>>> RFC 3326 says in 1.0:
>>>>   "[...] it can appear in any request
>>>>    within a dialog, in any CANCEL request and in any response whose
>>>>    status code explicitly allows the presence of this header field."
>>>>
>>>> To be honest, I believe Q.850 codes are much more common in 
>>> Responses 
>>>> than in requests.
>>> Googling
>>>
>>>      "sip/2.0" reason "q.850"
>>>
>>> turns up numerous examples of SIP responses using the Reason 
>>> header in the forbidden manner.
>>>
>>> I'd say that your draft formally allows what people are already doing.
>>>
>>> Dale
>>>
>>>
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 


From gonzalo.camarillo@ericsson.com  Tue Jul 21 01:05:17 2009
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Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi Paul,

it is good that we agree that some problems need both ends to 
collaborate. Now, let's discuss whether or not that is the case with 
*this* problem.

> I still haven't seen anything showing why this approach ends up being 
> "better" than the "centralized" approach. (Note that I don't agree with 
> the term "centralized", because it isn't really more centralized than 
> the "distributed" approach is. This would become clearer if we looked at 
> all the communications necessary to establish a call with disaggregated 
> media, and the various topologies of devices that might play in such 
> scenarios.)

well, 3pcc is centralized because you have a (central) controller that 
is in control of all SIP signalling. If that controller leaves the 
session, the whole session disappears. An approach where several devices 
have their own SIP dialog with the remote devices is not centralized 
because you can remove any of them and the other sessions will remain 
unaffected.

> Perhaps we should back off from the signaling mechanisms for the moment 
> and consider the use cases of interest in more detail, and the various 
> sorts of communications between them that are necessary to establish a 
> call that involves them.

The arguments have been discussed several times. The thing is that 3pcc 
is a complex mechanism that is not supported by most UAs. You can argue 
about its complexity, but the fact that most UAs do not support 3pcc is 
a fact.

There are folks that are willing to implement support for a correlation 
mechanism for dialogs but would not implement 3pcc because (they think) 
it is too complicated.

A second argument is that any device should be able to leave the session 
at any point. If you want the 3pcc controller to leave the session while 
the session remains established, you will need to use a complex 
combination of REFER and Replaces that very few (if any) UAs support.

Therefore, the requirement here is that the mechanism should not be so 
complicated that implementers do not actually want to implement it :o) 
... unfortunately, as we all know too well, we have a history of doing 
just that.

Cheers,

Gonzalo


From john.elwell@siemens-enterprise.com  Tue Jul 21 05:43:08 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [dispatch] Disaggregated Media in SIP
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Gonzalo,

I have already expressed opinions similar to Paul on several occasions.
Experience tells us we cannot rely on the remote UA supporting a
particular feature. Taking REFER as an example, we tend to end up with a
B2BUA terminating a REFER request, because:
- the remote UA doesn't support REFER;
- some UAs don't support REFER, so if the B2BUA sometimes needs to
terminate REFER to accommodate those UAs, it might as well always
terminate REFER;
- some UAs don't accept REFER as a matter of policy.

So B2BUAs tend to act as an anchor point and terminate REFER requests,
behaving in a 3PCC-like manner.

If the distributed form of disaggregated media goes ahead, I think that
is what will happen in practice, i.e., a B2BUA will need to act as the
anchor point. If UAs wanting to use disaggregated media are behind a
B2BUA that concentrates the multiple dialogs into a single ongoing
dialog towards the remote UA, that will work. It does indeed have the
benefit that one local UA can drop out, leaving the other local UA(s)
still in communication with the remote UA. Relying on the remote UA
supporting distributed disaggregated media I fear will prevent the
capability being used in many cases where it might be required.

As part of specifying distributed disaggregated media, we will need to
pay attention to all the interactions that can occur when both sides
attempt to use disaggregated media, particular when the two sides
simultaneously try to add or remove UAs. This sounds like a difficult
problem, but I may be wrong. Having an anchor point (thereby making it
centralised, in a way) would probably ease these problems.

John

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> Sent: 21 July 2009 09:05
> To: Paul Kyzivat
> Cc: dispatch@ietf.org; Henry Sinnreich
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
> Hi Paul,
>=20
> it is good that we agree that some problems need both ends to=20
> collaborate. Now, let's discuss whether or not that is the case with=20
> *this* problem.
>=20
> > I still haven't seen anything showing why this approach=20
> ends up being=20
> > "better" than the "centralized" approach. (Note that I=20
> don't agree with=20
> > the term "centralized", because it isn't really more=20
> centralized than=20
> > the "distributed" approach is. This would become clearer if=20
> we looked at=20
> > all the communications necessary to establish a call with=20
> disaggregated=20
> > media, and the various topologies of devices that might=20
> play in such=20
> > scenarios.)
>=20
> well, 3pcc is centralized because you have a (central)=20
> controller that=20
> is in control of all SIP signalling. If that controller leaves the=20
> session, the whole session disappears. An approach where=20
> several devices=20
> have their own SIP dialog with the remote devices is not centralized=20
> because you can remove any of them and the other sessions will remain=20
> unaffected.
>=20
> > Perhaps we should back off from the signaling mechanisms=20
> for the moment=20
> > and consider the use cases of interest in more detail, and=20
> the various=20
> > sorts of communications between them that are necessary to=20
> establish a=20
> > call that involves them.
>=20
> The arguments have been discussed several times. The thing is=20
> that 3pcc=20
> is a complex mechanism that is not supported by most UAs. You=20
> can argue=20
> about its complexity, but the fact that most UAs do not=20
> support 3pcc is=20
> a fact.
>=20
> There are folks that are willing to implement support for a=20
> correlation=20
> mechanism for dialogs but would not implement 3pcc because=20
> (they think)=20
> it is too complicated.
>=20
> A second argument is that any device should be able to leave=20
> the session=20
> at any point. If you want the 3pcc controller to leave the=20
> session while=20
> the session remains established, you will need to use a complex=20
> combination of REFER and Replaces that very few (if any) UAs support.
>=20
> Therefore, the requirement here is that the mechanism should=20
> not be so=20
> complicated that implementers do not actually want to=20
> implement it :o)=20
> ... unfortunately, as we all know too well, we have a history=20
> of doing=20
> just that.
>=20
> Cheers,
>=20
> Gonzalo
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From gonzalo.camarillo@ericsson.com  Tue Jul 21 07:51:29 2009
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Folks,

our charter includes the following milestone:

   Aug 2009   Proposed charter for SIP Overload WG

Volker (in the cc:) will be circulating a charter proposal for such a WG 
on this list shortly.

Additionally, according to its authors, the following draft should be 
ready for WGLC:

   draft-ietf-sipping-overload-design-02

Given that we are chartered to form a WG on SIP overload, we will 
include this draft as the first milestone for the new WG. Therefore, 
this draft will be WGLC in the new WG.

Cheers,

Gonzalo
DISPATCH co-chair

From gonzalo.camarillo@ericsson.com  Tue Jul 21 07:55:10 2009
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Cc: dispatch@ietf.org, Henry Sinnreich <hsinnrei@adobe.com>
Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi John,

many UAs already support multiple SIP dialogs in parallel. Adding a 
header that instruct them to treat the media sessions related to those 
dialogs as a single one is much easier than anything you have described 
in your email.

The point is that people are going to implement something like this 
because it is fairly simple and it is needed. We can either design a 
mechanism that meets their simplicity requirement or propose complex 
mechanisms that, as usual, will not get implemented.

Cheers,

Gonzalo

Elwell, John wrote:
> Gonzalo,
> 
> I have already expressed opinions similar to Paul on several occasions.
> Experience tells us we cannot rely on the remote UA supporting a
> particular feature. Taking REFER as an example, we tend to end up with a
> B2BUA terminating a REFER request, because:
> - the remote UA doesn't support REFER;
> - some UAs don't support REFER, so if the B2BUA sometimes needs to
> terminate REFER to accommodate those UAs, it might as well always
> terminate REFER;
> - some UAs don't accept REFER as a matter of policy.
> 
> So B2BUAs tend to act as an anchor point and terminate REFER requests,
> behaving in a 3PCC-like manner.
> 
> If the distributed form of disaggregated media goes ahead, I think that
> is what will happen in practice, i.e., a B2BUA will need to act as the
> anchor point. If UAs wanting to use disaggregated media are behind a
> B2BUA that concentrates the multiple dialogs into a single ongoing
> dialog towards the remote UA, that will work. It does indeed have the
> benefit that one local UA can drop out, leaving the other local UA(s)
> still in communication with the remote UA. Relying on the remote UA
> supporting distributed disaggregated media I fear will prevent the
> capability being used in many cases where it might be required.
> 
> As part of specifying distributed disaggregated media, we will need to
> pay attention to all the interactions that can occur when both sides
> attempt to use disaggregated media, particular when the two sides
> simultaneously try to add or remove UAs. This sounds like a difficult
> problem, but I may be wrong. Having an anchor point (thereby making it
> centralised, in a way) would probably ease these problems.
> 
> John
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>> Sent: 21 July 2009 09:05
>> To: Paul Kyzivat
>> Cc: dispatch@ietf.org; Henry Sinnreich
>> Subject: Re: [dispatch] Disaggregated Media in SIP
>>
>> Hi Paul,
>>
>> it is good that we agree that some problems need both ends to 
>> collaborate. Now, let's discuss whether or not that is the case with 
>> *this* problem.
>>
>>> I still haven't seen anything showing why this approach 
>> ends up being 
>>> "better" than the "centralized" approach. (Note that I 
>> don't agree with 
>>> the term "centralized", because it isn't really more 
>> centralized than 
>>> the "distributed" approach is. This would become clearer if 
>> we looked at 
>>> all the communications necessary to establish a call with 
>> disaggregated 
>>> media, and the various topologies of devices that might 
>> play in such 
>>> scenarios.)
>> well, 3pcc is centralized because you have a (central) 
>> controller that 
>> is in control of all SIP signalling. If that controller leaves the 
>> session, the whole session disappears. An approach where 
>> several devices 
>> have their own SIP dialog with the remote devices is not centralized 
>> because you can remove any of them and the other sessions will remain 
>> unaffected.
>>
>>> Perhaps we should back off from the signaling mechanisms 
>> for the moment 
>>> and consider the use cases of interest in more detail, and 
>> the various 
>>> sorts of communications between them that are necessary to 
>> establish a 
>>> call that involves them.
>> The arguments have been discussed several times. The thing is 
>> that 3pcc 
>> is a complex mechanism that is not supported by most UAs. You 
>> can argue 
>> about its complexity, but the fact that most UAs do not 
>> support 3pcc is 
>> a fact.
>>
>> There are folks that are willing to implement support for a 
>> correlation 
>> mechanism for dialogs but would not implement 3pcc because 
>> (they think) 
>> it is too complicated.
>>
>> A second argument is that any device should be able to leave 
>> the session 
>> at any point. If you want the 3pcc controller to leave the 
>> session while 
>> the session remains established, you will need to use a complex 
>> combination of REFER and Replaces that very few (if any) UAs support.
>>
>> Therefore, the requirement here is that the mechanism should 
>> not be so 
>> complicated that implementers do not actually want to 
>> implement it :o) 
>> ... unfortunately, as we all know too well, we have a history 
>> of doing 
>> just that.
>>
>> Cheers,
>>
>> Gonzalo
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>


From john.elwell@siemens-enterprise.com  Tue Jul 21 08:39:43 2009
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Date: Tue, 21 Jul 2009 16:35:14 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: dispatch@ietf.org, Henry Sinnreich <hsinnrei@adobe.com>
Subject: Re: [dispatch] Disaggregated Media in SIP
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Gonzalo,

Well, my concern is that it might not be as simple as described when
both sides use multiple UAs. If I am using 2 UAs, and you are using 2
UAs, presumably we end up with 4 dialogs - correct?

John=20

> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
> Sent: 21 July 2009 15:37
> To: Elwell, John
> Cc: Paul Kyzivat; dispatch@ietf.org; Henry Sinnreich
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
> Hi John,
>=20
> many UAs already support multiple SIP dialogs in parallel. Adding a=20
> header that instruct them to treat the media sessions related=20
> to those=20
> dialogs as a single one is much easier than anything you have=20
> described=20
> in your email.
>=20
> The point is that people are going to implement something like this=20
> because it is fairly simple and it is needed. We can either design a=20
> mechanism that meets their simplicity requirement or propose complex=20
> mechanisms that, as usual, will not get implemented.
>=20
> Cheers,
>=20
> Gonzalo
>=20
> Elwell, John wrote:
> > Gonzalo,
> >=20
> > I have already expressed opinions similar to Paul on=20
> several occasions.
> > Experience tells us we cannot rely on the remote UA supporting a
> > particular feature. Taking REFER as an example, we tend to=20
> end up with a
> > B2BUA terminating a REFER request, because:
> > - the remote UA doesn't support REFER;
> > - some UAs don't support REFER, so if the B2BUA sometimes needs to
> > terminate REFER to accommodate those UAs, it might as well always
> > terminate REFER;
> > - some UAs don't accept REFER as a matter of policy.
> >=20
> > So B2BUAs tend to act as an anchor point and terminate=20
> REFER requests,
> > behaving in a 3PCC-like manner.
> >=20
> > If the distributed form of disaggregated media goes ahead,=20
> I think that
> > is what will happen in practice, i.e., a B2BUA will need to=20
> act as the
> > anchor point. If UAs wanting to use disaggregated media are behind a
> > B2BUA that concentrates the multiple dialogs into a single ongoing
> > dialog towards the remote UA, that will work. It does=20
> indeed have the
> > benefit that one local UA can drop out, leaving the other=20
> local UA(s)
> > still in communication with the remote UA. Relying on the remote UA
> > supporting distributed disaggregated media I fear will prevent the
> > capability being used in many cases where it might be required.
> >=20
> > As part of specifying distributed disaggregated media, we=20
> will need to
> > pay attention to all the interactions that can occur when both sides
> > attempt to use disaggregated media, particular when the two sides
> > simultaneously try to add or remove UAs. This sounds like a=20
> difficult
> > problem, but I may be wrong. Having an anchor point=20
> (thereby making it
> > centralised, in a way) would probably ease these problems.
> >=20
> > John
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org=20
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> >> Sent: 21 July 2009 09:05
> >> To: Paul Kyzivat
> >> Cc: dispatch@ietf.org; Henry Sinnreich
> >> Subject: Re: [dispatch] Disaggregated Media in SIP
> >>
> >> Hi Paul,
> >>
> >> it is good that we agree that some problems need both ends to=20
> >> collaborate. Now, let's discuss whether or not that is the=20
> case with=20
> >> *this* problem.
> >>
> >>> I still haven't seen anything showing why this approach=20
> >> ends up being=20
> >>> "better" than the "centralized" approach. (Note that I=20
> >> don't agree with=20
> >>> the term "centralized", because it isn't really more=20
> >> centralized than=20
> >>> the "distributed" approach is. This would become clearer if=20
> >> we looked at=20
> >>> all the communications necessary to establish a call with=20
> >> disaggregated=20
> >>> media, and the various topologies of devices that might=20
> >> play in such=20
> >>> scenarios.)
> >> well, 3pcc is centralized because you have a (central)=20
> >> controller that=20
> >> is in control of all SIP signalling. If that controller leaves the=20
> >> session, the whole session disappears. An approach where=20
> >> several devices=20
> >> have their own SIP dialog with the remote devices is not=20
> centralized=20
> >> because you can remove any of them and the other sessions=20
> will remain=20
> >> unaffected.
> >>
> >>> Perhaps we should back off from the signaling mechanisms=20
> >> for the moment=20
> >>> and consider the use cases of interest in more detail, and=20
> >> the various=20
> >>> sorts of communications between them that are necessary to=20
> >> establish a=20
> >>> call that involves them.
> >> The arguments have been discussed several times. The thing is=20
> >> that 3pcc=20
> >> is a complex mechanism that is not supported by most UAs. You=20
> >> can argue=20
> >> about its complexity, but the fact that most UAs do not=20
> >> support 3pcc is=20
> >> a fact.
> >>
> >> There are folks that are willing to implement support for a=20
> >> correlation=20
> >> mechanism for dialogs but would not implement 3pcc because=20
> >> (they think)=20
> >> it is too complicated.
> >>
> >> A second argument is that any device should be able to leave=20
> >> the session=20
> >> at any point. If you want the 3pcc controller to leave the=20
> >> session while=20
> >> the session remains established, you will need to use a complex=20
> >> combination of REFER and Replaces that very few (if any)=20
> UAs support.
> >>
> >> Therefore, the requirement here is that the mechanism should=20
> >> not be so=20
> >> complicated that implementers do not actually want to=20
> >> implement it :o)=20
> >> ... unfortunately, as we all know too well, we have a history=20
> >> of doing=20
> >> just that.
> >>
> >> Cheers,
> >>
> >> Gonzalo
> >>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
>=20
>=20

From victor.pascual.avila@gmail.com  Tue Jul 21 09:17:09 2009
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Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
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As mentioned in the below message, the authors would appreciate
feedback as to whether this is helpful and going in the right
direction.

Thanks,
-Victor

2009/6/29 Elwell, John <john.elwell@siemens-enterprise.com>:
> There was an extensive discussion at IETF#74 on SIP Identity, where a goo=
d many views were put forward. The only consensus seemed to be to continue =
to discuss the topic.
>
> One of the themes during that discussion was to focus more on requirement=
s, which the authors have attempted to do in this new draft:
> http://www.ietf.org/internet-drafts/draft-elwell-dispatch-identity-reqs-0=
0.txt
>
> We would appreciate feedback as to whether this is helpful and going in t=
he right direction, as well as detailed comments.
>
> I had hoped to do this a lot earlier, to trigger a discussion in time to =
get something set up for DISPATCH at IETF#75, but I failed, and also I fail=
ed to talk to the many of you who have interest in the topic and expressed =
opinions in the past. But since the deadline is approaching, I thought it b=
est to post the draft and let discussion continue on that basis.
>
> John
>

From gonzalo.camarillo@ericsson.com  Tue Jul 21 09:23:59 2009
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Cc: dispatch@ietf.org, Henry Sinnreich <hsinnrei@adobe.com>
Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi,

> Well, my concern is that it might not be as simple as described when
> both sides use multiple UAs. 

there are two aspects: device coordination among a user's devices and 
dialog association.

> If I am using 2 UAs, and you are using 2
> UAs, presumably we end up with 4 dialogs - correct?

Not necessarily. The draft has several use cases. For example, you can 
use a pair of SIP-enabled speakers to play out my voice and a 
SIP-enabled giant screen while I send you audio from my mobile phone and 
video from an external SIP-enabled camera. In this case, there would be 
two dialogs: one from my camera to your screen and one from my mobile 
phone to your speakers.

Cheers,

Gonzalo


From AUDET@nortel.com  Tue Jul 21 10:12:48 2009
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From: "Francois Audet" <audet@nortel.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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I agree with Gonzalo.

I my opinion, we need a very loosely coupled mechanism for this. =
Something very
simple.

People who need a "centralized" approach will continue to use 3PCC. =
Typically,
those are not simple UAs like phones because of complexity, but Call =
Centers
and the likes. This is a different problem.

> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
> Sent: Tuesday, July 21, 2009 01:05
> To: Paul Kyzivat
> Cc: Audet, Francois (SC100:3055); Henry Sinnreich; dispatch@ietf.org
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
> Hi Paul,
>=20
> it is good that we agree that some problems need both ends to=20
> collaborate. Now, let's discuss whether or not that is the case with
> *this* problem.
>=20
> > I still haven't seen anything showing why this approach=20
> ends up being=20
> > "better" than the "centralized" approach. (Note that I don't agree=20
> > with the term "centralized", because it isn't really more=20
> centralized=20
> > than the "distributed" approach is. This would become clearer if we=20
> > looked at all the communications necessary to establish a call with=20
> > disaggregated media, and the various topologies of devices=20
> that might=20
> > play in such
> > scenarios.)
>=20
> well, 3pcc is centralized because you have a (central)=20
> controller that is in control of all SIP signalling. If that=20
> controller leaves the session, the whole session disappears.=20
> An approach where several devices have their own SIP dialog=20
> with the remote devices is not centralized because you can=20
> remove any of them and the other sessions will remain unaffected.
>=20
> > Perhaps we should back off from the signaling mechanisms for the=20
> > moment and consider the use cases of interest in more=20
> detail, and the=20
> > various sorts of communications between them that are necessary to=20
> > establish a call that involves them.
>=20
> The arguments have been discussed several times. The thing is=20
> that 3pcc is a complex mechanism that is not supported by=20
> most UAs. You can argue about its complexity, but the fact=20
> that most UAs do not support 3pcc is a fact.
>=20
> There are folks that are willing to implement support for a=20
> correlation mechanism for dialogs but would not implement=20
> 3pcc because (they think) it is too complicated.
>=20
> A second argument is that any device should be able to leave=20
> the session at any point. If you want the 3pcc controller to=20
> leave the session while the session remains established, you=20
> will need to use a complex combination of REFER and Replaces=20
> that very few (if any) UAs support.
>=20
> Therefore, the requirement here is that the mechanism should=20
> not be so complicated that implementers do not actually want=20
> to implement it :o) ... unfortunately, as we all know too=20
> well, we have a history of doing just that.
>=20
> Cheers,
>=20
> Gonzalo
>=20
>=20

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From: "Francois Audet" <audet@nortel.com>
To: <R.Jesske@telekom.de>, <Gonzalo.Camarillo@ericsson.com>
Cc: dispatch@ietf.org
Subject: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
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and explain why 3XX, 2XX and 1XX don't make sense.=20

> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> Sent: Monday, July 20, 2009 22:03
> To: Gonzalo.Camarillo@ericsson.com; Audet, Francois (SC100:3055)
> Cc: Worley, Dale (BL60:9D30); dispatch@ietf.org
> Subject: AW: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Hi Gonzalo,
> Thank you for your comments.
> You are correct. The used cases within the document shows=20
> where ISUP causes will be included.
> I think in such cases we should clearly state that SIP Reason=20
> should be excluded within SIP responses, to avoid contradictions.=20
>=20
> Then I will include that only within 4xx/5xx/6xx Responses=20
> the Reason header with an Q.850 Cause makes sense.
>=20
> There are requirements and three used cases described within=20
> the draft so I hope that fits.=20
>=20
> Best Regards
>=20
> Roland
> -----Urspr=FCngliche Nachricht-----
> Von: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> Gesendet: Montag, 20. Juli 2009 11:41
> An: Francois Audet
> Cc: Dale Worley; dispatch@ietf.org; Jesske, Roland
> Betreff: Re: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
>=20
> Hi,
>=20
> as Francois suggests, a document specifying the use of Reason=20
> header fields in responses needs to specify those things (see=20
> Francois' list below). Additionally, you should think of=20
> whether or not Reason header fields in responses can carry=20
> SIP status codes and what happens if they are different to=20
> the status code of the response.
>=20
> In short, the document cannot simply say that now it is OK to=20
> use Reason in responses. It needs to address the different=20
> situations a typical implementation may face.
>=20
> Cheers,
>=20
> Gonzalo
>=20
>=20
> Francois Audet wrote:
> > Again, the spec is very clear that it IS allowed.
> >=20
> > I believe the wishy-washy text about "status code=20
> explicitly allowing=20
> > it" was meant to exclude responses that were not appropriate, and=20
> > restricing it to effectively error responses.
> >=20
> > At the time this was written, I believe we were not clear on which=20
> > codes were supposed to be appropriate or not.
> >=20
> > Seems to me:
> > - Any Error response code should be allowed.
> > - I don't think 2XX makes sense.
> > - 3XX is controversial (as per the email quoted by Roland):=20
> seems to me it
> >   would be quite useful
> > - Provisional is interesting... Sounds like 199 error=20
> response to me...
> >=20
> >> -----Original Message-----
> >> From: Worley, Dale (BL60:9D30)
> >> Sent: Thursday, July 16, 2009 10:09
> >> To: Audet, Francois (SC100:3055)
> >> Cc: R.Jesske@telekom.de; dispatch@ietf.org
> >> Subject: RE: [dispatch] draft-jesske-sipping-etsi-ngn-reason-04
> >>
> >> On Thu, 2009-07-16 at 12:48 -0400, Audet, Francois=20
> (SC100:3055) wrote:
> >>> Hi Roland,
> >>>
> >>> You use case is very common. =20
> >>>
> >>> I believe you are incorrect in saying that "reasons are
> >> currently not
> >>> allowed in responses. Neither conditionally nor allowed".
> >>>
> >>> RFC 3326 says in 1.0:
> >>>   "[...] it can appear in any request
> >>>    within a dialog, in any CANCEL request and in any=20
> response whose
> >>>    status code explicitly allows the presence of this=20
> header field."
> >>>
> >>> To be honest, I believe Q.850 codes are much more common in
> >> Responses
> >>> than in requests.
> >> Googling
> >>
> >>      "sip/2.0" reason "q.850"
> >>
> >> turns up numerous examples of SIP responses using the=20
> Reason header=20
> >> in the forbidden manner.
> >>
> >> I'd say that your draft formally allows what people are=20
> already doing.
> >>
> >> Dale
> >>
> >>
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
>=20
>=20

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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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I would agree with Paul and John. There's a well established way of 
handling multiple media streams in SIP - part of SIP since the 
beginning. Don't tamper with that. If someone wants to disaggregate the 
streams they should be the ones to pay for it - by dealing with the 
added complexity and the changes to the existing model.

Plus I'd be very skeptical about claims that the alternative is simple. 
For example, how do you transfer an audio+video call that is actually 
two SIP dialogs? Now all of a sudden there are additional failure modes 
because the dialogs have to be transferred separately and can succeed or 
fail independently.

Existing UAs may already support simultaneous dialogs. If they're 
independent calls then that's not at all the same thing.

Thanks,
Anders

Gonzalo Camarillo wrote:
> Hi John,
> 
> many UAs already support multiple SIP dialogs in parallel. Adding a 
> header that instruct them to treat the media sessions related to those 
> dialogs as a single one is much easier than anything you have described 
> in your email.
> 
> The point is that people are going to implement something like this 
> because it is fairly simple and it is needed. We can either design a 
> mechanism that meets their simplicity requirement or propose complex 
> mechanisms that, as usual, will not get implemented.
> 
> Cheers,
> 
> Gonzalo
> 
> Elwell, John wrote:
>> Gonzalo,
>>
>> I have already expressed opinions similar to Paul on several occasions.
>> Experience tells us we cannot rely on the remote UA supporting a
>> particular feature. Taking REFER as an example, we tend to end up with a
>> B2BUA terminating a REFER request, because:
>> - the remote UA doesn't support REFER;
>> - some UAs don't support REFER, so if the B2BUA sometimes needs to
>> terminate REFER to accommodate those UAs, it might as well always
>> terminate REFER;
>> - some UAs don't accept REFER as a matter of policy.
>>
>> So B2BUAs tend to act as an anchor point and terminate REFER requests,
>> behaving in a 3PCC-like manner.
>>
>> If the distributed form of disaggregated media goes ahead, I think that
>> is what will happen in practice, i.e., a B2BUA will need to act as the
>> anchor point. If UAs wanting to use disaggregated media are behind a
>> B2BUA that concentrates the multiple dialogs into a single ongoing
>> dialog towards the remote UA, that will work. It does indeed have the
>> benefit that one local UA can drop out, leaving the other local UA(s)
>> still in communication with the remote UA. Relying on the remote UA
>> supporting distributed disaggregated media I fear will prevent the
>> capability being used in many cases where it might be required.
>>
>> As part of specifying distributed disaggregated media, we will need to
>> pay attention to all the interactions that can occur when both sides
>> attempt to use disaggregated media, particular when the two sides
>> simultaneously try to add or remove UAs. This sounds like a difficult
>> problem, but I may be wrong. Having an anchor point (thereby making it
>> centralised, in a way) would probably ease these problems.
>>
>> John
>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On 
>>> Behalf Of Gonzalo Camarillo
>>> Sent: 21 July 2009 09:05
>>> To: Paul Kyzivat
>>> Cc: dispatch@ietf.org; Henry Sinnreich
>>> Subject: Re: [dispatch] Disaggregated Media in SIP
>>>
>>> Hi Paul,
>>>
>>> it is good that we agree that some problems need both ends to 
>>> collaborate. Now, let's discuss whether or not that is the case with 
>>> *this* problem.
>>>
>>>> I still haven't seen anything showing why this approach 
>>> ends up being
>>>> "better" than the "centralized" approach. (Note that I 
>>> don't agree with
>>>> the term "centralized", because it isn't really more 
>>> centralized than
>>>> the "distributed" approach is. This would become clearer if 
>>> we looked at
>>>> all the communications necessary to establish a call with 
>>> disaggregated
>>>> media, and the various topologies of devices that might 
>>> play in such
>>>> scenarios.)
>>> well, 3pcc is centralized because you have a (central) controller 
>>> that is in control of all SIP signalling. If that controller leaves 
>>> the session, the whole session disappears. An approach where several 
>>> devices have their own SIP dialog with the remote devices is not 
>>> centralized because you can remove any of them and the other sessions 
>>> will remain unaffected.
>>>
>>>> Perhaps we should back off from the signaling mechanisms 
>>> for the moment
>>>> and consider the use cases of interest in more detail, and 
>>> the various
>>>> sorts of communications between them that are necessary to 
>>> establish a
>>>> call that involves them.
>>> The arguments have been discussed several times. The thing is that 
>>> 3pcc is a complex mechanism that is not supported by most UAs. You 
>>> can argue about its complexity, but the fact that most UAs do not 
>>> support 3pcc is a fact.
>>>
>>> There are folks that are willing to implement support for a 
>>> correlation mechanism for dialogs but would not implement 3pcc 
>>> because (they think) it is too complicated.
>>>
>>> A second argument is that any device should be able to leave the 
>>> session at any point. If you want the 3pcc controller to leave the 
>>> session while the session remains established, you will need to use a 
>>> complex combination of REFER and Replaces that very few (if any) UAs 
>>> support.
>>>
>>> Therefore, the requirement here is that the mechanism should not be 
>>> so complicated that implementers do not actually want to implement it 
>>> :o) ... unfortunately, as we all know too well, we have a history of 
>>> doing just that.
>>>
>>> Cheers,
>>>
>>> Gonzalo
>>>
>>> _______________________________________________
>>> 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 volkerh@bell-labs.com  Tue Jul 21 14:10:48 2009
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Subject: [dispatch] Charter Proposal for SIP Overload
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All,

below is a charter proposal for a WG on SIP Overload Control.

All comments, thoughts, feedback is very welcome!

Thanks,

Volker



SIP Overload Control WG Charter
-------------------------------

As with any network element, a Session Initiation Protocol (SIP) 
[RFC3261] server can suffer from overload when the number of SIP 
messages it receives exceeds the number of messages it can process. 
Overload is said to occur when a SIP server does not have sufficient 
resources to process all incoming SIP messages.  These resources can 
include CPU, memory, network bandwidth, input/output, or disk resources.

Overload can pose a serious problem for a network of SIP servers. During 
periods of overload, the throughput of a network of SIP servers can be 
significantly degraded.  In fact, overload may lead to a situation in 
which the throughput drops down to a small fraction of the original 
processing capacity.  This is often called congestion collapse.

To objective of a SIP overload control mechanism is to enable SIP 
servers to perform close to their capacity limit during times of overload.

The SIP protocol provides a limited mechanism for overload control 
through its 503 (Service Unavailable) response code and the Retry-After 
header.  However, this mechanism cannot prevent overload of a SIP server 
and it cannot prevent congestion collapse.  In fact, it may cause 
traffic to oscillate and to shift between SIP servers and thereby worsen 
an overload condition.  A detailed discussion of the SIP overload 
problem, the problems with the 503 (Service Unavailable) response code 
and the Retry-After header and the requirements for a SIP overload 
control mechanism can be found in [RFC5390].

The objective of the proposed working group is to develop mechanisms for 
SIP overload control. Two complementary approaches exist for handling 
overload situations:
- The first mechanism aims at preventing overload in SIP servers by 
adjusting the incoming load to a level that is acceptable for this 
server. This mechanism uses implicit and/or explicit feedback to 
determine when an overload condition occurs in a SIP server and a 
reduction in load is required.
- The second mechanism aims at preventing overload in SIP servers by 
distributing load control filters to SIP servers that throttle calls 
based on their destination domain, telephone number prefix or for a 
specific user. Such filters can be used, for example, to throttle calls 
to a hotline during a ticket-giveaway event.

Specifically, the proposed working group will develop the following 
deliverables:
1. A document describing key design considerations for a SIP overload 
control mechanism.
2. A specification for an SIP overload control mechanism based on 
implicit/explicit feedback.
3. A specification for a SIP load filtering mechanism.



From gunnar.hellstrom@omnitor.se  Tue Jul 21 14:47:46 2009
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
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Subject: [dispatch] Registration of the "real-time-text" Media Feature Tag
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A new draft draft-hellstrom-text-turntaking-02.txt  is available.
Since I am not sure what group is responsible for approving this kind of
registrations, I turn to the Dispatch group.=20
In an earlier version of this draft, there were more values defined for =
in
detail specifying what limitations in simultaneity a network component =
has.
This was regarded too complex, so in this version, the indication is =
limited
to two variants: full and restricted.  For further info, see the =
document.
=20
Details:
=20
Title : Registration of the Real-time-text Media Feature Tag

Author(s) : G. Hellstrom, A. Wijk

Filename : draft-hellstrom-text-turntaking-02.txt

Pages : 6

Date : 2009-07-12

This memo defines a new Media Feature Tag "real-time-text" for use in =
SIP
registration and session establishment. This is used to indicate if a =
device
capable of text communication has full real-time text capabilities or
limitations in its capabilities requiring the users to apply some
turn-taking habits.

A URL for this Internet-Draft is:

=20
<http://www.ietf.org/internet-drafts/draft-hellstrom-text-turntaking-02.t=
xt>
http://www.ietf.org/internet-drafts/draft-hellstrom-text-turntaking-02.tx=
t

=20
-------------------------------------------------------------------
Gunnar Hellstr=F6m
Omnitor
gunnar.hellstrom@omnitor.se
Tel: +46708204288
www.omnitor.se <http://www.omnitor.se/>=20
=20

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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5730.11" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D265503621-21072009><FONT size=3D2>A new draft=20
draft-hellstrom-text-turntaking-02.txt&nbsp; is =
available.</FONT></SPAN></DIV>
<DIV><SPAN class=3D265503621-21072009><FONT size=3D2>Since I am not sure =
what group=20
is responsible for approving this kind of registrations, I turn to the =
Dispatch=20
group. </FONT></SPAN></DIV>
<DIV><SPAN class=3D265503621-21072009><FONT size=3D2>In an earlier =
version of this=20
draft, there were more values defined for in detail specifying what =
limitations=20
in simultaneity a network component has. This was regarded too complex, =
so in=20
this version, the indication is limited to two variants: full and=20
restricted.&nbsp; For further info, see the =
document.</FONT></SPAN></DIV>
<DIV><SPAN class=3D265503621-21072009><FONT =
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D265503621-21072009><FONT =
size=3D2>Details:</FONT></SPAN></DIV>
<DIV><FONT face=3DArial><FONT face=3D"Times New Roman"=20
size=3D2></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT face=3D"Times New Roman" size=3D2>Title : =
Registration=20
of the Real-time-text Media Feature Tag</FONT></DIV>
<DIV>
<P><FONT size=3D2>Author(s) : G. Hellstrom, A. Wijk</FONT></P>
<P><FONT size=3D2>Filename : =
draft-hellstrom-text-turntaking-02.txt</FONT></P>
<P><FONT size=3D2>Pages : 6</FONT></P>
<P><FONT size=3D2>Date : 2009-07-12</FONT></P>
<P><FONT size=3D2>This memo defines a new Media Feature Tag =
"real-time-text" for=20
use in SIP registration and session establishment. This is used to =
indicate if a=20
device capable of text communication has full real-time text =
capabilities or=20
limitations in its capabilities requiring the users to apply some =
turn-taking=20
habits.</FONT></P>
<P><FONT size=3D2>A URL for this Internet-Draft is:</FONT></P>
<P><A=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-text-turntaki=
ng-02.txt=20
href=3D"http://www.ietf.org/internet-drafts/draft-hellstrom-text-turntaki=
ng-02.txt"><U=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-text-turntaki=
ng-02.txt><FONT=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-text-turntaki=
ng-02.txt=20
color=3D#0000ff=20
size=3D2>http://www.ietf.org/internet-drafts/draft-hellstrom-text-turntak=
ing-02.txt</U></FONT></A></P></DIV></FONT>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT face=3DArial=20
size=3D2>----------------------------------------------------------------=
---</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Gunnar =
Hellstr=F6m</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Omnitor</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2><A=20
href=3D"mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</=
A></FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Tel: =
+46708204288</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2><A=20
href=3D"http://www.omnitor.se/">www.omnitor.se</A></FONT></DIV>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV></BODY></HTML>

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From AUDET@nortel.com  Tue Jul 21 21:40:49 2009
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References: <4A5261E2.4050506@cisco.com><C677FFD1.48EB%hsinnrei@adobe.com>	<1ECE0EB50388174790F9694F77522CCF1ED91794@zrc2hxm0.corp.nortel.com>	<4A5348F5.5030200@cisco.com>	<1ECE0EB50388174790F9694F77522CCF1ED91D73@zrc2hxm0.corp.nortel.com>	<4A536DF9.4020906@cisco.com>	<1ECE0EB50388174790F9694F77522CCF1ED91E95@zrc2hxm0.corp.nortel.com>	<4A537B66.4000608@cisco.com><4A643F2D.5030300@ericsson.com>	<4A646BAF.7030007@cisco.com><4A6576B9.4020504@ericsson.com>	<0D5F89FAC29E2C41B98A6A762007F5D0022A3425@GBNTHT12009MSX.gb002.siemens.net><4A65D27F.1060604@ericsson.com> <4A661564.9000008@cisco.com>
From: "Francois Audet" <audet@nortel.com>
To: "Anders Kristensen" <ankriste@cisco.com>, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Well, that particular example is already supported in SIP with
the REFER method. This is not an issue.

It's other things, like, "please answer dialog X" or "please
kill dialog Y" that are an issue.=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Anders Kristensen
> Sent: Tuesday, July 21, 2009 12:22
> To: Gonzalo Camarillo
> Cc: Henry Sinnreich; dispatch@ietf.org
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
> I would agree with Paul and John. There's a well established=20
> way of handling multiple media streams in SIP - part of SIP=20
> since the beginning. Don't tamper with that. If someone wants=20
> to disaggregate the streams they should be the ones to pay=20
> for it - by dealing with the added complexity and the changes=20
> to the existing model.
>=20
> Plus I'd be very skeptical about claims that the alternative=20
> is simple.=20
> For example, how do you transfer an audio+video call that is=20
> actually two SIP dialogs? Now all of a sudden there are=20
> additional failure modes because the dialogs have to be=20
> transferred separately and can succeed or fail independently.
>=20
> Existing UAs may already support simultaneous dialogs. If=20
> they're independent calls then that's not at all the same thing.
>=20
> Thanks,
> Anders
>=20
> Gonzalo Camarillo wrote:
> > Hi John,
> >=20
> > many UAs already support multiple SIP dialogs in parallel. Adding a=20
> > header that instruct them to treat the media sessions=20
> related to those=20
> > dialogs as a single one is much easier than anything you have=20
> > described in your email.
> >=20
> > The point is that people are going to implement something like this=20
> > because it is fairly simple and it is needed. We can either=20
> design a=20
> > mechanism that meets their simplicity requirement or=20
> propose complex=20
> > mechanisms that, as usual, will not get implemented.
> >=20
> > Cheers,
> >=20
> > Gonzalo
> >=20
> > Elwell, John wrote:
> >> Gonzalo,
> >>
> >> I have already expressed opinions similar to Paul on=20
> several occasions.
> >> Experience tells us we cannot rely on the remote UA supporting a=20
> >> particular feature. Taking REFER as an example, we tend to end up=20
> >> with a B2BUA terminating a REFER request, because:
> >> - the remote UA doesn't support REFER;
> >> - some UAs don't support REFER, so if the B2BUA sometimes needs to=20
> >> terminate REFER to accommodate those UAs, it might as well always=20
> >> terminate REFER;
> >> - some UAs don't accept REFER as a matter of policy.
> >>
> >> So B2BUAs tend to act as an anchor point and terminate REFER=20
> >> requests, behaving in a 3PCC-like manner.
> >>
> >> If the distributed form of disaggregated media goes ahead, I think=20
> >> that is what will happen in practice, i.e., a B2BUA will=20
> need to act=20
> >> as the anchor point. If UAs wanting to use disaggregated media are=20
> >> behind a B2BUA that concentrates the multiple dialogs into=20
> a single=20
> >> ongoing dialog towards the remote UA, that will work. It=20
> does indeed=20
> >> have the benefit that one local UA can drop out, leaving the other=20
> >> local UA(s) still in communication with the remote UA.=20
> Relying on the=20
> >> remote UA supporting distributed disaggregated media I fear will=20
> >> prevent the capability being used in many cases where it=20
> might be required.
> >>
> >> As part of specifying distributed disaggregated media, we=20
> will need=20
> >> to pay attention to all the interactions that can occur when both=20
> >> sides attempt to use disaggregated media, particular when the two=20
> >> sides simultaneously try to add or remove UAs. This sounds like a=20
> >> difficult problem, but I may be wrong. Having an anchor point=20
> >> (thereby making it centralised, in a way) would probably=20
> ease these problems.
> >>
> >> John
> >>
> >>> -----Original Message-----
> >>> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org]=20
> >>> On Behalf Of Gonzalo Camarillo
> >>> Sent: 21 July 2009 09:05
> >>> To: Paul Kyzivat
> >>> Cc: dispatch@ietf.org; Henry Sinnreich
> >>> Subject: Re: [dispatch] Disaggregated Media in SIP
> >>>
> >>> Hi Paul,
> >>>
> >>> it is good that we agree that some problems need both ends to=20
> >>> collaborate. Now, let's discuss whether or not that is=20
> the case with
> >>> *this* problem.
> >>>
> >>>> I still haven't seen anything showing why this approach
> >>> ends up being
> >>>> "better" than the "centralized" approach. (Note that I
> >>> don't agree with
> >>>> the term "centralized", because it isn't really more
> >>> centralized than
> >>>> the "distributed" approach is. This would become clearer if
> >>> we looked at
> >>>> all the communications necessary to establish a call with
> >>> disaggregated
> >>>> media, and the various topologies of devices that might
> >>> play in such
> >>>> scenarios.)
> >>> well, 3pcc is centralized because you have a (central) controller=20
> >>> that is in control of all SIP signalling. If that=20
> controller leaves=20
> >>> the session, the whole session disappears. An approach=20
> where several=20
> >>> devices have their own SIP dialog with the remote devices is not=20
> >>> centralized because you can remove any of them and the other=20
> >>> sessions will remain unaffected.
> >>>
> >>>> Perhaps we should back off from the signaling mechanisms
> >>> for the moment
> >>>> and consider the use cases of interest in more detail, and
> >>> the various
> >>>> sorts of communications between them that are necessary to
> >>> establish a
> >>>> call that involves them.
> >>> The arguments have been discussed several times. The=20
> thing is that=20
> >>> 3pcc is a complex mechanism that is not supported by most=20
> UAs. You=20
> >>> can argue about its complexity, but the fact that most UAs do not=20
> >>> support 3pcc is a fact.
> >>>
> >>> There are folks that are willing to implement support for a=20
> >>> correlation mechanism for dialogs but would not implement 3pcc=20
> >>> because (they think) it is too complicated.
> >>>
> >>> A second argument is that any device should be able to leave the=20
> >>> session at any point. If you want the 3pcc controller to=20
> leave the=20
> >>> session while the session remains established, you will=20
> need to use=20
> >>> a complex combination of REFER and Replaces that very few=20
> (if any)=20
> >>> UAs support.
> >>>
> >>> Therefore, the requirement here is that the mechanism=20
> should not be=20
> >>> so complicated that implementers do not actually want to=20
> implement=20
> >>> it
> >>> :o) ... unfortunately, as we all know too well, we have a=20
> history of=20
> >>> doing just that.
> >>>
> >>> Cheers,
> >>>
> >>> Gonzalo
> >>>
> >>> _______________________________________________
> >>> 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
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From ankriste@cisco.com  Tue Jul 21 22:23:23 2009
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Francois Audet wrote:
> Well, that particular example is already supported in SIP with
> the REFER method. This is not an issue.

