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Subject: RE: Fw: [BLISS] Welcome! Main question - what services?
Date: Thu, 1 Feb 2007 10:09:00 +0100
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From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: <shida@ntt-at.com>, "Spencer Dawkins" <spencer@mcsr-labs.org>
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I don't think this statement helps.

What should happen is that if BLISS identifies the need for any of your
list, it develops the requirements and passes the requirements to each
of those groups you have mentioned.

Keith=20

> -----Original Message-----
> From: Shida Schubert [mailto:shida@ntt-at.com]=20
> Sent: 31 January 2007 21:34
> To: Spencer Dawkins
> Cc: bliss@ietf.org
> Subject: Re: Fw: [BLISS] Welcome! Main question - what services?
>=20
>=20
> Hi Spencer;
>=20
> Yes you are right. I will change the text to say..
>=20
> "SIP applications such as configuration, presence and IM, and=20
> session policy are clearly out of scope for BLISS and remain=20
> the sole province of SIPPING and SIMPLE."
>