Are you saying that in this example a single REFER will work because the 
triggered INVITE will contain both audio and media streams? Maybe so. 
However, other aspects of transfer flows will be impacted. The Replaces 
header field can refer to just one dialog, for example.

Anders

> 
> It's other things, like, "please answer dialog X" or "please
> kill dialog Y" that are an issue. 
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Anders Kristensen
>> Sent: Tuesday, July 21, 2009 12:22
>> To: Gonzalo Camarillo
>> Cc: Henry Sinnreich; dispatch@ietf.org
>> Subject: Re: [dispatch] Disaggregated Media in SIP
>>
>> I would agree with Paul and John. There's a well established 
>> way of handling multiple media streams in SIP - part of SIP 
>> since the beginning. Don't tamper with that. If someone wants 
>> to disaggregate the streams they should be the ones to pay 
>> for it - by dealing with the added complexity and the changes 
>> to the existing model.
>>
>> Plus I'd be very skeptical about claims that the alternative 
>> is simple. 
>> For example, how do you transfer an audio+video call that is 
>> actually two SIP dialogs? Now all of a sudden there are 
>> additional failure modes because the dialogs have to be 
>> transferred separately and can succeed or fail independently.
>>
>> Existing UAs may already support simultaneous dialogs. If 
>> they're independent calls then that's not at all the same thing.
>>
>> Thanks,
>> Anders
>>
>> Gonzalo Camarillo wrote:
>>> Hi John,
>>>
>>> many UAs already support multiple SIP dialogs in parallel. Adding a 
>>> header that instruct them to treat the media sessions 
>> related to those 
>>> dialogs as a single one is much easier than anything you have 
>>> described in your email.
>>>
>>> The point is that people are going to implement something like this 
>>> because it is fairly simple and it is needed. We can either 
>> design a 
>>> mechanism that meets their simplicity requirement or 
>> propose complex 
>>> mechanisms that, as usual, will not get implemented.
>>>
>>> Cheers,
>>>
>>> Gonzalo
>>>
>>> Elwell, John wrote:
>>>> Gonzalo,
>>>>
>>>> I have already expressed opinions similar to Paul on 
>> several occasions.
>>>> Experience tells us we cannot rely on the remote UA supporting a 
>>>> particular feature. Taking REFER as an example, we tend to end up 
>>>> with a B2BUA terminating a REFER request, because:
>>>> - the remote UA doesn't support REFER;
>>>> - some UAs don't support REFER, so if the B2BUA sometimes needs to 
>>>> terminate REFER to accommodate those UAs, it might as well always 
>>>> terminate REFER;
>>>> - some UAs don't accept REFER as a matter of policy.
>>>>
>>>> So B2BUAs tend to act as an anchor point and terminate REFER 
>>>> requests, behaving in a 3PCC-like manner.
>>>>
>>>> If the distributed form of disaggregated media goes ahead, I think 
>>>> that is what will happen in practice, i.e., a B2BUA will 
>> need to act 
>>>> as the anchor point. If UAs wanting to use disaggregated media are 
>>>> behind a B2BUA that concentrates the multiple dialogs into 
>> a single 
>>>> ongoing dialog towards the remote UA, that will work. It 
>> does indeed 
>>>> have the benefit that one local UA can drop out, leaving the other 
>>>> local UA(s) still in communication with the remote UA. 
>> Relying on the 
>>>> remote UA supporting distributed disaggregated media I fear will 
>>>> prevent the capability being used in many cases where it 
>> might be required.
>>>> As part of specifying distributed disaggregated media, we 
>> will need 
>>>> to pay attention to all the interactions that can occur when both 
>>>> sides attempt to use disaggregated media, particular when the two 
>>>> sides simultaneously try to add or remove UAs. This sounds like a 
>>>> difficult problem, but I may be wrong. Having an anchor point 
>>>> (thereby making it centralised, in a way) would probably 
>> ease these problems.
>>>> John
>>>>
>>>>> -----Original Message-----
>>>>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] 
>>>>> On Behalf Of Gonzalo Camarillo
>>>>> Sent: 21 July 2009 09:05
>>>>> To: Paul Kyzivat
>>>>> Cc: dispatch@ietf.org; Henry Sinnreich
>>>>> Subject: Re: [dispatch] Disaggregated Media in SIP
>>>>>
>>>>> Hi Paul,
>>>>>
>>>>> it is good that we agree that some problems need both ends to 
>>>>> collaborate. Now, let's discuss whether or not that is 
>> the case with
>>>>> *this* problem.
>>>>>
>>>>>> I still haven't seen anything showing why this approach
>>>>> ends up being
>>>>>> "better" than the "centralized" approach. (Note that I
>>>>> don't agree with
>>>>>> the term "centralized", because it isn't really more
>>>>> centralized than
>>>>>> the "distributed" approach is. This would become clearer if
>>>>> we looked at
>>>>>> all the communications necessary to establish a call with
>>>>> disaggregated
>>>>>> media, and the various topologies of devices that might
>>>>> play in such
>>>>>> scenarios.)
>>>>> well, 3pcc is centralized because you have a (central) controller 
>>>>> that is in control of all SIP signalling. If that 
>> controller leaves 
>>>>> the session, the whole session disappears. An approach 
>> where several 
>>>>> devices have their own SIP dialog with the remote devices is not 
>>>>> centralized because you can remove any of them and the other 
>>>>> sessions will remain unaffected.
>>>>>
>>>>>> Perhaps we should back off from the signaling mechanisms
>>>>> for the moment
>>>>>> and consider the use cases of interest in more detail, and
>>>>> the various
>>>>>> sorts of communications between them that are necessary to
>>>>> establish a
>>>>>> call that involves them.
>>>>> The arguments have been discussed several times. The 
>> thing is that 
>>>>> 3pcc is a complex mechanism that is not supported by most 
>> UAs. You 
>>>>> can argue about its complexity, but the fact that most UAs do not 
>>>>> support 3pcc is a fact.
>>>>>
>>>>> There are folks that are willing to implement support for a 
>>>>> correlation mechanism for dialogs but would not implement 3pcc 
>>>>> because (they think) it is too complicated.
>>>>>
>>>>> A second argument is that any device should be able to leave the 
>>>>> session at any point. If you want the 3pcc controller to 
>> leave the 
>>>>> session while the session remains established, you will 
>> need to use 
>>>>> a complex combination of REFER and Replaces that very few 
>> (if any) 
>>>>> UAs support.
>>>>>
>>>>> Therefore, the requirement here is that the mechanism 
>> should not be 
>>>>> so complicated that implementers do not actually want to 
>> implement 
>>>>> it
>>>>> :o) ... unfortunately, as we all know too well, we have a 
>> history of 
>>>>> doing just that.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Gonzalo
>>>>>
>>>>> _______________________________________________
>>>>> 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 gunnar.hellstrom@omnitor.se  Tue Jul 21 22:35:58 2009
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Subject: [dispatch] Presentation aspects of real-time text sessions
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A new draft on presentation of real-time text is available at
 =
<http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-06.txt>
http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-06.txt

It specifies details on how sessions with real-time text can be =
presented,
and how presentation protocol items can be used to control the =
presentation.
It also introduces some important rules to follow in order to maintain =
the
same contents of the conversation by the participants while they very =
well
may have selected different presentation layouts.

New in this version are details on how far back erasure actions are =
allowed,
and a new indicator differing soft-return and hard-return. There is also =
an
effort to align it with RFC5198 "Unicode Format for Network Interchange"

I have not found a suitable home for discussion of this draft in IETF, =
and
is therefore turning to Dispatch for advice.

Info on the draft:

-------------------------------------------------------------------------=
---
--------------=20

Title : Presentation of Text Conversation in realtime and en-bloc form

Author(s) : G. Hellstrom, et al.

Filename : draft-hellstrom-textpreview-06.txt

Pages : 17

Date : 2009-07-11

This specification defines methods for presentation of a text =
conversation
with focus on the real-time features. The aim is to give the =
participants in
a conversation a good opportunity to perceive the real-time flow of the
conversation and also provide a display of the history of the =
conversation
that makes it easy to read. Both two-party and multi-party situations =
are
defined.

A URL for this Internet-Draft is:

 =
<http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-06.txt>
http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-06.txt

-------------------------------------------------------------------
Gunnar Hellstr=F6m
Omnitor
gunnar.hellstrom@omnitor.se
Tel: +46708204288
www.omnitor.se <http://www.omnitor.se/>=20
=20

------=_NextPart_000_010D_01CA0A9B.DFA101C0
Content-Type: text/html;
	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5730.11" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D265095704-22072009><FONT face=3DArial>A new draft on =
presentation=20
of real-time text is available at</FONT></SPAN></DIV>
<DIV><SPAN class=3D265095704-22072009>
<P><A=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-0=
6.txt=20
href=3D"http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-0=
6.txt"><FONT=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-0=
6.txt=20
face=3DArial color=3D#0000ff=20
size=3D2>http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-=
06.txt</FONT></A></P>
<P><SPAN class=3D265095704-22072009><FONT face=3DArial>It&nbsp;specifies =
details on=20
how sessions with real-time text can be presented, and how presentation =
protocol=20
items can be used to control the presentation. It also introduces some =
important=20
rules to follow in order to maintain&nbsp;the same&nbsp;contents of the=20
conversation by the participants while&nbsp;they very well may have =
selected=20
different presentation layouts.</FONT></SPAN></P>
<P><SPAN class=3D265095704-22072009><FONT face=3DArial>New in this =
version are=20
details on how far back erasure actions are allowed, and a new indicator =

differing soft-return and hard-return. There is also an effort to align =
it with=20
RFC5198 "Unicode Format for Network Interchange"</FONT></SPAN></P>
<P><SPAN class=3D265095704-22072009><FONT face=3DArial>I have not found =
a suitable=20
home for&nbsp;discussion of this draft in IETF, and is therefore turning =
to=20
Dispatch for advice.</FONT></SPAN></P>
<P><SPAN class=3D265095704-22072009><FONT face=3DArial>Info on the=20
draft:</FONT></SPAN></P>
<P><SPAN class=3D265095704-22072009><FONT=20
face=3DArial>------------------------------------------------------------=
------------------------------</FONT>&nbsp;</SPAN></P>
<P></SPAN><FONT size=3D2>Title : Presentation of Text Conversation in =
realtime and=20
en-bloc form</FONT></P></DIV>
<DIV>
<P><FONT size=3D2>Author(s) : G. Hellstrom, et al.</FONT></P>
<P><FONT size=3D2>Filename : =
draft-hellstrom-textpreview-06.txt</FONT></P>
<P><FONT size=3D2>Pages : 17</FONT></P>
<P><FONT size=3D2>Date : 2009-07-11</FONT></P>
<P><FONT size=3D2>This specification defines methods for presentation of =
a text=20
conversation with focus on the real-time features. The aim is to give =
the=20
participants in a conversation a good opportunity to perceive the =
real-time flow=20
of the conversation and also provide a display of the history of the=20
conversation that makes it easy to read. Both two-party and multi-party=20
situations are defined.</FONT></P>
<P><FONT size=3D2>A URL for this Internet-Draft is:</FONT></P>
<P><A=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-0=
6.txt=20
href=3D"http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-0=
6.txt"><FONT=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-0=
6.txt=20
color=3D#0000ff=20
size=3D2>http://www.ietf.org/internet-drafts/draft-hellstrom-textpreview-=
06.txt</FONT></A></P></DIV>
<DIV><FONT face=3DArial=20
size=3D2>----------------------------------------------------------------=
---</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Gunnar =
Hellstr=F6m</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Omnitor</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2><A=20
href=3D"mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</=
A></FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Tel: =
+46708204288</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2><A=20
href=3D"http://www.omnitor.se/">www.omnitor.se</A></FONT></DIV>
<DIV><FONT face=3DArial></FONT><FONT =
face=3DArial></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_010D_01CA0A9B.DFA101C0--


From gonzalo.camarillo@ericsson.com  Tue Jul 21 22:59:30 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Anders Kristensen <ankriste@cisco.com>
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi,

> I would agree with Paul and John. There's a well established way of 
> handling multiple media streams in SIP - part of SIP since the 
> beginning. Don't tamper with that. 

yes, it exists in an imaginary world that some IETFers live in. If you 
check in the real world, no UAs support that type of functionality. If 
we IETF specs to become irrelevant, continuing living in such imaginary 
world and ignoring reality is the best way forward.

> Plus I'd be very skeptical about claims that the alternative is simple. 
> For example, how do you transfer an audio+video call that is actually 
> two SIP dialogs? Now all of a sudden there are additional failure modes 
> because the dialogs have to be transferred separately and can succeed or 
> fail independently.

We have mechanisms to deal with transfers of a single dialog. That is 
what you would do; exactly the same as in the 3pcc model: you transfer 
each dialog to its new location.

> Existing UAs may already support simultaneous dialogs. If they're 
> independent calls then that's not at all the same thing.

That is exactly my point. Existing UAs *already* support multiple 
dialogs. Presenting the video stream established by one dialog and the 
IM stream established by the other one as if they were a single session 
is *very* simple and, at the protocol level, it can also be implemented 
with a simple mechanism. All the alternatives given (e.g., 3pcc) are not 
currently supported and are way more complex.

Cheers,

Gonzalo


From AUDET@nortel.com  Tue Jul 21 23:13:31 2009
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From: "Francois Audet" <audet@nortel.com>
To: "Anders Kristensen" <ankriste@cisco.com>
Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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That is correct. I don't see separating audio and video for a single =
call
to be desireable (for example).

In my mind, the term "disaggregated media" is a misnomer. It's really
"loosely coupled control" that is useful.


> -----Original Message-----
> From: Anders Kristensen [mailto:ankriste@cisco.com]=20
> Sent: Tuesday, July 21, 2009 22:23
> To: Audet, Francois (SC100:3055)
> Cc: Gonzalo Camarillo; Henry Sinnreich; dispatch@ietf.org
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
>=20
>=20
> Francois Audet wrote:
> > Well, that particular example is already supported in SIP with the=20
> > REFER method. This is not an issue.
>=20
> Are you saying that in this example a single REFER will work=20
> because the triggered INVITE will contain both audio and=20
> media streams? Maybe so.=20
> However, other aspects of transfer flows will be impacted.=20
> The Replaces header field can refer to just one dialog, for example.
>=20
> Anders
>=20
> >=20
> > It's other things, like, "please answer dialog X" or "please kill=20
> > dialog Y" that are an issue.
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Anders Kristensen
> >> Sent: Tuesday, July 21, 2009 12:22
> >> To: Gonzalo Camarillo
> >> Cc: Henry Sinnreich; dispatch@ietf.org
> >> Subject: Re: [dispatch] Disaggregated Media in SIP
> >>
> >> I would agree with Paul and John. There's a well=20
> established way of=20
> >> handling multiple media streams in SIP - part of SIP since the=20
> >> beginning. Don't tamper with that. If someone wants to=20
> disaggregate=20
> >> the streams they should be the ones to pay for it - by=20
> dealing with=20
> >> the added complexity and the changes to the existing model.
> >>
> >> Plus I'd be very skeptical about claims that the alternative is=20
> >> simple.
> >> For example, how do you transfer an audio+video call that=20
> is actually=20
> >> two SIP dialogs? Now all of a sudden there are additional failure=20
> >> modes because the dialogs have to be transferred=20
> separately and can=20
> >> succeed or fail independently.
> >>
> >> Existing UAs may already support simultaneous dialogs. If they're=20
> >> independent calls then that's not at all the same thing.
> >>
> >> Thanks,
> >> Anders
> >>
> >> Gonzalo Camarillo wrote:
> >>> Hi John,
> >>>
> >>> many UAs already support multiple SIP dialogs in=20
> parallel. Adding a=20
> >>> header that instruct them to treat the media sessions
> >> related to those
> >>> dialogs as a single one is much easier than anything you have=20
> >>> described in your email.
> >>>
> >>> The point is that people are going to implement something=20
> like this=20
> >>> because it is fairly simple and it is needed. We can either
> >> design a
> >>> mechanism that meets their simplicity requirement or
> >> propose complex
> >>> mechanisms that, as usual, will not get implemented.
> >>>
> >>> Cheers,
> >>>
> >>> Gonzalo
> >>>
> >>> Elwell, John wrote:
> >>>> Gonzalo,
> >>>>
> >>>> I have already expressed opinions similar to Paul on
> >> several occasions.
> >>>> Experience tells us we cannot rely on the remote UA supporting a=20
> >>>> particular feature. Taking REFER as an example, we tend=20
> to end up=20
> >>>> with a B2BUA terminating a REFER request, because:
> >>>> - the remote UA doesn't support REFER;
> >>>> - some UAs don't support REFER, so if the B2BUA=20
> sometimes needs to=20
> >>>> terminate REFER to accommodate those UAs, it might as=20
> well always=20
> >>>> terminate REFER;
> >>>> - some UAs don't accept REFER as a matter of policy.
> >>>>
> >>>> So B2BUAs tend to act as an anchor point and terminate REFER=20
> >>>> requests, behaving in a 3PCC-like manner.
> >>>>
> >>>> If the distributed form of disaggregated media goes=20
> ahead, I think=20
> >>>> that is what will happen in practice, i.e., a B2BUA will
> >> need to act
> >>>> as the anchor point. If UAs wanting to use disaggregated=20
> media are=20
> >>>> behind a B2BUA that concentrates the multiple dialogs into
> >> a single
> >>>> ongoing dialog towards the remote UA, that will work. It
> >> does indeed
> >>>> have the benefit that one local UA can drop out, leaving=20
> the other=20
> >>>> local UA(s) still in communication with the remote UA.
> >> Relying on the
> >>>> remote UA supporting distributed disaggregated media I fear will=20
> >>>> prevent the capability being used in many cases where it
> >> might be required.
> >>>> As part of specifying distributed disaggregated media, we
> >> will need
> >>>> to pay attention to all the interactions that can occur=20
> when both=20
> >>>> sides attempt to use disaggregated media, particular=20
> when the two=20
> >>>> sides simultaneously try to add or remove UAs. This=20
> sounds like a=20
> >>>> difficult problem, but I may be wrong. Having an anchor point=20
> >>>> (thereby making it centralised, in a way) would probably
> >> ease these problems.
> >>>> John
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org]
> >>>>> On Behalf Of Gonzalo Camarillo
> >>>>> Sent: 21 July 2009 09:05
> >>>>> To: Paul Kyzivat
> >>>>> Cc: dispatch@ietf.org; Henry Sinnreich
> >>>>> Subject: Re: [dispatch] Disaggregated Media in SIP
> >>>>>
> >>>>> Hi Paul,
> >>>>>
> >>>>> it is good that we agree that some problems need both ends to=20
> >>>>> collaborate. Now, let's discuss whether or not that is
> >> the case with
> >>>>> *this* problem.
> >>>>>
> >>>>>> I still haven't seen anything showing why this approach
> >>>>> ends up being
> >>>>>> "better" than the "centralized" approach. (Note that I
> >>>>> don't agree with
> >>>>>> the term "centralized", because it isn't really more
> >>>>> centralized than
> >>>>>> the "distributed" approach is. This would become clearer if
> >>>>> we looked at
> >>>>>> all the communications necessary to establish a call with
> >>>>> disaggregated
> >>>>>> media, and the various topologies of devices that might
> >>>>> play in such
> >>>>>> scenarios.)
> >>>>> well, 3pcc is centralized because you have a (central)=20
> controller=20
> >>>>> that is in control of all SIP signalling. If that
> >> controller leaves
> >>>>> the session, the whole session disappears. An approach
> >> where several
> >>>>> devices have their own SIP dialog with the remote=20
> devices is not=20
> >>>>> centralized because you can remove any of them and the other=20
> >>>>> sessions will remain unaffected.
> >>>>>
> >>>>>> Perhaps we should back off from the signaling mechanisms
> >>>>> for the moment
> >>>>>> and consider the use cases of interest in more detail, and
> >>>>> the various
> >>>>>> sorts of communications between them that are necessary to
> >>>>> establish a
> >>>>>> call that involves them.
> >>>>> The arguments have been discussed several times. The
> >> thing is that
> >>>>> 3pcc is a complex mechanism that is not supported by most
> >> UAs. You
> >>>>> can argue about its complexity, but the fact that most=20
> UAs do not=20
> >>>>> support 3pcc is a fact.
> >>>>>
> >>>>> There are folks that are willing to implement support for a=20
> >>>>> correlation mechanism for dialogs but would not implement 3pcc=20
> >>>>> because (they think) it is too complicated.
> >>>>>
> >>>>> A second argument is that any device should be able to=20
> leave the=20
> >>>>> session at any point. If you want the 3pcc controller to
> >> leave the
> >>>>> session while the session remains established, you will
> >> need to use
> >>>>> a complex combination of REFER and Replaces that very few
> >> (if any)
> >>>>> UAs support.
> >>>>>
> >>>>> Therefore, the requirement here is that the mechanism
> >> should not be
> >>>>> so complicated that implementers do not actually want to
> >> implement
> >>>>> it
> >>>>> :o) ... unfortunately, as we all know too well, we have a
> >> history of
> >>>>> doing just that.
> >>>>>
> >>>>> Cheers,
> >>>>>
> >>>>> Gonzalo
> >>>>>
> >>>>> _______________________________________________
> >>>>> 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

From gonzalo.camarillo@ericsson.com  Tue Jul 21 23:39:17 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Francois Audet <audet@nortel.com>
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Cc: Anders Kristensen <ankriste@cisco.com>, Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi,

> That is correct. I don't see separating audio and video for a single call
> to be desireable (for example).

exactly. Regular UAs would continue establishing multi-stream sessions 
as usual (i.e., using a single dialog). You would only have multiple 
dialogs when multiple SIP-enabled devices are involved in the communication.

> In my mind, the term "disaggregated media" is a misnomer. It's really
> "loosely coupled control" that is useful.

yes, that would be a valid term as well.

Cheers,

Gonzalo


From michael@voip.co.uk  Wed Jul 22 01:13:19 2009
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Date: Wed, 22 Jul 2009 09:10:14 +0100
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From: Michael Procter <michael@voip.co.uk>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
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Hi John,

2009/6/29 Elwell, John <john.elwell@siemens-enterprise.com>:
> http://www.ietf.org/internet-drafts/draft-elwell-dispatch-identity-reqs-00.txt
>
> We would appreciate feedback as to whether this is helpful and going in the right direction, as well as detailed comments.

I think you have identified some useful fundamental properties of
caller/callee id in your requirements.  My only query is about Req 6:
"meeting similar requirements to those above" seems a little
ambiguous.  Did you intend for the calling user to receive the same
level of assurance concerning the providence of the callee ID, or
would you expect a lower level to be sufficient?

Best regards,

Michael

From gunnar.hellstrom@omnitor.se  Wed Jul 22 01:24:13 2009
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Subject: [dispatch] Real-time text interworking between PSTN and IP networks
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A new draft is available at
 <http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt>
http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt

When a feature is offered in one network, it is important to offer
interoperability with other networks.=20

Even if real-time text in IP networks overshadow what was avialable in =
PSTN
in terms of functionality, it is still possible to arrange =
interoperability
between real-time text in IP networks with the different forms of text
telephony that are available in PSTN networks ( e.g. TTY, EDT, V.21 =
based,
DTMF based etc ).  Issues on many levels appear when such =
interoperability
shall be arranged. e.g. on routing, media negotiating, indications to =
the
users, handling the limitations in media simultaneity in PSTN in =
contrast
with the full simultaneity possible in IP etc.=20

The draft gives guidance on many of these topics and can be seen as an
expansion on this topic from what was specified in RFC 5194..

In my view, it is a typical SIPPING topic, so therefore I turn to =
Dispatch
with the question of where to discuss it.

-------------------------------------------------------------------------=
---
--------------------- =20

Title : Real-time text interworking between PSTN and IP networks

Author(s) : G. Hellstrom, et al.

Filename : draft-hellstrom-txtgwy-01.txt

Pages : 16

Date : 2009-07-13

IP networks can support real-time text communication. SIP-based

real- time text is called Text-over-IP or ToIP. PSTN networks support
real-time text using textphones (or TTYs). When real-time text is =
supported
by different networks, gateways are needed to provide interoperability.
Real-time text capable gateways may also support real-time voice.

This specification describes procedures for interworking between ToIP =
and
PSTN textphones using a real-time text capable gateway (RTT gateway). It
also describes ways to route calls to RTT gateways for several call
scenarios.

Procedures that support the phased introduction of RTT gateways and
procedures that support the invocation of text channels at any time =
during
the call are included. Interworking of PSTN textphones that do not =
support
simultaneity of voice and text with IP User Agents that support =
simultaneous
voice and text is also described.

A URL for this Internet-Draft is:

 <http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt>
http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt

=20
-------------------------------------------------------------------
Gunnar Hellstr=F6m
Omnitor
gunnar.hellstrom@omnitor.se
Tel: +46708204288
www.omnitor.se <http://www.omnitor.se/>=20
=20

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<BODY>
<DIV><SPAN class=3D421591405-22072009><FONT face=3DArial>A new draft is =
available=20
at</FONT></SPAN></DIV>
<DIV><SPAN class=3D421591405-22072009>
<P><A =
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
=20
href=3D"http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
"><U=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
><FONT=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
=20
color=3D#0000ff><FONT=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
=20
face=3DArial>http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-0=
1.txt</FONT></U></FONT></A></P>
<P><SPAN class=3D421591405-22072009><FONT face=3DArial>When a feature is =
offered in=20
one network, it is important to offer interoperability with other =
networks.=20
</FONT></SPAN></P>
<P><SPAN class=3D421591405-22072009><FONT face=3DArial>Even if real-time =
text in IP=20
networks overshadow what was avialable in PSTN in terms of =
functionality, it is=20
still possible to arrange interoperability between real-time text in IP =
networks=20
with the different forms of text telephony that are available&nbsp;in =
PSTN=20
networks ( e.g. TTY, EDT, V.21 based, DTMF based&nbsp;etc =
).&nbsp;&nbsp;Issues=20
on many levels appear&nbsp;when such interoperability shall be arranged. =
e.g. on=20
routing, media negotiating, indications to the users, handling the =
limitations=20
in media simultaneity in PSTN in contrast with the full simultaneity =
possible in=20
IP etc.&nbsp;</FONT></SPAN></P>
<P><SPAN class=3D421591405-22072009><FONT face=3DArial>The draft gives =
guidance on=20
many of these topics and can be seen as an expansion on this topic from =
what was=20
specified in RFC 5194..</FONT></SPAN></P>
<P><SPAN class=3D421591405-22072009><FONT face=3DArial>In my view, it is =
a typical=20
SIPPING topic, so therefore I turn to Dispatch with the question of =
where to=20
discuss it.</FONT></SPAN></P>
<P><SPAN class=3D421591405-22072009><FONT=20
face=3DArial>------------------------------------------------------------=
-------------------------------------&nbsp;&nbsp;</FONT></SPAN></SPAN></P=
></DIV>
<DIV><FONT face=3DArial>Title : Real-time text interworking between PSTN =
and IP=20
networks</FONT></DIV>
<DIV>
<DIV>
<P><FONT face=3DArial>Author(s) : G. Hellstrom, et al.</FONT></P>
<P><FONT face=3DArial>Filename : =
draft-hellstrom-txtgwy-01.txt</FONT></P>
<P><FONT face=3DArial>Pages : 16</FONT></P>
<P><FONT face=3DArial>Date : 2009-07-13</FONT></P>
<P><FONT face=3DArial>IP networks can support real-time text =
communication.=20
SIP-based</FONT></P>
<P><FONT face=3DArial>real- time text is called Text-over-IP or ToIP. =
PSTN=20
networks support real-time text using textphones (or TTYs). When =
real-time text=20
is supported by different networks, gateways are needed to provide=20
interoperability. Real-time text capable gateways may also support =
real-time=20
voice.</FONT></P>
<P><FONT face=3DArial>This specification describes procedures for =
interworking=20
between ToIP and PSTN textphones using a real-time text capable gateway =
(RTT=20
gateway). It also describes ways to route calls to RTT gateways for =
several call=20
scenarios.</FONT></P>
<P><FONT face=3DArial>Procedures that support the phased introduction of =
RTT=20
gateways and procedures that support the invocation of text channels at =
any time=20
during the call are included. Interworking of PSTN textphones that do =
not=20
support simultaneity of voice and text with IP User Agents that support=20
simultaneous voice and text is also described.</FONT></P>
<P><FONT face=3DArial>A URL for this Internet-Draft is:</FONT></P>
<P><A =
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
=20
href=3D"http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
"><U=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
><FONT=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
=20
color=3D#0000ff><FONT=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-01.txt=
=20
face=3DArial>http://www.ietf.org/internet-drafts/draft-hellstrom-txtgwy-0=
1.txt</FONT></U></FONT></A></P></DIV></DIV>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT face=3DArial=20
size=3D2>----------------------------------------------------------------=
---</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Gunnar =
Hellstr=F6m</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Omnitor</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2><A=20
href=3D"mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</=
A></FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Tel: =
+46708204288</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2><A=20
href=3D"http://www.omnitor.se/">www.omnitor.se</A></FONT></DIV>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV></BODY></HTML>

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From john.elwell@siemens-enterprise.com  Wed Jul 22 01:35:22 2009
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Date: Wed, 22 Jul 2009 09:31:43 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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To: Michael Procter <michael@voip.co.uk>
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Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
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Michael,

Thanks for your comment. The intention was indeed to have the same level
of assurance. I am not sure how to reword it (without reproducing all
the previous requirements and adapting them to the connected identity
case).

John


> -----Original Message-----
> From: Michael Procter [mailto:michael@voip.co.uk]=20
> Sent: 22 July 2009 09:10
> To: Elwell, John
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
>=20
> Hi John,
>=20
> 2009/6/29 Elwell, John <john.elwell@siemens-enterprise.com>:
> >=20
> http://www.ietf.org/internet-drafts/draft-elwell-dispatch-iden
> tity-reqs-00.txt
> >
> > We would appreciate feedback as to whether this is helpful=20
> and going in the right direction, as well as detailed comments.
>=20
> I think you have identified some useful fundamental properties of
> caller/callee id in your requirements.  My only query is about Req 6:
> "meeting similar requirements to those above" seems a little
> ambiguous.  Did you intend for the calling user to receive the same
> level of assurance concerning the providence of the callee ID, or
> would you expect a lower level to be sufficient?
>=20
> Best regards,
>=20
> Michael
>=20

From michael@voip.co.uk  Wed Jul 22 02:05:25 2009
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2009/7/22 Elwell, John <john.elwell@siemens-enterprise.com>:
> Michael,
>
> Thanks for your comment. The intention was indeed to have the same level
> of assurance. I am not sure how to reword it (without reproducing all
> the previous requirements and adapting them to the connected identity
> case).

Ah.  I see your problem.  Reproducing them is certainly one option,
one that at least is explicit!

How about:

   REQ6 - It MUST be possible for a calling user to receive connected
   user identification meeting corresponding requirements to those above for
   caller identification.

But it may be just me reading more flexibility into 'similar' than I should.

Michael

From john.elwell@siemens-enterprise.com  Wed Jul 22 02:31:52 2009
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Date: Wed, 22 Jul 2009 10:30:21 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
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OK, if there is interest in a further rev of this, I will make that
change.

John=20

> -----Original Message-----
> From: Michael Procter [mailto:michael@voip.co.uk]=20
> Sent: 22 July 2009 10:05
> To: Elwell, John
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
>=20
> 2009/7/22 Elwell, John <john.elwell@siemens-enterprise.com>:
> > Michael,
> >
> > Thanks for your comment. The intention was indeed to have=20
> the same level
> > of assurance. I am not sure how to reword it (without=20
> reproducing all
> > the previous requirements and adapting them to the=20
> connected identity
> > case).
>=20
> Ah.  I see your problem.  Reproducing them is certainly one option,
> one that at least is explicit!
>=20
> How about:
>=20
>    REQ6 - It MUST be possible for a calling user to receive connected
>    user identification meeting corresponding requirements to=20
> those above for
>    caller identification.
>=20
> But it may be just me reading more flexibility into 'similar'=20
> than I should.
>=20
> Michael
>=20

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Subject: Re: [dispatch] Disaggregated Media in SIP
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Gonzalo,

I haven't seen any response to Anders' issue about what happens if both 
ends want to use disaggregated media.

Nor have I seen any explanation of where the control resides that 
initiates and coordinates the inclusion of multiple media sources into 
call. *Something* must be capable of saying "I want device X to join 
call Y using media Z". That is still a centralized point of control even 
if it isn't directly in the resulting sip signaling flow. Either it is 
using some exotic form of REFER to cause this to happen, or else it is 
using some kind of private signaling.

Also, if a multimedia call is received by a UAS, and the UAS needs to 
use disaggregated media to cope with the offered media types, how will 
that work? Will some of the media be refused initially, and then offered 
back in a new INVITE going in the reverse direction?

There are a *lot* of details here that need to be worked out for this 
approach to be functional. OTOH, if the "centralized" approach is used 
then it all devolves to use of stuff we already have.

	Thanks,
	Paul

Gonzalo Camarillo wrote:
> Hi,
> 
>> That is correct. I don't see separating audio and video for a single call
>> to be desireable (for example).
> 
> exactly. Regular UAs would continue establishing multi-stream sessions 
> as usual (i.e., using a single dialog). You would only have multiple 
> dialogs when multiple SIP-enabled devices are involved in the 
> communication.
> 
>> In my mind, the term "disaggregated media" is a misnomer. It's really
>> "loosely coupled control" that is useful.
> 
> yes, that would be a valid term as well.
> 
> Cheers,
> 
> Gonzalo
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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Subject: Re: [dispatch] Disaggregated Media in SIP
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Gonzalo Camarillo wrote:
> Hi,
> 
>> I would agree with Paul and John. There's a well established way of 
>> handling multiple media streams in SIP - part of SIP since the 
>> beginning. Don't tamper with that. 
> 
> yes, it exists in an imaginary world that some IETFers live in. If you 
> check in the real world, no UAs support that type of functionality. If 
> we IETF specs to become irrelevant, continuing living in such imaginary 
> world and ignoring reality is the best way forward.

I had this functionality working in about 2002. We didn't productize it, 
but it was functional.

	Thanks,
	Paul

>> Plus I'd be very skeptical about claims that the alternative is 
>> simple. For example, how do you transfer an audio+video call that is 
>> actually two SIP dialogs? Now all of a sudden there are additional 
>> failure modes because the dialogs have to be transferred separately 
>> and can succeed or fail independently.
> 
> We have mechanisms to deal with transfers of a single dialog. That is 
> what you would do; exactly the same as in the 3pcc model: you transfer 
> each dialog to its new location.
> 
>> Existing UAs may already support simultaneous dialogs. If they're 
>> independent calls then that's not at all the same thing.
> 
> That is exactly my point. Existing UAs *already* support multiple 
> dialogs. Presenting the video stream established by one dialog and the 
> IM stream established by the other one as if they were a single session 
> is *very* simple and, at the protocol level, it can also be implemented 
> with a simple mechanism. All the alternatives given (e.g., 3pcc) are not 
> currently supported and are way more complex.
> 
> Cheers,
> 
> Gonzalo
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
Cc: Anders Kristensen <ankriste@cisco.com>, Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Wednesday, July 22, 2009 07:09
> To: Gonzalo Camarillo
> Cc: Audet, Francois (SC100:3055); Anders Kristensen; Henry=20
> Sinnreich; dispatch@ietf.org
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
> Gonzalo,
>=20
> I haven't seen any response to Anders' issue about what=20
> happens if both ends want to use disaggregated media.
>=20
> Nor have I seen any explanation of where the control resides=20
> that initiates and coordinates the inclusion of multiple=20
> media sources into call. *Something* must be capable of=20
> saying "I want device X to join call Y using media Z". That=20
> is still a centralized point of control even if it isn't=20
> directly in the resulting sip signaling flow. Either it is=20
> using some exotic form of REFER to cause this to happen, or=20
> else it is using some kind of private signaling.

I don't think we need/want the "...using media Z" part.

I truely believe that this makes it too complicated.

Again, loose coupling.

From gonzalo.camarillo@ericsson.com  Wed Jul 22 08:56:37 2009
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Hi Paul,

> I had this functionality working in about 2002. We didn't productize it, 
> but it was functional.

yes, I also have a lot of proof-of-concept prototypes in my lab that 
implement stuff that never makes it into a product for different reasons 
;o)... as a matter of fact, I can have a student implement a header that 
correlates two dialogs on top of a UA that already supports multiple 
dialogs in no time.

However, this is not what we are discussing here. We are discussing 
specifications that will see actual deployment.

Cheers,

Gonzalo


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Subject: Re: [dispatch] Disaggregated Media in SIP
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Francois Audet wrote:
>  
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Wednesday, July 22, 2009 07:09
>> To: Gonzalo Camarillo
>> Cc: Audet, Francois (SC100:3055); Anders Kristensen; Henry 
>> Sinnreich; dispatch@ietf.org
>> Subject: Re: [dispatch] Disaggregated Media in SIP
>>
>> Gonzalo,
>>
>> I haven't seen any response to Anders' issue about what 
>> happens if both ends want to use disaggregated media.
>>
>> Nor have I seen any explanation of where the control resides 
>> that initiates and coordinates the inclusion of multiple 
>> media sources into call. *Something* must be capable of 
>> saying "I want device X to join call Y using media Z". That 
>> is still a centralized point of control even if it isn't 
>> directly in the resulting sip signaling flow. Either it is 
>> using some exotic form of REFER to cause this to happen, or 
>> else it is using some kind of private signaling.
> 
> I don't think we need/want the "...using media Z" part.

Well, as I understand them, most or all of the use cases in the draft 
would require some sort of instruction to the secondary device to tell 
it what media to use, because it would otherwise use media that the use 
case assumes it will not use.

> I truely believe that this makes it too complicated.

All the more reason to ensure we agree on a goal set of use cases so we 
can be sure there is a mechanism sufficiently rich to support them 
without being unnecessarily complicated. Its no good to do something 
simple if it doesn't solve a problem.

	Thanks,
	Paul

> Again, loose coupling.
> 

From alan@sipstation.com  Wed Jul 22 10:07:39 2009
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Subject: [dispatch] Proposed Charter for CCUS - Call Control UUI for SIP
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For discussion during DISPATCH next week.  Comments most welcome.

Thanks,
Alan

- - - - - -

The Call Control UUI for SIP  (CCUS) working group is chartered to 
define a SIP mechanism for transporting call control related 
user-to-user (UUI) information between User Agents.

The ISDN User to User Information Service, defined by ITU-T Q.931, is 
extensively deployed in the PSTN today, supporting such applications as 
contact centers, call centers, and automatic call distributors (ACDs).  
A major barrier to the movement of these applications to SIP is the lack 
of a standard mechanism to transport this information in SIP.

Call control UUI is user information conveyed between user agents during 
call control operations.  As a result, the information must be conveyed 
with the INVITE transaction, and must survive proxy retargeting, 
redirection, and transfers.  The mechanism must utilize a minimum of SIP 
extensions as it will need to be supported by many simple SIP user 
agents such as PSTN gateways.  The mechanism must interwork with the 
existing ISDN service but should also be extensible for use by other 
applications and non-ISDN protocols.

While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control UUI 
can be generated by a number of protocols that are not ISUP, and as such 
it is architecturally unreasonable to interwork these protocols at the 
SIP / ISUP gateway. That is, existing SIP-T approaches defined in 
RFC3372 do not meet the requirements.

Mechanisms based on the SIP INFO method, RFC2976, will not be considered 
by the working group since the UUI contents carry information that must 
be conveyed during session setup at the user agent - the information 
must be conveyed with the INVITE transaction.  The information must be 
passed with the session setup request (INVITE), responses to that 
INVITE, or session termination requests.  As a result, it is not 
possible to use INFO in these cases.

The group will produce:

- A problem statement and requirements document for implementing a SIP 
call control UUI mechanism

- A specification of the SIP extension to best meet those requirements.

Goals and Milestones
====================

Dec 09 - Problem statement and requirements document to IESG 
(Informational)
Mar 10 - SIP call control UUI specification to IESG (PS)


From AUDET@nortel.com  Wed Jul 22 10:34:37 2009
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Thread-Topic: [dispatch] Disaggregated Media in SIP
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References: <4A5261E2.4050506@cisco.com><C677FFD1.48EB%hsinnrei@adobe.com>	<1ECE0EB50388174790F9694F77522CCF1ED91794@zrc2hxm0.corp.nortel.com>	<4A5348F5.5030200@cisco.com>	<1ECE0EB50388174790F9694F77522CCF1ED91D73@zrc2hxm0.corp.nortel.com>	<4A536DF9.4020906@cisco.com>	<1ECE0EB50388174790F9694F77522CCF1ED91E95@zrc2hxm0.corp.nortel.com>	<4A537B66.4000608@cisco.com><4A643F2D.5030300@ericsson.com>	<4A646BAF.7030007@cisco.com><4A6576B9.4020504@ericsson.com>	<0D5F89FAC29E2C41B98A6A762007F5D0022A3425@GBNTHT12009MSX.gb002.siemens.net><4A65D27F.1060604@ericsson.com>	<4A661564.9000008@cisco.com>	<1ECE0EB50388174790F9694F77522CCF1F1A37AE@zrc2hxm0.corp.nortel.com>	<4A66A239.4090806@cisco.com>	<1ECE0EB50388174790F9694F77522CCF1F1A37DC@zrc2hxm0.corp.nortel.com> <4A66AFF3.7040500@ericsson.com> <4A671D75.6000907@cisco.com> <1ECE0EB50388174790F9694F77522CCF1F1A3B82@zrc2hxm0.corp.nortel.com> <4A674376.8020106@cisco.com>
From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: Anders Kristensen <ankriste@cisco.com>, Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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The more I hear from the mailing list, the more I think we=20
might have more than one goal.

So we may need multiple solutions.=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Wednesday, July 22, 2009 09:51
> To: Audet, Francois (SC100:3055)
> Cc: Gonzalo Camarillo; Anders Kristensen; Henry Sinnreich;=20
> dispatch@ietf.org
> Subject: Re: [dispatch] Disaggregated Media in SIP
>=20
>=20
>=20
> Francois Audet wrote:
> > =20
> >=20
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Wednesday, July 22, 2009 07:09
> >> To: Gonzalo Camarillo
> >> Cc: Audet, Francois (SC100:3055); Anders Kristensen; Henry=20
> Sinnreich;=20
> >> dispatch@ietf.org
> >> Subject: Re: [dispatch] Disaggregated Media in SIP
> >>
> >> Gonzalo,
> >>
> >> I haven't seen any response to Anders' issue about what happens if=20
> >> both ends want to use disaggregated media.
> >>
> >> Nor have I seen any explanation of where the control resides that=20
> >> initiates and coordinates the inclusion of multiple media sources=20
> >> into call. *Something* must be capable of saying "I want=20
> device X to=20
> >> join call Y using media Z". That is still a centralized point of=20
> >> control even if it isn't directly in the resulting sip signaling=20
> >> flow. Either it is using some exotic form of REFER to=20
> cause this to=20
> >> happen, or else it is using some kind of private signaling.
> >=20
> > I don't think we need/want the "...using media Z" part.
>=20
> Well, as I understand them, most or all of the use cases in=20
> the draft would require some sort of instruction to the=20
> secondary device to tell it what media to use, because it=20
> would otherwise use media that the use case assumes it will not use.
>=20
> > I truely believe that this makes it too complicated.
>=20
> All the more reason to ensure we agree on a goal set of use=20
> cases so we can be sure there is a mechanism sufficiently=20
> rich to support them without being unnecessarily complicated.=20
> Its no good to do something simple if it doesn't solve a problem.
>=20
> 	Thanks,
> 	Paul
>=20
> > Again, loose coupling.
> >=20
>=20

From mary.barnes@nortel.com  Wed Jul 22 11:14:55 2009
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Hi folks,

The final agenda for IETF-75 for the DISPATCH WG session is available:
http://www.ietf.org/proceedings/09jul/agenda/dispatch.html

As mentioned previously, in order for the f2f meeting time to be
effective, we should be having ongoing discussion of these topics prior
to the meeting - at a minimum folks need to have thoroughly reviewed the
materials for each topic. Alan has just posted an updated cc-uui
proposal, please take a look at that. Additional feedback on Scott's 3rd
party authorization proposal would be extremely useful. There is now a
document for session recording, which has only received one set of
comments.=20

So, please review the materials and provided feedback on the mailing
list.

Regards,
Mary
DISPATCH WG co-chair

From spromano@unina.it  Wed Jul 22 11:46:27 2009
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Hi Mary and others,

you might also be interested in having a look at the following draft =20
we submitted on july 6th and which was not advertised on the list:

http://tools.ietf.org/html/draft-romano-dcon-recording-00

"Session Recording for Conferences using SMIL"

Cheers,

Simon


Quoting Mary Barnes <mary.barnes@nortel.com>:

> Hi folks,
>
> The final agenda for IETF-75 for the DISPATCH WG session is available:
> http://www.ietf.org/proceedings/09jul/agenda/dispatch.html
>
> As mentioned previously, in order for the f2f meeting time to be
> effective, we should be having ongoing discussion of these topics prior
> to the meeting - at a minimum folks need to have thoroughly reviewed the
> materials for each topic. Alan has just posted an updated cc-uui
> proposal, please take a look at that. Additional feedback on Scott's 3rd
> party authorization proposal would be extremely useful. There is now a
> document for session recording, which has only received one set of
> comments.
>
> So, please review the materials and provided feedback on the mailing
> list.
>
> Regards,
> Mary
> DISPATCH WG co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>



--=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

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


From john.elwell@siemens-enterprise.com  Thu Jul 23 00:34:00 2009
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Date: Thu, 23 Jul 2009 07:53:23 +0100
From: "Elwell, John" <john.elwell@siemens-enterprise.com>
In-reply-to: <4A6740CD.3070609@sipstation.com>
To: Alan Johnston <alan@sipstation.com>, dispatch@ietf.org
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Alan,

What is meant exactly by "must survive ... transfers". Surviving
retargeting and redirection I am comfortable with, but transfers?

Would it be worth enumerating the particular ISDN services that it is
intended to work with? I believe it is just UUI1.

John=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
> Sent: 22 July 2009 17:40
> To: dispatch@ietf.org
> Subject: [dispatch] Proposed Charter for CCUS - Call Control=20
> UUI for SIP
>=20
> For discussion during DISPATCH next week.  Comments most welcome.
>=20
> Thanks,
> Alan
>=20
> - - - - - -
>=20
> The Call Control UUI for SIP  (CCUS) working group is chartered to=20
> define a SIP mechanism for transporting call control related=20
> user-to-user (UUI) information between User Agents.
>=20
> The ISDN User to User Information Service, defined by ITU-T Q.931, is=20
> extensively deployed in the PSTN today, supporting such=20
> applications as=20
> contact centers, call centers, and automatic call=20
> distributors (ACDs). =20
> A major barrier to the movement of these applications to SIP=20
> is the lack=20
> of a standard mechanism to transport this information in SIP.
>=20
> Call control UUI is user information conveyed between user=20
> agents during=20
> call control operations.  As a result, the information must=20
> be conveyed=20
> with the INVITE transaction, and must survive proxy retargeting,=20
> redirection, and transfers.  The mechanism must utilize a=20
> minimum of SIP=20
> extensions as it will need to be supported by many simple SIP user=20
> agents such as PSTN gateways.  The mechanism must interwork with the=20
> existing ISDN service but should also be extensible for use by other=20
> applications and non-ISDN protocols.
>=20
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call=20
> control UUI=20
> can be generated by a number of protocols that are not ISUP,=20
> and as such=20
> it is architecturally unreasonable to interwork these=20
> protocols at the=20
> SIP / ISUP gateway. That is, existing SIP-T approaches defined in=20
> RFC3372 do not meet the requirements.
>=20
> Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> considered=20
> by the working group since the UUI contents carry information=20
> that must=20
> be conveyed during session setup at the user agent - the information=20
> must be conveyed with the INVITE transaction.  The=20
> information must be=20
> passed with the session setup request (INVITE), responses to that=20
> INVITE, or session termination requests.  As a result, it is not=20
> possible to use INFO in these cases.
>=20
> The group will produce:
>=20
> - A problem statement and requirements document for=20
> implementing a SIP=20
> call control UUI mechanism
>=20
> - A specification of the SIP extension to best meet those=20
> requirements.
>=20
> Goals and Milestones
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> Dec 09 - Problem statement and requirements document to IESG=20
> (Informational)
> Mar 10 - SIP call control UUI specification to IESG (PS)
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

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From: Leon Portman <Leon.Portman@nice.com>
To: "spromano@unina.it" <spromano@unina.it>, Mary Barnes <mary.barnes@nortel.com>
Date: Thu, 23 Jul 2009 10:52:01 +0300
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Simon, Hi

It will be very interesting to see how we can combine both works: generic s=
ession recording protocol and your work regarding conferencing recording.
>From my point of view both these draft complement each other: SRP is more f=
ocused on interaction between recording  system and communication system an=
d DCON recording is focused on how actually conference resource will be all=
ocated and inserted to the call in order to be able to forward media to rec=
ording - in the terms of SRP is how to implement conference based recording=
 client. Obviously there are also other important use cases for recording w=
here conferencing is not valuable option because another component that alr=
eady involved in the media path can do the forking: B2BUA, gateway, end poi=
nts.

Using SMIL for capturing metadata for the recording is also very interestin=
g concept and well worth discussion.

Regards

Leon



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of spromano@unina.it
Sent: Wednesday, July 22, 2009 9:34 PM
To: Mary Barnes
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Agenda for IETF-75

Hi Mary and others,

you might also be interested in having a look at the following draft =20
we submitted on july 6th and which was not advertised on the list:

http://tools.ietf.org/html/draft-romano-dcon-recording-00

"Session Recording for Conferences using SMIL"

Cheers,

Simon


Quoting Mary Barnes <mary.barnes@nortel.com>:

> Hi folks,
>
> The final agenda for IETF-75 for the DISPATCH WG session is available:
> http://www.ietf.org/proceedings/09jul/agenda/dispatch.html
>
> As mentioned previously, in order for the f2f meeting time to be
> effective, we should be having ongoing discussion of these topics prior
> to the meeting - at a minimum folks need to have thoroughly reviewed the
> materials for each topic. Alan has just posted an updated cc-uui
> proposal, please take a look at that. Additional feedback on Scott's 3rd
> party authorization proposal would be extremely useful. There is now a
> document for session recording, which has only received one set of
> comments.
>
> So, please review the materials and provided feedback on the mailing
> list.
>
> Regards,
> Mary
> DISPATCH WG co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>



--=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

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

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

From Leon.Portman@nice.com  Thu Jul 23 04:30:20 2009
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From: Leon Portman <Leon.Portman@nice.com>
To: "spromano@unina.it" <spromano@unina.it>
Date: Thu, 23 Jul 2009 14:24:39 +0300
Thread-Topic: [dispatch] Agenda for IETF-75
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Yes, of course. I will be only until Tuesday so Sunday, Monday and Tuesday =
morning are all open.

Regards and looking forward

Leon

-----Original Message-----
From: spromano@unina.it [mailto:spromano@unina.it]=20
Sent: Thursday, July 23, 2009 1:50 PM
To: Leon Portman
Cc: Mary Barnes; dispatch@ietf.org
Subject: RE: [dispatch] Agenda for IETF-75

Hi Leon,

I totally agree with your proposal and I'm looking forward to working =20
together on the topic. What about discussing this f2f next week in =20
Stocholm?

Cheers,

Simon

Quoting Leon Portman <Leon.Portman@nice.com>:

> Simon, Hi
>
> It will be very interesting to see how we can combine both works:  =20
> generic session recording protocol and your work regarding  =20
> conferencing recording.
> From my point of view both these draft complement each other: SRP is =20
>  more focused on interaction between recording  system and  =20
> communication system and DCON recording is focused on how actually  =20
> conference resource will be allocated and inserted to the call in  =20
> order to be able to forward media to recording - in the terms of SRP =20
>  is how to implement conference based recording client. Obviously  =20
> there are also other important use cases for recording where  =20
> conferencing is not valuable option because another component that  =20
> already involved in the media path can do the forking: B2BUA,  =20
> gateway, end points.
>
> Using SMIL for capturing metadata for the recording is also very  =20
> interesting concept and well worth discussion.
>
> Regards
>
> Leon
>
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]  =20
> On Behalf Of spromano@unina.it
> Sent: Wednesday, July 22, 2009 9:34 PM
> To: Mary Barnes
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Agenda for IETF-75
>
> Hi Mary and others,
>
> you might also be interested in having a look at the following draft
> we submitted on july 6th and which was not advertised on the list:
>
> http://tools.ietf.org/html/draft-romano-dcon-recording-00
>
> "Session Recording for Conferences using SMIL"
>
> Cheers,
>
> Simon
>
>
> Quoting Mary Barnes <mary.barnes@nortel.com>:
>
>> Hi folks,
>>
>> The final agenda for IETF-75 for the DISPATCH WG session is available:
>> http://www.ietf.org/proceedings/09jul/agenda/dispatch.html
>>
>> As mentioned previously, in order for the f2f meeting time to be
>> effective, we should be having ongoing discussion of these topics prior
>> to the meeting - at a minimum folks need to have thoroughly reviewed the
>> materials for each topic. Alan has just posted an updated cc-uui
>> proposal, please take a look at that. Additional feedback on Scott's 3rd
>> party authorization proposal would be extremely useful. There is now a
>> document for session recording, which has only received one set of
>> comments.
>>
>> So, please review the materials and provided feedback on the mailing
>> list.
>>
>> Regards,
>> Mary
>> DISPATCH WG co-chair
>> _______________________________________________
>> 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
>
>      <<Molti mi dicono che lo scoraggiamento =E8 l'alibi degli
>     idioti. Ci rifletto un istante; e mi scoraggio>>. Magritte.
>                           oooO
>     ~~~~~~~~~~~~~~~~~~~~~~(   )~~ Oooo~~~~~~~~~~~~~~~~~~~~~~~~~
>                            \ (    (   )
>                             \_)    ) /
>                                   (_/
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>



--=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

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


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Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi Paul,

> Nor have I seen any explanation of where the control resides that 
> initiates and coordinates the inclusion of multiple media sources
> into call. *Something* must be capable of saying "I want device X to
> join call Y using media Z". That is still a centralized point of
> control even if it isn't directly in the resulting sip signaling
> flow. Either it is using some exotic form of REFER to cause this to
> happen, or else it is using some kind of private signaling.

yes, you could use REFER as you suggest, although (as Francois also
pointed out) I would remove the "using media Z" bit. When we designed
REFER, we agreed not to include that type of stuff because applications
typically know what to include based on the context in which the REFER
is received.

You could also use some type of private signalling if you want. There is
nothing that precludes that. If you want to use your proprietary 
protocol for local coordination, you can do that.

> Also, if a multimedia call is received by a UAS, and the UAS needs to
>  use disaggregated media to cope with the offered media types, how
> will that work? Will some of the media be refused initially, and then
> offered back in a new INVITE going in the reverse direction?

Yes, this could be implemented like that.

> There are a *lot* of details here that need to be worked out for this
>  approach to be functional.

I would have loved to discuss all those details earlier (we submitted
our first draft about this a long time ago) but folks did not want to
even consider using something else than 3pcc. Once we agree that 3pcc is
not the only way to do things, we can work out the details.

Note that when we introduced SIP more than 10 years ago many people did
not like the fact that media was simply loosely coupled with the
signalling. They wanted a tighter coupling... it seems we will have to
go through similar discussions once more ;o)

You wrote in another message:

> All the more reason to ensure we agree on a goal set of use cases so
> we can be sure there is a mechanism sufficiently rich to support them
> without being unnecessarily complicated. Its no good to do something
> simple if it doesn't solve a problem.

Yes, agreeing on use cases would be great. That's why the draft has a 
section with use cases. Fell free to propose new ones or send comments 
on the existing ones.

Thanks,

Gonzalo


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Hi Mary,

Sorry, I should have answered this mail earlier.

On Sat, Jul 4, 2009 at 5:34 PM, Mary Barnes<mary.barnes@nortel.com> wrote:
> Xavier,

> Can you point the WG to past ML discussions (in any WG) for this document=
?
> I don't recall nor could I find any email threads that indicate this docu=
ment had
> ever been on the radar for the SIPPING WG.
I agree I haven't seen either any discussion on the topic before.


> the usage of this "service indicator" is not clear to me,
> perhaps because it's based on IMS functionality.
Yes, such service indicator would indeed be useful in IMS networks,
but it may also be of interest for other networks too. As I see it,
this a way to exchange/manage information between services when hosted
on different servers in the path of a SIP message. I would thus rather
see such an activity -if any- in the scope of dispatch, and not in the
scope of ecrit of atoca (maybe other examples should thus be provided).

Cheers,
Xavier

> Regards,
> Mary.
> DISPATCH WG co-chair
>
> -----Original Message-----
> From: xavier.marjou@gmail.com [mailto:xavier.marjou@gmail.com] On Behalf =
Of Xavier Marjou
> Sent: Friday, July 03, 2009 10:06 AM
> To: dispatch@ietf.org; Barnes, Mary (RICH2:AR00)
> Subject: draft-shen-interaction-ind
>
> Hi,
>
> As it seems there will be time for additional topics, I have a
> suggestion:=A0I would be interested by a presentation of the following
> draft:
> http://tools.ietf.org/html/draft-shen-interaction-ind-05.txt
>
>
> Cheers,
> Xavier
>
> On Fri, Jun 26, 2009 at 12:51 AM, Mary Barnes <mary.barnes@nortel.com> wr=
ote:
>>
>> Hi folks,
>>
>> We have uploaded a revised (tentative) agenda for IETF-75 for the
>> DISPATCH WG session:
>> http://www.ietf.org/proceedings/09jul/agenda/dispatch.html
>>
>> Due to popular demand, we have added a slot for Session Recording.
>> Note, that the current agenda has us meeting on Thursday afternoon,
>> but Cullen has asked for that to be changed to Monday. As always, the
>> agenda is subject to change up until the meeting.
>>
>> As a reminder any new documents related to the agenda topics must be
>> submitted by July 6th and any updates to current docs by July 13th.
>> And, most importantly we really should be seeing traffic on the list
>> on these topics before the meeting so we can have focused discussions
>> with conclusions and a well defined way forward.
>>
>> Please send any comments on the proposed agenda to the list and/or
>> Gonzalo and myself.
>>
>> Regards,
>> Mary
>> DISPATCH WG co-chair
>> _______________________________________________
>> 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 alan@sipstation.com  Thu Jul 23 05:56:55 2009
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Subject: Re: [dispatch] Proposed Charter for CCUS - Call Control UUI for SIP
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John,

Thanks for your comments.  Transfers means the ability to pass UUI in a 
Refer-To URI in a REFER.  This requirement and the use cases are in:

     http://tools.ietf.org/html/draft-johnston-dispatch-sip-cc-uui

which is the same set of requirements and use cases we discussed in 
SIPPING in the past.  I'm not trying to add or change these 
requirements, just summarize them at a high level in the charter text.

And yes, we should say that this is ISDN UUS Service 1.

Thanks,
Alan

Elwell, John wrote:
> Alan,
>
> What is meant exactly by "must survive ... transfers". Surviving
> retargeting and redirection I am comfortable with, but transfers?
>
> Would it be worth enumerating the particular ISDN services that it is
> intended to work with? I believe it is just UUI1.
>
> John 
>
>   
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
>> Sent: 22 July 2009 17:40
>> To: dispatch@ietf.org
>> Subject: [dispatch] Proposed Charter for CCUS - Call Control 
>> UUI for SIP
>>
>> For discussion during DISPATCH next week.  Comments most welcome.
>>
>> Thanks,
>> Alan
>>
>> - - - - - -
>>
>> The Call Control UUI for SIP  (CCUS) working group is chartered to 
>> define a SIP mechanism for transporting call control related 
>> user-to-user (UUI) information between User Agents.
>>
>> The ISDN User to User Information Service, defined by ITU-T Q.931, is 
>> extensively deployed in the PSTN today, supporting such 
>> applications as 
>> contact centers, call centers, and automatic call 
>> distributors (ACDs).  
>> A major barrier to the movement of these applications to SIP 
>> is the lack 
>> of a standard mechanism to transport this information in SIP.
>>
>> Call control UUI is user information conveyed between user 
>> agents during 
>> call control operations.  As a result, the information must 
>> be conveyed 
>> with the INVITE transaction, and must survive proxy retargeting, 
>> redirection, and transfers.  The mechanism must utilize a 
>> minimum of SIP 
>> extensions as it will need to be supported by many simple SIP user 
>> agents such as PSTN gateways.  The mechanism must interwork with the 
>> existing ISDN service but should also be extensible for use by other 
>> applications and non-ISDN protocols.
>>
>> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call 
>> control UUI 
>> can be generated by a number of protocols that are not ISUP, 
>> and as such 
>> it is architecturally unreasonable to interwork these 
>> protocols at the 
>> SIP / ISUP gateway. That is, existing SIP-T approaches defined in 
>> RFC3372 do not meet the requirements.
>>
>> Mechanisms based on the SIP INFO method, RFC2976, will not be 
>> considered 
>> by the working group since the UUI contents carry information 
>> that must 
>> be conveyed during session setup at the user agent - the information 
>> must be conveyed with the INVITE transaction.  The 
>> information must be 
>> passed with the session setup request (INVITE), responses to that 
>> INVITE, or session termination requests.  As a result, it is not 
>> possible to use INFO in these cases.
>>
>> The group will produce:
>>
>> - A problem statement and requirements document for 
>> implementing a SIP 
>> call control UUI mechanism
>>
>> - A specification of the SIP extension to best meet those 
>> requirements.
>>
>> Goals and Milestones
>> ====================
>>
>> Dec 09 - Problem statement and requirements document to IESG 
>> (Informational)
>> Mar 10 - SIP call control UUI specification to IESG (PS)
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>     
>
>   


From spromano@unina.it  Thu Jul 23 06:10:37 2009
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To: Leon Portman <Leon.Portman@nice.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Agenda for IETF-75
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Hi Leon,

I totally agree with your proposal and I'm looking forward to working =20
together on the topic. What about discussing this f2f next week in =20
Stocholm?

Cheers,

Simon

Quoting Leon Portman <Leon.Portman@nice.com>:

> Simon, Hi
>
> It will be very interesting to see how we can combine both works:  =20
> generic session recording protocol and your work regarding  =20
> conferencing recording.
> From my point of view both these draft complement each other: SRP is =20
>  more focused on interaction between recording  system and  =20
> communication system and DCON recording is focused on how actually  =20
> conference resource will be allocated and inserted to the call in  =20
> order to be able to forward media to recording - in the terms of SRP =20
>  is how to implement conference based recording client. Obviously  =20
> there are also other important use cases for recording where  =20
> conferencing is not valuable option because another component that  =20
> already involved in the media path can do the forking: B2BUA,  =20
> gateway, end points.
>
> Using SMIL for capturing metadata for the recording is also very  =20
> interesting concept and well worth discussion.
>
> Regards
>
> Leon
>
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]  =20
> On Behalf Of spromano@unina.it
> Sent: Wednesday, July 22, 2009 9:34 PM
> To: Mary Barnes
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Agenda for IETF-75
>
> Hi Mary and others,
>
> you might also be interested in having a look at the following draft
> we submitted on july 6th and which was not advertised on the list:
>
> http://tools.ietf.org/html/draft-romano-dcon-recording-00
>
> "Session Recording for Conferences using SMIL"
>
> Cheers,
>
> Simon
>
>
> Quoting Mary Barnes <mary.barnes@nortel.com>:
>
>> Hi folks,
>>
>> The final agenda for IETF-75 for the DISPATCH WG session is available:
>> http://www.ietf.org/proceedings/09jul/agenda/dispatch.html
>>
>> As mentioned previously, in order for the f2f meeting time to be
>> effective, we should be having ongoing discussion of these topics prior
>> to the meeting - at a minimum folks need to have thoroughly reviewed the
>> materials for each topic. Alan has just posted an updated cc-uui
>> proposal, please take a look at that. Additional feedback on Scott's 3rd
>> party authorization proposal would be extremely useful. There is now a
>> document for session recording, which has only received one set of
>> comments.
>>
>> So, please review the materials and provided feedback on the mailing
>> list.
>>
>> Regards,
>> Mary
>> DISPATCH WG co-chair
>> _______________________________________________
>> 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
>
>      <<Molti mi dicono che lo scoraggiamento =E8 l'alibi degli
>     idioti. Ci rifletto un istante; e mi scoraggio>>. Magritte.
>                           oooO
>     ~~~~~~~~~~~~~~~~~~~~~~(   )~~ Oooo~~~~~~~~~~~~~~~~~~~~~~~~~
>                            \ (    (   )
>                             \_)    ) /
>                                   (_/
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>



--=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

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


From gonzalo.camarillo@ericsson.com  Thu Jul 23 06:17:55 2009
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Subject: Re: [dispatch] Preconditions and intermediaries
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Hi Paul,

yes, you are right. The draft assumes a particular setting (e.g., the 
UAS is the answerer, the UAC reserves the resources and informs the UAS, 
and the UAS starts using the new parameters). I can fix that and add all 
the other possible scenarios for completeness.

However, before putting any more cycles on this, I would like to know 
whether or not people think this is important. If so, I can edit the 
draft so that we can progress it. On the other hand, if folks are not 
interested in this, I'd rather spend my cycles on something else.

Thanks,

Gonzalo

Paul Kyzivat wrote:
>> 2.  Starting Using the Media Parameters Subject to Preconditions
> ...
>>    However, the fact that the UAs can start using the new media
>>    parameters does not mean that they need to start using them
>>    immediately.  When preconditions are used, the UAS decides when to
>>    start using them.  
> 
> Why do you say that?
> 
> For one thing offerer and answerer have more significance than UAC and 
> UAS in any such decision.
> 
> For another, QoS needs to be obtained by both ends before it can be used 
> by either end. The order in which it is obtained, and signaled that it 
> has been obtained, is indeterminate.
> 
> When one end has been told that the other end has obtained it, and knows 
> that it has obtained it, then it can start using it.
> 
> But that doesn't change the point about signaling the intent to start 
> using.
> 
>     Thanks,
>     Paul
> 


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Cc: Anders Kristensen <ankriste@cisco.com>, dispatch@ietf.org, Henry Sinnreich <hsinnrei@adobe.com>
Subject: Re: [dispatch] Disaggregated Media in SIP
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Gonzalo Camarillo wrote:
> Hi Paul,
> 
>> Nor have I seen any explanation of where the control resides that 
>> initiates and coordinates the inclusion of multiple media sources
>> into call. *Something* must be capable of saying "I want device X to
>> join call Y using media Z". That is still a centralized point of
>> control even if it isn't directly in the resulting sip signaling
>> flow. Either it is using some exotic form of REFER to cause this to
>> happen, or else it is using some kind of private signaling.
> 
> yes, you could use REFER as you suggest, although (as Francois also
> pointed out) I would remove the "using media Z" bit. When we designed
> REFER, we agreed not to include that type of stuff because applications
> typically know what to include based on the context in which the REFER
> is received.
> 
> You could also use some type of private signalling if you want. There is
> nothing that precludes that. If you want to use your proprietary 
> protocol for local coordination, you can do that.
> 
>> Also, if a multimedia call is received by a UAS, and the UAS needs to
>>  use disaggregated media to cope with the offered media types, how
>> will that work? Will some of the media be refused initially, and then
>> offered back in a new INVITE going in the reverse direction?
> 
> Yes, this could be implemented like that.
> 
>> There are a *lot* of details here that need to be worked out for this
>>  approach to be functional.
> 
> I would have loved to discuss all those details earlier (we submitted
> our first draft about this a long time ago) but folks did not want to
> even consider using something else than 3pcc. Once we agree that 3pcc is
> not the only way to do things, we can work out the details.

I find the *use cases* interesting and worthy of discussing independent 
of the mechanism used to achieve them. Either approach requires some 
coordination among the disaggregated devices. With the 3pcc approach 
there is the possibility of using sip to do the coordination, but its 
still worth investigation since that may not provide the dieal user 
experience.

So I'm not really conceding that the proposed solution is necessary, or 
even desirable to address the use cases. But I do think that 
investigating the use cases is a good way to get beyond individual 
opinions on the subject.

> Note that when we introduced SIP more than 10 years ago many people did
> not like the fact that media was simply loosely coupled with the
> signalling. They wanted a tighter coupling... it seems we will have to
> go through similar discussions once more ;o)
> 
> You wrote in another message:
> 
>> All the more reason to ensure we agree on a goal set of use cases so
>> we can be sure there is a mechanism sufficiently rich to support them
>> without being unnecessarily complicated. Its no good to do something
>> simple if it doesn't solve a problem.
> 
> Yes, agreeing on use cases would be great. That's why the draft has a 
> section with use cases. Fell free to propose new ones or send comments 
> on the existing ones.

I didn't comment because I liked them.

	Thanks,
	Paul

From dworley@nortel.com  Thu Jul 23 11:14:30 2009
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Subject: Re: [dispatch] Disaggregated Media in SIP
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On Tue, 2009-07-21 at 11:05 +0300, Gonzalo Camarillo wrote:
> well, 3pcc is centralized because you have a (central) controller that 
> is in control of all SIP signalling. If that controller leaves the 
> session, the whole session disappears. An approach where several devices 
> have their own SIP dialog with the remote devices is not centralized 
> because you can remove any of them and the other sessions will remain 
> unaffected.

It's true that if the implementation is a single dialog, then the UA at
each end must remain in the call.  For a UA to replace itself with
another UA would require a transfer-like process.

The difficulty (under either implementation) is to transfer the "state"
information (which media stream goes to which destination) from one
controller to another.  That need is obvious in the centralized
approach, but it seems to me that it would also be needed in the
decentralized approach.

> The arguments have been discussed several times. The thing is that 3pcc 
> is a complex mechanism that is not supported by most UAs. You can argue 
> about its complexity, but the fact that most UAs do not support 3pcc is 
> a fact.

This does not seem to be true to me.  Phones that support MOH from an
external server are doing 3pcc.  In addition, we expect phones to
support REFER and INVITE-with-Replaces.

> A second argument is that any device should be able to leave the session 
> at any point. If you want the 3pcc controller to leave the session while 
> the session remains established, you will need to use a complex 
> combination of REFER and Replaces that very few (if any) UAs support.

As a matter of course, we expect our customers to buy phones that
support both of these.

I suspect the use case that truly distinguishes these implementations is
where there is a "controller" at one end, that is, a UA that understands
the aggregation, but there is no controller at the other end.

Dale



From dworley@nortel.com  Thu Jul 23 11:43:01 2009
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Cc: sip-ops@ietf.org, dispatch@ietf.org
Subject: Re: [dispatch] [sip-ops] I-D Action:draft-kuthan-dispatch-diagrevived-00.txt
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On Fri, 2009-07-03 at 15:40 +0200, Victor Pascual Avila wrote:
> A temporal URL for this Internet-Draft is:
> http://www.ietf.org/proceedings/staging/draft-kuthan-dispatch-diagrevived-00.txt

Now at http://www.ietf.org/id/draft-kuthan-dispatch-diagrevived-00.txt

Dale



From gonzalo.camarillo@ericsson.com  Thu Jul 23 14:50:05 2009
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Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi Paul,

yes, let's focus on the use cases and how they could be implemented.

Cheers,

Gonzalo

Paul Kyzivat wrote:
> 
> 
> Gonzalo Camarillo wrote:
>> Hi Paul,
>>
>>> Nor have I seen any explanation of where the control resides that 
>>> initiates and coordinates the inclusion of multiple media sources
>>> into call. *Something* must be capable of saying "I want device X to
>>> join call Y using media Z". That is still a centralized point of
>>> control even if it isn't directly in the resulting sip signaling
>>> flow. Either it is using some exotic form of REFER to cause this to
>>> happen, or else it is using some kind of private signaling.
>>
>> yes, you could use REFER as you suggest, although (as Francois also
>> pointed out) I would remove the "using media Z" bit. When we designed
>> REFER, we agreed not to include that type of stuff because applications
>> typically know what to include based on the context in which the REFER
>> is received.
>>
>> You could also use some type of private signalling if you want. There is
>> nothing that precludes that. If you want to use your proprietary 
>> protocol for local coordination, you can do that.
>>
>>> Also, if a multimedia call is received by a UAS, and the UAS needs to
>>>  use disaggregated media to cope with the offered media types, how
>>> will that work? Will some of the media be refused initially, and then
>>> offered back in a new INVITE going in the reverse direction?
>>
>> Yes, this could be implemented like that.
>>
>>> There are a *lot* of details here that need to be worked out for this
>>>  approach to be functional.
>>
>> I would have loved to discuss all those details earlier (we submitted
>> our first draft about this a long time ago) but folks did not want to
>> even consider using something else than 3pcc. Once we agree that 3pcc is
>> not the only way to do things, we can work out the details.
> 
> I find the *use cases* interesting and worthy of discussing independent 
> of the mechanism used to achieve them. Either approach requires some 
> coordination among the disaggregated devices. With the 3pcc approach 
> there is the possibility of using sip to do the coordination, but its 
> still worth investigation since that may not provide the dieal user 
> experience.
> 
> So I'm not really conceding that the proposed solution is necessary, or 
> even desirable to address the use cases. But I do think that 
> investigating the use cases is a good way to get beyond individual 
> opinions on the subject.
> 
>> Note that when we introduced SIP more than 10 years ago many people did
>> not like the fact that media was simply loosely coupled with the
>> signalling. They wanted a tighter coupling... it seems we will have to
>> go through similar discussions once more ;o)
>>
>> You wrote in another message:
>>
>>> All the more reason to ensure we agree on a goal set of use cases so
>>> we can be sure there is a mechanism sufficiently rich to support them
>>> without being unnecessarily complicated. Its no good to do something
>>> simple if it doesn't solve a problem.
>>
>> Yes, agreeing on use cases would be great. That's why the draft has a 
>> section with use cases. Fell free to propose new ones or send comments 
>> on the existing ones.
> 
> I didn't comment because I liked them.
> 
>     Thanks,
>     Paul


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Dale Worley wrote:
> On Tue, 2009-07-21 at 11:05 +0300, Gonzalo Camarillo wrote:
>   
>> well, 3pcc is centralized because you have a (central) controller that 
>> is in control of all SIP signalling. If that controller leaves the 
>> session, the whole session disappears. An approach where several devices 
>> have their own SIP dialog with the remote devices is not centralized 
>> because you can remove any of them and the other sessions will remain 
>> unaffected.
>>     
>
> It's true that if the implementation is a single dialog, then the UA at
> each end must remain in the call.  For a UA to replace itself with
> another UA would require a transfer-like process.
>
> The difficulty (under either implementation) is to transfer the "state"
> information (which media stream goes to which destination) from one
> controller to another.  That need is obvious in the centralized
> approach, but it seems to me that it would also be needed in the
> decentralized approach.
>   
the need or not to transfer the "state" really depends on how the 
decentralized approach will be designed.

for example:
In a "loosely coupled control" approach, as Francois is proposing, you 
don't have any "state" information that
need to be transferred from one to another because you don't have a real 
controller.
Another possibility would be to have a "fully distributed approach", 
where the "information state" does not reside
in one single UA, but is in somehow spread among all the UAs involved in 
the disaggregate media.


/Sal

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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Disaggregated Media in SIP
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Salvatore Loreto wrote:
> Dale Worley wrote:
>> On Tue, 2009-07-21 at 11:05 +0300, Gonzalo Camarillo wrote:
>>  
>>> well, 3pcc is centralized because you have a (central) controller 
>>> that is in control of all SIP signalling. If that controller leaves 
>>> the session, the whole session disappears. An approach where several 
>>> devices have their own SIP dialog with the remote devices is not 
>>> centralized because you can remove any of them and the other sessions 
>>> will remain unaffected.
>>>     
>>
>> It's true that if the implementation is a single dialog, then the UA at
>> each end must remain in the call.  For a UA to replace itself with
>> another UA would require a transfer-like process.
>>
>> The difficulty (under either implementation) is to transfer the "state"
>> information (which media stream goes to which destination) from one
>> controller to another.  That need is obvious in the centralized
>> approach, but it seems to me that it would also be needed in the
>> decentralized approach.
>>   
> the need or not to transfer the "state" really depends on how the 
> decentralized approach will be designed.
> 
> for example:
> In a "loosely coupled control" approach, as Francois is proposing, you 
> don't have any "state" information that
> need to be transferred from one to another because you don't have a real 
> controller.

I don't understand that.

Suppose Alice calls Bob using her voice phone. Then she arranges for her 
laptop to join the call with Bob to add voice. (Bob has a video phone.)

Then Bob decides to do a consultative transfer of the call to Charlie. 
Because Bob's phone is a video phone, its UI presumably treats this as a 
single call, even if there are actually two calls behind the scenes. So 
Bob first puts the call with Alice on hold. It already has to know to do 
this on two separate calls. Then it calls Charlie, who also has a video 
phone. So there is an audio+video call from Bob to Charlie. Then Bob 
asks to complete the transfer. What does it do? Normally it would send a 
REFER to Charlie's phone, asking it to do an INVITE/Replaces to Alice. 
But there are two calls to Alice.

The "state" of the call, that it is distributed across two dialogs, is 
important info that needs to be transferred to Charlie given this 
implementation approach. With the "centralized" approach that state 
resides in Alice's realm and is of no concern to either Bob or Charlie.

	Thanks,
	Paul

> Another possibility would be to have a "fully distributed approach", 
> where the "information state" does not reside
> in one single UA, but is in somehow spread among all the UAs involved in 
> the disaggregate media.
> 
> 
> /Sal
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From victor.pascual.avila@gmail.com  Fri Jul 24 07:34:49 2009
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Subject: Re: [dispatch] [sip-ops] I-D Action:draft-kuthan-dispatch-diagrevived-00.txt
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On Thu, Jul 23, 2009 at 8:40 PM, Dale Worley<dworley@nortel.com> wrote:
> On Fri, 2009-07-03 at 15:40 +0200, Victor Pascual Avila wrote:
>> A temporal URL for this Internet-Draft is:
>> http://www.ietf.org/proceedings/staging/draft-kuthan-dispatch-diagrevive=
d-00.txt
>
> Now at http://www.ietf.org/id/draft-kuthan-dispatch-diagrevived-00.txt

Thanks for the update, Dale.

We would appreciate feedback as to whether this is helpful and going
in the right direction, as well as detailed comments-- there are some
open issues (Appendix B) that we would like to get discussed (Topology
Hiding, Amount of troubleshooting information, Selective logging and
B2BUA traversal).

Cheers,
--=20
Victor Pascual =C3=81vila

From alan@sipstation.com  Fri Jul 24 11:29:01 2009
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Here are my comments on this draft.

Overall, I think this draft is an excellent start at the requirements 
for this important topic.  I look forward to future discussions of 
charter text so we can get started on the work.

Section 2:

"  Another, related IETF draft is draft-wing-sipping-srtp-key
   [I-D.wing-sipping-srtp-key], which describes an approach for
   providing SRTP session keys to a recorder-type server.  That draft
   focuses on the mechanism to provide the SRTP session key, rather than
   the mechanism to invoke and sustain recording sessions themselves."

draft-wing-sipping-srtp-key actually has a lot of discussion, including 
call flows showing how recording sessions can be established.  Also, 
this document has four recording modes discussed, which I think should 
be discussed in this document.  Two are mentioned without really 
explaining them, and two others are implied by various requirements.

Section 3:

" Recording Session: The session created between an RC and RS for the
   purpose of recording a Recorded Session.  The Recorded Session may
   itself be based on SIP, but it is a separate session from the
   Recording Session.

   Recorded Session: A session created between a UAC and UAS that is
   recorded by the RC and RS systems."


These two terms only differ by their tense, yet I think one refers to a 
signaling session and the other refers to a media session.  If so, 
better terms and definitions are needed.  If not, we need a term for the 
actual recorded media.  Also, there are (at least) two sessions here - 
the one between the UAs and the Recording Session - in a few places I 
suspect the two are being confused. 

"  Dynamic Recording: Dynamic recording is a mode of recording where the
   recording sessions are established on an as needed basis.  The length
   of these sessions is typically same as the length of the actual media
   sessions.

   Persistent Recording: Persistent recording is a mode of recording
   where the recording sessions are established at system start-up and
   kept-alive from that point on.  The length of these sessions is
   independent from the length of the actual recorded media sessions.
   Persistent recording sessions avoid issues such as media clipping
   that can occur due to delays in recording session establishment."

These terms need more explanation as I can think of multiple things they 
might mean.  For example, when you say "session" do you mean a signaling 
session or media session?  When you talk about avoiding media clipping, 
do you mean that media is always sent by the RC to the RS regardless of 
whether there is any media? 

Figures could use Figure numbers and labels.  Also, it would be worth 
showing which is the RC and which is the RS in them.

Section 5

"   o REQ-001 The mechanism MUST support the ability to perform
   persistent (always-on) or dynamic (on-demand) Recording Sessions."

I'm not sure that the Persistent and Dynamic Recording as defined above 
correspond to the Always On and On Demand modes defined in 
http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#section-4.

"   o REQ-002 The mechanism MUST support the ability to perform
   persistent Recording Sessions, such that the connection and
   attributes of media in the Recording Session are not dynamically
   signaled for each Recorded Session before it can be recorded.  Call
   details and metadata will still be signaled, but can be post-
   correlated to the recorded media.  There will still need to be a
   means of correlating the recorded media connection/packets to the
   Recorded Session, however this may be on a permanent filter-type
   basis, such as based on a SIP AoR of an agent that is always
   recorded, or based on identifiers in the recorded media itself."


Without fully understanding Persistent Recording, I can't evaluate this 
requirement. 

"   o REQ-009 The mechanism MUST support the ability to deliver mixed
   media streams to the RS.  The RS MAY be informed about the
   composition of the mixed streams through session metadata.

   Note: A mixed media stream is where several recorded sessions are
   carried in a single recording session.  A mixed media stream is
   typically produced by a mixer function."


I'm not clear what mixed media means here - does it mean a mix of media 
types - synchronized audio and video from the same session, or do you 
mean multiple media streams from different sessions?  The note seems to 
indicate that this relates only to a conference call where multiple 
sources (SSRCs) have been combined into a single RTP session.

"   o REQ-013 The mechanism MUST support the ability to inject tones into
   the Recorded Session either from the RS or from the RC."

Recorded Session seems to be defined as the media between the RC and 
RS.  Is that where the injected tones should be?  Or should then be 
mixed in the media between the two UAs?

"   o REQ-021 The mechanism MUST support the ability for the RC to only
   perform media replication to the RS, without needing to decode or mix
   audio/video/etc., and without needing to be an RTP agent.  This will
   allow the RC to replicate media at layers below RTP.  Clearly, such
   an RC mode would not be able to provide transcoding, or media
   injection from the RS back into the Recorded Session."

Does this mean that you think that RTP is unsuitable for transporting 
media between the RS and RC?  Clearly it isn't for IM, but I would think 
that many of the functions of RTP are needed for audio and video media.  
To understand this requirement will take more discussion.

Thanks,
Alan

From dean.willis@softarmor.com  Sat Jul 25 01:58:13 2009
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Alan Johnston wrote:
> John,
> 
> Thanks for your comments.  Transfers means the ability to pass UUI in a
> Refer-To URI in a REFER. 

So, IF we had a way to pass user-to-user parameters in a request URI,
then we would not need a specialized mechanism for UUI.

We have an effort underway to learn how to pass parameters, but people
keep getting confused about why we need it.

So here's a question: Can we solve the UUI case using History Info? If
so, then we're basically done; we just need to register a standard
parameter for UUI. If not, then I argue that History-Info is not an
adequate method for parameter preservation end-to-end, and we need to
revisit that solution.

In absolutely no case should we be developing yet another
application-specific SIP extension header for CC-UUI. The application is
NOT part of the signaling.

--
Dean


From drage@alcatel-lucent.com  Sat Jul 25 02:37:38 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Dean Willis <dean.willis@softarmor.com>, Alan Johnston <alan@sipstation.com>
Date: Sat, 25 Jul 2009 11:37:43 +0200
Thread-Topic: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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At the higher level I think the semantics of this line of reasoning are bro=
ken.

Surely, the overall requirement is that in certain forms of transfer, the U=
UI that went to the original called user should be delivered to the transfe=
rred user. Like a number of headers, this is information that the calling u=
ser put in the package because he wanted them delivered to the final destin=
ation.

That has nothing to do with recording the Request-URI of the calling user, =
which is what I believe the semantics of the History-Info relate to. That i=
s because it has nothing to do with the Request-URI in the first place. Sur=
ely we place URI parameters in the Request-URI because they are something t=
o do with the URI therein contained.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Dean Willis
> Sent: Saturday, July 25, 2009 9:58 AM
> To: Alan Johnston
> Cc: dispatch@ietf.org
> Subject: [dispatch] CCUS and History-Info (was Proposed=20
> Charter for CCUS - Call Control UUI for SIP)
>=20
> Alan Johnston wrote:
> > John,
> >=20
> > Thanks for your comments.  Transfers means the ability to=20
> pass UUI in=20
> > a Refer-To URI in a REFER.
>=20
> So, IF we had a way to pass user-to-user parameters in a=20
> request URI, then we would not need a specialized mechanism for UUI.
>=20
> We have an effort underway to learn how to pass parameters,=20
> but people keep getting confused about why we need it.
>=20
> So here's a question: Can we solve the UUI case using History=20
> Info? If so, then we're basically done; we just need to=20
> register a standard parameter for UUI. If not, then I argue=20
> that History-Info is not an adequate method for parameter=20
> preservation end-to-end, and we need to revisit that solution.
>=20
> In absolutely no case should we be developing yet another=20
> application-specific SIP extension header for CC-UUI. The=20
> application is NOT part of the signaling.
>=20
> --
> Dean
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From alan@sipstation.com  Sat Jul 25 07:46:54 2009
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Dean,

Thanks for your comments.

We did consider URI parameters as a mechanism.  If we used this 
approach, we would still need SIP extensions for those URI parameters 
and a new SIP option tag to indicate support for it.  As such, we'd 
still need a mechanism to be standardized.  I think in DISPATCH we are 
supposed to be discussing scope and charter text, and I think the 
current charter text is mechanism agnostic, so this approach would still 
be covered, correct?

In terms of History-Info, I don't think we've considered that before.  
There were two problems with the URI parameters.  One was the 
retargeting problem but the other is semantics.  Last time I read RFC 
3261, the Request-URI contained information about the destination 
resource, and sometimes information about the next hop to that 
resource.  To put user-to-user data about the originator in the request 
into the Request-URI seems like a big change to SIP semantics that we'd 
need to think carefully about before diving into.  We would definitely 
need a SIP option tag to say that we are reversing the meaning of 
Request-URI.

As for History-Info, it doesn't feel very right - how is UUI history 
about the call?  If anything, it is future information about the call.  
Also, wouldn't every proxy that might retarget in the path have to 
support it, or the UUI would be lost?  If so, we'd need to use 
Proxy-Require: history which is both ugly and not permitted by RFC 4244.

Anyway, I look forward to more mechanism discussions in the future, but 
for now lets focus on the charter discussions.

Thanks,
Alan

Dean Willis wrote:
> Alan Johnston wrote:
>   
>> John,
>>
>> Thanks for your comments.  Transfers means the ability to pass UUI in a
>> Refer-To URI in a REFER. 
>>     
>
> So, IF we had a way to pass user-to-user parameters in a request URI,
> then we would not need a specialized mechanism for UUI.
>
> We have an effort underway to learn how to pass parameters, but people
> keep getting confused about why we need it.
>
> So here's a question: Can we solve the UUI case using History Info? If
> so, then we're basically done; we just need to register a standard
> parameter for UUI. If not, then I argue that History-Info is not an
> adequate method for parameter preservation end-to-end, and we need to
> revisit that solution.
>
> In absolutely no case should we be developing yet another
> application-specific SIP extension header for CC-UUI. The application is
> NOT part of the signaling.
>
> --
> Dean
>
>
>   


From HKaplan@acmepacket.com  Sat Jul 25 13:27:45 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Alan Johnston <alan@sipstation.com>, Dean Willis <dean.willis@softarmor.com>
Date: Sat, 25 Jul 2009 16:27:42 -0400
Thread-Topic: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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I think the retargeting and middlebox support issues are not really germane=
 as it pertains to H-I.  H-I records the req-uri's even through retargeting=
, so the UUI would be available to the UAS.  And it would survive even if P=
roxies don't support H-I, because the first UAC can insert the first H-I of=
 the req-uri, which would simply be mandated for UUI-supporting UAC's if we=
 went that way.=20

But the real problem is the semantics as you say - to me a request URI is i=
nformation about the targeted resource or how to reach it, not information =
about the source resource.  Since H-I is just a list of req-uri's, it's not=
 the right place.

Really the technically *right* place would be in MIME, afaict - it's opaque=
 application data just like geoloc, right?  But pragmatically, the issues/p=
roblems with doing it in MIME were already covered and debated in sipping, =
and iirc consensus was: put it in a header.

ISTM that we always complain about lack of running code, and here we have a=
 case where there is running code from multiple vendors, and it's not funda=
mentally broken/un-sound.  It ain't broken, why do we need to fix it?

-hadriel

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Alan Johnston
> Sent: Saturday, July 25, 2009 10:45 AM
>=20
> In terms of History-Info, I don't think we've considered that before.
> There were two problems with the URI parameters.  One was the
> retargeting problem but the other is semantics.  Last time I read RFC
> 3261, the Request-URI contained information about the destination
> resource, and sometimes information about the next hop to that
> resource.  To put user-to-user data about the originator in the request
> into the Request-URI seems like a big change to SIP semantics that we'd
> need to think carefully about before diving into.  We would definitely
> need a SIP option tag to say that we are reversing the meaning of
> Request-URI.
>=20
> As for History-Info, it doesn't feel very right - how is UUI history
> about the call?  If anything, it is future information about the call.
> Also, wouldn't every proxy that might retarget in the path have to
> support it, or the UUI would be lost?  If so, we'd need to use
> Proxy-Require: history which is both ugly and not permitted by RFC 4244.
>=20
> Anyway, I look forward to more mechanism discussions in the future, but
> for now lets focus on the charter discussions.

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To: Paul Kyzivat <pkyzivat@cisco.com>
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Just to respond to this bit of the message:

> > Not quite.  We are not making the assumption that both are=20
> gateways,=20
> > although that is possible.  Most likely, one element is SIP enabled.
> > However, it is not a fully intelligent SIP endpoint - the basic=20
> > software and application (probably many years old) is=20
> unchanged but a=20
> > SIP interface (SIP trunk) has been added.  As such, the SIP=20
> stack does=20
> > not know anything about the UUI as it is implementing=20
> exactly the ISDN=20
> > UUI service - the information is just pushed up the stack=20
> to another=20
> > application, and this application knows nothing about SIP. =20
> It still=20
> > thinks it is using an ISDN trunk.
>=20
> Then in the way I meant the question, both ends *are* gateways.
>=20
> But then if someone wishes to build native sip gear to plug=20
> into this environment, then it will still have to understand=20
> this isdn encoding.=20
> Perhaps that is the cruel truth, unless all the equipment is=20
> replaced at once.=20

I've said this before and I'll say it again. The bit being delivered to the=
 end UE is not "isdn encoding". If it was ISDN encoding, then the ISDN / SI=
P gateway would be able to interpret it and map it into SIP. Rather it is s=
ome application sitting above the ISDN call control layer in the calling te=
rminal, which is wishing to communicate with its peer application running i=
n the destination SIP UA, or gateway.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Saturday, July 04, 2009 12:39 AM
> To: Alan Johnston
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] I-D=20
> Action:draft-johnston-sipping-cc-uui-08.txt
>=20
> Hi Alan,
>=20
> responses inline. I've trimmed out the parts that don't=20
> require more discussion.
>=20
> 	Thanks,
> 	Paul
>=20
> Alan Johnston wrote:
> > Paul,
> >=20
> > Thanks for your comments.  See mine below.
> >=20
> > Thanks,
> > Alan
> >=20
> > Paul Kyzivat wrote:
> >> Hi Alan,
> >>
> >> I took a quick look at this doc and the requirements doc=20
> as well. I=20
> >> have a few thoughts:
> >>
> >> From section 1:
> >>
> >>>    In the future, where both endpoints are intelligent=20
> SIP user agents,
> >>>    it may be possible for them to understand and=20
> interpret the UUI data.
> >>>    There may be some cases where the UUI information is=20
> relevant to SIP.
> >>>    In this case, it might be worthwhile attempting to map=20
> UUI data to an
> >>>    appropriate SIP header field or to standardize a new=20
> header field.
> >>>    However, the requirements and use cases for this are=20
> different enough
> >>>    from those described in this document that these two situations
> >>>    should be examined separately.  This document looks only at the
> >>>    requirements and mechanisms for replicating the=20
> existing, widely used
> >>>    and deployed ISDN UUI service.
> >>
> >> This *still* troubles me!
> >> Is the assumption here that *both* the UAC and UAS are=20
> gateways, so=20
> >> that this mechanism is solely for the tunneling of Q.931 and Q.763=20
> >> UUI message over SIP?
> >>
> >> I get the impression that an important use case is where at least
> >> *one* end (probably the UAS) is a SIP device. In that case it will=20
> >> already be *necessary* for it to understand and interpret UUI data.
> >> And where there is redundancy between SIP data and the UUI=20
> data, it=20
> >> will need to resolve the discrepancy.
> >=20
> > Not quite.  We are not making the assumption that both are=20
> gateways,=20
> > although that is possible.  Most likely, one element is SIP enabled.
> > However, it is not a fully intelligent SIP endpoint - the basic=20
> > software and application (probably many years old) is=20
> unchanged but a=20
> > SIP interface (SIP trunk) has been added.  As such, the SIP=20
> stack does=20
> > not know anything about the UUI as it is implementing=20
> exactly the ISDN=20
> > UUI service - the information is just pushed up the stack=20
> to another=20
> > application, and this application knows nothing about SIP. =20
> It still=20
> > thinks it is using an ISDN trunk.
>=20
> Then in the way I meant the question, both ends *are* gateways.
>=20
> But then if someone wishes to build native sip gear to plug=20
> into this environment, then it will still have to understand=20
> this isdn encoding.=20
> Perhaps that is the cruel truth, unless all the equipment is=20
> replaced at once.
>=20
> >>> 5.  Syntax for UUI Header Field
> >> ...
> >>>        UUI         =3D "User-to-User" HCOLON uui-data=20
> *(SEMI uui-param)
> >>>        uui-data    =3D token
> >>>        uui-param   =3D enc-param | cont-param | purp-param=20
> | generic-param
> >>>        enc-param   =3D "encoding=3D" ("hex" | token)
> >>>        cont-param  =3D "content=3D" ("isdn-uui" | token)
> >>>        purp-param  =3D "purpose=3D" ("isdn-interwork" | token)
> >>
> >> At the very least, I would request that this be expanded a=20
> little bit=20
> >> for future flexibility, by permitting the uui-data to be either a=20
> >> token or a string. While the encoding you have chosen works as a=20
> >> token, other encodings may need the additional flexibility=20
> of a string.
> >>
> >>        uui-data    =3D token | quoted-string
> >=20
> > OK, I guess I'm OK with that.  For the isdn-interwork=20
> purpose, we'll=20
> > require the token, but other applications could utilize the=20
> quoted string.
>=20
> I would hope that when the token encoding is sufficient, then=20
> either form would be accepted. Or it would perhaps be ok to=20
> specify for a particular encoding (hex) that the token=20
> encoding must be used. I don't see how the purpose has=20
> anything to do with it.
>=20
> >>> 6.3.  Registration of SIP Option Tag
> >>
> >> This section registers the new option tag. But I find=20
> almost nothing=20
> >> that defines the semantics of the option. (Section 5 does=20
> include the
> >> following:
> >>
> >>>    A UA that supports this feature and the "uui" option=20
> tag MUST support
> >>>    the call flows in [johnston-dispatch-sip-cc-uui]...
> >>
> >> but that is pretty thin from a normative perspective.
> >>
> >> Of particular note, does support for the option tag mean=20
> support for=20
> >> the particular encoding, content, and purpose included in this=20
> >> document? Or does it only mean support for the header and=20
> params in=20
> >> the abstract? (That wouldn't be very useful.)
> >=20
> > Yes, this needs clarification.  I'd suggest we define the=20
> option tag=20
> > to be uui-isdn which means it understands the header field and the=20
> > isdn-interwork purpose, and the call flows, including escaping into=20
> > redirection and Refer-To URIs.
>=20
> I'd like to hear some other opinions on whether to have tag=20
> per param for tag, or a tag for the combination of param=20
> values. I think I'm ok with what you propose.
>=20
> >> I don't see how it can mean support for some other yet to=20
> be defined=20
> >> values for those. Yet if it only means support for the=20
> current ones,=20
> >> then how will one indicate support for future ones?
> >>
> >> The only way out of this that I can see is to have=20
> separate options,=20
> >> either for particular values of each parameter, or for particular=20
> >> combinations of values of all the parameters.
> >=20
> > New applications using this header field would have the option of=20
> > defining a new SIP option tag which would mean support for=20
> the header=20
> > field and their new purpose.
> >=20
> >> So you might have options uui-purpose-isdn-interwork,=20
> >> uui-content-isdn, and uui-encoding-hex. Or you might just=20
> have option=20
> >> uui-isdn-interwork that means the combination.
> >=20
> > I think the latter.  I see it as a hierarchy - the application or=20
> > purpose is the top one.  Each purpose has a number of=20
> contents that it=20
> > supports.  Each content has a number of encoding schemes it=20
> supports.
>=20
> I guess I'm ok with that, pending nailing down the definition=20
> of "purpose".
>=20
> >> I also am not finding much explanation of the semantics of the=20
> >> content and purpose parameters. I can only extrapolate=20
> from a single=20
> >> example of each, which I'm having trouble doing.
> >>
> >> I'm guessing that 'content' must refer to the precise=20
> syntax of the=20
> >> contained data, after decoding. So presumably it must map=20
> onto some=20
> >> particular spec. For isdn-uui, would that be [Q931] or=20
> [Q763]? If so,=20
> >> shouldn't it refer to something specific in that document?
> >=20
> > Not really.  In this case, isdn-uui effectively means "opaque data"=20
> > which is how ISDN defines the UUI - it is completely undefined,=20
> > necessarily so  by the strict layering used in the ISDN application.
>=20
> I don't believe you!
>=20
> If that is the case, then the UAC and UAS must have a private=20
> side agreement about what the isdn-uui content means. Is that=20
> really what you mean?
>=20
> You already state that the first byte is a protocol=20
> discriminator. There must also be something, somewhere, that=20
> specifies the mapping from a particular value for that byte=20
> to a particular protocol/format. If there is, then that=20
> should be part of the definition of this content.
>=20
> >> 'purpose' is even less clear to me. Does it refer to why=20
> it is being=20
> >> included? Or what is to be done with it? If received by a=20
> UA that is=20
> >> not an ISDN gateway, how can it decide if this is=20
> something it should=20
> >> be trying to process? Is it still ISDN interworking if neither the=20
> >> UAC nor UAS is connected to an ISDN network?
> >=20
> > The purpose is the application using the UUI.  Others have=20
> expressed=20
> > an interest in carrying other data in the header field, in=20
> which case=20
> > a new purpose would be defined.  I'll see if I can think of=20
> an example one.
>=20
> So are you saying that a particular call center application=20
> might constitute a distinct "purpose"?
>=20
> > Basically, if two UAs both support the header but have different=20
> > applications generating and consuming the UUI, it will not=20
> work.  This=20
> > is not a SIP failure, and there is nothing SIP can do about=20
> it.  But=20
> > this purpose header field allows the UAs to detect this condition.
>=20
> In that case, perhaps "isdn-interwork" isn't really a single=20
> purpose at all, unless any two pieces of equipment that=20
> support that purpose can then interwork without knowing any=20
> more about each other.
>=20
> >> Suppose I was developing a sophisticated UA that=20
> potentially might be=20
> >> usable by a call center operator. How should=20
> purpose=3Disdn-interwork=20
> >> affect the operation of this device?
> >=20
> > To make it work in today's contact center applications, supporting=20
> > isdn-interwork would likely make it work in an application,=20
> provided=20
> > the appropriate call center application also ran on it. =20
> Some contact=20
> > centers do not use ISDN UUI, instead they use all kinds of really,=20
> > really ugly CTI protocols running in parallel.
>=20
> Yes, I know! :-(
>=20
> > This is a way of moving
> > those to the ISDN model, and from there to a more=20
> intelligent endpoint=20
> > SIP model.
> >=20
> >>     Thanks,
> >>     Paul
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From Leon.Portman@nice.com  Sat Jul 25 19:10:13 2009
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From: Leon Portman <Leon.Portman@nice.com>
To: Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Sun, 26 Jul 2009 05:10:08 +0300
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Alan, Hi

Thanks for the comments.

Please see below my answers

Regards

Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Alan Johnston
Sent: Friday, July 24, 2009 9:29 PM
To: dispatch@ietf.org
Subject: [dispatch] Comments on draft-jain-dispatch-session-recording-proto=
col-req-00

Here are my comments on this draft.

Overall, I think this draft is an excellent start at the requirements=20
for this important topic.  I look forward to future discussions of=20
charter text so we can get started on the work.

Section 2:

"  Another, related IETF draft is draft-wing-sipping-srtp-key
   [I-D.wing-sipping-srtp-key], which describes an approach for
   providing SRTP session keys to a recorder-type server.  That draft
   focuses on the mechanism to provide the SRTP session key, rather than
   the mechanism to invoke and sustain recording sessions themselves."

draft-wing-sipping-srtp-key actually has a lot of discussion, including=20
call flows showing how recording sessions can be established.  Also,=20
this document has four recording modes discussed, which I think should=20
be discussed in this document.  Two are mentioned without really=20
explaining them, and two others are implied by various requirements.

[LeonP] Yes, for the protocol draft, we will need to combine all of the app=
roaches and make sure that we covered all use cases

Section 3:

" Recording Session: The session created between an RC and RS for the
   purpose of recording a Recorded Session.  The Recorded Session may
   itself be based on SIP, but it is a separate session from the
   Recording Session.

   Recorded Session: A session created between a UAC and UAS that is
   recorded by the RC and RS systems."


These two terms only differ by their tense, yet I think one refers to a=20
signaling session and the other refers to a media session.  If so,=20
better terms and definitions are needed.  If not, we need a term for the=20
actual recorded media.  Also, there are (at least) two sessions here -=20
the one between the UAs and the Recording Session - in a few places I=20
suspect the two are being confused.=20

[LeonP] Actually Recording Session relates to both SIP and Media that is fo=
rked to the recording server and Recorded Session is actually the original =
conversation between two UAs. These two are exactly the two sessions that y=
ou are describing.

"  Dynamic Recording: Dynamic recording is a mode of recording where the
   recording sessions are established on an as needed basis.  The length
   of these sessions is typically same as the length of the actual media
   sessions.

   Persistent Recording: Persistent recording is a mode of recording
   where the recording sessions are established at system start-up and
   kept-alive from that point on.  The length of these sessions is
   independent from the length of the actual recorded media sessions.
   Persistent recording sessions avoid issues such as media clipping
   that can occur due to delays in recording session establishment."

These terms need more explanation as I can think of multiple things they=20
might mean.  For example, when you say "session" do you mean a signaling=20
session or media session?  When you talk about avoiding media clipping,=20
do you mean that media is always sent by the RC to the RS regardless of=20
whether there is any media?=20

[LeonP] Yes, by session we mean both legs: signaling and media. And in Pers=
istent Recording mode, the Recording SIP session supposed to stay connected=
 even there is not actual connected sessions for the specific recorded UA.=
=20

Figures could use Figure numbers and labels.  Also, it would be worth=20
showing which is the RC and which is the RS in them.

Section 5

"   o REQ-001 The mechanism MUST support the ability to perform
   persistent (always-on) or dynamic (on-demand) Recording Sessions."

I'm not sure that the Persistent and Dynamic Recording as defined above=20
correspond to the Always On and On Demand modes defined in=20
http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#section-4.

[LeonP] It is different in the way that the Recording Session is establishe=
d only once during initialization (or login events) and then only media is =
forwarded per each call in order to minimize the clipping.

"   o REQ-002 The mechanism MUST support the ability to perform
   persistent Recording Sessions, such that the connection and
   attributes of media in the Recording Session are not dynamically
   signaled for each Recorded Session before it can be recorded.  Call
   details and metadata will still be signaled, but can be post-
   correlated to the recorded media.  There will still need to be a
   means of correlating the recorded media connection/packets to the
   Recorded Session, however this may be on a permanent filter-type
   basis, such as based on a SIP AoR of an agent that is always
   recorded, or based on identifiers in the recorded media itself."


Without fully understanding Persistent Recording, I can't evaluate this=20
requirement.=20

"   o REQ-009 The mechanism MUST support the ability to deliver mixed
   media streams to the RS.  The RS MAY be informed about the
   composition of the mixed streams through session metadata.

   Note: A mixed media stream is where several recorded sessions are
   carried in a single recording session.  A mixed media stream is
   typically produced by a mixer function."


I'm not clear what mixed media means here - does it mean a mix of media=20
types - synchronized audio and video from the same session, or do you=20
mean multiple media streams from different sessions?  The note seems to=20
indicate that this relates only to a conference call where multiple=20
sources (SSRCs) have been combined into a single RTP session.

 [LeonP] It is specific trading floor environment requirements where multip=
le handset and speakers for specific turrets will be mixed on the turret le=
vel for recording by one recording channel.


"   o REQ-013 The mechanism MUST support the ability to inject tones into
   the Recorded Session either from the RS or from the RC."

Recorded Session seems to be defined as the media between the RC and=20
RS.  Is that where the injected tones should be?  Or should then be=20
mixed in the media between the two UAs?
[LeonP] we have multiple options here. Preferable solution that one of the =
UAs will inject them (triggered by policy or parameters in SRP), but there =
are also configuration where RS will be required to create them and send ba=
ck to RC (like talking clock for example)

"   o REQ-021 The mechanism MUST support the ability for the RC to only
   perform media replication to the RS, without needing to decode or mix
   audio/video/etc., and without needing to be an RTP agent.  This will
   allow the RC to replicate media at layers below RTP.  Clearly, such
   an RC mode would not be able to provide transcoding, or media
   injection from the RS back into the Recorded Session."

Does this mean that you think that RTP is unsuitable for transporting=20
media between the RS and RC?  Clearly it isn't for IM, but I would think=20
that many of the functions of RTP are needed for audio and video media. =20
To understand this requirement will take more discussion.

[LeonP] Actually it was Hadriel requirement. I believe it only means that R=
C does not required to implement RTP stack, it can just fork RTP packets on=
 the network level (by hardware)

Thanks,
Alan
_______________________________________________
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From gonzalo.camarillo@ericsson.com  Sun Jul 26 04:22:12 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: Henry Sinnreich <hsinnrei@adobe.com>, dispatch@ietf.org
Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi,

> I guess the one way I could buy into your "distributed" approach is to 
> simply model it as a conference, where the focus is at one "end". But I 
> think we discussed that once.

that is how it is modeled when only one of the ends is distributed. What 
we discussed was that the semantics of join were intended for 
conferences and that even though this was very similar, it was not the 
same thing... but the model is valid.

Cheers,

Gonzalo

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Thread-Topic: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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From: "Francois Audet" <audet@nortel.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Alan Johnston" <alan@sipstation.com>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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I agree with both Alan and Hadriel.=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Hadriel Kaplan
> Sent: Saturday, July 25, 2009 13:28
> To: Alan Johnston; Dean Willis
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] CCUS and History-Info (was Proposed=20
> Charter for CCUS - Call Control UUI for SIP)
>=20
>=20
> I think the retargeting and middlebox support issues are not=20
> really germane as it pertains to H-I.  H-I records the=20
> req-uri's even through retargeting, so the UUI would be=20
> available to the UAS.  And it would survive even if Proxies=20
> don't support H-I, because the first UAC can insert the first=20
> H-I of the req-uri, which would simply be mandated for=20
> UUI-supporting UAC's if we went that way.=20
>=20
> But the real problem is the semantics as you say - to me a=20
> request URI is information about the targeted resource or how=20
> to reach it, not information about the source resource. =20
> Since H-I is just a list of req-uri's, it's not the right place.
>=20
> Really the technically *right* place would be in MIME, afaict=20
> - it's opaque application data just like geoloc, right?  But=20
> pragmatically, the issues/problems with doing it in MIME were=20
> already covered and debated in sipping, and iirc consensus=20
> was: put it in a header.
>=20
> ISTM that we always complain about lack of running code, and=20
> here we have a case where there is running code from multiple=20
> vendors, and it's not fundamentally broken/un-sound.  It=20
> ain't broken, why do we need to fix it?
>=20
> -hadriel
>=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On=20
> > Behalf Of Alan Johnston
> > Sent: Saturday, July 25, 2009 10:45 AM
> >=20
> > In terms of History-Info, I don't think we've considered=20
> that before.
> > There were two problems with the URI parameters.  One was the=20
> > retargeting problem but the other is semantics.  Last time=20
> I read RFC=20
> > 3261, the Request-URI contained information about the destination=20
> > resource, and sometimes information about the next hop to that=20
> > resource.  To put user-to-user data about the originator in the=20
> > request into the Request-URI seems like a big change to SIP=20
> semantics=20
> > that we'd need to think carefully about before diving into.=20
>  We would=20
> > definitely need a SIP option tag to say that we are reversing the=20
> > meaning of Request-URI.
> >=20
> > As for History-Info, it doesn't feel very right - how is=20
> UUI history=20
> > about the call?  If anything, it is future information=20
> about the call.
> > Also, wouldn't every proxy that might retarget in the path have to=20
> > support it, or the UUI would be lost?  If so, we'd need to use
> > Proxy-Require: history which is both ugly and not permitted=20
> by RFC 4244.
> >=20
> > Anyway, I look forward to more mechanism discussions in the future,=20
> > but for now lets focus on the charter discussions.
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From alan@sipstation.com  Mon Jul 27 01:07:48 2009
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Subject: Re: [dispatch] Comments on	draft-jain-dispatch-session-recording-protocol-req-00
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Leon,

Thanks for your replies - see a few comments below.

Thanks,
Alan

Leon Portman wrote:
> Alan, Hi
>
> Thanks for the comments.
>
> Please see below my answers
>
> Regards
>
> Leon
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
> Sent: Friday, July 24, 2009 9:29 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Comments on draft-jain-dispatch-session-recording-protocol-req-00
>
> Here are my comments on this draft.
>
> Overall, I think this draft is an excellent start at the requirements 
> for this important topic.  I look forward to future discussions of 
> charter text so we can get started on the work.
>
> Section 2:
>
> "  Another, related IETF draft is draft-wing-sipping-srtp-key
>    [I-D.wing-sipping-srtp-key], which describes an approach for
>    providing SRTP session keys to a recorder-type server.  That draft
>    focuses on the mechanism to provide the SRTP session key, rather than
>    the mechanism to invoke and sustain recording sessions themselves."
>
> draft-wing-sipping-srtp-key actually has a lot of discussion, including 
> call flows showing how recording sessions can be established.  Also, 
> this document has four recording modes discussed, which I think should 
> be discussed in this document.  Two are mentioned without really 
> explaining them, and two others are implied by various requirements.
>
> [LeonP] Yes, for the protocol draft, we will need to combine all of the approaches and make sure that we covered all use cases
>
> Section 3:
>
> " Recording Session: The session created between an RC and RS for the
>    purpose of recording a Recorded Session.  The Recorded Session may
>    itself be based on SIP, but it is a separate session from the
>    Recording Session.
>
>    Recorded Session: A session created between a UAC and UAS that is
>    recorded by the RC and RS systems."
>
>
> These two terms only differ by their tense, yet I think one refers to a 
> signaling session and the other refers to a media session.  If so, 
> better terms and definitions are needed.  If not, we need a term for the 
> actual recorded media.  Also, there are (at least) two sessions here - 
> the one between the UAs and the Recording Session - in a few places I 
> suspect the two are being confused. 
>
> [LeonP] Actually Recording Session relates to both SIP and Media that is forked to the recording server and Recorded Session is actually the original conversation between two UAs. These two are exactly the two sessions that you are describing.
>   
OK - I didn't get that from the definitions, so you probably want to 
reword these and maybe reference a figure. 


> "  Dynamic Recording: Dynamic recording is a mode of recording where the
>    recording sessions are established on an as needed basis.  The length
>    of these sessions is typically same as the length of the actual media
>    sessions.
>
>    Persistent Recording: Persistent recording is a mode of recording
>    where the recording sessions are established at system start-up and
>    kept-alive from that point on.  The length of these sessions is
>    independent from the length of the actual recorded media sessions.
>    Persistent recording sessions avoid issues such as media clipping
>    that can occur due to delays in recording session establishment."
>
> These terms need more explanation as I can think of multiple things they 
> might mean.  For example, when you say "session" do you mean a signaling 
> session or media session?  When you talk about avoiding media clipping, 
> do you mean that media is always sent by the RC to the RS regardless of 
> whether there is any media? 
>
> [LeonP] Yes, by session we mean both legs: signaling and media. And in Persistent Recording mode, the Recording SIP session supposed to stay connected even there is not actual connected sessions for the specific recorded UA. 
>   

OK, you need to reword these definitions to this is very clear.

> Figures could use Figure numbers and labels.  Also, it would be worth 
> showing which is the RC and which is the RS in them.
>
> Section 5
>
> "   o REQ-001 The mechanism MUST support the ability to perform
>    persistent (always-on) or dynamic (on-demand) Recording Sessions."
>
> I'm not sure that the Persistent and Dynamic Recording as defined above 
> correspond to the Always On and On Demand modes defined in 
> http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#section-4.
>
> [LeonP] It is different in the way that the Recording Session is established only once during initialization (or login events) and then only media is forwarded per each call in order to minimize the clipping.
>   

That's what I thought you meant, and this is different from the 
definitions in my draft.  We should try to get these definitions sorted 
out this week.

> "   o REQ-002 The mechanism MUST support the ability to perform
>    persistent Recording Sessions, such that the connection and
>    attributes of media in the Recording Session are not dynamically
>    signaled for each Recorded Session before it can be recorded.  Call
>    details and metadata will still be signaled, but can be post-
>    correlated to the recorded media.  There will still need to be a
>    means of correlating the recorded media connection/packets to the
>    Recorded Session, however this may be on a permanent filter-type
>    basis, such as based on a SIP AoR of an agent that is always
>    recorded, or based on identifiers in the recorded media itself."
>
>
> Without fully understanding Persistent Recording, I can't evaluate this 
> requirement. 
>
> "   o REQ-009 The mechanism MUST support the ability to deliver mixed
>    media streams to the RS.  The RS MAY be informed about the
>    composition of the mixed streams through session metadata.
>
>    Note: A mixed media stream is where several recorded sessions are
>    carried in a single recording session.  A mixed media stream is
>    typically produced by a mixer function."
>
>
> I'm not clear what mixed media means here - does it mean a mix of media 
> types - synchronized audio and video from the same session, or do you 
> mean multiple media streams from different sessions?  The note seems to 
> indicate that this relates only to a conference call where multiple 
> sources (SSRCs) have been combined into a single RTP session.
>
>  [LeonP] It is specific trading floor environment requirements where multiple handset and speakers for specific turrets will be mixed on the turret level for recording by one recording channel.
>   

OK - since this mixing is done by the endpoint (turret), this is an 
example of  where the UA is acting as the RC, right?  This use case 
should be described in the document.

>
> "   o REQ-013 The mechanism MUST support the ability to inject tones into
>    the Recorded Session either from the RS or from the RC."
>
> Recorded Session seems to be defined as the media between the RC and 
> RS.  Is that where the injected tones should be?  Or should then be 
> mixed in the media between the two UAs?
> [LeonP] we have multiple options here. Preferable solution that one of the UAs will inject them (triggered by policy or parameters in SRP), but there are also configuration where RS will be required to create them and send back to RC (like talking clock for example)
>   

So, we are talking about the RS sending media to the RC which must be 
inserted into the media session between the two UAs?  This use case 
needs to be discussed elsewhere outside of this requirement.

> "   o REQ-021 The mechanism MUST support the ability for the RC to only
>    perform media replication to the RS, without needing to decode or mix
>    audio/video/etc., and without needing to be an RTP agent.  This will
>    allow the RC to replicate media at layers below RTP.  Clearly, such
>    an RC mode would not be able to provide transcoding, or media
>    injection from the RS back into the Recorded Session."
>
> Does this mean that you think that RTP is unsuitable for transporting 
> media between the RS and RC?  Clearly it isn't for IM, but I would think 
> that many of the functions of RTP are needed for audio and video media.  
> To understand this requirement will take more discussion.
>
> [LeonP] Actually it was Hadriel requirement. I believe it only means that RC does not required to implement RTP stack, it can just fork RTP packets on the network level (by hardware)
>   

Yes, I think I get that when I reread the requirement.  This is a very 
important requirement that has a huge impact on the scalability of the 
system.

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


From dean.willis@softarmor.com  Mon Jul 27 01:42:31 2009
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Alan Johnston wrote:
> Dean,
> 
> Thanks for your comments.
> 
> We did consider URI parameters as a mechanism.  If we used this
> approach, we would still need SIP extensions for those URI parameters
> and a new SIP option tag to indicate support for it.  As such, we'd
> still need a mechanism to be standardized.  I think in DISPATCH we are
> supposed to be discussing scope and charter text, and I think the
> current charter text is mechanism agnostic, so this approach would still
> be covered, correct?

Part of what DISPATCH has to do is decide whether the problem requires
substantive new specification to perform, and if so, to shape the
charter of that work.

My belief is that if SIP provides a mechanism for user-to-user parameter
passing as an inherent mechanism, then call-control UUI should use that
mechanism. This means that there is no need for a SIP-extending working
group to be formed to complete that work. Instead, it would be
appropriate to document the parameter names and values using an
individual draft.

So I think my comments are very appropriate to the charter discussion in
 DISPATCH. I'm saying that we do NOT need to charter a working group for
this work as I understand it. More strongly said, since History-Info is
our currently agreed mechanism for meeting the SIPCORE charter
deliverable of UAC to UAS parameter passing, and since ISDN call control
proposes a new method of UAC to UAS parameter passing, then THIS
PROPSOED WORK IS IN DIRECT CONFLICT WITH CHARTERED WORK IN SIPCORE.

Now, it is arguable that we don't really have an effective mechanism for
parameter passing using History-Info. If so, then the charter we should
be talking about is for a generic UAC to UAS parameter passing
mechanism, which call-control UUI could then utilize. This work could be
done in SIPCORE (in fact, I believe that's the place for it, as we're
talking about a very central and universal aspect of SIP), or in a new
working group. That conversation we can certainly have in DISPATCH.

> 
> In terms of History-Info, I don't think we've considered that before. 
> There were two problems with the URI parameters.  One was the
> retargeting problem but the other is semantics.  Last time I read RFC
> 3261, the Request-URI contained information about the destination
> resource, and sometimes information about the next hop to that
> resource.  To put user-to-user data about the originator in the request
> into the Request-URI seems like a big change to SIP semantics that we'd
> need to think carefully about before diving into.  We would definitely
> need a SIP option tag to say that we are reversing the meaning of
> Request-URI.

...

> As for History-Info, it doesn't feel very right - how is UUI history
> about the call?  If anything, it is future information about the call. 
> Also, wouldn't every proxy that might retarget in the path have to
> support it, or the UUI would be lost?  If so, we'd need to use
> Proxy-Require: history which is both ugly and not permitted by RFC 4244.

We currently have SIPCORE working group consensus on History-Info being
the mechanism for delivery of user-supplied data from UAC to UAS, that
having been the goal of the original UA Loose route work.

This is not a "reversal of request-URI semantics"; rather it is the
original intent of the SIP Request URI, which is modeled on the HTTP
request URI. The HTTP request URI is HTTP's normal mechanism for the
delivery of application data from UAC to UAS. HTTP server applications
don't have ready access to new HTTP headers (one would have to rewrite
Apache for every such protocol extension), but they do have easy access
to the request URI.

There's nothing special about ISDN call control data in its interaction
with SIP. It does NOT interact with the SIP state machine in any way.
Extending SIP explicitly to support it is a serious violation of
protocol layering. Admittedly, SIP is full of such cross-layer issues,
but why do we insist on adding more?

Call Control UUI is a "poster child", or in other words a perfect use
case, for UAC to UAS parameter passing. If our UAC to UAS
parameter-passing mechanism doesn't work for it, then we need to back up
and take another look at the parameter-passing mechanism rather than
continue to extend SIP with application-specific usages that should be
outside of its scope.

--
Dean

From dean.willis@softarmor.com  Mon Jul 27 01:50:06 2009
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Hadriel Kaplan wrote:

> I think the retargeting and middlebox support issues are not really
> But the real problem is the semantics as you say - to me a request
> URI is information about the targeted resource or how to reach it,
> not information about the source resource.  Since H-I is just a list
> of req-uri's, it's not the right place.

I disagree. SIP is based on HTTP The Request URI encodes the request
being made by the UAC to the UAS. The CCUUI data is not germane to SIP,
it's just stuff being passed along the chain from UAC to UAS.

Now, it might be reasonable to universally declare that SIP is, in fact,
NOT HTTP, and that it uses a different parameter passing mechanism. If
so, then what we would probably want to do is define a new header field,
perhaps "User-Parameters:", that is reserved for user-to-user data
consumed by applications. I'd be okay with that.


> Really the technically *right* place would be in MIME, afaict - it's
> opaque application data just like geoloc, right?  But pragmatically,
> the issues/problems with doing it in MIME were already covered and
> debated in sipping, and iirc consensus was: put it in a header.
> 
> ISTM that we always complain about lack of running code, and here we
> have a case where there is running code from multiple vendors, and
> it's not fundamentally broken/un-sound.  It ain't broken, why do we
> need to fix it?

The same reason we shoud object to all layer-violation hacks:
because it is architecturally unsound, inconsistent with the current SIP
design approach, and leads to further unnecessary complexity in an
already too-complex protocol.


--
Dean

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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Dean,

So basically, you want a different name for the header field?   
Otherwise the semantics seem identical to what is proposed.

Thanks,
Alan



On Jul 27, 2009, at 10:48 AM, Dean Willis <dean.willis@softarmor.com>  
wrote:

> Hadriel Kaplan wrote:
>
>> I think the retargeting and middlebox support issues are not really
>> But the real problem is the semantics as you say - to me a request
>> URI is information about the targeted resource or how to reach it,
>> not information about the source resource.  Since H-I is just a list
>> of req-uri's, it's not the right place.
>
> I disagree. SIP is based on HTTP The Request URI encodes the request
> being made by the UAC to the UAS. The CCUUI data is not germane to  
> SIP,
> it's just stuff being passed along the chain from UAC to UAS.
>
> Now, it might be reasonable to universally declare that SIP is, in  
> fact,
> NOT HTTP, and that it uses a different parameter passing mechanism. If
> so, then what we would probably want to do is define a new header  
> field,
> perhaps "User-Parameters:", that is reserved for user-to-user data
> consumed by applications. I'd be okay with that.
>
>
>> Really the technically *right* place would be in MIME, afaict - it's
>> opaque application data just like geoloc, right?  But pragmatically,
>> the issues/problems with doing it in MIME were already covered and
>> debated in sipping, and iirc consensus was: put it in a header.
>>
>> ISTM that we always complain about lack of running code, and here we
>> have a case where there is running code from multiple vendors, and
>> it's not fundamentally broken/un-sound.  It ain't broken, why do we
>> need to fix it?
>
> The same reason we shoud object to all layer-violation hacks:
> because it is architecturally unsound, inconsistent with the current  
> SIP
> design approach, and leads to further unnecessary complexity in an
> already too-complex protocol.
>
>
> --
> Dean

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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Monday, July 27, 2009 4:49 AM
> To: Hadriel Kaplan
>=20
> I disagree. SIP is based on HTTP The Request URI encodes the request
> being made by the UAC to the UAS.=20

Then you might as well argue the request-URI should encode MIME bodies.  Th=
e UUI is not a part of the resource being addressed in any way, and thus no=
t a part of a Uniform Resource Identifier.

=20
> Now, it might be reasonable to universally declare that SIP is, in fact,
> NOT HTTP, and that it uses a different parameter passing mechanism.=20

Well, I'll probably be flamed, but afaict it is not HTTP in most every impo=
rtant way.  It does a LOT of things differently.  That ship has sailed.  It=
 is far closer to email (with some weird combo of SMTP and POP3), if anythi=
ng.=20


> If
> so, then what we would probably want to do is define a new header field,
> perhaps "User-Parameters:", that is reserved for user-to-user data
> consumed by applications. I'd be okay with that.

That was actually how I interpreted the semantic notion of the User-to-User=
 header.

-hadriel

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Subject: Re: [dispatch] I-D Action:draft-johnston-sipping-cc-uui-08.txt
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Keith,

If this is *not* ISDN encoding, then where is the indication of what 
encoding is being used? Are you saying that this is a private agreement 
between the UAC and UAS, neither signaled nor negotiated? How can that 
possibly work?

I thought the first byte of the value was a registered format indicator.

	Thanks,
	Paul

DRAGE, Keith (Keith) wrote:
> Just to respond to this bit of the message:
> 
>>> Not quite.  We are not making the assumption that both are 
>> gateways, 
>>> although that is possible.  Most likely, one element is SIP enabled.
>>> However, it is not a fully intelligent SIP endpoint - the basic 
>>> software and application (probably many years old) is 
>> unchanged but a 
>>> SIP interface (SIP trunk) has been added.  As such, the SIP 
>> stack does 
>>> not know anything about the UUI as it is implementing 
>> exactly the ISDN 
>>> UUI service - the information is just pushed up the stack 
>> to another 
>>> application, and this application knows nothing about SIP.  
>> It still 
>>> thinks it is using an ISDN trunk.
>> Then in the way I meant the question, both ends *are* gateways.
>>
>> But then if someone wishes to build native sip gear to plug 
>> into this environment, then it will still have to understand 
>> this isdn encoding. 
>> Perhaps that is the cruel truth, unless all the equipment is 
>> replaced at once. 
> 
> I've said this before and I'll say it again. The bit being delivered to the end UE is not "isdn encoding". If it was ISDN encoding, then the ISDN / SIP gateway would be able to interpret it and map it into SIP. Rather it is some application sitting above the ISDN call control layer in the calling terminal, which is wishing to communicate with its peer application running in the destination SIP UA, or gateway.
> 
> regards
> 
> Keith
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: Saturday, July 04, 2009 12:39 AM
>> To: Alan Johnston
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] I-D 
>> Action:draft-johnston-sipping-cc-uui-08.txt
>>
>> Hi Alan,
>>
>> responses inline. I've trimmed out the parts that don't 
>> require more discussion.
>>
>> 	Thanks,
>> 	Paul
>>
>> Alan Johnston wrote:
>>> Paul,
>>>
>>> Thanks for your comments.  See mine below.
>>>
>>> Thanks,
>>> Alan
>>>
>>> Paul Kyzivat wrote:
>>>> Hi Alan,
>>>>
>>>> I took a quick look at this doc and the requirements doc 
>> as well. I 
>>>> have a few thoughts:
>>>>
>>>> From section 1:
>>>>
>>>>>    In the future, where both endpoints are intelligent 
>> SIP user agents,
>>>>>    it may be possible for them to understand and 
>> interpret the UUI data.
>>>>>    There may be some cases where the UUI information is 
>> relevant to SIP.
>>>>>    In this case, it might be worthwhile attempting to map 
>> UUI data to an
>>>>>    appropriate SIP header field or to standardize a new 
>> header field.
>>>>>    However, the requirements and use cases for this are 
>> different enough
>>>>>    from those described in this document that these two situations
>>>>>    should be examined separately.  This document looks only at the
>>>>>    requirements and mechanisms for replicating the 
>> existing, widely used
>>>>>    and deployed ISDN UUI service.
>>>> This *still* troubles me!
>>>> Is the assumption here that *both* the UAC and UAS are 
>> gateways, so 
>>>> that this mechanism is solely for the tunneling of Q.931 and Q.763 
>>>> UUI message over SIP?
>>>>
>>>> I get the impression that an important use case is where at least
>>>> *one* end (probably the UAS) is a SIP device. In that case it will 
>>>> already be *necessary* for it to understand and interpret UUI data.
>>>> And where there is redundancy between SIP data and the UUI 
>> data, it 
>>>> will need to resolve the discrepancy.
>>> Not quite.  We are not making the assumption that both are 
>> gateways, 
>>> although that is possible.  Most likely, one element is SIP enabled.
>>> However, it is not a fully intelligent SIP endpoint - the basic 
>>> software and application (probably many years old) is 
>> unchanged but a 
>>> SIP interface (SIP trunk) has been added.  As such, the SIP 
>> stack does 
>>> not know anything about the UUI as it is implementing 
>> exactly the ISDN 
>>> UUI service - the information is just pushed up the stack 
>> to another 
>>> application, and this application knows nothing about SIP.  
>> It still 
>>> thinks it is using an ISDN trunk.
>> Then in the way I meant the question, both ends *are* gateways.
>>
>> But then if someone wishes to build native sip gear to plug 
>> into this environment, then it will still have to understand 
>> this isdn encoding. 
>> Perhaps that is the cruel truth, unless all the equipment is 
>> replaced at once.
>>
>>>>> 5.  Syntax for UUI Header Field
>>>> ...
>>>>>        UUI         = "User-to-User" HCOLON uui-data 
>> *(SEMI uui-param)
>>>>>        uui-data    = token
>>>>>        uui-param   = enc-param | cont-param | purp-param 
>> | generic-param
>>>>>        enc-param   = "encoding=" ("hex" | token)
>>>>>        cont-param  = "content=" ("isdn-uui" | token)
>>>>>        purp-param  = "purpose=" ("isdn-interwork" | token)
>>>> At the very least, I would request that this be expanded a 
>> little bit 
>>>> for future flexibility, by permitting the uui-data to be either a 
>>>> token or a string. While the encoding you have chosen works as a 
>>>> token, other encodings may need the additional flexibility 
>> of a string.
>>>>        uui-data    = token | quoted-string
>>> OK, I guess I'm OK with that.  For the isdn-interwork 
>> purpose, we'll 
>>> require the token, but other applications could utilize the 
>> quoted string.
>>
>> I would hope that when the token encoding is sufficient, then 
>> either form would be accepted. Or it would perhaps be ok to 
>> specify for a particular encoding (hex) that the token 
>> encoding must be used. I don't see how the purpose has 
>> anything to do with it.
>>
>>>>> 6.3.  Registration of SIP Option Tag
>>>> This section registers the new option tag. But I find 
>> almost nothing 
>>>> that defines the semantics of the option. (Section 5 does 
>> include the
>>>> following:
>>>>
>>>>>    A UA that supports this feature and the "uui" option 
>> tag MUST support
>>>>>    the call flows in [johnston-dispatch-sip-cc-uui]...
>>>> but that is pretty thin from a normative perspective.
>>>>
>>>> Of particular note, does support for the option tag mean 
>> support for 
>>>> the particular encoding, content, and purpose included in this 
>>>> document? Or does it only mean support for the header and 
>> params in 
>>>> the abstract? (That wouldn't be very useful.)
>>> Yes, this needs clarification.  I'd suggest we define the 
>> option tag 
>>> to be uui-isdn which means it understands the header field and the 
>>> isdn-interwork purpose, and the call flows, including escaping into 
>>> redirection and Refer-To URIs.
>> I'd like to hear some other opinions on whether to have tag 
>> per param for tag, or a tag for the combination of param 
>> values. I think I'm ok with what you propose.
>>
>>>> I don't see how it can mean support for some other yet to 
>> be defined 
>>>> values for those. Yet if it only means support for the 
>> current ones, 
>>>> then how will one indicate support for future ones?
>>>>
>>>> The only way out of this that I can see is to have 
>> separate options, 
>>>> either for particular values of each parameter, or for particular 
>>>> combinations of values of all the parameters.
>>> New applications using this header field would have the option of 
>>> defining a new SIP option tag which would mean support for 
>> the header 
>>> field and their new purpose.
>>>
>>>> So you might have options uui-purpose-isdn-interwork, 
>>>> uui-content-isdn, and uui-encoding-hex. Or you might just 
>> have option 
>>>> uui-isdn-interwork that means the combination.
>>> I think the latter.  I see it as a hierarchy - the application or 
>>> purpose is the top one.  Each purpose has a number of 
>> contents that it 
>>> supports.  Each content has a number of encoding schemes it 
>> supports.
>>
>> I guess I'm ok with that, pending nailing down the definition 
>> of "purpose".
>>
>>>> I also am not finding much explanation of the semantics of the 
>>>> content and purpose parameters. I can only extrapolate 
>> from a single 
>>>> example of each, which I'm having trouble doing.
>>>>
>>>> I'm guessing that 'content' must refer to the precise 
>> syntax of the 
>>>> contained data, after decoding. So presumably it must map 
>> onto some 
>>>> particular spec. For isdn-uui, would that be [Q931] or 
>> [Q763]? If so, 
>>>> shouldn't it refer to something specific in that document?
>>> Not really.  In this case, isdn-uui effectively means "opaque data" 
>>> which is how ISDN defines the UUI - it is completely undefined, 
>>> necessarily so  by the strict layering used in the ISDN application.
>> I don't believe you!
>>
>> If that is the case, then the UAC and UAS must have a private 
>> side agreement about what the isdn-uui content means. Is that 
>> really what you mean?
>>
>> You already state that the first byte is a protocol 
>> discriminator. There must also be something, somewhere, that 
>> specifies the mapping from a particular value for that byte 
>> to a particular protocol/format. If there is, then that 
>> should be part of the definition of this content.
>>
>>>> 'purpose' is even less clear to me. Does it refer to why 
>> it is being 
>>>> included? Or what is to be done with it? If received by a 
>> UA that is 
>>>> not an ISDN gateway, how can it decide if this is 
>> something it should 
>>>> be trying to process? Is it still ISDN interworking if neither the 
>>>> UAC nor UAS is connected to an ISDN network?
>>> The purpose is the application using the UUI.  Others have 
>> expressed 
>>> an interest in carrying other data in the header field, in 
>> which case 
>>> a new purpose would be defined.  I'll see if I can think of 
>> an example one.
>>
>> So are you saying that a particular call center application 
>> might constitute a distinct "purpose"?
>>
>>> Basically, if two UAs both support the header but have different 
>>> applications generating and consuming the UUI, it will not 
>> work.  This 
>>> is not a SIP failure, and there is nothing SIP can do about 
>> it.  But 
>>> this purpose header field allows the UAs to detect this condition.
>> In that case, perhaps "isdn-interwork" isn't really a single 
>> purpose at all, unless any two pieces of equipment that 
>> support that purpose can then interwork without knowing any 
>> more about each other.
>>
>>>> Suppose I was developing a sophisticated UA that 
>> potentially might be 
>>>> usable by a call center operator. How should 
>> purpose=isdn-interwork 
>>>> affect the operation of this device?
>>> To make it work in today's contact center applications, supporting 
>>> isdn-interwork would likely make it work in an application, 
>> provided 
>>> the appropriate call center application also ran on it.  
>> Some contact 
>>> centers do not use ISDN UUI, instead they use all kinds of really, 
>>> really ugly CTI protocols running in parallel.
>> Yes, I know! :-(
>>
>>> This is a way of moving
>>> those to the ISDN model, and from there to a more 
>> intelligent endpoint 
>>> SIP model.
>>>
>>>>     Thanks,
>>>>     Paul
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

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Subject: Re: [dispatch] Disaggregated Media in SIP
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Gonzalo Camarillo wrote:
> Hi,
> 
>> I guess the one way I could buy into your "distributed" approach is to 
>> simply model it as a conference, where the focus is at one "end". But 
>> I think we discussed that once.
> 
> that is how it is modeled when only one of the ends is distributed. What 
> we discussed was that the semantics of join were intended for 
> conferences and that even though this was very similar, it was not the 
> same thing... but the model is valid.

If the model is valid, then I think the mechanisms that go with it 
should be valid too.

The conferencing mechanism could be used for either the "distributed" or 
"centralized" approach. Then there needs to be a focus somewhere, and 
the difference between the approaches is where it resides - at your own 
end, or at the "other" end.

	Thanks,
	Paul

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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Alan Johnston wrote:
> Dean,
> 
> So basically, you want a different name for the header field?  Otherwise 
> the semantics seem identical to what is proposed.

There at least needs to be some interoperable way to specify the format 
of the data being passed. We have that with MIME, but when using a 
header the mechanism needs to be reinvented, along with a way to say "I 
don't understand that".

	Thanks,
	Paul

> Thanks,
> Alan
> 
> 
> 
> On Jul 27, 2009, at 10:48 AM, Dean Willis <dean.willis@softarmor.com> 
> wrote:
> 
>> Hadriel Kaplan wrote:
>>
>>> I think the retargeting and middlebox support issues are not really
>>> But the real problem is the semantics as you say - to me a request
>>> URI is information about the targeted resource or how to reach it,
>>> not information about the source resource.  Since H-I is just a list
>>> of req-uri's, it's not the right place.
>>
>> I disagree. SIP is based on HTTP The Request URI encodes the request
>> being made by the UAC to the UAS. The CCUUI data is not germane to SIP,
>> it's just stuff being passed along the chain from UAC to UAS.
>>
>> Now, it might be reasonable to universally declare that SIP is, in fact,
>> NOT HTTP, and that it uses a different parameter passing mechanism. If
>> so, then what we would probably want to do is define a new header field,
>> perhaps "User-Parameters:", that is reserved for user-to-user data
>> consumed by applications. I'd be okay with that.
>>
>>
>>> Really the technically *right* place would be in MIME, afaict - it's
>>> opaque application data just like geoloc, right?  But pragmatically,
>>> the issues/problems with doing it in MIME were already covered and
>>> debated in sipping, and iirc consensus was: put it in a header.
>>>
>>> ISTM that we always complain about lack of running code, and here we
>>> have a case where there is running code from multiple vendors, and
>>> it's not fundamentally broken/un-sound.  It ain't broken, why do we
>>> need to fix it?
>>
>> The same reason we shoud object to all layer-violation hacks:
>> because it is architecturally unsound, inconsistent with the current SIP
>> design approach, and leads to further unnecessary complexity in an
>> already too-complex protocol.
>>
>>
>> -- 
>> Dean
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 27 Jul 2009 16:12:33 +0200
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1st octet of the payload.

Keith=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Monday, July 27, 2009 2:14 PM
> To: DRAGE, Keith (Keith)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] I-D=20
> Action:draft-johnston-sipping-cc-uui-08.txt
>=20
> Keith,
>=20
> If this is *not* ISDN encoding, then where is the indication=20
> of what encoding is being used? Are you saying that this is a=20
> private agreement between the UAC and UAS, neither signaled=20
> nor negotiated? How can that possibly work?
>=20
> I thought the first byte of the value was a registered format=20
> indicator.
>=20
> 	Thanks,
> 	Paul
>=20
> DRAGE, Keith (Keith) wrote:
> > Just to respond to this bit of the message:
> >=20
> >>> Not quite.  We are not making the assumption that both are
> >> gateways,
> >>> although that is possible.  Most likely, one element is=20
> SIP enabled.
> >>> However, it is not a fully intelligent SIP endpoint - the basic=20
> >>> software and application (probably many years old) is
> >> unchanged but a
> >>> SIP interface (SIP trunk) has been added.  As such, the SIP
> >> stack does
> >>> not know anything about the UUI as it is implementing
> >> exactly the ISDN
> >>> UUI service - the information is just pushed up the stack
> >> to another
> >>> application, and this application knows nothing about SIP. =20
> >> It still
> >>> thinks it is using an ISDN trunk.
> >> Then in the way I meant the question, both ends *are* gateways.
> >>
> >> But then if someone wishes to build native sip gear to=20
> plug into this=20
> >> environment, then it will still have to understand this isdn=20
> >> encoding.
> >> Perhaps that is the cruel truth, unless all the equipment=20
> is replaced=20
> >> at once.
> >=20
> > I've said this before and I'll say it again. The bit being=20
> delivered to the end UE is not "isdn encoding". If it was=20
> ISDN encoding, then the ISDN / SIP gateway would be able to=20
> interpret it and map it into SIP. Rather it is some=20
> application sitting above the ISDN call control layer in the=20
> calling terminal, which is wishing to communicate with its=20
> peer application running in the destination SIP UA, or gateway.
> >=20
> > regards
> >=20
> > Keith
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> >> Sent: Saturday, July 04, 2009 12:39 AM
> >> To: Alan Johnston
> >> Cc: dispatch@ietf.org
> >> Subject: Re: [dispatch] I-D
> >> Action:draft-johnston-sipping-cc-uui-08.txt
> >>
> >> Hi Alan,
> >>
> >> responses inline. I've trimmed out the parts that don't=20
> require more=20
> >> discussion.
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >> Alan Johnston wrote:
> >>> Paul,
> >>>
> >>> Thanks for your comments.  See mine below.
> >>>
> >>> Thanks,
> >>> Alan
> >>>
> >>> Paul Kyzivat wrote:
> >>>> Hi Alan,
> >>>>
> >>>> I took a quick look at this doc and the requirements doc
> >> as well. I
> >>>> have a few thoughts:
> >>>>
> >>>> From section 1:
> >>>>
> >>>>>    In the future, where both endpoints are intelligent
> >> SIP user agents,
> >>>>>    it may be possible for them to understand and
> >> interpret the UUI data.
> >>>>>    There may be some cases where the UUI information is
> >> relevant to SIP.
> >>>>>    In this case, it might be worthwhile attempting to map
> >> UUI data to an
> >>>>>    appropriate SIP header field or to standardize a new
> >> header field.
> >>>>>    However, the requirements and use cases for this are
> >> different enough
> >>>>>    from those described in this document that these two=20
> situations
> >>>>>    should be examined separately.  This document looks=20
> only at the
> >>>>>    requirements and mechanisms for replicating the
> >> existing, widely used
> >>>>>    and deployed ISDN UUI service.
> >>>> This *still* troubles me!
> >>>> Is the assumption here that *both* the UAC and UAS are
> >> gateways, so
> >>>> that this mechanism is solely for the tunneling of Q.931=20
> and Q.763=20
> >>>> UUI message over SIP?
> >>>>
> >>>> I get the impression that an important use case is where at least
> >>>> *one* end (probably the UAS) is a SIP device. In that=20
> case it will=20
> >>>> already be *necessary* for it to understand and=20
> interpret UUI data.
> >>>> And where there is redundancy between SIP data and the UUI
> >> data, it
> >>>> will need to resolve the discrepancy.
> >>> Not quite.  We are not making the assumption that both are
> >> gateways,
> >>> although that is possible.  Most likely, one element is=20
> SIP enabled.
> >>> However, it is not a fully intelligent SIP endpoint - the basic=20
> >>> software and application (probably many years old) is
> >> unchanged but a
> >>> SIP interface (SIP trunk) has been added.  As such, the SIP
> >> stack does
> >>> not know anything about the UUI as it is implementing
> >> exactly the ISDN
> >>> UUI service - the information is just pushed up the stack
> >> to another
> >>> application, and this application knows nothing about SIP. =20
> >> It still
> >>> thinks it is using an ISDN trunk.
> >> Then in the way I meant the question, both ends *are* gateways.
> >>
> >> But then if someone wishes to build native sip gear to=20
> plug into this=20
> >> environment, then it will still have to understand this isdn=20
> >> encoding.
> >> Perhaps that is the cruel truth, unless all the equipment=20
> is replaced=20
> >> at once.
> >>
> >>>>> 5.  Syntax for UUI Header Field
> >>>> ...
> >>>>>        UUI         =3D "User-to-User" HCOLON uui-data=20
> >> *(SEMI uui-param)
> >>>>>        uui-data    =3D token
> >>>>>        uui-param   =3D enc-param | cont-param | purp-param=20
> >> | generic-param
> >>>>>        enc-param   =3D "encoding=3D" ("hex" | token)
> >>>>>        cont-param  =3D "content=3D" ("isdn-uui" | token)
> >>>>>        purp-param  =3D "purpose=3D" ("isdn-interwork" | token)
> >>>> At the very least, I would request that this be expanded a
> >> little bit
> >>>> for future flexibility, by permitting the uui-data to be=20
> either a=20
> >>>> token or a string. While the encoding you have chosen works as a=20
> >>>> token, other encodings may need the additional flexibility
> >> of a string.
> >>>>        uui-data    =3D token | quoted-string
> >>> OK, I guess I'm OK with that.  For the isdn-interwork
> >> purpose, we'll
> >>> require the token, but other applications could utilize the
> >> quoted string.
> >>
> >> I would hope that when the token encoding is sufficient,=20
> then either=20
> >> form would be accepted. Or it would perhaps be ok to specify for a=20
> >> particular encoding (hex) that the token encoding must be used. I=20
> >> don't see how the purpose has anything to do with it.
> >>
> >>>>> 6.3.  Registration of SIP Option Tag
> >>>> This section registers the new option tag. But I find
> >> almost nothing
> >>>> that defines the semantics of the option. (Section 5 does
> >> include the
> >>>> following:
> >>>>
> >>>>>    A UA that supports this feature and the "uui" option
> >> tag MUST support
> >>>>>    the call flows in [johnston-dispatch-sip-cc-uui]...
> >>>> but that is pretty thin from a normative perspective.
> >>>>
> >>>> Of particular note, does support for the option tag mean
> >> support for
> >>>> the particular encoding, content, and purpose included in this=20
> >>>> document? Or does it only mean support for the header and
> >> params in
> >>>> the abstract? (That wouldn't be very useful.)
> >>> Yes, this needs clarification.  I'd suggest we define the
> >> option tag
> >>> to be uui-isdn which means it understands the header=20
> field and the=20
> >>> isdn-interwork purpose, and the call flows, including=20
> escaping into=20
> >>> redirection and Refer-To URIs.
> >> I'd like to hear some other opinions on whether to have=20
> tag per param=20
> >> for tag, or a tag for the combination of param values. I=20
> think I'm ok=20
> >> with what you propose.
> >>
> >>>> I don't see how it can mean support for some other yet to
> >> be defined
> >>>> values for those. Yet if it only means support for the
> >> current ones,
> >>>> then how will one indicate support for future ones?
> >>>>
> >>>> The only way out of this that I can see is to have
> >> separate options,
> >>>> either for particular values of each parameter, or for=20
> particular=20
> >>>> combinations of values of all the parameters.
> >>> New applications using this header field would have the option of=20
> >>> defining a new SIP option tag which would mean support for
> >> the header
> >>> field and their new purpose.
> >>>
> >>>> So you might have options uui-purpose-isdn-interwork,=20
> >>>> uui-content-isdn, and uui-encoding-hex. Or you might just
> >> have option
> >>>> uui-isdn-interwork that means the combination.
> >>> I think the latter.  I see it as a hierarchy - the application or=20
> >>> purpose is the top one.  Each purpose has a number of
> >> contents that it
> >>> supports.  Each content has a number of encoding schemes it
> >> supports.
> >>
> >> I guess I'm ok with that, pending nailing down the definition of=20
> >> "purpose".
> >>
> >>>> I also am not finding much explanation of the semantics of the=20
> >>>> content and purpose parameters. I can only extrapolate
> >> from a single
> >>>> example of each, which I'm having trouble doing.
> >>>>
> >>>> I'm guessing that 'content' must refer to the precise
> >> syntax of the
> >>>> contained data, after decoding. So presumably it must map
> >> onto some
> >>>> particular spec. For isdn-uui, would that be [Q931] or
> >> [Q763]? If so,
> >>>> shouldn't it refer to something specific in that document?
> >>> Not really.  In this case, isdn-uui effectively means=20
> "opaque data"=20
> >>> which is how ISDN defines the UUI - it is completely undefined,=20
> >>> necessarily so  by the strict layering used in the ISDN=20
> application.
> >> I don't believe you!
> >>
> >> If that is the case, then the UAC and UAS must have a private side=20
> >> agreement about what the isdn-uui content means. Is that=20
> really what=20
> >> you mean?
> >>
> >> You already state that the first byte is a protocol discriminator.=20
> >> There must also be something, somewhere, that specifies=20
> the mapping=20
> >> from a particular value for that byte to a particular=20
> >> protocol/format. If there is, then that should be part of the=20
> >> definition of this content.
> >>
> >>>> 'purpose' is even less clear to me. Does it refer to why
> >> it is being
> >>>> included? Or what is to be done with it? If received by a
> >> UA that is
> >>>> not an ISDN gateway, how can it decide if this is
> >> something it should
> >>>> be trying to process? Is it still ISDN interworking if=20
> neither the=20
> >>>> UAC nor UAS is connected to an ISDN network?
> >>> The purpose is the application using the UUI.  Others have
> >> expressed
> >>> an interest in carrying other data in the header field, in
> >> which case
> >>> a new purpose would be defined.  I'll see if I can think of
> >> an example one.
> >>
> >> So are you saying that a particular call center application might=20
> >> constitute a distinct "purpose"?
> >>
> >>> Basically, if two UAs both support the header but have different=20
> >>> applications generating and consuming the UUI, it will not
> >> work.  This
> >>> is not a SIP failure, and there is nothing SIP can do about
> >> it.  But
> >>> this purpose header field allows the UAs to detect this condition.
> >> In that case, perhaps "isdn-interwork" isn't really a=20
> single purpose=20
> >> at all, unless any two pieces of equipment that support=20
> that purpose=20
> >> can then interwork without knowing any more about each other.
> >>
> >>>> Suppose I was developing a sophisticated UA that
> >> potentially might be
> >>>> usable by a call center operator. How should
> >> purpose=3Disdn-interwork
> >>>> affect the operation of this device?
> >>> To make it work in today's contact center applications,=20
> supporting=20
> >>> isdn-interwork would likely make it work in an application,
> >> provided
> >>> the appropriate call center application also ran on it. =20
> >> Some contact
> >>> centers do not use ISDN UUI, instead they use all kinds=20
> of really,=20
> >>> really ugly CTI protocols running in parallel.
> >> Yes, I know! :-(
> >>
> >>> This is a way of moving
> >>> those to the ISDN model, and from there to a more
> >> intelligent endpoint
> >>> SIP model.
> >>>
> >>>>     Thanks,
> >>>>     Paul
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> =

From alan@sipstation.com  Mon Jul 27 07:43:40 2009
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Paul Kyzivat wrote:
>
>
> Alan Johnston wrote:
>> Dean,
>>
>> So basically, you want a different name for the header field?  
>> Otherwise the semantics seem identical to what is proposed.
>
> There at least needs to be some interoperable way to specify the 
> format of the data being passed. We have that with MIME, but when 
> using a header the mechanism needs to be reinvented, along with a way 
> to say "I don't understand that".
This is an excellent point.

The latest version of the mechanism draft has a way of specifying the 
purpose (application) and encoding, based on this need.

The requirements currently say that a UAC has to be able to discover if 
a UAS supports the UUI  *mechanism*.   The obvious way to do this is a 
SIP option tag.  But do we need a requirement for a UAC to discover if a 
UAS supports a particular purpose/application?  If the same UUI is 
encoded in different ways, do we need a way to say "you need to be able 
to support *one* of the data objects?

Thanks,
Alan


>
>     Thanks,
>     Paul
>
>> Thanks,
>> Alan
>>
>>
>>
>> On Jul 27, 2009, at 10:48 AM, Dean Willis <dean.willis@softarmor.com> 
>> wrote:
>>
>>> Hadriel Kaplan wrote:
>>>
>>>> I think the retargeting and middlebox support issues are not really
>>>> But the real problem is the semantics as you say - to me a request
>>>> URI is information about the targeted resource or how to reach it,
>>>> not information about the source resource.  Since H-I is just a list
>>>> of req-uri's, it's not the right place.
>>>
>>> I disagree. SIP is based on HTTP The Request URI encodes the request
>>> being made by the UAC to the UAS. The CCUUI data is not germane to SIP,
>>> it's just stuff being passed along the chain from UAC to UAS.
>>>
>>> Now, it might be reasonable to universally declare that SIP is, in 
>>> fact,
>>> NOT HTTP, and that it uses a different parameter passing mechanism. If
>>> so, then what we would probably want to do is define a new header 
>>> field,
>>> perhaps "User-Parameters:", that is reserved for user-to-user data
>>> consumed by applications. I'd be okay with that.
>>>
>>>
>>>> Really the technically *right* place would be in MIME, afaict - it's
>>>> opaque application data just like geoloc, right?  But pragmatically,
>>>> the issues/problems with doing it in MIME were already covered and
>>>> debated in sipping, and iirc consensus was: put it in a header.
>>>>
>>>> ISTM that we always complain about lack of running code, and here we
>>>> have a case where there is running code from multiple vendors, and
>>>> it's not fundamentally broken/un-sound.  It ain't broken, why do we
>>>> need to fix it?
>>>
>>> The same reason we shoud object to all layer-violation hacks:
>>> because it is architecturally unsound, inconsistent with the current 
>>> SIP
>>> design approach, and leads to further unnecessary complexity in an
>>> already too-complex protocol.
>>>
>>>
>>> -- 
>>> Dean
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>


From Even.roni@huawei.com  Mon Jul 27 08:00:33 2009
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Date: Mon, 27 Jul 2009 17:58:26 +0300
From: Roni Even <Even.roni@huawei.com>
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To: 'Alan Johnston' <alan@sipstation.com>, 'Leon Portman' <Leon.Portman@nice.com>
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Subject: Re: [dispatch] Comments	on	draft-jain-dispatch-session-recording-protocol-req-00
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Hi,
I am not sure how REQ-021 will work if you want to address packet loss and
control using RTCP. I suggest you look at the Topologies RFC 5117 for
directions
Roni Even
> 
> > "   o REQ-021 The mechanism MUST support the ability for the RC to
> only
> >    perform media replication to the RS, without needing to decode or
> mix
> >    audio/video/etc., and without needing to be an RTP agent.  This
> will
> >    allow the RC to replicate media at layers below RTP.  Clearly,
> such
> >    an RC mode would not be able to provide transcoding, or media
> >    injection from the RS back into the Recorded Session."
> >
> > Does this mean that you think that RTP is unsuitable for transporting
> > media between the RS and RC?  Clearly it isn't for IM, but I would
> think
> > that many of the functions of RTP are needed for audio and video
> media.
> > To understand this requirement will take more discussion.
> >
> > [LeonP] Actually it was Hadriel requirement. I believe it only means
> that RC does not required to implement RTP stack, it can just fork RTP
> packets on the network level (by hardware)
> >
> 
> Yes, I think I get that when I reread the requirement.  This is a very
> important requirement that has a huge impact on the scalability of the
> system.
> 
> > Thanks,
> > Alan
> > _______________________________________________
> > 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 drage@alcatel-lucent.com  Mon Jul 27 08:08:12 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Alan Johnston <alan@sipstation.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 27 Jul 2009 17:07:17 +0200
Thread-Topic: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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And just to note that this needs to map into the equivalent ISDN features.

These are implicit, where I am saying: "if the place where this is delivere=
d to cannot understand it, then throw it away",

and explicit, where I am saying: "if the place where this is delivered cann=
ot understand it, then fail the call".

It doesn't need anything else, and there is no discovery mechanism in the I=
SDN (apart from by making calls containing UUI explicit and seeing what fai=
ls).

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Alan Johnston
> Sent: Monday, July 27, 2009 3:43 PM
> To: Paul Kyzivat
> Cc: dispatch@ietf.org; Hadriel Kaplan
> Subject: Re: [dispatch] CCUS and History-Info (was Proposed=20
> Charter for CCUS - Call Control UUI for SIP)
>=20
> Paul Kyzivat wrote:
> >
> >
> > Alan Johnston wrote:
> >> Dean,
> >>
> >> So basically, you want a different name for the header field? =20
> >> Otherwise the semantics seem identical to what is proposed.
> >
> > There at least needs to be some interoperable way to specify the=20
> > format of the data being passed. We have that with MIME, but when=20
> > using a header the mechanism needs to be reinvented, along=20
> with a way=20
> > to say "I don't understand that".
> This is an excellent point.
>=20
> The latest version of the mechanism draft has a way of=20
> specifying the purpose (application) and encoding, based on this need.
>=20
> The requirements currently say that a UAC has to be able to=20
> discover if=20
> a UAS supports the UUI  *mechanism*.   The obvious way to do=20
> this is a=20
> SIP option tag.  But do we need a requirement for a UAC to=20
> discover if a UAS supports a particular purpose/application? =20
> If the same UUI is encoded in different ways, do we need a=20
> way to say "you need to be able to support *one* of the data objects?
>=20
> Thanks,
> Alan
>=20
>=20
> >
> >     Thanks,
> >     Paul
> >
> >> Thanks,
> >> Alan
> >>
> >>
> >>
> >> On Jul 27, 2009, at 10:48 AM, Dean Willis=20
> <dean.willis@softarmor.com>
> >> wrote:
> >>
> >>> Hadriel Kaplan wrote:
> >>>
> >>>> I think the retargeting and middlebox support issues are=20
> not really=20
> >>>> But the real problem is the semantics as you say - to me=20
> a request=20
> >>>> URI is information about the targeted resource or how to=20
> reach it,=20
> >>>> not information about the source resource.  Since H-I is just a=20
> >>>> list of req-uri's, it's not the right place.
> >>>
> >>> I disagree. SIP is based on HTTP The Request URI encodes=20
> the request=20
> >>> being made by the UAC to the UAS. The CCUUI data is not=20
> germane to=20
> >>> SIP, it's just stuff being passed along the chain from UAC to UAS.
> >>>
> >>> Now, it might be reasonable to universally declare that=20
> SIP is, in=20
> >>> fact, NOT HTTP, and that it uses a different parameter passing=20
> >>> mechanism. If so, then what we would probably want to do=20
> is define a=20
> >>> new header field, perhaps "User-Parameters:", that is=20
> reserved for=20
> >>> user-to-user data consumed by applications. I'd be okay with that.
> >>>
> >>>
> >>>> Really the technically *right* place would be in MIME, afaict -=20
> >>>> it's opaque application data just like geoloc, right?  But=20
> >>>> pragmatically, the issues/problems with doing it in MIME were=20
> >>>> already covered and debated in sipping, and iirc=20
> consensus was: put it in a header.
> >>>>
> >>>> ISTM that we always complain about lack of running code,=20
> and here=20
> >>>> we have a case where there is running code from multiple=20
> vendors,=20
> >>>> and it's not fundamentally broken/un-sound.  It ain't=20
> broken, why=20
> >>>> do we need to fix it?
> >>>
> >>> The same reason we shoud object to all layer-violation hacks:
> >>> because it is architecturally unsound, inconsistent with=20
> the current=20
> >>> SIP design approach, and leads to further unnecessary=20
> complexity in=20
> >>> an already too-complex protocol.
> >>>
> >>>
> >>> --
> >>> Dean
> >> _______________________________________________
> >> 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
> =

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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] I-D Action:draft-johnston-sipping-cc-uui-08.txt
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DRAGE, Keith (Keith) wrote:
> 1st octet of the payload.

Fine. But what does the first octet of the payload *mean*?

Aren't the encodings corresponding to particular values of that octet 
defined somewhere? If so, how many of the 256 possible values are 
defined, and who controls how new values can be defined?

I'm pretty sure it isn't IETF or IANA.

	Thanks,
	Paul

> Keith 
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Monday, July 27, 2009 2:14 PM
>> To: DRAGE, Keith (Keith)
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] I-D 
>> Action:draft-johnston-sipping-cc-uui-08.txt
>>
>> Keith,
>>
>> If this is *not* ISDN encoding, then where is the indication 
>> of what encoding is being used? Are you saying that this is a 
>> private agreement between the UAC and UAS, neither signaled 
>> nor negotiated? How can that possibly work?
>>
>> I thought the first byte of the value was a registered format 
>> indicator.
>>
>> 	Thanks,
>> 	Paul
>>
>> DRAGE, Keith (Keith) wrote:
>>> Just to respond to this bit of the message:
>>>
>>>>> Not quite.  We are not making the assumption that both are
>>>> gateways,
>>>>> although that is possible.  Most likely, one element is 
>> SIP enabled.
>>>>> However, it is not a fully intelligent SIP endpoint - the basic 
>>>>> software and application (probably many years old) is
>>>> unchanged but a
>>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
>>>> stack does
>>>>> not know anything about the UUI as it is implementing
>>>> exactly the ISDN
>>>>> UUI service - the information is just pushed up the stack
>>>> to another
>>>>> application, and this application knows nothing about SIP.  
>>>> It still
>>>>> thinks it is using an ISDN trunk.
>>>> Then in the way I meant the question, both ends *are* gateways.
>>>>
>>>> But then if someone wishes to build native sip gear to 
>> plug into this 
>>>> environment, then it will still have to understand this isdn 
>>>> encoding.
>>>> Perhaps that is the cruel truth, unless all the equipment 
>> is replaced 
>>>> at once.
>>> I've said this before and I'll say it again. The bit being 
>> delivered to the end UE is not "isdn encoding". If it was 
>> ISDN encoding, then the ISDN / SIP gateway would be able to 
>> interpret it and map it into SIP. Rather it is some 
>> application sitting above the ISDN call control layer in the 
>> calling terminal, which is wishing to communicate with its 
>> peer application running in the destination SIP UA, or gateway.
>>> regards
>>>
>>> Keith
>>>
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org
>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>>>> Sent: Saturday, July 04, 2009 12:39 AM
>>>> To: Alan Johnston
>>>> Cc: dispatch@ietf.org
>>>> Subject: Re: [dispatch] I-D
>>>> Action:draft-johnston-sipping-cc-uui-08.txt
>>>>
>>>> Hi Alan,
>>>>
>>>> responses inline. I've trimmed out the parts that don't 
>> require more 
>>>> discussion.
>>>>
>>>> 	Thanks,
>>>> 	Paul
>>>>
>>>> Alan Johnston wrote:
>>>>> Paul,
>>>>>
>>>>> Thanks for your comments.  See mine below.
>>>>>
>>>>> Thanks,
>>>>> Alan
>>>>>
>>>>> Paul Kyzivat wrote:
>>>>>> Hi Alan,
>>>>>>
>>>>>> I took a quick look at this doc and the requirements doc
>>>> as well. I
>>>>>> have a few thoughts:
>>>>>>
>>>>>> From section 1:
>>>>>>
>>>>>>>    In the future, where both endpoints are intelligent
>>>> SIP user agents,
>>>>>>>    it may be possible for them to understand and
>>>> interpret the UUI data.
>>>>>>>    There may be some cases where the UUI information is
>>>> relevant to SIP.
>>>>>>>    In this case, it might be worthwhile attempting to map
>>>> UUI data to an
>>>>>>>    appropriate SIP header field or to standardize a new
>>>> header field.
>>>>>>>    However, the requirements and use cases for this are
>>>> different enough
>>>>>>>    from those described in this document that these two 
>> situations
>>>>>>>    should be examined separately.  This document looks 
>> only at the
>>>>>>>    requirements and mechanisms for replicating the
>>>> existing, widely used
>>>>>>>    and deployed ISDN UUI service.
>>>>>> This *still* troubles me!
>>>>>> Is the assumption here that *both* the UAC and UAS are
>>>> gateways, so
>>>>>> that this mechanism is solely for the tunneling of Q.931 
>> and Q.763 
>>>>>> UUI message over SIP?
>>>>>>
>>>>>> I get the impression that an important use case is where at least
>>>>>> *one* end (probably the UAS) is a SIP device. In that 
>> case it will 
>>>>>> already be *necessary* for it to understand and 
>> interpret UUI data.
>>>>>> And where there is redundancy between SIP data and the UUI
>>>> data, it
>>>>>> will need to resolve the discrepancy.
>>>>> Not quite.  We are not making the assumption that both are
>>>> gateways,
>>>>> although that is possible.  Most likely, one element is 
>> SIP enabled.
>>>>> However, it is not a fully intelligent SIP endpoint - the basic 
>>>>> software and application (probably many years old) is
>>>> unchanged but a
>>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
>>>> stack does
>>>>> not know anything about the UUI as it is implementing
>>>> exactly the ISDN
>>>>> UUI service - the information is just pushed up the stack
>>>> to another
>>>>> application, and this application knows nothing about SIP.  
>>>> It still
>>>>> thinks it is using an ISDN trunk.
>>>> Then in the way I meant the question, both ends *are* gateways.
>>>>
>>>> But then if someone wishes to build native sip gear to 
>> plug into this 
>>>> environment, then it will still have to understand this isdn 
>>>> encoding.
>>>> Perhaps that is the cruel truth, unless all the equipment 
>> is replaced 
>>>> at once.
>>>>
>>>>>>> 5.  Syntax for UUI Header Field
>>>>>> ...
>>>>>>>        UUI         = "User-to-User" HCOLON uui-data 
>>>> *(SEMI uui-param)
>>>>>>>        uui-data    = token
>>>>>>>        uui-param   = enc-param | cont-param | purp-param 
>>>> | generic-param
>>>>>>>        enc-param   = "encoding=" ("hex" | token)
>>>>>>>        cont-param  = "content=" ("isdn-uui" | token)
>>>>>>>        purp-param  = "purpose=" ("isdn-interwork" | token)
>>>>>> At the very least, I would request that this be expanded a
>>>> little bit
>>>>>> for future flexibility, by permitting the uui-data to be 
>> either a 
>>>>>> token or a string. While the encoding you have chosen works as a 
>>>>>> token, other encodings may need the additional flexibility
>>>> of a string.
>>>>>>        uui-data    = token | quoted-string
>>>>> OK, I guess I'm OK with that.  For the isdn-interwork
>>>> purpose, we'll
>>>>> require the token, but other applications could utilize the
>>>> quoted string.
>>>>
>>>> I would hope that when the token encoding is sufficient, 
>> then either 
>>>> form would be accepted. Or it would perhaps be ok to specify for a 
>>>> particular encoding (hex) that the token encoding must be used. I 
>>>> don't see how the purpose has anything to do with it.
>>>>
>>>>>>> 6.3.  Registration of SIP Option Tag
>>>>>> This section registers the new option tag. But I find
>>>> almost nothing
>>>>>> that defines the semantics of the option. (Section 5 does
>>>> include the
>>>>>> following:
>>>>>>
>>>>>>>    A UA that supports this feature and the "uui" option
>>>> tag MUST support
>>>>>>>    the call flows in [johnston-dispatch-sip-cc-uui]...
>>>>>> but that is pretty thin from a normative perspective.
>>>>>>
>>>>>> Of particular note, does support for the option tag mean
>>>> support for
>>>>>> the particular encoding, content, and purpose included in this 
>>>>>> document? Or does it only mean support for the header and
>>>> params in
>>>>>> the abstract? (That wouldn't be very useful.)
>>>>> Yes, this needs clarification.  I'd suggest we define the
>>>> option tag
>>>>> to be uui-isdn which means it understands the header 
>> field and the 
>>>>> isdn-interwork purpose, and the call flows, including 
>> escaping into 
>>>>> redirection and Refer-To URIs.
>>>> I'd like to hear some other opinions on whether to have 
>> tag per param 
>>>> for tag, or a tag for the combination of param values. I 
>> think I'm ok 
>>>> with what you propose.
>>>>
>>>>>> I don't see how it can mean support for some other yet to
>>>> be defined
>>>>>> values for those. Yet if it only means support for the
>>>> current ones,
>>>>>> then how will one indicate support for future ones?
>>>>>>
>>>>>> The only way out of this that I can see is to have
>>>> separate options,
>>>>>> either for particular values of each parameter, or for 
>> particular 
>>>>>> combinations of values of all the parameters.
>>>>> New applications using this header field would have the option of 
>>>>> defining a new SIP option tag which would mean support for
>>>> the header
>>>>> field and their new purpose.
>>>>>
>>>>>> So you might have options uui-purpose-isdn-interwork, 
>>>>>> uui-content-isdn, and uui-encoding-hex. Or you might just
>>>> have option
>>>>>> uui-isdn-interwork that means the combination.
>>>>> I think the latter.  I see it as a hierarchy - the application or 
>>>>> purpose is the top one.  Each purpose has a number of
>>>> contents that it
>>>>> supports.  Each content has a number of encoding schemes it
>>>> supports.
>>>>
>>>> I guess I'm ok with that, pending nailing down the definition of 
>>>> "purpose".
>>>>
>>>>>> I also am not finding much explanation of the semantics of the 
>>>>>> content and purpose parameters. I can only extrapolate
>>>> from a single
>>>>>> example of each, which I'm having trouble doing.
>>>>>>
>>>>>> I'm guessing that 'content' must refer to the precise
>>>> syntax of the
>>>>>> contained data, after decoding. So presumably it must map
>>>> onto some
>>>>>> particular spec. For isdn-uui, would that be [Q931] or
>>>> [Q763]? If so,
>>>>>> shouldn't it refer to something specific in that document?
>>>>> Not really.  In this case, isdn-uui effectively means 
>> "opaque data" 
>>>>> which is how ISDN defines the UUI - it is completely undefined, 
>>>>> necessarily so  by the strict layering used in the ISDN 
>> application.
>>>> I don't believe you!
>>>>
>>>> If that is the case, then the UAC and UAS must have a private side 
>>>> agreement about what the isdn-uui content means. Is that 
>> really what 
>>>> you mean?
>>>>
>>>> You already state that the first byte is a protocol discriminator. 
>>>> There must also be something, somewhere, that specifies 
>> the mapping 
>>>> from a particular value for that byte to a particular 
>>>> protocol/format. If there is, then that should be part of the 
>>>> definition of this content.
>>>>
>>>>>> 'purpose' is even less clear to me. Does it refer to why
>>>> it is being
>>>>>> included? Or what is to be done with it? If received by a
>>>> UA that is
>>>>>> not an ISDN gateway, how can it decide if this is
>>>> something it should
>>>>>> be trying to process? Is it still ISDN interworking if 
>> neither the 
>>>>>> UAC nor UAS is connected to an ISDN network?
>>>>> The purpose is the application using the UUI.  Others have
>>>> expressed
>>>>> an interest in carrying other data in the header field, in
>>>> which case
>>>>> a new purpose would be defined.  I'll see if I can think of
>>>> an example one.
>>>>
>>>> So are you saying that a particular call center application might 
>>>> constitute a distinct "purpose"?
>>>>
>>>>> Basically, if two UAs both support the header but have different 
>>>>> applications generating and consuming the UUI, it will not
>>>> work.  This
>>>>> is not a SIP failure, and there is nothing SIP can do about
>>>> it.  But
>>>>> this purpose header field allows the UAs to detect this condition.
>>>> In that case, perhaps "isdn-interwork" isn't really a 
>> single purpose 
>>>> at all, unless any two pieces of equipment that support 
>> that purpose 
>>>> can then interwork without knowing any more about each other.
>>>>
>>>>>> Suppose I was developing a sophisticated UA that
>>>> potentially might be
>>>>>> usable by a call center operator. How should
>>>> purpose=isdn-interwork
>>>>>> affect the operation of this device?
>>>>> To make it work in today's contact center applications, 
>> supporting 
>>>>> isdn-interwork would likely make it work in an application,
>>>> provided
>>>>> the appropriate call center application also ran on it.  
>>>> Some contact
>>>>> centers do not use ISDN UUI, instead they use all kinds 
>> of really, 
>>>>> really ugly CTI protocols running in parallel.
>>>> Yes, I know! :-(
>>>>
>>>>> This is a way of moving
>>>>> those to the ISDN model, and from there to a more
>>>> intelligent endpoint
>>>>> SIP model.
>>>>>
>>>>>>     Thanks,
>>>>>>     Paul
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>

From chris.labarbera@citi.com  Mon Jul 27 08:10:49 2009
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From: "Labarbera, Chris " <chris.labarbera@citi.com>
To: Roni Even <Even.roni@huawei.com>, "'Alan Johnston'" <alan@sipstation.com>,  "'Leon Portman'" <Leon.Portman@nice.com>
Date: Mon, 27 Jul 2009 10:10:20 -0500
Thread-Topic: [dispatch]	Comments	on draft-jain-dispatch-session-recording-protocol-req-00
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We need to look at this in the big picture. If you want to address packet l=
oss and control using RTCP, you will most likely use the services of a sess=
ion border controller to act as the B2BUA and leverage the services of the =
SBC to manage RTCP and MOS scores for the media. The whole context of the d=
ispatch recording method is to provide a forking point at a large ingress p=
oint such as a SBC. In this model, a SBC can deliver 15,000 SIP sessions to=
 a series of enterprise PBX/ACD systems and the dispatch method can be used=
 to selectively fork media to a voice recorder at great scale and capacity.=
 The overall goal of this model is to provide an effective and scalable sol=
ution for recording in high availability (transparently) without putting an=
y additional demand on the PBX systems downstream from the recorder.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Roni Even
Sent: Monday, July 27, 2009 10:58 AM
To: 'Alan Johnston'; 'Leon Portman'
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Comments on draft-jain-dispatch-session-recording-p=
rotocol-req-00

Hi,
I am not sure how REQ-021 will work if you want to address packet loss and
control using RTCP. I suggest you look at the Topologies RFC 5117 for
directions
Roni Even
>
> > "   o REQ-021 The mechanism MUST support the ability for the RC to
> only
> >    perform media replication to the RS, without needing to decode or
> mix
> >    audio/video/etc., and without needing to be an RTP agent.  This
> will
> >    allow the RC to replicate media at layers below RTP.  Clearly,
> such
> >    an RC mode would not be able to provide transcoding, or media
> >    injection from the RS back into the Recorded Session."
> >
> > Does this mean that you think that RTP is unsuitable for transporting
> > media between the RS and RC?  Clearly it isn't for IM, but I would
> think
> > that many of the functions of RTP are needed for audio and video
> media.
> > To understand this requirement will take more discussion.
> >
> > [LeonP] Actually it was Hadriel requirement. I believe it only means
> that RC does not required to implement RTP stack, it can just fork RTP
> packets on the network level (by hardware)
> >
>
> Yes, I think I get that when I reread the requirement.  This is a very
> important requirement that has a huge impact on the scalability of the
> system.
>
> > Thanks,
> > Alan
> > _______________________________________________
> > 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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Room 300 over lunch on Friday

From mohamed.boucadair@orange-ftgroup.com  Mon Jul 27 13:12:18 2009
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=20
Dear all,

A new version of the atypes media feature tag draft has been submitted.

This draft has been discussed in SIPPING mailing list: =
http://www.ietf.org/mail-archive/web/sipping/current/msg16718.html.

Comments and suggestions are more than welcome.

Cheers,
Med



-----Message d'origine-----
De : i-d-announce-bounces@ietf.org =
[mailto:i-d-announce-bounces@ietf.org] De la part de =
Internet-Drafts@ietf.org
Envoy=E9 : lundi 27 juillet 2009 20:15
=C0 : i-d-announce@ietf.org
Objet : I-D Action:draft-boucadair-dispatch-ipv6-atypes-00.txt=20

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

	Title           : The atypes media feature tag for Session Initiation =
Protocol (SIP)
	Author(s)       : M. Boucadair, et al.
	Filename        : draft-boucadair-dispatch-ipv6-atypes-00.txt
	Pages           : 16
	Date            : 2009-07-27

This specification defines a new media feature tag called atypes.
This new media feature tag indicates the IP address type capabilities of =
the UA (User Agent) and can aid the routing process and ease the =
invocation of required functions when heterogeneous (i.e.  IPv4 and
IPv6) parties are involved in a given SIP session.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-boucadair-dispatch-ipv6-atypes-=
00.txt

Internet-Drafts are also available by anonymous FTP at:
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From dean.willis@softarmor.com  Mon Jul 27 15:24:41 2009
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Date: Mon, 27 Jul 2009 17:24:33 -0500
From: Dean Willis <dean.willis@softarmor.com>
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Hadriel Kaplan wrote:
> 
>> -----Original Message----- From: Dean Willis
>> [mailto:dean.willis@softarmor.com] Sent: Monday, July 27, 2009 4:49
>> AM To: Hadriel Kaplan
>> 
>> I disagree. SIP is based on HTTP The Request URI encodes the
>> request being made by the UAC to the UAS.
> 
> Then you might as well argue the request-URI should encode MIME
> bodies.  The UUI is not a part of the resource being addressed in any
> way, and thus not a part of a Uniform Resource Identifier.

I take it you've never written an HTTP application using something like
the Common Gateway Interface (CGI) for parameter passing. Perhaps taking
some time to learn the history here would be helpful for you, so that
you stop making patently incorrect assertions about what can be in a
request URI.

Let's look at the request-URI used by the IETF tools team's draft search
page.

Take, for example: http://tools.ietf.org/id/?doc=common

Everything up to the first / is network routing information -- it finds
the host responsible for the application.

The /id/ part identifies the application. That might arguably be
"application routing", but it is not NETWORK routing. It doesn't create
Via header fields in the SIP request.

The "?doc=common" part is parameters being delivered to the "id"
application. The question mark is the common CGI delimiter for
parameters. This delimits a parameter having the name "doc" and taking
the value "common", thereby telling the id application to search for all
docs having the string "common" in their name.

Note that, in your parelance, "id" and "common" are NOT part of the
resource being addressed in any way. Yet, they are encoded into the
resource, in such a way that they can easily be presented in writing,
pasted onto a business card, or whatever else you might want to do with
a URI.

Nowhere in the HTTP spec (nor in the HTTCP CGI spec at
http://hoohoo.ncsa.illinois.edu/cgi/interface.html) is "doc" or "common"
defined. Rather, the specification defines a syntax for parameter names
and parameter values, and leaves the actual choice of parameter names
and encoding of values up to the application. This has allowed many
millions of HTTP applications to be developed without having to amend
the HTTP specification for each one of them.

Back when we were working on the SIP that eventually became RFC 2543, we
intended SIP to support a CGI-type invocation model. We just didn't do
it right, for a number of reasons mostly having to do with failed
communication between the people driving the spec.

> 
> 
>> Now, it might be reasonable to universally declare that SIP is, in
>> fact, NOT HTTP, and that it uses a different parameter passing
>> mechanism.
> 
> Well, I'll probably be flamed, but afaict it is not HTTP in most
> every important way.  It does a LOT of things differently.  That ship
> has sailed.  It is far closer to email (with some weird combo of SMTP
> and POP3), if anything.

Possibly true. But we need SOME equivalent of HTTP's parameter passing
model, else we are stuck with having to extend SIP again and again for
every possible application, and that's just a ridiculous idea.

> 
> 
>> If so, then what we would probably want to do is define a new
>> header field, perhaps "User-Parameters:", that is reserved for
>> user-to-user data consumed by applications. I'd be okay with that.
> 
> That was actually how I interpreted the semantic notion of the
> User-to-User header.

That is, I think, the intent in Alan's current draft. He's not at all
far off from what I think would be a reasonable solution. There's some
guidance needed on semantic negotiation and how to say "I don't
understand the parameters you have sent me", but it should be pretty
straightforward to remedy.

I find this approach much more attractive for user-to-user parameter
passing than the idea that we encode the parameters HTTP-style in the
request URI, then have the UAS wade backwards thru the History-Info
stack until it finds them.

But our current consensus is to do just exactly that: so I'm using
CC-UUI to challenge that consensus. Either we can use request-URI
parameters and History-Info for parameter passing (in which case we
should do CC-UUI that way), or we develop the more robust mechanism that
SIP has long needed (and that Alan has a good start on). I'm in favor of
the latter, but will accept either.

What I won't accept is yet another application-specific SIP extension
header that works for CC-UUI but doesn't solve the general problem,
coupled with a head-in-sand insistence that we've solved the general
problem using History-Info, despite having proved (by the need to
develop yet another special extension for CC-UUI) that we in fact have
NOT solved the general UAC-to-UAS parameter passing problem.

--
Dean




From dean.willis@softarmor.com  Mon Jul 27 15:30:47 2009
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Alan Johnston wrote:

> The requirements currently say that a UAC has to be able to discover if
> a UAS supports the UUI  *mechanism*.   The obvious way to do this is a
> SIP option tag.  But do we need a requirement for a UAC to discover if a
> UAS supports a particular purpose/application?  If the same UUI is
> encoded in different ways, do we need a way to say "you need to be able
> to support *one* of the data objects?

We certainly need a way to say "I understand the parameter you have sent
me, but the value you gave it doesn't make any sense".

It might also be ok to have a criticality indicator on a parameter, and
I can envision a subset-selection indication that wouldn't be too hairy,
but I'm having a hard time coming up witha real use case.

It would be a lot easier to write the applications to have positive
resopnse: that is, if the UAS understands the parameter in question and
the value makes sense, it says so affirmatively in the response.
Further, it might indicate which of several alternatives it preferred.
This provides a fail-safe, backward-compatible mechanism that gives
debugging feedback, but at much less complexity cost.

--
Dean

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Thread-Topic: New version of "draft-avasasarala-dispatch-comm-diversion-info" draft submitted
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
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Subject: [dispatch] New version of "draft-avasasarala-dispatch-comm-diversion-info" draft submitted
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Hi All

We have submitted an updated version of
draft-avasasarala-dispatch-comm-diversion-info=20

It can be accessed at:
http://www.ietf.org/staging/draft-avasarala-dispatch-comm-div-notificati
on-01.txt

This draft proposes a SIP Event package for Communication Diversions
Notification Information and conforms to procedures and schema described
in 3GPP TS 24.604.

Regards
Ranjit


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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">We have submitted an =
updated version of draft-avasasarala-dispatch-comm-diversion-info =
</FONT>
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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">It can be accessed =
at: </FONT><A =
HREF=3D"http://www.ietf.org/staging/draft-avasarala-dispatch-comm-div-not=
ification-01.txt"><U></U><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">This draft proposes a =
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and conforms to procedures and schema described in 3GPP TS =
24.604.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Regards</FONT>

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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Vijay Gurbani <vkg@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Tue, 28 Jul 2009 00:30:25 -0700
Thread-Topic: [dispatch] Session Recording in SIP
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> -----Original Message-----
> From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]
> Sent: Monday, July 20, 2009 5:51 AM
> To: Doken, Serhad; Vijay Gurbani; dispatch@ietf.org
> Subject: RE: [dispatch] Session Recording in SIP
>=20
> Hi,
>=20
> Thanks for your detailed comments. Please see inline:
>=20
> > -----Original Message-----
> > From: Doken, Serhad [mailto:sdoken@qualcomm.com]
> > Sent: Friday, July 17, 2009 3:56 AM
> > To: Jain, Rajnish; Vijay Gurbani; dispatch@ietf.org
> > Subject: RE: [dispatch] Session Recording in SIP
> >
> > I read this draft and following are my comments :
> >
> > Section 3 Definitions
> >
> > Perhaps Recording Server(RS) should simply be called Recorder. I see
> the
> > note on the last sentence however as long as definitions refer to
> Client and
> > Server, it will always imply to the reader to think that RS is the
> server.
>=20
> Naming these things is always a bit tricky. If we called the RS simply
> the Recorder given your reasoning, then the RC should also be named
> differently (because the RC can act as the "server side" of the SRP
> signaling). The RS naming is a bit akin to the term "Media Server". If
> there is enough confusion on RS then, we can consider renaming it.

Indeed, I think RC could be renamed as well since it is not a pure client a=
nd can be a server at times. Using, Recording Client and Recording Server a=
t section 1 before their definitions are given in Section 2 adds to the con=
fusion. I would offer something like Recorder and (Recorded Media) Forker w=
hich is self explanatory.

>=20
> > The statement that SRP may be SIP is an understatement since the
> definition
> > for RC already referred to as a SIP UA/B2BUA.
>=20
> This will be clarified in subsequent revisions when we'll say that SRP
> is indeed SIP.
>=20
> > Dynamic Recording, if the recording session is started and ended at
> random
> > points on demand, its length will not be the same as actual media
> session.
>=20
> Ok. We can clarify this.
>=20
> > Persistent Recording, it is not clear what is meant by "system start
> up". Is
> > "system" referring to RC or RS ?
>=20
> It meant that when both devices become functional. We can clarify this.

It'll probably need more than functional, at a minimum registered/can make-=
receive calls/process media.

>=20
> > Depending on the answer, I am curious how are these persistent
> recording
> > sessions are correlated to many active media sessions and what would
> be
> > mechanism to keep them alive ?
>=20
> The data over the call metadata interface will offer the correlation. A
> simple way to keep SRP persistent sessions alive is through the SIP
> session-timers.
>=20
> What is the number of persistent recording
> > sessions per RC/RS ?
>=20
> That's an implementation-dependent choice.

Obviously, however as I referred to in my subsequent question below, this m=
ay have implications scaling up. If I have a device with high number of sim=
ultaneous calls capability, media forker almost needs to keep that many per=
sistent recording sessions and refresh them all the time by session timers.=
 A bunch of race conditions may occur and it may be a drain of resources on=
 the device that is doing the media forking especially when there is no act=
ual session. Moreover, there will be certain assumptions made for such pers=
istent recording sessions such as the codec used for the media session etc.
=20
>=20
> > Perhaps implications of this should be covered somewhere else, maybe
> in the
> > requirements section ?
> >
> > Section 5 Requirements
> >
> > REQ-006 Not a good idea to talk about the solution here and how some
> RC
> > implementations handle the problem.
>=20
> That note is only informative text. I guess we can take it out.
>=20
> > Perhaps REQ-010 and 011 can be combined into one by just saying "over
> a
> > single or separate".
>=20
> Fine. Will change this.
>=20
> > REQ-018 can be extended to say that the mechanism MUST support the
> ability
> > to find out the associated recorded session(s) given that one(for
> instance
> > RS) is only aware of recording session(s). I'd rather prefer this to
> be a
> > separate requirement.
>=20
> The RS is aware of the recorded session metadata. REQ-018 is a bit
> broad in the sense that it talks about correlation in either direction
> (i.e. from recorded sessions to recording sessions and vice-versa). I
> think your comment is about splitting the two directions, right?

Yes

>=20
> > I have another requirement in mind : The mechanism MUST support the
> ability
> > to setup recording sessions that will record different conversation
> > directions of the Recorded session(s) from RC(s) and they too should
> be
> > correlated accordingly. For instance, while an agent is talking to a
> > customer, there should be a way to setup two recording sessions, one
> > carrying the media from agent towards customer and vice versa for the
> other.
> > Furthermore, these recording sessions should be marked as such in
> SRP.
>=20
> Yes, this is needed. This requirement was actually in there and somehow
> got removed in one of the edits. We'll add this.
>=20
> > REQ-021 talks about a mode where RC just forks media to RS. However,
> there
> > is value in studying transcoding, media incompatibility use cases and
> see if
> > there are requirements they bring given that RS/RCs are/will most
> likely
> > have different media capabilities.
>=20
> REQ-021 is primarily about a middle-box type of RC that forks the
> media. The RC vendors in this case want their boxes to be optimal and
> scalable by doing as little media processing as possible. REQ-021 isn't
> precluding the use cases that you mentioned, but I think it'll make
> sense to address such use cases in this document.
>=20
> > If recorded session media is stopped/temporarily interrupted due to
> problems
> > or features invoked on it(recording session is put on hold or started
> a
> > conferencing or transfer operation), how does that affect the already
> setup
> > recording session(s) ?
>=20
> I think that'll largely depend on the RC/RS implementations and the
> nature of the SRP sessions. For instance, if the recording sessions are
> persistent and the RC puts recorded session media on hold, then it RC
> MAY reINVITE the RS and put the recording session media on hold or it
> MAY just temporarily stop sending such media (silence suppression) to
> the RS.

In that case, the recorded session and the persistent recording sessions ne=
ed to be synchronized back during resume which may bring the media clipping=
 problem that persistent recording attempted to solve in the first case.

Regards,
Serhad Doken

>=20
> > Nit :
> >
> > Section2, sentence that starts with "that draft focuses on the
> mechanism to
> > provide the SRTP...." is odd. It is clear that srtp-key draft is not
> for
> > session recording, so no need to state the obvious.
>=20
> Ok. We can change some wording here.
>=20
> Thanks,
> Raj
>=20
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From gonzalo.camarillo@ericsson.com  Tue Jul 28 00:44:38 2009
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Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300 over Lunch on	Friday
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Hi,

since we are now allowed to bring food into the building, we will run 
this meeting from 12:00 to 13:00 so that people can either have a quick 
lunch from 11:30 to 12:00 or grab some food somewhere and bring it to 
the meeting.

Cheers,

Gonzalo


Robert Sparks wrote:
> Room 300 over lunch on Friday
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


From sdoken@qualcomm.com  Tue Jul 28 00:47:27 2009
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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: Leon Portman <Leon.Portman@nice.com>, Alan Johnston <alan@sipstation.com>,  "dispatch@ietf.org" <dispatch@ietf.org>
Date: Tue, 28 Jul 2009 00:47:04 -0700
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> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Leon Portman
> Sent: Saturday, July 25, 2009 7:10 PM
> To: Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Comments on draft-jain-dispatch-session-
> recording-protocol-req-00
>=20
> Alan, Hi
>=20
> Thanks for the comments.
>=20
> Please see below my answers
>=20
> Regards
>=20
> Leon
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Alan Johnston
> Sent: Friday, July 24, 2009 9:29 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Comments on draft-jain-dispatch-session-recording-
> protocol-req-00
>=20
> Here are my comments on this draft.
>=20
> Overall, I think this draft is an excellent start at the requirements
> for this important topic.  I look forward to future discussions of
> charter text so we can get started on the work.
>=20
> Section 2:
>=20
> "  Another, related IETF draft is draft-wing-sipping-srtp-key
>    [I-D.wing-sipping-srtp-key], which describes an approach for
>    providing SRTP session keys to a recorder-type server.  That draft
>    focuses on the mechanism to provide the SRTP session key, rather
> than
>    the mechanism to invoke and sustain recording sessions themselves."
>=20
> draft-wing-sipping-srtp-key actually has a lot of discussion, including
> call flows showing how recording sessions can be established.  Also,
> this document has four recording modes discussed, which I think should
> be discussed in this document.  Two are mentioned without really
> explaining them, and two others are implied by various requirements.
>=20
> [LeonP] Yes, for the protocol draft, we will need to combine all of the
> approaches and make sure that we covered all use cases
>=20
> Section 3:
>=20
> " Recording Session: The session created between an RC and RS for the
>    purpose of recording a Recorded Session.  The Recorded Session may
>    itself be based on SIP, but it is a separate session from the
>    Recording Session.
>=20
>    Recorded Session: A session created between a UAC and UAS that is
>    recorded by the RC and RS systems."
>=20
>=20
> These two terms only differ by their tense, yet I think one refers to a
> signaling session and the other refers to a media session.  If so,
> better terms and definitions are needed.  If not, we need a term for
> the
> actual recorded media.  Also, there are (at least) two sessions here -
> the one between the UAs and the Recording Session - in a few places I
> suspect the two are being confused.
>=20
> [LeonP] Actually Recording Session relates to both SIP and Media that
> is forked to the recording server and Recorded Session is actually the
> original conversation between two UAs. These two are exactly the two
> sessions that you are describing.
>=20
> "  Dynamic Recording: Dynamic recording is a mode of recording where
> the
>    recording sessions are established on an as needed basis.  The
> length
>    of these sessions is typically same as the length of the actual
> media
>    sessions.
>=20
>    Persistent Recording: Persistent recording is a mode of recording
>    where the recording sessions are established at system start-up and
>    kept-alive from that point on.  The length of these sessions is
>    independent from the length of the actual recorded media sessions.
>    Persistent recording sessions avoid issues such as media clipping
>    that can occur due to delays in recording session establishment."
>=20
> These terms need more explanation as I can think of multiple things
> they
> might mean.  For example, when you say "session" do you mean a
> signaling
> session or media session?  When you talk about avoiding media clipping,
> do you mean that media is always sent by the RC to the RS regardless of
> whether there is any media?
>=20
> [LeonP] Yes, by session we mean both legs: signaling and media. And in
> Persistent Recording mode, the Recording SIP session supposed to stay
> connected even there is not actual connected sessions for the specific
> recorded UA.
>=20
> Figures could use Figure numbers and labels.  Also, it would be worth
> showing which is the RC and which is the RS in them.
>=20
> Section 5
>=20
> "   o REQ-001 The mechanism MUST support the ability to perform
>    persistent (always-on) or dynamic (on-demand) Recording Sessions."
>=20
> I'm not sure that the Persistent and Dynamic Recording as defined above
> correspond to the Always On and On Demand modes defined in
> http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#section-4.
>=20
> [LeonP] It is different in the way that the Recording Session is
> established only once during initialization (or login events) and then
> only media is forwarded per each call in order to minimize the
> clipping.

I see the "Always On" Recording as a separate third mode of Recording where=
 for a particular one or set of devices(possibly set via provisioning), all=
 calls are recorded from start to end(by setting up recording sessions) and=
 no persistent recording sessions are kept for them.

>=20
> "   o REQ-002 The mechanism MUST support the ability to perform
>    persistent Recording Sessions, such that the connection and
>    attributes of media in the Recording Session are not dynamically
>    signaled for each Recorded Session before it can be recorded.  Call
>    details and metadata will still be signaled, but can be post-
>    correlated to the recorded media.  There will still need to be a
>    means of correlating the recorded media connection/packets to the
>    Recorded Session, however this may be on a permanent filter-type
>    basis, such as based on a SIP AoR of an agent that is always
>    recorded, or based on identifiers in the recorded media itself."
>=20
>=20
> Without fully understanding Persistent Recording, I can't evaluate this
> requirement.
>=20
> "   o REQ-009 The mechanism MUST support the ability to deliver mixed
>    media streams to the RS.  The RS MAY be informed about the
>    composition of the mixed streams through session metadata.
>=20
>    Note: A mixed media stream is where several recorded sessions are
>    carried in a single recording session.  A mixed media stream is
>    typically produced by a mixer function."
>=20
>=20
> I'm not clear what mixed media means here - does it mean a mix of media
> types - synchronized audio and video from the same session, or do you
> mean multiple media streams from different sessions?  The note seems to
> indicate that this relates only to a conference call where multiple
> sources (SSRCs) have been combined into a single RTP session.
>=20
>  [LeonP] It is specific trading floor environment requirements where
> multiple handset and speakers for specific turrets will be mixed on the
> turret level for recording by one recording channel.
>=20
>=20
> "   o REQ-013 The mechanism MUST support the ability to inject tones
> into
>    the Recorded Session either from the RS or from the RC."
>=20
> Recorded Session seems to be defined as the media between the RC and
> RS.  Is that where the injected tones should be?  Or should then be
> mixed in the media between the two UAs?
> [LeonP] we have multiple options here. Preferable solution that one of
> the UAs will inject them (triggered by policy or parameters in SRP),
> but there are also configuration where RS will be required to create
> them and send back to RC (like talking clock for example)
>=20
> "   o REQ-021 The mechanism MUST support the ability for the RC to only
>    perform media replication to the RS, without needing to decode or
> mix
>    audio/video/etc., and without needing to be an RTP agent.  This will
>    allow the RC to replicate media at layers below RTP.  Clearly, such
>    an RC mode would not be able to provide transcoding, or media
>    injection from the RS back into the Recorded Session."
>=20
> Does this mean that you think that RTP is unsuitable for transporting
> media between the RS and RC?  Clearly it isn't for IM, but I would
> think
> that many of the functions of RTP are needed for audio and video media.
> To understand this requirement will take more discussion.
>=20
> [LeonP] Actually it was Hadriel requirement. I believe it only means
> that RC does not required to implement RTP stack, it can just fork RTP
> packets on the network level (by hardware)

Most such middlebox media forkers will have RTP stack implemented anyway. T=
hough packet replication can be done at a lower layer, it'd probably still =
need the smarts to know the replicated packet is actually a RTP packet to d=
etermine which packet to replicate.

>=20
> Thanks,
> Alan
> _______________________________________________
> 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 spencer@wonderhamster.org  Tue Jul 28 00:47:50 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
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Just to provide some logistics:

We'll meet from 12 until 12:50 or so. Please use the 11:30-12:00 time to
grab lunch. .SE has made arrangements to allow us to bring food into the
conference center (not into the large conference rooms, but the rest of the
building is OK), so you can pick up lunch and munch while we chat.

Robert's goals for this session are:

- To discuss feedback we gotten so far,
- To refine the charter, and
- To organize the next steps of work

Please prepare for the meeting with these thoughts in mind.

I also note that Robert's milestone dates on the proposed charter are 
currently:

Oct 09 - Problem statement, motivation, and use cases WGLC
Nov 09 - Problem statement, motivation, and use cases to IESG 
(Informational)
Jan 10 - SIP Common Log Format specification WGLC
Feb 10 - SIP Common Log Format specification to IESG (PS)

There's not a lot of time between now and October/November, so please come 
prepared to make progress!

Thanks,

Spencer



From gonzalo.camarillo@ericsson.com  Tue Jul 28 00:56:02 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [dispatch] Disaggregated Media in SIP
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Hi Paul,

yes, what you say is totally correct when one of the endpoints is not 
distributed. That case should be relatively-easy to resolve. The case 
where both ends are distributed is, of course, more "interesting" ;o)

Cheers,

Gonzalo

Paul Kyzivat wrote:
> 
> 
> Gonzalo Camarillo wrote:
>> Hi,
>>
>>> I guess the one way I could buy into your "distributed" approach is 
>>> to simply model it as a conference, where the focus is at one "end". 
>>> But I think we discussed that once.
>>
>> that is how it is modeled when only one of the ends is distributed. 
>> What we discussed was that the semantics of join were intended for 
>> conferences and that even though this was very similar, it was not the 
>> same thing... but the model is valid.
> 
> If the model is valid, then I think the mechanisms that go with it 
> should be valid too.
> 
> The conferencing mechanism could be used for either the "distributed" or 
> "centralized" approach. Then there needs to be a focus somewhere, and 
> the difference between the approaches is where it resides - at your own 
> end, or at the "other" end.
> 
>     Thanks,
>     Paul


From AUDET@nortel.com  Tue Jul 28 01:24:04 2009
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References: <20090702211501.52B5B3A6B6C@core3.amsl.com>	<4A4D3E55.80307@cisco.com><4A4E2E1E.1050509@sipstation.com> <4A4E96B1.1080005@cisco.com><EDC0A1AE77C57744B664A310A0B23AE2070BDB30@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com><4A6DA80F.2070003@cisco.com><EDC0A1AE77C57744B664A310A0B23AE2070BDF6C@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com> <4A6DC33F.7050108@cisco.com>
From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, "Alan Johnston" <alan@sipstation.com>
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Subject: Re: [dispatch] I-D Action:draft-johnston-sipping-cc-uui-08.txt
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You could always read the appropriate ITU specs:
http://www.itu.int/rec/T-REC-Q.957.1-199607-I/en

and

http://www.itu.int/rec/T-REC-Q.931-199805-I/en
(section 4.5.30)

And then, there would be all the "national" values,
and, even more importantly, vendor specific values. Large operators
and equipment manufacturers often have their own specificatons
for use of UUS.

UUS is more more complicated than just sending a blob in an=20
information element.

And as I said yesterday, there is more to it than just carrying the UUI =
IE
blob.

There is also Facility IE for example. The other thing is that UUS may =
be=20
related to what specific ISDN message carries the information. I don't =
think=20
this is addressed in the draft. So it may impose restrictions on when it =

is appropriate to use this draft, versus, for example, the alternative =
of=20
tunnelling the whole ISDN messages (=E0-la RFC 3204).

I believe the intent of this draft is to cover ONLY the=20
UUIE case, not the others (again, like Facility). Perhaps I'm wrong,
but at the very least, it needs to be clarified in the document.

Realistically, just defining the blob for carrying UUI will NOT allow
us to have any interoperability. It will just allow us to carry the =
blob.

Somebody else, perhaps other SDOs, or even just the vendors, would have
to define the interop procedures.

If it's something that is only meant to be vendor-specific, then I'm
not even sure we need any standardization. One can use one of the=20
existing mechanisms in SIP (e.g., URI parameter, MIME bodies, =
proprietary
headers) to achieve this.


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Monday, July 27, 2009 08:10
> To: DRAGE, Keith (Keith)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] I-D=20
> Action:draft-johnston-sipping-cc-uui-08.txt
>=20
>=20
>=20
> DRAGE, Keith (Keith) wrote:
> > 1st octet of the payload.
>=20
> Fine. But what does the first octet of the payload *mean*?
>=20
> Aren't the encodings corresponding to particular values of=20
> that octet defined somewhere? If so, how many of the 256=20
> possible values are defined, and who controls how new values=20
> can be defined?
>=20
> I'm pretty sure it isn't IETF or IANA.
>=20
> 	Thanks,
> 	Paul
>=20
> > Keith
> >=20
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Monday, July 27, 2009 2:14 PM
> >> To: DRAGE, Keith (Keith)
> >> Cc: dispatch@ietf.org
> >> Subject: Re: [dispatch] I-D
> >> Action:draft-johnston-sipping-cc-uui-08.txt
> >>
> >> Keith,
> >>
> >> If this is *not* ISDN encoding, then where is the=20
> indication of what=20
> >> encoding is being used? Are you saying that this is a private=20
> >> agreement between the UAC and UAS, neither signaled nor=20
> negotiated?=20
> >> How can that possibly work?
> >>
> >> I thought the first byte of the value was a registered format=20
> >> indicator.
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >> DRAGE, Keith (Keith) wrote:
> >>> Just to respond to this bit of the message:
> >>>
> >>>>> Not quite.  We are not making the assumption that both are
> >>>> gateways,
> >>>>> although that is possible.  Most likely, one element is
> >> SIP enabled.
> >>>>> However, it is not a fully intelligent SIP endpoint - the basic=20
> >>>>> software and application (probably many years old) is
> >>>> unchanged but a
> >>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
> >>>> stack does
> >>>>> not know anything about the UUI as it is implementing
> >>>> exactly the ISDN
> >>>>> UUI service - the information is just pushed up the stack
> >>>> to another
> >>>>> application, and this application knows nothing about SIP. =20
> >>>> It still
> >>>>> thinks it is using an ISDN trunk.
> >>>> Then in the way I meant the question, both ends *are* gateways.
> >>>>
> >>>> But then if someone wishes to build native sip gear to
> >> plug into this
> >>>> environment, then it will still have to understand this isdn=20
> >>>> encoding.
> >>>> Perhaps that is the cruel truth, unless all the equipment
> >> is replaced
> >>>> at once.
> >>> I've said this before and I'll say it again. The bit being
> >> delivered to the end UE is not "isdn encoding". If it was ISDN=20
> >> encoding, then the ISDN / SIP gateway would be able to=20
> interpret it=20
> >> and map it into SIP. Rather it is some application sitting=20
> above the=20
> >> ISDN call control layer in the calling terminal, which is=20
> wishing to=20
> >> communicate with its peer application running in the=20
> destination SIP=20
> >> UA, or gateway.
> >>> regards
> >>>
> >>> Keith
> >>>
> >>>> -----Original Message-----
> >>>> From: dispatch-bounces@ietf.org
> >>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> >>>> Sent: Saturday, July 04, 2009 12:39 AM
> >>>> To: Alan Johnston
> >>>> Cc: dispatch@ietf.org
> >>>> Subject: Re: [dispatch] I-D
> >>>> Action:draft-johnston-sipping-cc-uui-08.txt
> >>>>
> >>>> Hi Alan,
> >>>>
> >>>> responses inline. I've trimmed out the parts that don't
> >> require more
> >>>> discussion.
> >>>>
> >>>> 	Thanks,
> >>>> 	Paul
> >>>>
> >>>> Alan Johnston wrote:
> >>>>> Paul,
> >>>>>
> >>>>> Thanks for your comments.  See mine below.
> >>>>>
> >>>>> Thanks,
> >>>>> Alan
> >>>>>
> >>>>> Paul Kyzivat wrote:
> >>>>>> Hi Alan,
> >>>>>>
> >>>>>> I took a quick look at this doc and the requirements doc
> >>>> as well. I
> >>>>>> have a few thoughts:
> >>>>>>
> >>>>>> From section 1:
> >>>>>>
> >>>>>>>    In the future, where both endpoints are intelligent
> >>>> SIP user agents,
> >>>>>>>    it may be possible for them to understand and
> >>>> interpret the UUI data.
> >>>>>>>    There may be some cases where the UUI information is
> >>>> relevant to SIP.
> >>>>>>>    In this case, it might be worthwhile attempting to map
> >>>> UUI data to an
> >>>>>>>    appropriate SIP header field or to standardize a new
> >>>> header field.
> >>>>>>>    However, the requirements and use cases for this are
> >>>> different enough
> >>>>>>>    from those described in this document that these two
> >> situations
> >>>>>>>    should be examined separately.  This document looks
> >> only at the
> >>>>>>>    requirements and mechanisms for replicating the
> >>>> existing, widely used
> >>>>>>>    and deployed ISDN UUI service.
> >>>>>> This *still* troubles me!
> >>>>>> Is the assumption here that *both* the UAC and UAS are
> >>>> gateways, so
> >>>>>> that this mechanism is solely for the tunneling of Q.931
> >> and Q.763
> >>>>>> UUI message over SIP?
> >>>>>>
> >>>>>> I get the impression that an important use case is=20
> where at least
> >>>>>> *one* end (probably the UAS) is a SIP device. In that
> >> case it will
> >>>>>> already be *necessary* for it to understand and
> >> interpret UUI data.
> >>>>>> And where there is redundancy between SIP data and the UUI
> >>>> data, it
> >>>>>> will need to resolve the discrepancy.
> >>>>> Not quite.  We are not making the assumption that both are
> >>>> gateways,
> >>>>> although that is possible.  Most likely, one element is
> >> SIP enabled.
> >>>>> However, it is not a fully intelligent SIP endpoint - the basic=20
> >>>>> software and application (probably many years old) is
> >>>> unchanged but a
> >>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
> >>>> stack does
> >>>>> not know anything about the UUI as it is implementing
> >>>> exactly the ISDN
> >>>>> UUI service - the information is just pushed up the stack
> >>>> to another
> >>>>> application, and this application knows nothing about SIP. =20
> >>>> It still
> >>>>> thinks it is using an ISDN trunk.
> >>>> Then in the way I meant the question, both ends *are* gateways.
> >>>>
> >>>> But then if someone wishes to build native sip gear to
> >> plug into this
> >>>> environment, then it will still have to understand this isdn=20
> >>>> encoding.
> >>>> Perhaps that is the cruel truth, unless all the equipment
> >> is replaced
> >>>> at once.
> >>>>
> >>>>>>> 5.  Syntax for UUI Header Field
> >>>>>> ...
> >>>>>>>        UUI         =3D "User-to-User" HCOLON uui-data=20
> >>>> *(SEMI uui-param)
> >>>>>>>        uui-data    =3D token
> >>>>>>>        uui-param   =3D enc-param | cont-param | purp-param=20
> >>>> | generic-param
> >>>>>>>        enc-param   =3D "encoding=3D" ("hex" | token)
> >>>>>>>        cont-param  =3D "content=3D" ("isdn-uui" | token)
> >>>>>>>        purp-param  =3D "purpose=3D" ("isdn-interwork" | token)
> >>>>>> At the very least, I would request that this be expanded a
> >>>> little bit
> >>>>>> for future flexibility, by permitting the uui-data to be
> >> either a
> >>>>>> token or a string. While the encoding you have chosen=20
> works as a=20
> >>>>>> token, other encodings may need the additional flexibility
> >>>> of a string.
> >>>>>>        uui-data    =3D token | quoted-string
> >>>>> OK, I guess I'm OK with that.  For the isdn-interwork
> >>>> purpose, we'll
> >>>>> require the token, but other applications could utilize the
> >>>> quoted string.
> >>>>
> >>>> I would hope that when the token encoding is sufficient,
> >> then either
> >>>> form would be accepted. Or it would perhaps be ok to=20
> specify for a=20
> >>>> particular encoding (hex) that the token encoding must=20
> be used. I=20
> >>>> don't see how the purpose has anything to do with it.
> >>>>
> >>>>>>> 6.3.  Registration of SIP Option Tag
> >>>>>> This section registers the new option tag. But I find
> >>>> almost nothing
> >>>>>> that defines the semantics of the option. (Section 5 does
> >>>> include the
> >>>>>> following:
> >>>>>>
> >>>>>>>    A UA that supports this feature and the "uui" option
> >>>> tag MUST support
> >>>>>>>    the call flows in [johnston-dispatch-sip-cc-uui]...
> >>>>>> but that is pretty thin from a normative perspective.
> >>>>>>
> >>>>>> Of particular note, does support for the option tag mean
> >>>> support for
> >>>>>> the particular encoding, content, and purpose included in this=20
> >>>>>> document? Or does it only mean support for the header and
> >>>> params in
> >>>>>> the abstract? (That wouldn't be very useful.)
> >>>>> Yes, this needs clarification.  I'd suggest we define the
> >>>> option tag
> >>>>> to be uui-isdn which means it understands the header
> >> field and the
> >>>>> isdn-interwork purpose, and the call flows, including
> >> escaping into
> >>>>> redirection and Refer-To URIs.
> >>>> I'd like to hear some other opinions on whether to have
> >> tag per param
> >>>> for tag, or a tag for the combination of param values. I
> >> think I'm ok
> >>>> with what you propose.
> >>>>
> >>>>>> I don't see how it can mean support for some other yet to
> >>>> be defined
> >>>>>> values for those. Yet if it only means support for the
> >>>> current ones,
> >>>>>> then how will one indicate support for future ones?
> >>>>>>
> >>>>>> The only way out of this that I can see is to have
> >>>> separate options,
> >>>>>> either for particular values of each parameter, or for
> >> particular
> >>>>>> combinations of values of all the parameters.
> >>>>> New applications using this header field would have the=20
> option of=20
> >>>>> defining a new SIP option tag which would mean support for
> >>>> the header
> >>>>> field and their new purpose.
> >>>>>
> >>>>>> So you might have options uui-purpose-isdn-interwork,=20
> >>>>>> uui-content-isdn, and uui-encoding-hex. Or you might just
> >>>> have option
> >>>>>> uui-isdn-interwork that means the combination.
> >>>>> I think the latter.  I see it as a hierarchy - the=20
> application or=20
> >>>>> purpose is the top one.  Each purpose has a number of
> >>>> contents that it
> >>>>> supports.  Each content has a number of encoding schemes it
> >>>> supports.
> >>>>
> >>>> I guess I'm ok with that, pending nailing down the definition of=20
> >>>> "purpose".
> >>>>
> >>>>>> I also am not finding much explanation of the semantics of the=20
> >>>>>> content and purpose parameters. I can only extrapolate
> >>>> from a single
> >>>>>> example of each, which I'm having trouble doing.
> >>>>>>
> >>>>>> I'm guessing that 'content' must refer to the precise
> >>>> syntax of the
> >>>>>> contained data, after decoding. So presumably it must map
> >>>> onto some
> >>>>>> particular spec. For isdn-uui, would that be [Q931] or
> >>>> [Q763]? If so,
> >>>>>> shouldn't it refer to something specific in that document?
> >>>>> Not really.  In this case, isdn-uui effectively means
> >> "opaque data"=20
> >>>>> which is how ISDN defines the UUI - it is completely undefined,=20
> >>>>> necessarily so  by the strict layering used in the ISDN
> >> application.
> >>>> I don't believe you!
> >>>>
> >>>> If that is the case, then the UAC and UAS must have a=20
> private side=20
> >>>> agreement about what the isdn-uui content means. Is that
> >> really what
> >>>> you mean?
> >>>>
> >>>> You already state that the first byte is a protocol=20
> discriminator.=20
> >>>> There must also be something, somewhere, that specifies
> >> the mapping
> >>>> from a particular value for that byte to a particular=20
> >>>> protocol/format. If there is, then that should be part of the=20
> >>>> definition of this content.
> >>>>
> >>>>>> 'purpose' is even less clear to me. Does it refer to why
> >>>> it is being
> >>>>>> included? Or what is to be done with it? If received by a
> >>>> UA that is
> >>>>>> not an ISDN gateway, how can it decide if this is
> >>>> something it should
> >>>>>> be trying to process? Is it still ISDN interworking if
> >> neither the
> >>>>>> UAC nor UAS is connected to an ISDN network?
> >>>>> The purpose is the application using the UUI.  Others have
> >>>> expressed
> >>>>> an interest in carrying other data in the header field, in
> >>>> which case
> >>>>> a new purpose would be defined.  I'll see if I can think of
> >>>> an example one.
> >>>>
> >>>> So are you saying that a particular call center=20
> application might=20
> >>>> constitute a distinct "purpose"?
> >>>>
> >>>>> Basically, if two UAs both support the header but have=20
> different=20
> >>>>> applications generating and consuming the UUI, it will not
> >>>> work.  This
> >>>>> is not a SIP failure, and there is nothing SIP can do about
> >>>> it.  But
> >>>>> this purpose header field allows the UAs to detect this=20
> condition.
> >>>> In that case, perhaps "isdn-interwork" isn't really a
> >> single purpose
> >>>> at all, unless any two pieces of equipment that support
> >> that purpose
> >>>> can then interwork without knowing any more about each other.
> >>>>
> >>>>>> Suppose I was developing a sophisticated UA that
> >>>> potentially might be
> >>>>>> usable by a call center operator. How should
> >>>> purpose=3Disdn-interwork
> >>>>>> affect the operation of this device?
> >>>>> To make it work in today's contact center applications,
> >> supporting
> >>>>> isdn-interwork would likely make it work in an application,
> >>>> provided
> >>>>> the appropriate call center application also ran on it. =20
> >>>> Some contact
> >>>>> centers do not use ISDN UUI, instead they use all kinds
> >> of really,
> >>>>> really ugly CTI protocols running in parallel.
> >>>> Yes, I know! :-(
> >>>>
> >>>>> This is a way of moving
> >>>>> those to the ISDN model, and from there to a more
> >>>> intelligent endpoint
> >>>>> SIP model.
> >>>>>
> >>>>>>     Thanks,
> >>>>>>     Paul
> >>>> _______________________________________________
> >>>> 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

From salvatore.loreto@ericsson.com  Tue Jul 28 01:28:06 2009
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [dispatch] Disaggregated Media in SIP
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Paul Kyzivat wrote:
>
>
> Salvatore Loreto wrote:
>> Dale Worley wrote:
>>> On Tue, 2009-07-21 at 11:05 +0300, Gonzalo Camarillo wrote:
>>>  
>>>> well, 3pcc is centralized because you have a (central) controller 
>>>> that is in control of all SIP signalling. If that controller leaves 
>>>> the session, the whole session disappears. An approach where 
>>>> several devices have their own SIP dialog with the remote devices 
>>>> is not centralized because you can remove any of them and the other 
>>>> sessions will remain unaffected.
>>>>     
>>>
>>> It's true that if the implementation is a single dialog, then the UA at
>>> each end must remain in the call.  For a UA to replace itself with
>>> another UA would require a transfer-like process.
>>>
>>> The difficulty (under either implementation) is to transfer the "state"
>>> information (which media stream goes to which destination) from one
>>> controller to another.  That need is obvious in the centralized
>>> approach, but it seems to me that it would also be needed in the
>>> decentralized approach.
>>>   
>> the need or not to transfer the "state" really depends on how the 
>> decentralized approach will be designed.
>>
>> for example:
>> In a "loosely coupled control" approach, as Francois is proposing, 
>> you don't have any "state" information that
>> need to be transferred from one to another because you don't have a 
>> real controller.
>
> I don't understand that.
>
> Suppose Alice calls Bob using her voice phone. Then she arranges for 
> her laptop to join the call with Bob to add voice. (Bob has a video 
> phone.)
>
> Then Bob decides to do a consultative transfer of the call to Charlie. 
> Because Bob's phone is a video phone, its UI presumably treats this as 
> a single call, even if there are actually two calls behind the scenes. 
> So Bob first puts the call with Alice on hold. It already has to know 
> to do this on two separate calls. Then it calls Charlie, who also has 
> a video phone. So there is an audio+video call from Bob to Charlie. 
> Then Bob asks to complete the transfer. What does it do? Normally it 
> would send a REFER to Charlie's phone, asking it to do an 
> INVITE/Replaces to Alice. But there are two calls to Alice.
>
> The "state" of the call, that it is distributed across two dialogs, is 
> important info that needs to be transferred to Charlie given this 
> implementation approach. With the "centralized" approach that state 
> resides in Alice's realm and is of no concern to either Bob or Charlie.

in the "disaggregated media" approach the state of the call will still 
reside in Alice's realm,
however there will be some difference with the "centralized" approach 
because you don't have a controller and the state of the call
is distributed among the different UAs present in the Alice's realm.


Thanks
/Sal

From drage@alcatel-lucent.com  Tue Jul 28 01:39:49 2009
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To: Francois Audet <audet@nortel.com>, Paul Kyzivat <pkyzivat@cisco.com>, Alan Johnston <alan@sipstation.com>
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Thread-Topic: [dispatch] I-D Action:draft-johnston-sipping-cc-uui-08.txt
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If people read ITU-T Recommendation Q.957.1 (which incidently is the correc=
t reference that should appear in the charter proposal and not Q.931) the k=
ey works you are looking for are UUS service 1 implicit. This is "implicit"=
 in that the service is requested merely by the presence of the data to be =
transferred and does not involve the use of the Facility information elemen=
t to control this.

Q.957.1 the references Q.931 for the coding of the information elements req=
uired by the procedures in Q.957.1.

(The procedures in ITU-T Recommendation Q.931 defines two different usages:
-       the mapping of user data in X.29 into Q.931 packet data.
-       the user to user bearer service which essentially defines a call th=
at consists of the control signalling only (plus encapsulated data in the s=
ignalling) and has no separate bearer path).

regards

Keith

> -----Original Message-----
> From: Francois Audet [mailto:audet@nortel.com]
> Sent: Tuesday, July 28, 2009 9:23 AM
> To: Paul Kyzivat; DRAGE, Keith (Keith); Alan Johnston
> Cc: dispatch@ietf.org
> Subject: RE: [dispatch] I-D
> Action:draft-johnston-sipping-cc-uui-08.txt
>
> You could always read the appropriate ITU specs:
> http://www.itu.int/rec/T-REC-Q.957.1-199607-I/en
>
> and
>
> http://www.itu.int/rec/T-REC-Q.931-199805-I/en
> (section 4.5.30)
>
> And then, there would be all the "national" values, and, even
> more importantly, vendor specific values. Large operators and
> equipment manufacturers often have their own specificatons
> for use of UUS.
>
> UUS is more more complicated than just sending a blob in an
> information element.
>
> And as I said yesterday, there is more to it than just
> carrying the UUI IE blob.
>
> There is also Facility IE for example. The other thing is
> that UUS may be related to what specific ISDN message carries
> the information. I don't think this is addressed in the
> draft. So it may impose restrictions on when it is
> appropriate to use this draft, versus, for example, the
> alternative of tunnelling the whole ISDN messages (=E0-la RFC 3204).
>
> I believe the intent of this draft is to cover ONLY the UUIE
> case, not the others (again, like Facility). Perhaps I'm
> wrong, but at the very least, it needs to be clarified in the
> document.
>
> Realistically, just defining the blob for carrying UUI will
> NOT allow us to have any interoperability. It will just allow
> us to carry the blob.
>
> Somebody else, perhaps other SDOs, or even just the vendors,
> would have to define the interop procedures.
>
> If it's something that is only meant to be vendor-specific,
> then I'm not even sure we need any standardization. One can
> use one of the existing mechanisms in SIP (e.g., URI
> parameter, MIME bodies, proprietary
> headers) to achieve this.
>
>
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> > Sent: Monday, July 27, 2009 08:10
> > To: DRAGE, Keith (Keith)
> > Cc: dispatch@ietf.org
> > Subject: Re: [dispatch] I-D
> > Action:draft-johnston-sipping-cc-uui-08.txt
> >
> >
> >
> > DRAGE, Keith (Keith) wrote:
> > > 1st octet of the payload.
> >
> > Fine. But what does the first octet of the payload *mean*?
> >
> > Aren't the encodings corresponding to particular values of
> > that octet defined somewhere? If so, how many of the 256
> > possible values are defined, and who controls how new values
> > can be defined?
> >
> > I'm pretty sure it isn't IETF or IANA.
> >
> >     Thanks,
> >     Paul
> >
> > > Keith
> > >
> > >> -----Original Message-----
> > >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > >> Sent: Monday, July 27, 2009 2:14 PM
> > >> To: DRAGE, Keith (Keith)
> > >> Cc: dispatch@ietf.org
> > >> Subject: Re: [dispatch] I-D
> > >> Action:draft-johnston-sipping-cc-uui-08.txt
> > >>
> > >> Keith,
> > >>
> > >> If this is *not* ISDN encoding, then where is the
> > indication of what
> > >> encoding is being used? Are you saying that this is a private
> > >> agreement between the UAC and UAS, neither signaled nor
> > negotiated?
> > >> How can that possibly work?
> > >>
> > >> I thought the first byte of the value was a registered format
> > >> indicator.
> > >>
> > >>  Thanks,
> > >>  Paul
> > >>
> > >> DRAGE, Keith (Keith) wrote:
> > >>> Just to respond to this bit of the message:
> > >>>
> > >>>>> Not quite.  We are not making the assumption that both are
> > >>>> gateways,
> > >>>>> although that is possible.  Most likely, one element is
> > >> SIP enabled.
> > >>>>> However, it is not a fully intelligent SIP endpoint -
> the basic
> > >>>>> software and application (probably many years old) is
> > >>>> unchanged but a
> > >>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
> > >>>> stack does
> > >>>>> not know anything about the UUI as it is implementing
> > >>>> exactly the ISDN
> > >>>>> UUI service - the information is just pushed up the stack
> > >>>> to another
> > >>>>> application, and this application knows nothing about SIP.
> > >>>> It still
> > >>>>> thinks it is using an ISDN trunk.
> > >>>> Then in the way I meant the question, both ends *are* gateways.
> > >>>>
> > >>>> But then if someone wishes to build native sip gear to
> > >> plug into this
> > >>>> environment, then it will still have to understand this isdn
> > >>>> encoding.
> > >>>> Perhaps that is the cruel truth, unless all the equipment
> > >> is replaced
> > >>>> at once.
> > >>> I've said this before and I'll say it again. The bit being
> > >> delivered to the end UE is not "isdn encoding". If it was ISDN
> > >> encoding, then the ISDN / SIP gateway would be able to
> > interpret it
> > >> and map it into SIP. Rather it is some application sitting
> > above the
> > >> ISDN call control layer in the calling terminal, which is
> > wishing to
> > >> communicate with its peer application running in the
> > destination SIP
> > >> UA, or gateway.
> > >>> regards
> > >>>
> > >>> Keith
> > >>>
> > >>>> -----Original Message-----
> > >>>> From: dispatch-bounces@ietf.org
> > >>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> > >>>> Sent: Saturday, July 04, 2009 12:39 AM
> > >>>> To: Alan Johnston
> > >>>> Cc: dispatch@ietf.org
> > >>>> Subject: Re: [dispatch] I-D
> > >>>> Action:draft-johnston-sipping-cc-uui-08.txt
> > >>>>
> > >>>> Hi Alan,
> > >>>>
> > >>>> responses inline. I've trimmed out the parts that don't
> > >> require more
> > >>>> discussion.
> > >>>>
> > >>>>        Thanks,
> > >>>>        Paul
> > >>>>
> > >>>> Alan Johnston wrote:
> > >>>>> Paul,
> > >>>>>
> > >>>>> Thanks for your comments.  See mine below.
> > >>>>>
> > >>>>> Thanks,
> > >>>>> Alan
> > >>>>>
> > >>>>> Paul Kyzivat wrote:
> > >>>>>> Hi Alan,
> > >>>>>>
> > >>>>>> I took a quick look at this doc and the requirements doc
> > >>>> as well. I
> > >>>>>> have a few thoughts:
> > >>>>>>
> > >>>>>> From section 1:
> > >>>>>>
> > >>>>>>>    In the future, where both endpoints are intelligent
> > >>>> SIP user agents,
> > >>>>>>>    it may be possible for them to understand and
> > >>>> interpret the UUI data.
> > >>>>>>>    There may be some cases where the UUI information is
> > >>>> relevant to SIP.
> > >>>>>>>    In this case, it might be worthwhile attempting to map
> > >>>> UUI data to an
> > >>>>>>>    appropriate SIP header field or to standardize a new
> > >>>> header field.
> > >>>>>>>    However, the requirements and use cases for this are
> > >>>> different enough
> > >>>>>>>    from those described in this document that these two
> > >> situations
> > >>>>>>>    should be examined separately.  This document looks
> > >> only at the
> > >>>>>>>    requirements and mechanisms for replicating the
> > >>>> existing, widely used
> > >>>>>>>    and deployed ISDN UUI service.
> > >>>>>> This *still* troubles me!
> > >>>>>> Is the assumption here that *both* the UAC and UAS are
> > >>>> gateways, so
> > >>>>>> that this mechanism is solely for the tunneling of Q.931
> > >> and Q.763
> > >>>>>> UUI message over SIP?
> > >>>>>>
> > >>>>>> I get the impression that an important use case is
> > where at least
> > >>>>>> *one* end (probably the UAS) is a SIP device. In that
> > >> case it will
> > >>>>>> already be *necessary* for it to understand and
> > >> interpret UUI data.
> > >>>>>> And where there is redundancy between SIP data and the UUI
> > >>>> data, it
> > >>>>>> will need to resolve the discrepancy.
> > >>>>> Not quite.  We are not making the assumption that both are
> > >>>> gateways,
> > >>>>> although that is possible.  Most likely, one element is
> > >> SIP enabled.
> > >>>>> However, it is not a fully intelligent SIP endpoint -
> the basic
> > >>>>> software and application (probably many years old) is
> > >>>> unchanged but a
> > >>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
> > >>>> stack does
> > >>>>> not know anything about the UUI as it is implementing
> > >>>> exactly the ISDN
> > >>>>> UUI service - the information is just pushed up the stack
> > >>>> to another
> > >>>>> application, and this application knows nothing about SIP.
> > >>>> It still
> > >>>>> thinks it is using an ISDN trunk.
> > >>>> Then in the way I meant the question, both ends *are* gateways.
> > >>>>
> > >>>> But then if someone wishes to build native sip gear to
> > >> plug into this
> > >>>> environment, then it will still have to understand this isdn
> > >>>> encoding.
> > >>>> Perhaps that is the cruel truth, unless all the equipment
> > >> is replaced
> > >>>> at once.
> > >>>>
> > >>>>>>> 5.  Syntax for UUI Header Field
> > >>>>>> ...
> > >>>>>>>        UUI         =3D "User-to-User" HCOLON uui-data
> > >>>> *(SEMI uui-param)
> > >>>>>>>        uui-data    =3D token
> > >>>>>>>        uui-param   =3D enc-param | cont-param | purp-param
> > >>>> | generic-param
> > >>>>>>>        enc-param   =3D "encoding=3D" ("hex" | token)
> > >>>>>>>        cont-param  =3D "content=3D" ("isdn-uui" | token)
> > >>>>>>>        purp-param  =3D "purpose=3D" ("isdn-interwork" | token)
> > >>>>>> At the very least, I would request that this be expanded a
> > >>>> little bit
> > >>>>>> for future flexibility, by permitting the uui-data to be
> > >> either a
> > >>>>>> token or a string. While the encoding you have chosen
> > works as a
> > >>>>>> token, other encodings may need the additional flexibility
> > >>>> of a string.
> > >>>>>>        uui-data    =3D token | quoted-string
> > >>>>> OK, I guess I'm OK with that.  For the isdn-interwork
> > >>>> purpose, we'll
> > >>>>> require the token, but other applications could utilize the
> > >>>> quoted string.
> > >>>>
> > >>>> I would hope that when the token encoding is sufficient,
> > >> then either
> > >>>> form would be accepted. Or it would perhaps be ok to
> > specify for a
> > >>>> particular encoding (hex) that the token encoding must
> > be used. I
> > >>>> don't see how the purpose has anything to do with it.
> > >>>>
> > >>>>>>> 6.3.  Registration of SIP Option Tag
> > >>>>>> This section registers the new option tag. But I find
> > >>>> almost nothing
> > >>>>>> that defines the semantics of the option. (Section 5 does
> > >>>> include the
> > >>>>>> following:
> > >>>>>>
> > >>>>>>>    A UA that supports this feature and the "uui" option
> > >>>> tag MUST support
> > >>>>>>>    the call flows in [johnston-dispatch-sip-cc-uui]...
> > >>>>>> but that is pretty thin from a normative perspective.
> > >>>>>>
> > >>>>>> Of particular note, does support for the option tag mean
> > >>>> support for
> > >>>>>> the particular encoding, content, and purpose
> included in this
> > >>>>>> document? Or does it only mean support for the header and
> > >>>> params in
> > >>>>>> the abstract? (That wouldn't be very useful.)
> > >>>>> Yes, this needs clarification.  I'd suggest we define the
> > >>>> option tag
> > >>>>> to be uui-isdn which means it understands the header
> > >> field and the
> > >>>>> isdn-interwork purpose, and the call flows, including
> > >> escaping into
> > >>>>> redirection and Refer-To URIs.
> > >>>> I'd like to hear some other opinions on whether to have
> > >> tag per param
> > >>>> for tag, or a tag for the combination of param values. I
> > >> think I'm ok
> > >>>> with what you propose.
> > >>>>
> > >>>>>> I don't see how it can mean support for some other yet to
> > >>>> be defined
> > >>>>>> values for those. Yet if it only means support for the
> > >>>> current ones,
> > >>>>>> then how will one indicate support for future ones?
> > >>>>>>
> > >>>>>> The only way out of this that I can see is to have
> > >>>> separate options,
> > >>>>>> either for particular values of each parameter, or for
> > >> particular
> > >>>>>> combinations of values of all the parameters.
> > >>>>> New applications using this header field would have the
> > option of
> > >>>>> defining a new SIP option tag which would mean support for
> > >>>> the header
> > >>>>> field and their new purpose.
> > >>>>>
> > >>>>>> So you might have options uui-purpose-isdn-interwork,
> > >>>>>> uui-content-isdn, and uui-encoding-hex. Or you might just
> > >>>> have option
> > >>>>>> uui-isdn-interwork that means the combination.
> > >>>>> I think the latter.  I see it as a hierarchy - the
> > application or
> > >>>>> purpose is the top one.  Each purpose has a number of
> > >>>> contents that it
> > >>>>> supports.  Each content has a number of encoding schemes it
> > >>>> supports.
> > >>>>
> > >>>> I guess I'm ok with that, pending nailing down the
> definition of
> > >>>> "purpose".
> > >>>>
> > >>>>>> I also am not finding much explanation of the
> semantics of the
> > >>>>>> content and purpose parameters. I can only extrapolate
> > >>>> from a single
> > >>>>>> example of each, which I'm having trouble doing.
> > >>>>>>
> > >>>>>> I'm guessing that 'content' must refer to the precise
> > >>>> syntax of the
> > >>>>>> contained data, after decoding. So presumably it must map
> > >>>> onto some
> > >>>>>> particular spec. For isdn-uui, would that be [Q931] or
> > >>>> [Q763]? If so,
> > >>>>>> shouldn't it refer to something specific in that document?
> > >>>>> Not really.  In this case, isdn-uui effectively means
> > >> "opaque data"
> > >>>>> which is how ISDN defines the UUI - it is completely
> undefined,
> > >>>>> necessarily so  by the strict layering used in the ISDN
> > >> application.
> > >>>> I don't believe you!
> > >>>>
> > >>>> If that is the case, then the UAC and UAS must have a
> > private side
> > >>>> agreement about what the isdn-uui content means. Is that
> > >> really what
> > >>>> you mean?
> > >>>>
> > >>>> You already state that the first byte is a protocol
> > discriminator.
> > >>>> There must also be something, somewhere, that specifies
> > >> the mapping
> > >>>> from a particular value for that byte to a particular
> > >>>> protocol/format. If there is, then that should be part of the
> > >>>> definition of this content.
> > >>>>
> > >>>>>> 'purpose' is even less clear to me. Does it refer to why
> > >>>> it is being
> > >>>>>> included? Or what is to be done with it? If received by a
> > >>>> UA that is
> > >>>>>> not an ISDN gateway, how can it decide if this is
> > >>>> something it should
> > >>>>>> be trying to process? Is it still ISDN interworking if
> > >> neither the
> > >>>>>> UAC nor UAS is connected to an ISDN network?
> > >>>>> The purpose is the application using the UUI.  Others have
> > >>>> expressed
> > >>>>> an interest in carrying other data in the header field, in
> > >>>> which case
> > >>>>> a new purpose would be defined.  I'll see if I can think of
> > >>>> an example one.
> > >>>>
> > >>>> So are you saying that a particular call center
> > application might
> > >>>> constitute a distinct "purpose"?
> > >>>>
> > >>>>> Basically, if two UAs both support the header but have
> > different
> > >>>>> applications generating and consuming the UUI, it will not
> > >>>> work.  This
> > >>>>> is not a SIP failure, and there is nothing SIP can do about
> > >>>> it.  But
> > >>>>> this purpose header field allows the UAs to detect this
> > condition.
> > >>>> In that case, perhaps "isdn-interwork" isn't really a
> > >> single purpose
> > >>>> at all, unless any two pieces of equipment that support
> > >> that purpose
> > >>>> can then interwork without knowing any more about each other.
> > >>>>
> > >>>>>> Suppose I was developing a sophisticated UA that
> > >>>> potentially might be
> > >>>>>> usable by a call center operator. How should
> > >>>> purpose=3Disdn-interwork
> > >>>>>> affect the operation of this device?
> > >>>>> To make it work in today's contact center applications,
> > >> supporting
> > >>>>> isdn-interwork would likely make it work in an application,
> > >>>> provided
> > >>>>> the appropriate call center application also ran on it.
> > >>>> Some contact
> > >>>>> centers do not use ISDN UUI, instead they use all kinds
> > >> of really,
> > >>>>> really ugly CTI protocols running in parallel.
> > >>>> Yes, I know! :-(
> > >>>>
> > >>>>> This is a way of moving
> > >>>>> those to the ISDN model, and from there to a more
> > >>>> intelligent endpoint
> > >>>>> SIP model.
> > >>>>>
> > >>>>>>     Thanks,
> > >>>>>>     Paul
> > >>>> _______________________________________________
> > >>>> 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] Disaggregated Media in SIP
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Gonzalo Camarillo wrote:
> Hi Paul,
> 
> yes, what you say is totally correct when one of the endpoints is not 
> distributed. That case should be relatively-easy to resolve. The case 
> where both ends are distributed is, of course, more "interesting" ;o)

I don't understand at all how the "distributed" model works when both 
ends want to be distributed.

	Thanks,
	Paul

> Cheers,
> 
> Gonzalo
> 
> Paul Kyzivat wrote:
>>
>>
>> Gonzalo Camarillo wrote:
>>> Hi,
>>>
>>>> I guess the one way I could buy into your "distributed" approach is 
>>>> to simply model it as a conference, where the focus is at one "end". 
>>>> But I think we discussed that once.
>>>
>>> that is how it is modeled when only one of the ends is distributed. 
>>> What we discussed was that the semantics of join were intended for 
>>> conferences and that even though this was very similar, it was not 
>>> the same thing... but the model is valid.
>>
>> If the model is valid, then I think the mechanisms that go with it 
>> should be valid too.
>>
>> The conferencing mechanism could be used for either the "distributed" 
>> or "centralized" approach. Then there needs to be a focus somewhere, 
>> and the difference between the approaches is where it resides - at 
>> your own end, or at the "other" end.
>>
>>     Thanks,
>>     Paul
> 
> 

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Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action:draft-johnston-sipping-cc-uui-08.txt
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Francois Audet wrote:
> You could always read the appropriate ITU specs:
> http://www.itu.int/rec/T-REC-Q.957.1-199607-I/en
> 
> and
> 
> http://www.itu.int/rec/T-REC-Q.931-199805-I/en
> (section 4.5.30)

How would I know those are the "appropriate" specs to read?
In the doc, Q.957 isn't mentioned at all. Q.931 is referenced as an 
informative document in the list of references, and referenced 
non-normatively in the first paragraph of the Overview.

There is no linkage at all of encoding/content/purpose params to either 
of these specs.

> And then, there would be all the "national" values,
> and, even more importantly, vendor specific values. Large operators
> and equipment manufacturers often have their own specificatons
> for use of UUS.

So, how would the use of national and/or vendor specific values be 
indicated in this header?

> UUS is more more complicated than just sending a blob in an 
> information element.
> 
> And as I said yesterday, there is more to it than just carrying the UUI IE
> blob.
> 
> There is also Facility IE for example. The other thing is that UUS may be 
> related to what specific ISDN message carries the information. I don't think 
> this is addressed in the draft. So it may impose restrictions on when it 
> is appropriate to use this draft, versus, for example, the alternative of 
> tunnelling the whole ISDN messages (à-la RFC 3204).
> 
> I believe the intent of this draft is to cover ONLY the 
> UUIE case, not the others (again, like Facility). Perhaps I'm wrong,
> but at the very least, it needs to be clarified in the document.
> 
> Realistically, just defining the blob for carrying UUI will NOT allow
> us to have any interoperability. It will just allow us to carry the blob.

That is a big part of my concern.

I am looking for the direct linkage from the values in the header to the 
specs that define what those values mean.

> Somebody else, perhaps other SDOs, or even just the vendors, would have
> to define the interop procedures.

That would be a problem.

> If it's something that is only meant to be vendor-specific, then I'm
> not even sure we need any standardization. One can use one of the 
> existing mechanisms in SIP (e.g., URI parameter, MIME bodies, proprietary
> headers) to achieve this.

I agree.

	Thanks,
	Paul

>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: Monday, July 27, 2009 08:10
>> To: DRAGE, Keith (Keith)
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] I-D 
>> Action:draft-johnston-sipping-cc-uui-08.txt
>>
>>
>>
>> DRAGE, Keith (Keith) wrote:
>>> 1st octet of the payload.
>> Fine. But what does the first octet of the payload *mean*?
>>
>> Aren't the encodings corresponding to particular values of 
>> that octet defined somewhere? If so, how many of the 256 
>> possible values are defined, and who controls how new values 
>> can be defined?
>>
>> I'm pretty sure it isn't IETF or IANA.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Keith
>>>
>>>> -----Original Message-----
>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>> Sent: Monday, July 27, 2009 2:14 PM
>>>> To: DRAGE, Keith (Keith)
>>>> Cc: dispatch@ietf.org
>>>> Subject: Re: [dispatch] I-D
>>>> Action:draft-johnston-sipping-cc-uui-08.txt
>>>>
>>>> Keith,
>>>>
>>>> If this is *not* ISDN encoding, then where is the 
>> indication of what 
>>>> encoding is being used? Are you saying that this is a private 
>>>> agreement between the UAC and UAS, neither signaled nor 
>> negotiated? 
>>>> How can that possibly work?
>>>>
>>>> I thought the first byte of the value was a registered format 
>>>> indicator.
>>>>
>>>> 	Thanks,
>>>> 	Paul
>>>>
>>>> DRAGE, Keith (Keith) wrote:
>>>>> Just to respond to this bit of the message:
>>>>>
>>>>>>> Not quite.  We are not making the assumption that both are
>>>>>> gateways,
>>>>>>> although that is possible.  Most likely, one element is
>>>> SIP enabled.
>>>>>>> However, it is not a fully intelligent SIP endpoint - the basic 
>>>>>>> software and application (probably many years old) is
>>>>>> unchanged but a
>>>>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
>>>>>> stack does
>>>>>>> not know anything about the UUI as it is implementing
>>>>>> exactly the ISDN
>>>>>>> UUI service - the information is just pushed up the stack
>>>>>> to another
>>>>>>> application, and this application knows nothing about SIP.  
>>>>>> It still
>>>>>>> thinks it is using an ISDN trunk.
>>>>>> Then in the way I meant the question, both ends *are* gateways.
>>>>>>
>>>>>> But then if someone wishes to build native sip gear to
>>>> plug into this
>>>>>> environment, then it will still have to understand this isdn 
>>>>>> encoding.
>>>>>> Perhaps that is the cruel truth, unless all the equipment
>>>> is replaced
>>>>>> at once.
>>>>> I've said this before and I'll say it again. The bit being
>>>> delivered to the end UE is not "isdn encoding". If it was ISDN 
>>>> encoding, then the ISDN / SIP gateway would be able to 
>> interpret it 
>>>> and map it into SIP. Rather it is some application sitting 
>> above the 
>>>> ISDN call control layer in the calling terminal, which is 
>> wishing to 
>>>> communicate with its peer application running in the 
>> destination SIP 
>>>> UA, or gateway.
>>>>> regards
>>>>>
>>>>> Keith
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: dispatch-bounces@ietf.org
>>>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>>>>>> Sent: Saturday, July 04, 2009 12:39 AM
>>>>>> To: Alan Johnston
>>>>>> Cc: dispatch@ietf.org
>>>>>> Subject: Re: [dispatch] I-D
>>>>>> Action:draft-johnston-sipping-cc-uui-08.txt
>>>>>>
>>>>>> Hi Alan,
>>>>>>
>>>>>> responses inline. I've trimmed out the parts that don't
>>>> require more
>>>>>> discussion.
>>>>>>
>>>>>> 	Thanks,
>>>>>> 	Paul
>>>>>>
>>>>>> Alan Johnston wrote:
>>>>>>> Paul,
>>>>>>>
>>>>>>> Thanks for your comments.  See mine below.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Alan
>>>>>>>
>>>>>>> Paul Kyzivat wrote:
>>>>>>>> Hi Alan,
>>>>>>>>
>>>>>>>> I took a quick look at this doc and the requirements doc
>>>>>> as well. I
>>>>>>>> have a few thoughts:
>>>>>>>>
>>>>>>>> From section 1:
>>>>>>>>
>>>>>>>>>    In the future, where both endpoints are intelligent
>>>>>> SIP user agents,
>>>>>>>>>    it may be possible for them to understand and
>>>>>> interpret the UUI data.
>>>>>>>>>    There may be some cases where the UUI information is
>>>>>> relevant to SIP.
>>>>>>>>>    In this case, it might be worthwhile attempting to map
>>>>>> UUI data to an
>>>>>>>>>    appropriate SIP header field or to standardize a new
>>>>>> header field.
>>>>>>>>>    However, the requirements and use cases for this are
>>>>>> different enough
>>>>>>>>>    from those described in this document that these two
>>>> situations
>>>>>>>>>    should be examined separately.  This document looks
>>>> only at the
>>>>>>>>>    requirements and mechanisms for replicating the
>>>>>> existing, widely used
>>>>>>>>>    and deployed ISDN UUI service.
>>>>>>>> This *still* troubles me!
>>>>>>>> Is the assumption here that *both* the UAC and UAS are
>>>>>> gateways, so
>>>>>>>> that this mechanism is solely for the tunneling of Q.931
>>>> and Q.763
>>>>>>>> UUI message over SIP?
>>>>>>>>
>>>>>>>> I get the impression that an important use case is 
>> where at least
>>>>>>>> *one* end (probably the UAS) is a SIP device. In that
>>>> case it will
>>>>>>>> already be *necessary* for it to understand and
>>>> interpret UUI data.
>>>>>>>> And where there is redundancy between SIP data and the UUI
>>>>>> data, it
>>>>>>>> will need to resolve the discrepancy.
>>>>>>> Not quite.  We are not making the assumption that both are
>>>>>> gateways,
>>>>>>> although that is possible.  Most likely, one element is
>>>> SIP enabled.
>>>>>>> However, it is not a fully intelligent SIP endpoint - the basic 
>>>>>>> software and application (probably many years old) is
>>>>>> unchanged but a
>>>>>>> SIP interface (SIP trunk) has been added.  As such, the SIP
>>>>>> stack does
>>>>>>> not know anything about the UUI as it is implementing
>>>>>> exactly the ISDN
>>>>>>> UUI service - the information is just pushed up the stack
>>>>>> to another
>>>>>>> application, and this application knows nothing about SIP.  
>>>>>> It still
>>>>>>> thinks it is using an ISDN trunk.
>>>>>> Then in the way I meant the question, both ends *are* gateways.
>>>>>>
>>>>>> But then if someone wishes to build native sip gear to
>>>> plug into this
>>>>>> environment, then it will still have to understand this isdn 
>>>>>> encoding.
>>>>>> Perhaps that is the cruel truth, unless all the equipment
>>>> is replaced
>>>>>> at once.
>>>>>>
>>>>>>>>> 5.  Syntax for UUI Header Field
>>>>>>>> ...
>>>>>>>>>        UUI         = "User-to-User" HCOLON uui-data 
>>>>>> *(SEMI uui-param)
>>>>>>>>>        uui-data    = token
>>>>>>>>>        uui-param   = enc-param | cont-param | purp-param 
>>>>>> | generic-param
>>>>>>>>>        enc-param   = "encoding=" ("hex" | token)
>>>>>>>>>        cont-param  = "content=" ("isdn-uui" | token)
>>>>>>>>>        purp-param  = "purpose=" ("isdn-interwork" | token)
>>>>>>>> At the very least, I would request that this be expanded a
>>>>>> little bit
>>>>>>>> for future flexibility, by permitting the uui-data to be
>>>> either a
>>>>>>>> token or a string. While the encoding you have chosen 
>> works as a 
>>>>>>>> token, other encodings may need the additional flexibility
>>>>>> of a string.
>>>>>>>>        uui-data    = token | quoted-string
>>>>>>> OK, I guess I'm OK with that.  For the isdn-interwork
>>>>>> purpose, we'll
>>>>>>> require the token, but other applications could utilize the
>>>>>> quoted string.
>>>>>>
>>>>>> I would hope that when the token encoding is sufficient,
>>>> then either
>>>>>> form would be accepted. Or it would perhaps be ok to 
>> specify for a 
>>>>>> particular encoding (hex) that the token encoding must 
>> be used. I 
>>>>>> don't see how the purpose has anything to do with it.
>>>>>>
>>>>>>>>> 6.3.  Registration of SIP Option Tag
>>>>>>>> This section registers the new option tag. But I find
>>>>>> almost nothing
>>>>>>>> that defines the semantics of the option. (Section 5 does
>>>>>> include the
>>>>>>>> following:
>>>>>>>>
>>>>>>>>>    A UA that supports this feature and the "uui" option
>>>>>> tag MUST support
>>>>>>>>>    the call flows in [johnston-dispatch-sip-cc-uui]...
>>>>>>>> but that is pretty thin from a normative perspective.
>>>>>>>>
>>>>>>>> Of particular note, does support for the option tag mean
>>>>>> support for
>>>>>>>> the particular encoding, content, and purpose included in this 
>>>>>>>> document? Or does it only mean support for the header and
>>>>>> params in
>>>>>>>> the abstract? (That wouldn't be very useful.)
>>>>>>> Yes, this needs clarification.  I'd suggest we define the
>>>>>> option tag
>>>>>>> to be uui-isdn which means it understands the header
>>>> field and the
>>>>>>> isdn-interwork purpose, and the call flows, including
>>>> escaping into
>>>>>>> redirection and Refer-To URIs.
>>>>>> I'd like to hear some other opinions on whether to have
>>>> tag per param
>>>>>> for tag, or a tag for the combination of param values. I
>>>> think I'm ok
>>>>>> with what you propose.
>>>>>>
>>>>>>>> I don't see how it can mean support for some other yet to
>>>>>> be defined
>>>>>>>> values for those. Yet if it only means support for the
>>>>>> current ones,
>>>>>>>> then how will one indicate support for future ones?
>>>>>>>>
>>>>>>>> The only way out of this that I can see is to have
>>>>>> separate options,
>>>>>>>> either for particular values of each parameter, or for
>>>> particular
>>>>>>>> combinations of values of all the parameters.
>>>>>>> New applications using this header field would have the 
>> option of 
>>>>>>> defining a new SIP option tag which would mean support for
>>>>>> the header
>>>>>>> field and their new purpose.
>>>>>>>
>>>>>>>> So you might have options uui-purpose-isdn-interwork, 
>>>>>>>> uui-content-isdn, and uui-encoding-hex. Or you might just
>>>>>> have option
>>>>>>>> uui-isdn-interwork that means the combination.
>>>>>>> I think the latter.  I see it as a hierarchy - the 
>> application or 
>>>>>>> purpose is the top one.  Each purpose has a number of
>>>>>> contents that it
>>>>>>> supports.  Each content has a number of encoding schemes it
>>>>>> supports.
>>>>>>
>>>>>> I guess I'm ok with that, pending nailing down the definition of 
>>>>>> "purpose".
>>>>>>
>>>>>>>> I also am not finding much explanation of the semantics of the 
>>>>>>>> content and purpose parameters. I can only extrapolate
>>>>>> from a single
>>>>>>>> example of each, which I'm having trouble doing.
>>>>>>>>
>>>>>>>> I'm guessing that 'content' must refer to the precise
>>>>>> syntax of the
>>>>>>>> contained data, after decoding. So presumably it must map
>>>>>> onto some
>>>>>>>> particular spec. For isdn-uui, would that be [Q931] or
>>>>>> [Q763]? If so,
>>>>>>>> shouldn't it refer to something specific in that document?
>>>>>>> Not really.  In this case, isdn-uui effectively means
>>>> "opaque data" 
>>>>>>> which is how ISDN defines the UUI - it is completely undefined, 
>>>>>>> necessarily so  by the strict layering used in the ISDN
>>>> application.
>>>>>> I don't believe you!
>>>>>>
>>>>>> If that is the case, then the UAC and UAS must have a 
>> private side 
>>>>>> agreement about what the isdn-uui content means. Is that
>>>> really what
>>>>>> you mean?
>>>>>>
>>>>>> You already state that the first byte is a protocol 
>> discriminator. 
>>>>>> There must also be something, somewhere, that specifies
>>>> the mapping
>>>>>> from a particular value for that byte to a particular 
>>>>>> protocol/format. If there is, then that should be part of the 
>>>>>> definition of this content.
>>>>>>
>>>>>>>> 'purpose' is even less clear to me. Does it refer to why
>>>>>> it is being
>>>>>>>> included? Or what is to be done with it? If received by a
>>>>>> UA that is
>>>>>>>> not an ISDN gateway, how can it decide if this is
>>>>>> something it should
>>>>>>>> be trying to process? Is it still ISDN interworking if
>>>> neither the
>>>>>>>> UAC nor UAS is connected to an ISDN network?
>>>>>>> The purpose is the application using the UUI.  Others have
>>>>>> expressed
>>>>>>> an interest in carrying other data in the header field, in
>>>>>> which case
>>>>>>> a new purpose would be defined.  I'll see if I can think of
>>>>>> an example one.
>>>>>>
>>>>>> So are you saying that a particular call center 
>> application might 
>>>>>> constitute a distinct "purpose"?
>>>>>>
>>>>>>> Basically, if two UAs both support the header but have 
>> different 
>>>>>>> applications generating and consuming the UUI, it will not
>>>>>> work.  This
>>>>>>> is not a SIP failure, and there is nothing SIP can do about
>>>>>> it.  But
>>>>>>> this purpose header field allows the UAs to detect this 
>> condition.
>>>>>> In that case, perhaps "isdn-interwork" isn't really a
>>>> single purpose
>>>>>> at all, unless any two pieces of equipment that support
>>>> that purpose
>>>>>> can then interwork without knowing any more about each other.
>>>>>>
>>>>>>>> Suppose I was developing a sophisticated UA that
>>>>>> potentially might be
>>>>>>>> usable by a call center operator. How should
>>>>>> purpose=isdn-interwork
>>>>>>>> affect the operation of this device?
>>>>>>> To make it work in today's contact center applications,
>>>> supporting
>>>>>>> isdn-interwork would likely make it work in an application,
>>>>>> provided
>>>>>>> the appropriate call center application also ran on it.  
>>>>>> Some contact
>>>>>>> centers do not use ISDN UUI, instead they use all kinds
>>>> of really,
>>>>>>> really ugly CTI protocols running in parallel.
>>>>>> Yes, I know! :-(
>>>>>>
>>>>>>> This is a way of moving
>>>>>>> those to the ISDN model, and from there to a more
>>>>>> intelligent endpoint
>>>>>>> SIP model.
>>>>>>>
>>>>>>>>     Thanks,
>>>>>>>>     Paul
>>>>>> _______________________________________________
>>>>>> 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 gonzalo.camarillo@ericsson.com  Tue Jul 28 06:32:53 2009
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Date: Tue, 28 Jul 2009 15:32:46 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi Paul,

yes, I agree we need to add use cases where both ends are distributed to 
the draft so that we understand this type of scenario in a better way.

Cheers,

Gonzalo


Paul Kyzivat wrote:
> 
> 
> Gonzalo Camarillo wrote:
>> Hi Paul,
>>
>> yes, what you say is totally correct when one of the endpoints is not 
>> distributed. That case should be relatively-easy to resolve. The case 
>> where both ends are distributed is, of course, more "interesting" ;o)
> 
> I don't understand at all how the "distributed" model works when both 
> ends want to be distributed.
> 
>     Thanks,
>     Paul
> 
>> Cheers,
>>
>> Gonzalo
>>
>> Paul Kyzivat wrote:
>>>
>>>
>>> Gonzalo Camarillo wrote:
>>>> Hi,
>>>>
>>>>> I guess the one way I could buy into your "distributed" approach is 
>>>>> to simply model it as a conference, where the focus is at one 
>>>>> "end". But I think we discussed that once.
>>>>
>>>> that is how it is modeled when only one of the ends is distributed. 
>>>> What we discussed was that the semantics of join were intended for 
>>>> conferences and that even though this was very similar, it was not 
>>>> the same thing... but the model is valid.
>>>
>>> If the model is valid, then I think the mechanisms that go with it 
>>> should be valid too.
>>>
>>> The conferencing mechanism could be used for either the "distributed" 
>>> or "centralized" approach. Then there needs to be a focus somewhere, 
>>> and the difference between the approaches is where it resides - at 
>>> your own end, or at the "other" end.
>>>
>>>     Thanks,
>>>     Paul
>>
>>


From HKaplan@acmepacket.com  Tue Jul 28 08:41:52 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Tue, 28 Jul 2009 11:41:49 -0400
Thread-Topic: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Monday, July 27, 2009 6:25 PM
>=20
> Hadriel Kaplan wrote:
> >
> >> -----Original Message----- From: Dean Willis
> >> [mailto:dean.willis@softarmor.com] Sent: Monday, July 27, 2009 4:49
> >> AM To: Hadriel Kaplan
> >>
> >> I disagree. SIP is based on HTTP The Request URI encodes the
> >> request being made by the UAC to the UAS.
> >
> > Then you might as well argue the request-URI should encode MIME
> > bodies.  The UUI is not a part of the resource being addressed in any
> > way, and thus not a part of a Uniform Resource Identifier.
>=20
> I take it you've never written an HTTP application using something like
> the Common Gateway Interface (CGI) for parameter passing. Perhaps taking
> some time to learn the history here would be helpful for you, so that
> you stop making patently incorrect assertions about what can be in a
> request URI.

Au contraire, mon ami.
I was not talking about the history of HTTP, nor do I care. (and to be fair=
, the "history" of HTTP goes back before the term URI was ever invented, if=
 I recall).

I am talking about the request URI in SIP.  And I'm not talking about what =
can/cannot be there - you can put whatever you want in it - I'm talking abo=
ut what we should semantically expect/prescribe to be there from a standard=
s usage perspective.

=20
> Let's look at the request-URI used by the IETF tools team's draft search
> page.
>=20
> Take, for example: http://tools.ietf.org/id/?doc=3Dcommon
> [snip]
> Note that, in your parelance, "id" and "common" are NOT part of the
> resource being addressed in any way.=20

I disagree.  They are absolutely part of the resource being addressed.  The=
y are not a filename/directory/route identifier, but that doesn't matter.


> Yet, they are encoded into the
> resource, in such a way that they can easily be presented in writing,
> pasted onto a business card, or whatever else you might want to do with
> a URI.

And that's exactly it!  I cannot go around adding "/id/?doc=3Dcommon" to th=
e end of every HTTP URI/URL and expect anything to work.  Why?  Because it =
only has meaning to tools.ietf.org, the *target* of the HTTP request.  UUI =
is different.  I can go add the same UUI value to every request I send rega=
rdless of the target, because it's a property of, or based on, the *sender*=
. (well, mostly... it's proprietary of course, so who the heck knows what's=
 in there)

And while *anyone* can go add "/id/?doc=3Dcommon" to the tools.ietf.org URI=
, if it were on a business card, for example - the same is not true of a UU=
I's value.  You could not put the UUI's hex value as a param into a busines=
s card and have anyone/everyone use it to you and expect it to work right. =
(again, generally speaking, ignoring the fact it's a proprietary blob)=20

-hadriel=20

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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Hadriel Kaplan wrote:
> 

> And that's exactly it!  I cannot go around adding "/id/?doc=common"
> to the end of every HTTP URI/URL and expect anything to work.  Why?
> Because it only has meaning to tools.ietf.org, the *target* of the
> HTTP request.  UUI is different.  I can go add the same UUI value to
> every request I send regardless of the target, because it's a
> property of, or based on, the *sender*. (well, mostly... it's
> proprietary of course, so who the heck knows what's in there)


Horsefeathers. It has nothing to do with the sender (UAC). Understanding
the data is all up to the receiver (UAS). If UUI data were printed on a
business card and entered on my Nokia phone (which I assure you, doesn't
understand UUI)  it would still be delivered to the receiver. If the
receiver understands it, it can act on it with complete independence
from any property of the sender.

> And while *anyone* can go add "/id/?doc=common" to the tools.ietf.org
> URI, if it were on a business card, for example - the same is not
> true of a UUI's value.  You could not put the UUI's hex value as a
> param into a business card and have anyone/everyone use it to you and
> expect it to work right. (again, generally speaking, ignoring the
> fact it's a proprietary blob)


That sort of depends on the UUI application, doesn't it?

--
Dean

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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Hadriel,

I don't understand what you are trying to say at all.

I *think* your are saying that UUI is like my scribbling any random 
garbage, in any notation I want, onto my business card and handing it to 
somebody, in expectation that they will be able to do something useful 
with it.

I indeed *might* be able to do that, if I know precisely who I am 
handing the card to, and what sorts of things they understand, and the 
notations they will be able to process. But of course in this case I 
have to put this info into the INVITE before I know who I will end up 
talking to.

	Thanks,
	Paul

Dean Willis wrote:
> Hadriel Kaplan wrote:
> 
>> And that's exactly it!  I cannot go around adding "/id/?doc=common"
>> to the end of every HTTP URI/URL and expect anything to work.  Why?
>> Because it only has meaning to tools.ietf.org, the *target* of the
>> HTTP request.  UUI is different.  I can go add the same UUI value to
>> every request I send regardless of the target, because it's a
>> property of, or based on, the *sender*. (well, mostly... it's
>> proprietary of course, so who the heck knows what's in there)
> 
> 
> Horsefeathers. It has nothing to do with the sender (UAC). Understanding
> the data is all up to the receiver (UAS). If UUI data were printed on a
> business card and entered on my Nokia phone (which I assure you, doesn't
> understand UUI)  it would still be delivered to the receiver. If the
> receiver understands it, it can act on it with complete independence
> from any property of the sender.
> 
>> And while *anyone* can go add "/id/?doc=common" to the tools.ietf.org
>> URI, if it were on a business card, for example - the same is not
>> true of a UUI's value.  You could not put the UUI's hex value as a
>> param into a business card and have anyone/everyone use it to you and
>> expect it to work right. (again, generally speaking, ignoring the
>> fact it's a proprietary blob)
> 
> 
> That sort of depends on the UUI application, doesn't it?
> 
> --
> Dean
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From HKaplan@acmepacket.com  Tue Jul 28 22:10:34 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Dean Willis <dean.willis@softarmor.com>
Date: Wed, 29 Jul 2009 01:10:29 -0400
Thread-Topic: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, July 28, 2009 9:14 PM
> To: Dean Willis
>=20
> Hadriel,
> I don't understand what you are trying to say at all.
> I *think* your are saying that UUI is like my scribbling any random
> garbage, in any notation I want, onto my business card and handing it to
> somebody, in expectation that they will be able to do something useful
> with it.

Nope, I said *if* we were to put it into the request-URI, then that's what =
it would be like.  I don't want to put it into the request-URI, because I t=
hink the req-URI is the wrong semantic.  Dean's saying we should put it in =
the req-uri, because HTTP would.=20

=20
> I indeed *might* be able to do that, if I know precisely who I am
> handing the card to, and what sorts of things they understand, and the
> notations they will be able to process. But of course in this case I
> have to put this info into the INVITE before I know who I will end up
> talking to.

Yup.


> Dean Willis wrote:
> > Hadriel Kaplan wrote:
> >
> >> And that's exactly it!  I cannot go around adding "/id/?doc=3Dcommon"
> >> to the end of every HTTP URI/URL and expect anything to work.  Why?
> >> Because it only has meaning to tools.ietf.org, the *target* of the
> >> HTTP request.  UUI is different.  I can go add the same UUI value to
> >> every request I send regardless of the target, because it's a
> >> property of, or based on, the *sender*. (well, mostly... it's
> >> proprietary of course, so who the heck knows what's in there)
> >
> >
> > Horsefeathers. It has nothing to do with the sender (UAC). Understandin=
g
> > the data is all up to the receiver (UAS). If UUI data were printed on a
> > business card and entered on my Nokia phone (which I assure you, doesn'=
t
> > understand UUI)  it would still be delivered to the receiver. If the
> > receiver understands it, it can act on it with complete independence
> > from any property of the sender.
> >
> >> And while *anyone* can go add "/id/?doc=3Dcommon" to the tools.ietf.or=
g
> >> URI, if it were on a business card, for example - the same is not
> >> true of a UUI's value.  You could not put the UUI's hex value as a
> >> param into a business card and have anyone/everyone use it to you and
> >> expect it to work right. (again, generally speaking, ignoring the
> >> fact it's a proprietary blob)
> >
> >
> > That sort of depends on the UUI application, doesn't it?
> >
> > --
> > Dean
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >

From HKaplan@acmepacket.com  Tue Jul 28 22:34:58 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Wed, 29 Jul 2009 01:34:57 -0400
Thread-Topic: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Tuesday, July 28, 2009 7:49 PM
> To: Hadriel Kaplan
>=20
> Horsefeathers. It has nothing to do with the sender (UAC). Understanding
> the data is all up to the receiver (UAS). If UUI data were printed on a
> business card and entered on my Nokia phone (which I assure you, doesn't
> understand UUI)  it would still be delivered to the receiver. If the
> receiver understands it, it can act on it with complete independence
> from any property of the sender.

Poppycock.  What's *in* the UUI can change based on who's sending it, even =
if it's accessing the same target resource.  The fact that the receiving UA=
S' application above SIP has to understand it to use it, doesn't mean a hec=
k of a lot.  There's are plenty of existing SIP headers that I could enter =
into your Nokia phone (were it to let me) that its SIP stack needs to know =
nothing about to send the INVITE, that only the receiver has to understand.=
  Although I guess your point is that we were wrong to do that before, so w=
e should stop doing it. (right?)


> > And while *anyone* can go add "/id/?doc=3Dcommon" to the tools.ietf.org
> > URI, if it were on a business card, for example - the same is not
> > true of a UUI's value.  You could not put the UUI's hex value as a
> > param into a business card and have anyone/everyone use it to you and
> > expect it to work right. (again, generally speaking, ignoring the
> > fact it's a proprietary blob)
>=20
>=20
> That sort of depends on the UUI application, doesn't it?

Unfortunately yes. It's an opaque blob. It can have anything in it. =20

-hadriel

From dean.willis@softarmor.com  Wed Jul 29 06:05:13 2009
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Subject: Re: [dispatch] CCUS and History-Info (was Proposed Charter for CCUS - Call Control UUI for SIP)
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Hadriel Kaplan wrote:
> 
> Poppycock.  What's *in* the UUI can change based on who's sending it,
> even if it's accessing the same target resource.  The fact that the
> receiving UAS' application above SIP has to understand it to use it,
> doesn't mean a heck of a lot.  There's are plenty of existing SIP
> headers that I could enter into your Nokia phone (were it to let me)
> that its SIP stack needs to know nothing about to send the INVITE,
> that only the receiver has to understand.  Although I guess your
> point is that we were wrong to do that before, so we should stop
> doing it. (right?)
> 

Right. We totally molested the poodle by putting so much
application-specific cruft into SIP extension header fields. HTTP has
been able to support new applications without having to make a protocol
extension for every new application because it does NOT put application
data into HTTP extensions. Rather, it uses HTTP as a transport for the
application data, which is what SIP should have done and what we really
meant to do back when starting SIP.

HTTP has two places to transport application-specific stuff: in the
request URI, and in payloads. Application parameters are generally sent
using the request URI.

We have two alternatives to choose from:

1) Fix SIP so that we can use HTTP-style request URI parameterization.
Currently, SIPCORE has consensus to do exactly this, using history-info.

2) Give up on using HTTP-style request-URI parameterization and develop
an alternative. Adding application-specific SIP extensions for every
different application is not a reasonable alternative. Dedicating a
specific new extension field for all user-to-user parameters might be,
and that's essentially what Alan has proposed.

Pick one and use it.

--
Dean


> 
>>> And while *anyone* can go add "/id/?doc=common" to the
>>> tools.ietf.org URI, if it were on a business card, for example -
>>> the same is not true of a UUI's value.  You could not put the
>>> UUI's hex value as a param into a business card and have
>>> anyone/everyone use it to you and expect it to work right.
>>> (again, generally speaking, ignoring the fact it's a proprietary
>>> blob)
>> 
>> That sort of depends on the UUI application, doesn't it?
> 
> Unfortunately yes. It's an opaque blob. It can have anything in it.
> 
> 
> -hadriel
> 


From spencer@wonderhamster.org  Wed Jul 29 22:17:50 2009
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Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300 overLunch on Friday
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The SIP-CLF DISPATCH ad hoc on Friday is intended to form a working group,
and we'll be noodling over charter text that Robert has produced (in
consultation with others) during the meeting.

Just to make sure we're all on the same page, here's the charter text we'll
be discussing - please look it over BEFORE the meeting (and if you want to 
talk about it on the sip-clf@ietf.org mailing list before the meeting, 
that's fine, too):

The SIP Common Log Format (CLF) working group is chartered to define a
standard logging format for systems processing SIP messages.

Well-known web servers such as Apache and web proxies like Squid support
event logging using a common log format. The logs produced using these
de-facto standard formats are invaluable to system administrators for
trouble-shooting a server and tool writers to craft tools that mine the log
files to produce reports and trends and to search for a certain SIP message
or messages, a transaction or a related set of transactions. Furthermore,
these log records can also be used to train anomaly detection systems and
feed events into a security event management system.

The Session Initiation Protocol does not have a common log format. Diverse
element provide distinct log formats making it complex to produce tools to
analyze them.

The CLF working group will produce a format suitable for logging from any
SIP element. The format will anticipate the need to search, merge, and
summarize the log records from diverse elements. The format will anticipate
the need to correlate messages from multiple elements related to a given
request (that may fork) or a given dialog. The format will take SIP's
extensibility into consideration, providing a way to represent SIP message
components that are defined in the future. The format will anticipate being
used both for off-line analysis and on-line real-time processing
applications. The working group will consider the need for efficient
creation of records and the need for efficient processing of the records.

The working group will identify the fields to appear in a log record and
provide one or more formats for encoding those fields. The working group is
not pre-constrained to producing either a bit-field oriented or
text-oriented format, and may choose to provide both. If the group chooses
to specify both, it must be possible to mechanically translate between the
formats without loss of information.

Specifying the mechanics of exchanging, transporting, and storing SIP
Common Log Format records is explicitly out of scope. Specifying a real-time
transfer mechanism for heuristic analysis is explicitly out of scope.

The group will generate:

A problem statement enunciating the motivation, and use cases for a SIP
Common Log Format. This analysis will identify the required minimal
information that must appear in any record.

A specification of the SIP Common Log Format record

The group will consider providing one or more reference implementations for
decoding a CLF record.

One more piece of logistics ...

We'll need at least one, and preferably two, scribes and a jabber scribe for
our CLF session on Friday during lunch.

We'll call for volunteers at the meeting if necessary - we can't have the
meeting without producing minutes - but it would be great if you could
consider volunteering NOW.

We have about an hour to get through some pretty important discussions, and
every minute we DON'T spend gazing meaningfully at the attendees (us) and
looking down at computers trying not to make eye contact with the chairs
(you) is a minute that we can use more productively!

Thanks,

Spencer and Theo


From tom.taylor@rogers.com  Thu Jul 30 00:03:22 2009
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Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300 overLunch on Friday
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Small comments below.

Spencer Dawkins wrote:
> The SIP-CLF DISPATCH ad hoc on Friday is intended to form a working group,
> and we'll be noodling over charter text that Robert has produced (in
> consultation with others) during the meeting.
> 
> Just to make sure we're all on the same page, here's the charter text we'll
> be discussing - please look it over BEFORE the meeting (and if you want 
> to talk about it on the sip-clf@ietf.org mailing list before the 
> meeting, that's fine, too):
> 
> The SIP Common Log Format (CLF) working group is chartered to define a
> standard logging format for systems processing SIP messages.
> 
> Well-known web servers such as Apache and web proxies like Squid support
> event logging using a common log format. The logs produced using these
> de-facto standard formats are invaluable to system administrators for
> trouble-shooting a server and tool writers to craft tools that mine the log
> files to produce reports and trends and to search for a certain SIP message
> or messages, a transaction or a related set of transactions. Furthermore,
> these log records can also be used to train anomaly detection systems and
> feed events into a security event management system.

[PTT] Since this is a description of existing usage, I would drop "SIP" four 
lines up.
> 
> The Session Initiation Protocol does not have a common log format. Diverse
> element provide distinct log formats making it complex to produce tools to
> analyze them.
[PTT] element -> elements
> 
...

From vkg@alcatel-lucent.com  Thu Jul 30 00:49:32 2009
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Tom Taylor wrote:
> Small comments below.
[...]
> [PTT] Since this is a description of existing usage, I would drop "SIP" 
> four lines up.

Yes, I agree.  I was about to make the same comment as well.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
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From: "Jain, Rajnish" <Rajnish.Jain@ipc.com>
To: "Doken, Serhad" <sdoken@qualcomm.com>, Vijay Gurbani <vkg@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 30 Jul 2009 06:58:24 -0400
Thread-Topic: [dispatch] Session Recording in SIP
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Subject: Re: [dispatch] Session Recording in SIP
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Hi Serhad,

Comments inline:

> > Naming these things is always a bit tricky. If we called the RS simply
> > the Recorder given your reasoning, then the RC should also be named
> > differently (because the RC can act as the "server side" of the SRP
> > signaling). The RS naming is a bit akin to the term "Media Server". If
> > there is enough confusion on RS then, we can consider renaming it.
>
> Indeed, I think RC could be renamed as well since it is not a pure client
> and can be a server at times. Using, Recording Client and Recording Serve=
r
> at section 1 before their definitions are given in Section 2 adds to the
> confusion. I would offer something like Recorder and (Recorded Media) For=
ker
> which is self explanatory.

The RC/RS naming is intended to be _logical_ as opposed to the client serve=
r roles in an INVITE transaction. I'll try to gather design team's input on=
 this.

> > What is the number of persistent recording
> > > sessions per RC/RS ?
> >
> > That's an implementation-dependent choice.
>
> Obviously, however as I referred to in my subsequent question below, this
> may have implications scaling up. If I have a device with high number of
> simultaneous calls capability, media forker almost needs to keep that man=
y
> persistent recording sessions and refresh them all the time by session
> timers. A bunch of race conditions may occur and it may be a drain of
> resources on the device that is doing the media forking especially when
> there is no actual session. Moreover, there will be certain assumptions m=
ade
> for such persistent recording sessions such as the codec used for the med=
ia
> session etc.

These appear to be implementation issues. The SRP mechanism must allow for =
scalable implementations, but I don't think we can state that RC and RS mus=
t support 'n' number of session.


> > > If recorded session media is stopped/temporarily interrupted due to
> > problems
> > > or features invoked on it(recording session is put on hold or started
> > a
> > > conferencing or transfer operation), how does that affect the already
> > setup
> > > recording session(s) ?
> >
> > I think that'll largely depend on the RC/RS implementations and the
> > nature of the SRP sessions. For instance, if the recording sessions are
> > persistent and the RC puts recorded session media on hold, then it RC
> > MAY reINVITE the RS and put the recording session media on hold or it
> > MAY just temporarily stop sending such media (silence suppression) to
> > the RS.
>
> In that case, the recorded session and the persistent recording sessions
> need to be synchronized back during resume which may bring the media
> clipping problem that persistent recording attempted to solve in the firs=
t
> case.

Persistent recording sessions will likely not use re-INVITEs for recording =
session on recorded session hold/resume. They'll simply pause/resume record=
ing session RTP. The metadata events will slightly lag behind this RTP acti=
vity, but there will be timestamps in those events to achieve synchronizati=
on.

Thanks,
Raj



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From: "Jain, Rajnish" <Rajnish.Jain@ipc.com>
To: "Doken, Serhad" <sdoken@qualcomm.com>, Leon Portman <Leon.Portman@nice.com>, Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
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Thread-Topic: [dispatch]	Comments	on draft-jain-dispatch-session-recording-protocol-req-00
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Subject: Re: [dispatch] Comments	on	draft-jain-dispatch-session-recording-protocol-req-00
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> > I'm not sure that the Persistent and Dynamic Recording as defined above
> > correspond to the Always On and On Demand modes defined in
> > http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#section-4.
> >
> > [LeonP] It is different in the way that the Recording Session is
> > established only once during initialization (or login events) and then
> > only media is forwarded per each call in order to minimize the
> > clipping.
>
> I see the "Always On" Recording as a separate third mode of Recording whe=
re
> for a particular one or set of devices(possibly set via provisioning), al=
l
> calls are recorded from start to end(by setting up recording sessions) an=
d
> no persistent recording sessions are kept for them.

I guess this is semantics, but wouldn't that still be "On Demand"? In your =
scenario, it just so happens that the RC/RS have been configured to record =
all calls for a given end-point, but the recording sessions are still estab=
lished on a per call basis.

Thanks,
Raj



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To: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, "Doken, Serhad" <sdoken@qualcomm.com>, "Leon Portman" <Leon.Portman@nice.com>, "Alan Johnston" <alan@sipstation.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Comments	on	draft-jain-dispatch-session-recording-protocol-req-00
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Pl. see inline ...=20

>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of Jain, Rajnish
>Sent: Thursday, July 30, 2009 4:47 PM
>To: Doken, Serhad; Leon Portman; Alan Johnston; dispatch@ietf.org
>Subject: Re: [dispatch]Comments on=20
>draft-jain-dispatch-session-recording-protocol-req-00
>
>>
>> I see the "Always On" Recording as a separate third mode of=20
>Recording=20
>> where for a particular one or set of devices(possibly set via=20
>> provisioning), all calls are recorded from start to end(by=20
>setting up=20
>> recording sessions) and no persistent recording sessions are=20
>kept for them.
>
>I guess this is semantics, but wouldn't that still be "On=20
>Demand"? In your scenario, it just so happens that the RC/RS=20
>have been configured to record all calls for a given=20
>end-point, but the recording sessions are still established on=20
>a per call basis.

This is similar to a recording anchor where during recording either RC
or RS is designated as the recording anchor. For example in a call
center, the external caller is the recording anchor and recording
continues even when call is transferred to another agent, call is put on
hold or another agent is conferenced, the recording from the caller
device does not stop. The recording anchor also provides the metadata to
the RS to keep track of different service interactions.

Thanks
>
>Thanks,
>Raj

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From: "Jain, Rajnish" <Rajnish.Jain@ipc.com>
To: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>, "Doken, Serhad" <sdoken@qualcomm.com>, Leon Portman <Leon.Portman@nice.com>, Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 30 Jul 2009 11:00:06 -0400
Thread-Topic: [dispatch]Comments	on draft-jain-dispatch-session-recording-protocol-req-00
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Subject: Re: [dispatch] Comments	on	draft-jain-dispatch-session-recording-protocol-req-00
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Hi Sanjay,

Pls. see inline:

> -----Original Message-----
> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]
> Sent: Thursday, July 30, 2009 10:36 AM

> >I guess this is semantics, but wouldn't that still be "On
> >Demand"? In your scenario, it just so happens that the RC/RS
> >have been configured to record all calls for a given
> >end-point, but the recording sessions are still established on
> >a per call basis.
>
> This is similar to a recording anchor where during recording either RC
> or RS is designated as the recording anchor. For example in a call
> center, the external caller is the recording anchor and recording
> continues even when call is transferred to another agent, call is put on
> hold or another agent is conferenced, the recording from the caller
> device does not stop. The recording anchor also provides the metadata to
> the RS to keep track of different service interactions.

That's a good use case. One way to look at this is that the RC is anchored =
on the end-point that represents the external caller. This is also known as=
 line-side recording. Another way to look at this is that the RCs are ancho=
red on the agent UAs and the metadata is used to correlate and construct th=
e overall call. This is also known as station-side recording.

I'd imagine that this use case (regardless of where the RC is anchored) cou=
ld still be addressed through either On-Demand or Always-on recording modes=
. I don't see this as a separate recording mode, unless I'm missing somethi=
ng.

Thanks,
Raj


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To: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Vijay Gurbani <vkg@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>
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Hi Rajnish,

> -----Original Message-----
> From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]
> Sent: Thursday, July 30, 2009 3:58 AM
> To: Doken, Serhad; Vijay Gurbani; dispatch@ietf.org
> Subject: RE: [dispatch] Session Recording in SIP
>=20
> Hi Serhad,
>=20
> Comments inline:
>=20
> > > Naming these things is always a bit tricky. If we called the RS
> simply
> > > the Recorder given your reasoning, then the RC should also be named
> > > differently (because the RC can act as the "server side" of the SRP
> > > signaling). The RS naming is a bit akin to the term "Media Server".
> If
> > > there is enough confusion on RS then, we can consider renaming it.
> >
> > Indeed, I think RC could be renamed as well since it is not a pure
> client
> > and can be a server at times. Using, Recording Client and Recording
> Server
> > at section 1 before their definitions are given in Section 2 adds to
> the
> > confusion. I would offer something like Recorder and (Recorded Media)
> Forker
> > which is self explanatory.
>=20
> The RC/RS naming is intended to be _logical_ as opposed to the client
> server roles in an INVITE transaction. I'll try to gather design team's
> input on this.
>=20
> > > What is the number of persistent recording
> > > > sessions per RC/RS ?
> > >
> > > That's an implementation-dependent choice.
> >
> > Obviously, however as I referred to in my subsequent question below,
> this
> > may have implications scaling up. If I have a device with high number
> of
> > simultaneous calls capability, media forker almost needs to keep that
> many
> > persistent recording sessions and refresh them all the time by
> session
> > timers. A bunch of race conditions may occur and it may be a drain of
> > resources on the device that is doing the media forking especially
> when
> > there is no actual session. Moreover, there will be certain
> assumptions made
> > for such persistent recording sessions such as the codec used for the
> media
> > session etc.
>=20
> These appear to be implementation issues. The SRP mechanism must allow
> for scalable implementations, but I don't think we can state that RC
> and RS must support 'n' number of session.
>=20

It could be more than implementation issues especially for the enviroments =
where the media forker (RC) is limited in its capabilities such as memory, =
power, bandwidth. For such cases it is preferable that the Recorder(RS) set=
s up the persistent recording sessions.

>=20
> > > > If recorded session media is stopped/temporarily interrupted due
> to
> > > problems
> > > > or features invoked on it(recording session is put on hold or
> started
> > > a
> > > > conferencing or transfer operation), how does that affect the
> already
> > > setup
> > > > recording session(s) ?
> > >
> > > I think that'll largely depend on the RC/RS implementations and the
> > > nature of the SRP sessions. For instance, if the recording sessions
> are
> > > persistent and the RC puts recorded session media on hold, then it
> RC
> > > MAY reINVITE the RS and put the recording session media on hold or
> it
> > > MAY just temporarily stop sending such media (silence suppression)
> to
> > > the RS.
> >
> > In that case, the recorded session and the persistent recording
> sessions
> > need to be synchronized back during resume which may bring the media
> > clipping problem that persistent recording attempted to solve in the
> first
> > case.
>=20
> Persistent recording sessions will likely not use re-INVITEs for
> recording session on recorded session hold/resume. They'll simply
> pause/resume recording session RTP. The metadata events will slightly
> lag behind this RTP activity, but there will be timestamps in those
> events to achieve synchronization.

That is interesting. Since we are requirement phase, I will hold off my tho=
ughts till the solution draft.

Thanks,
Serhad Doken

>=20
> Thanks,
> Raj
>=20
>=20
>=20
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From: "Doken, Serhad" <sdoken@qualcomm.com>
To: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Leon Portman <Leon.Portman@nice.com>, Alan Johnston <alan@sipstation.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Fri, 31 Jul 2009 00:54:30 -0700
Thread-Topic: [dispatch]	Comments	on draft-jain-dispatch-session-recording-protocol-req-00
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Subject: Re: [dispatch] Comments	on	draft-jain-dispatch-session-recording-protocol-req-00
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> -----Original Message-----
> From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]
> Sent: Thursday, July 30, 2009 4:17 AM
> To: Doken, Serhad; Leon Portman; Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Comments on draft-jain-dispatch-session-
> recording-protocol-req-00
>=20
> > > I'm not sure that the Persistent and Dynamic Recording as defined
> above
> > > correspond to the Always On and On Demand modes defined in
> > > http://tools.ietf.org/html/draft-wing-sipping-srtp-key-04#section-
> 4.
> > >
> > > [LeonP] It is different in the way that the Recording Session is
> > > established only once during initialization (or login events) and
> then
> > > only media is forwarded per each call in order to minimize the
> > > clipping.
> >
> > I see the "Always On" Recording as a separate third mode of Recording
> where
> > for a particular one or set of devices(possibly set via
> provisioning), all
> > calls are recorded from start to end(by setting up recording
> sessions) and
> > no persistent recording sessions are kept for them.
>=20
> I guess this is semantics, but wouldn't that still be "On Demand"? In
> your scenario, it just so happens that the RC/RS have been configured
> to record all calls for a given end-point, but the recording sessions
> are still established on a per call basis.

The nuance I was thinking in my scenario is RC or RS avoids the need notify=
 the other via signaling to kick off the recording session which I consider=
 this notification process as on demand. So, yes in the end, I agree it is =
mostly semantics.

>=20
> Thanks,
> Raj
>=20
>=20
>=20
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From spencer@wonderhamster.org  Fri Jul 31 01:56:35 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "dispatch mailing list" <dispatch@ietf.org>
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Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300 overLunchon Friday
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Just to let people know - Mary has graciously posted the SIP-CLF slides to 
the meeting materials page (since this is a DISPATCH ad hoc, they're under 
DISPATCH).

URLs are

http://www.ietf.org/proceedings/75/slides/dispatch-6.ppt
http://www.ietf.org/proceedings/75/slides/dispatch-5.pdf

See you in about an hour...

Spencer


> The SIP-CLF DISPATCH ad hoc on Friday is intended to form a working group,
> and we'll be noodling over charter text that Robert has produced (in
> consultation with others) during the meeting.
>
> Just to make sure we're all on the same page, here's the charter text 
> we'll
> be discussing - please look it over BEFORE the meeting (and if you want to 
> talk about it on the sip-clf@ietf.org mailing list before the meeting, 
> that's fine, too):
>
> The SIP Common Log Format (CLF) working group is chartered to define a
> standard logging format for systems processing SIP messages.
>
> Well-known web servers such as Apache and web proxies like Squid support
> event logging using a common log format. The logs produced using these
> de-facto standard formats are invaluable to system administrators for
> trouble-shooting a server and tool writers to craft tools that mine the 
> log
> files to produce reports and trends and to search for a certain SIP 
> message
> or messages, a transaction or a related set of transactions. Furthermore,
> these log records can also be used to train anomaly detection systems and
> feed events into a security event management system.
>
> The Session Initiation Protocol does not have a common log format. Diverse
> element provide distinct log formats making it complex to produce tools to
> analyze them.
>
> The CLF working group will produce a format suitable for logging from any
> SIP element. The format will anticipate the need to search, merge, and
> summarize the log records from diverse elements. The format will 
> anticipate
> the need to correlate messages from multiple elements related to a given
> request (that may fork) or a given dialog. The format will take SIP's
> extensibility into consideration, providing a way to represent SIP message
> components that are defined in the future. The format will anticipate 
> being
> used both for off-line analysis and on-line real-time processing
> applications. The working group will consider the need for efficient
> creation of records and the need for efficient processing of the records.
>
> The working group will identify the fields to appear in a log record and
> provide one or more formats for encoding those fields. The working group 
> is
> not pre-constrained to producing either a bit-field oriented or
> text-oriented format, and may choose to provide both. If the group chooses
> to specify both, it must be possible to mechanically translate between the
> formats without loss of information.
>
> Specifying the mechanics of exchanging, transporting, and storing SIP
> Common Log Format records is explicitly out of scope. Specifying a 
> real-time
> transfer mechanism for heuristic analysis is explicitly out of scope.
>
> The group will generate:
>
> A problem statement enunciating the motivation, and use cases for a SIP
> Common Log Format. This analysis will identify the required minimal
> information that must appear in any record.
>
> A specification of the SIP Common Log Format record
>
> The group will consider providing one or more reference implementations 
> for
> decoding a CLF record.
>
> One more piece of logistics ...
>
> We'll need at least one, and preferably two, scribes and a jabber scribe 
> for
> our CLF session on Friday during lunch.
>
> We'll call for volunteers at the meeting if necessary - we can't have the
> meeting without producing minutes - but it would be great if you could
> consider volunteering NOW.
>
> We have about an hour to get through some pretty important discussions, 
> and
> every minute we DON'T spend gazing meaningfully at the attendees (us) and
> looking down at computers trying not to make eye contact with the chairs
> (you) is a minute that we can use more productively!
>
> Thanks,
>
> Spencer and Theo
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch 


From rjsparks@nostrum.com  Fri Jul 31 03:07:19 2009
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Subject: [dispatch] Next revision for the proposed CLF charter
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The SIP Common Log Format (CLF) working group is chartered to define
a standard logging format for systems processing SIP messages.

Well-known web servers such as Apache and web proxies like Squid
support event logging using a common log format.  The logs produced
using these de-facto standard formats are invaluable to system
administrators for trouble-shooting a server and tool writers to
craft tools that mine the log files to produce reports and trends
and to search for a certain message or messages, a transaction
or a related set of transactions.  Furthermore, these log records
can also be used to train anomaly detection systems and feed events
into a security event management system.

The Session Initiation Protocol does not have a common log
format. Diverse elements provide distinct log formats making
it complex to produce tools to analyze them.

The CLF working group will produce a format suitable for logging
from any SIP element. The format will anticipate the need to
search, merge, and summarize the log records from diverse elements.
The format will anticipate the need to correlate messages from
multiple elements related to a given request (that may fork) or a
given dialog. The format will take SIP's extensibility into
consideration, providing a way to represent SIP message components
that are defined in the future.  The format will anticipate being
used both for off-line analysis and on-line real-time processing
applications. The working group will consider the need for
efficient creation of records and the need for efficient processing
of the records.

The working group will identify the fields to appear in a log
record and provide one or more formats for encoding those fields.
The working group is not pre-constrained to producing either a
bit-field oriented or text-oriented format, and may choose to
provide both. If the group chooses to specify both, it must be
possible to mechanically translate between the formats without loss
of information.

Specifying the mechanics of exchanging, transporting, and storing
SIP Common Log Format records is explicitly out of scope. Specifying
a real-time transfer mechanism for heuristic analysis is explicitly
out of scope.

The group will generate:

- A problem statement enunciating the motivation,
and use cases for a SIP Common Log Format. This analysis
will identify the required minimal information that must
appear in any record.

- A specification of the SIP Common Log Format record

The group will consider providing one or more reference
implementations for decoding a CLF record.

Goals and Milestones
===========================

Oct 09 - Problem statement, motivation, and use cases
          WGLC
Nov 09 - Problem statement, motivation, and use cases
          to IESG (Informational)
Jan 10 - SIP Common Log Format specification
          WGLC
Feb 10 - SIP Common Log Format specification
          to IESG (PS)


From Jan.Seedorf@nw.neclab.eu  Fri Jul 31 04:04:44 2009
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Thread-Topic: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300 overLunchon Friday
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References: <8889A362-7C1C-4712-94CC-3A0C9B6B6F79@nostrum.com><D2BB23476B464D7199467669882A6757@china.huawei.com> <2D74BB779EAF4B5CBCF19030A9BA3E70@china.huawei.com>
From: "Jan Seedorf" <Jan.Seedorf@nw.neclab.eu>
To: "Spencer Dawkins" <spencer@wonderhamster.org>, "dispatch mailing list" <dispatch@ietf.org>
Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300 overLunchon Friday
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Dear Spencer,

Are the slides from the bof available? And who was the second chair of =
the bof?

 - Jan

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Spencer Dawkins
> Sent: Donnerstag, 30. Juli 2009 00:14
> To: dispatch mailing list
> Cc: clf@ietf.org
> Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300
> overLunchon Friday
>=20
> The SIP-CLF DISPATCH ad hoc on Friday is intended to form a working
> group,
> and we'll be noodling over charter text that Robert has produced (in
> consultation with others) during the meeting.
>=20
> Just to make sure we're all on the same page, here's the charter text
> we'll
> be discussing - please look it over BEFORE the meeting (and if you =
want
> to
> talk about it on the sip-clf@ietf.org mailing list before the meeting,
> that's fine, too):
>=20
> The SIP Common Log Format (CLF) working group is chartered to define a
> standard logging format for systems processing SIP messages.
>=20
> Well-known web servers such as Apache and web proxies like Squid
> support
> event logging using a common log format. The logs produced using these
> de-facto standard formats are invaluable to system administrators for
> trouble-shooting a server and tool writers to craft tools that mine =
the
> log
> files to produce reports and trends and to search for a certain SIP
> message
> or messages, a transaction or a related set of transactions.
> Furthermore,
> these log records can also be used to train anomaly detection systems
> and
> feed events into a security event management system.
>=20
> The Session Initiation Protocol does not have a common log format.
> Diverse
> element provide distinct log formats making it complex to produce =
tools
> to
> analyze them.
>=20
> The CLF working group will produce a format suitable for logging from
> any
> SIP element. The format will anticipate the need to search, merge, and
> summarize the log records from diverse elements. The format will
> anticipate
> the need to correlate messages from multiple elements related to a
> given
> request (that may fork) or a given dialog. The format will take SIP's
> extensibility into consideration, providing a way to represent SIP
> message
> components that are defined in the future. The format will anticipate
> being
> used both for off-line analysis and on-line real-time processing
> applications. The working group will consider the need for efficient
> creation of records and the need for efficient processing of the
> records.
>=20
> The working group will identify the fields to appear in a log record
> and
> provide one or more formats for encoding those fields. The working
> group is
> not pre-constrained to producing either a bit-field oriented or
> text-oriented format, and may choose to provide both. If the group
> chooses
> to specify both, it must be possible to mechanically translate between
> the
> formats without loss of information.
>=20
> Specifying the mechanics of exchanging, transporting, and storing SIP
> Common Log Format records is explicitly out of scope. Specifying a
> real-time
> transfer mechanism for heuristic analysis is explicitly out of scope.
>=20
> The group will generate:
>=20
> A problem statement enunciating the motivation, and use cases for a =
SIP
> Common Log Format. This analysis will identify the required minimal
> information that must appear in any record.
>=20
> A specification of the SIP Common Log Format record
>=20
> The group will consider providing one or more reference =
implementations
> for
> decoding a CLF record.
>=20
> One more piece of logistics ...
>=20
> We'll need at least one, and preferably two, scribes and a jabber
> scribe for
> our CLF session on Friday during lunch.
>=20
> We'll call for volunteers at the meeting if necessary - we can't have
> the
> meeting without producing minutes - but it would be great if you could
> consider volunteering NOW.
>=20
> We have about an hour to get through some pretty important =
discussions,
> and
> every minute we DON'T spend gazing meaningfully at the attendees (us)
> and
> looking down at computers trying not to make eye contact with the
> chairs
> (you) is a minute that we can use more productively!
>=20
> Thanks,
>=20
> Spencer and Theo
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From mary.barnes@nortel.com  Fri Jul 31 04:12:37 2009
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Thread-Topic: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300overLunchon Friday
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References: <8889A362-7C1C-4712-94CC-3A0C9B6B6F79@nostrum.com><D2BB23476B464D7199467669882A6757@china.huawei.com><2D74BB779EAF4B5CBCF19030A9BA3E70@china.huawei.com> <B94940C43117794C9432FF5FF0CCB506B6D3DC@VENUS.office>
From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Jan Seedorf" <Jan.Seedorf@nw.neclab.eu>, "Spencer Dawkins" <spencer@wonderhamster.org>, "dispatch mailing list" <dispatch@ietf.org>
Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300overLunchon Friday
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The charts are in the meeting materials for IETF-75 under the DISPATCH =
working group.  Theo was the second chair.

Mary.=20


-----Original Message-----
From: dispatch-bounces@ietf.org on behalf of Jan Seedorf
Sent: Fri 7/31/2009 6:04 AM
To: Spencer Dawkins; dispatch mailing list
Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room =
300overLunchon Friday
=20
Dear Spencer,

Are the slides from the bof available? And who was the second chair of =
the bof?

 - Jan

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Spencer Dawkins
> Sent: Donnerstag, 30. Juli 2009 00:14
> To: dispatch mailing list
> Cc: clf@ietf.org
> Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room 300
> overLunchon Friday
>=20
> The SIP-CLF DISPATCH ad hoc on Friday is intended to form a working
> group,
> and we'll be noodling over charter text that Robert has produced (in
> consultation with others) during the meeting.
>=20
> Just to make sure we're all on the same page, here's the charter text
> we'll
> be discussing - please look it over BEFORE the meeting (and if you =
want
> to
> talk about it on the sip-clf@ietf.org mailing list before the meeting,
> that's fine, too):
>=20
> The SIP Common Log Format (CLF) working group is chartered to define a
> standard logging format for systems processing SIP messages.
>=20
> Well-known web servers such as Apache and web proxies like Squid
> support
> event logging using a common log format. The logs produced using these
> de-facto standard formats are invaluable to system administrators for
> trouble-shooting a server and tool writers to craft tools that mine =
the
> log
> files to produce reports and trends and to search for a certain SIP
> message
> or messages, a transaction or a related set of transactions.
> Furthermore,
> these log records can also be used to train anomaly detection systems
> and
> feed events into a security event management system.
>=20
> The Session Initiation Protocol does not have a common log format.
> Diverse
> element provide distinct log formats making it complex to produce =
tools
> to
> analyze them.
>=20
> The CLF working group will produce a format suitable for logging from
> any
> SIP element. The format will anticipate the need to search, merge, and
> summarize the log records from diverse elements. The format will
> anticipate
> the need to correlate messages from multiple elements related to a
> given
> request (that may fork) or a given dialog. The format will take SIP's
> extensibility into consideration, providing a way to represent SIP
> message
> components that are defined in the future. The format will anticipate
> being
> used both for off-line analysis and on-line real-time processing
> applications. The working group will consider the need for efficient
> creation of records and the need for efficient processing of the
> records.
>=20
> The working group will identify the fields to appear in a log record
> and
> provide one or more formats for encoding those fields. The working
> group is
> not pre-constrained to producing either a bit-field oriented or
> text-oriented format, and may choose to provide both. If the group
> chooses
> to specify both, it must be possible to mechanically translate between
> the
> formats without loss of information.
>=20
> Specifying the mechanics of exchanging, transporting, and storing SIP
> Common Log Format records is explicitly out of scope. Specifying a
> real-time
> transfer mechanism for heuristic analysis is explicitly out of scope.
>=20
> The group will generate:
>=20
> A problem statement enunciating the motivation, and use cases for a =
SIP
> Common Log Format. This analysis will identify the required minimal
> information that must appear in any record.
>=20
> A specification of the SIP Common Log Format record
>=20
> The group will consider providing one or more reference =
implementations
> for
> decoding a CLF record.
>=20
> One more piece of logistics ...
>=20
> We'll need at least one, and preferably two, scribes and a jabber
> scribe for
> our CLF session on Friday during lunch.
>=20
> We'll call for volunteers at the meeting if necessary - we can't have
> the
> meeting without producing minutes - but it would be great if you could
> consider volunteering NOW.
>=20
> We have about an hour to get through some pretty important =
discussions,
> and
> every minute we DON'T spend gazing meaningfully at the attendees (us)
> and
> looking down at computers trying not to make eye contact with the
> chairs
> (you) is a minute that we can use more productively!
>=20
> Thanks,
>=20
> Spencer and Theo
>=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


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<P><FONT SIZE=3D2>The charts are in the meeting materials for IETF-75 =
under the DISPATCH working group.&nbsp; Theo was the second chair.<BR>
<BR>
Mary.<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: dispatch-bounces@ietf.org on behalf of Jan Seedorf<BR>
Sent: Fri 7/31/2009 6:04 AM<BR>
To: Spencer Dawkins; dispatch mailing list<BR>
Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room =
300overLunchon Friday<BR>
<BR>
Dear Spencer,<BR>
<BR>
Are the slides from the bof available? And who was the second chair of =
the bof?<BR>
<BR>
&nbsp;- Jan<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: dispatch-bounces@ietf.org [<A =
HREF=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.or=
g</A>] On<BR>
&gt; Behalf Of Spencer Dawkins<BR>
&gt; Sent: Donnerstag, 30. Juli 2009 00:14<BR>
&gt; To: dispatch mailing list<BR>
&gt; Cc: clf@ietf.org<BR>
&gt; Subject: Re: [dispatch] Room for DISPATCH ad-hoc on CLF : Room =
300<BR>
&gt; overLunchon Friday<BR>
&gt;<BR>
&gt; The SIP-CLF DISPATCH ad hoc on Friday is intended to form a =
working<BR>
&gt; group,<BR>
&gt; and we'll be noodling over charter text that Robert has produced =
(in<BR>
&gt; consultation with others) during the meeting.<BR>
&gt;<BR>
&gt; Just to make sure we're all on the same page, here's the charter =
text<BR>
&gt; we'll<BR>
&gt; be discussing - please look it over BEFORE the meeting (and if you =
want<BR>
&gt; to<BR>
&gt; talk about it on the sip-clf@ietf.org mailing list before the =
meeting,<BR>
&gt; that's fine, too):<BR>
&gt;<BR>
&gt; The SIP Common Log Format (CLF) working group is chartered to =
define a<BR>
&gt; standard logging format for systems processing SIP messages.<BR>
&gt;<BR>
&gt; Well-known web servers such as Apache and web proxies like =
Squid<BR>
&gt; support<BR>
&gt; event logging using a common log format. The logs produced using =
these<BR>
&gt; de-facto standard formats are invaluable to system administrators =
for<BR>
&gt; trouble-shooting a server and tool writers to craft tools that mine =
the<BR>
&gt; log<BR>
&gt; files to produce reports and trends and to search for a certain =
SIP<BR>
&gt; message<BR>
&gt; or messages, a transaction or a related set of transactions.<BR>
&gt; Furthermore,<BR>
&gt; these log records can also be used to train anomaly detection =
systems<BR>
&gt; and<BR>
&gt; feed events into a security event management system.<BR>
&gt;<BR>
&gt; The Session Initiation Protocol does not have a common log =
format.<BR>
&gt; Diverse<BR>
&gt; element provide distinct log formats making it complex to produce =
tools<BR>
&gt; to<BR>
&gt; analyze them.<BR>
&gt;<BR>
&gt; The CLF working group will produce a format suitable for logging =
from<BR>
&gt; any<BR>
&gt; SIP element. The format will anticipate the need to search, merge, =
and<BR>
&gt; summarize the log records from diverse elements. The format =
will<BR>
&gt; anticipate<BR>
&gt; the need to correlate messages from multiple elements related to =
a<BR>
&gt; given<BR>
&gt; request (that may fork) or a given dialog. The format will take =
SIP's<BR>
&gt; extensibility into consideration, providing a way to represent =
SIP<BR>
&gt; message<BR>
&gt; components that are defined in the future. The format will =
anticipate<BR>
&gt; being<BR>
&gt; used both for off-line analysis and on-line real-time =
processing<BR>
&gt; applications. The working group will consider the need for =
efficient<BR>
&gt; creation of records and the need for efficient processing of =
the<BR>
&gt; records.<BR>
&gt;<BR>
&gt; The working group will identify the fields to appear in a log =
record<BR>
&gt; and<BR>
&gt; provide one or more formats for encoding those fields. The =
working<BR>
&gt; group is<BR>
&gt; not pre-constrained to producing either a bit-field oriented or<BR>
&gt; text-oriented format, and may choose to provide both. If the =
group<BR>
&gt; chooses<BR>
&gt; to specify both, it must be possible to mechanically translate =
between<BR>
&gt; the<BR>
&gt; formats without loss of information.<BR>
&gt;<BR>
&gt; Specifying the mechanics of exchanging, transporting, and storing =
SIP<BR>
&gt; Common Log Format records is explicitly out of scope. Specifying =
a<BR>
&gt; real-time<BR>
&gt; transfer mechanism for heuristic analysis is explicitly out of =
scope.<BR>
&gt;<BR>
&gt; The group will generate:<BR>
&gt;<BR>
&gt; A problem statement enunciating the motivation, and use cases for a =
SIP<BR>
&gt; Common Log Format. This analysis will identify the required =
minimal<BR>
&gt; information that must appear in any record.<BR>
&gt;<BR>
&gt; A specification of the SIP Common Log Format record<BR>
&gt;<BR>
&gt; The group will consider providing one or more reference =
implementations<BR>
&gt; for<BR>
&gt; decoding a CLF record.<BR>
&gt;<BR>
&gt; One more piece of logistics ...<BR>
&gt;<BR>
&gt; We'll need at least one, and preferably two, scribes and a =
jabber<BR>
&gt; scribe for<BR>
&gt; our CLF session on Friday during lunch.<BR>
&gt;<BR>
&gt; We'll call for volunteers at the meeting if necessary - we can't =
have<BR>
&gt; the<BR>
&gt; meeting without producing minutes - but it would be great if you =
could<BR>
&gt; consider volunteering NOW.<BR>
&gt;<BR>
&gt; We have about an hour to get through some pretty important =
discussions,<BR>
&gt; and<BR>
&gt; every minute we DON'T spend gazing meaningfully at the attendees =
(us)<BR>
&gt; and<BR>
&gt; looking down at computers trying not to make eye contact with =
the<BR>
&gt; chairs<BR>
&gt; (you) is a minute that we can use more productively!<BR>
&gt;<BR>
&gt; Thanks,<BR>
&gt;<BR>
&gt; Spencer and Theo<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; dispatch mailing list<BR>
&gt; dispatch@ietf.org<BR>
&gt; <A =
HREF=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>
_______________________________________________<BR>
dispatch mailing list<BR>
dispatch@ietf.org<BR>
<A =
HREF=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR>
<BR>
</FONT>
</P>

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From rjsparks@nostrum.com  Fri Jul 31 05:29:59 2009
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Subject: Re: [dispatch] [sip-clf] Next revision for the proposed CLF charter
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For those not in the ad-hoc - this was the text before the meeting we  
had today.

We got a lot of good feedback (minutes should be in soon). Expect a  
revision
of this in a few days (probably sometime week after next).

RjS

On Jul 31, 2009, at 12:07 PM, Robert Sparks wrote:

> The SIP Common Log Format (CLF) working group is chartered to define
> a standard logging format for systems processing SIP messages.
>
> Well-known web servers such as Apache and web proxies like Squid
> support event logging using a common log format.  The logs produced
> using these de-facto standard formats are invaluable to system
> administrators for trouble-shooting a server and tool writers to
> craft tools that mine the log files to produce reports and trends
> and to search for a certain message or messages, a transaction
> or a related set of transactions.  Furthermore, these log records
> can also be used to train anomaly detection systems and feed events
> into a security event management system.
>
> The Session Initiation Protocol does not have a common log
> format. Diverse elements provide distinct log formats making
> it complex to produce tools to analyze them.
>
> The CLF working group will produce a format suitable for logging
> from any SIP element. The format will anticipate the need to
> search, merge, and summarize the log records from diverse elements.
> The format will anticipate the need to correlate messages from
> multiple elements related to a given request (that may fork) or a
> given dialog. The format will take SIP's extensibility into
> consideration, providing a way to represent SIP message components
> that are defined in the future.  The format will anticipate being
> used both for off-line analysis and on-line real-time processing
> applications. The working group will consider the need for
> efficient creation of records and the need for efficient processing
> of the records.
>
> The working group will identify the fields to appear in a log
> record and provide one or more formats for encoding those fields.
> The working group is not pre-constrained to producing either a
> bit-field oriented or text-oriented format, and may choose to
> provide both. If the group chooses to specify both, it must be
> possible to mechanically translate between the formats without loss
> of information.
>
> Specifying the mechanics of exchanging, transporting, and storing
> SIP Common Log Format records is explicitly out of scope. Specifying
> a real-time transfer mechanism for heuristic analysis is explicitly
> out of scope.
>
> The group will generate:
>
> - A problem statement enunciating the motivation,
> and use cases for a SIP Common Log Format. This analysis
> will identify the required minimal information that must
> appear in any record.
>
> - A specification of the SIP Common Log Format record
>
> The group will consider providing one or more reference
> implementations for decoding a CLF record.
>
> Goals and Milestones
> ===========================
>
> Oct 09 - Problem statement, motivation, and use cases
>         WGLC
> Nov 09 - Problem statement, motivation, and use cases
>         to IESG (Informational)
> Jan 10 - SIP Common Log Format specification
>         WGLC
> Feb 10 - SIP Common Log Format specification
>         to IESG (PS)
>
> _______________________________________________
> sip-clf mailing list
> sip-clf@ietf.org
> https://www.ietf.org/mailman/listinfo/sip-clf


From akoba@nttv6.net  Fri Jul 31 21:21:05 2009
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Date: Sat, 01 Aug 2009 13:15:39 +0900
From: Atsushi Kobayashi <akoba@nttv6.net>
To: Robert Sparks <rjsparks@nostrum.com>
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Cc: sip-clf@ietf.org, dispatch mailing list <dispatch@ietf.org>
Subject: Re: [dispatch] [sip-clf] Next revision for the proposed CLF charter
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Dear all,

I have one question.
Does this charter include the media description part, i.e. SDP?
I understood this motivation, however regarding security and trouble
shooting, we need to track signaling and media as well.
If SIP-CLF outputs the media description, we can correlate SIP signaling
and media traffic data. The media traffic data may be outputted by IPFIX
or other protocols.

Otherwise, is it future work?

Regards,
Atsushi


On Fri, 31 Jul 2009 12:07:14 +0200
Robert Sparks <rjsparks@nostrum.com> wrote:

> The SIP Common Log Format (CLF) working group is chartered to define
> a standard logging format for systems processing SIP messages.
> 
> Well-known web servers such as Apache and web proxies like Squid
> support event logging using a common log format.  The logs produced
> using these de-facto standard formats are invaluable to system
> administrators for trouble-shooting a server and tool writers to
> craft tools that mine the log files to produce reports and trends
> and to search for a certain message or messages, a transaction
> or a related set of transactions.  Furthermore, these log records
> can also be used to train anomaly detection systems and feed events
> into a security event management system.
> 
> The Session Initiation Protocol does not have a common log
> format. Diverse elements provide distinct log formats making
> it complex to produce tools to analyze them.
> 
> The CLF working group will produce a format suitable for logging
> from any SIP element. The format will anticipate the need to
> search, merge, and summarize the log records from diverse elements.
> The format will anticipate the need to correlate messages from
> multiple elements related to a given request (that may fork) or a
> given dialog. The format will take SIP's extensibility into
> consideration, providing a way to represent SIP message components
> that are defined in the future.  The format will anticipate being
> used both for off-line analysis and on-line real-time processing
> applications. The working group will consider the need for
> efficient creation of records and the need for efficient processing
> of the records.
> 
> The working group will identify the fields to appear in a log
> record and provide one or more formats for encoding those fields.
> The working group is not pre-constrained to producing either a
> bit-field oriented or text-oriented format, and may choose to
> provide both. If the group chooses to specify both, it must be
> possible to mechanically translate between the formats without loss
> of information.
> 
> Specifying the mechanics of exchanging, transporting, and storing
> SIP Common Log Format records is explicitly out of scope. Specifying
> a real-time transfer mechanism for heuristic analysis is explicitly
> out of scope.
> 
> The group will generate:
> 
> - A problem statement enunciating the motivation,
> and use cases for a SIP Common Log Format. This analysis
> will identify the required minimal information that must
> appear in any record.
> 
> - A specification of the SIP Common Log Format record
> 
> The group will consider providing one or more reference
> implementations for decoding a CLF record.
> 
> Goals and Milestones
> ===========================
> 
> Oct 09 - Problem statement, motivation, and use cases
>           WGLC
> Nov 09 - Problem statement, motivation, and use cases
>           to IESG (Informational)
> Jan 10 - SIP Common Log Format specification
>           WGLC
> Feb 10 - SIP Common Log Format specification
>           to IESG (PS)
> 
> _______________________________________________
> sip-clf mailing list
> sip-clf@ietf.org
> https://www.ietf.org/mailman/listinfo/sip-clf

--- 
Atsushi KOBAYASHI  <akoba@nttv6.net>
NTT Information Sharing Platform Lab.
tel:+81-(0)422-59-3978 fax:+81-(0)422-59-5637

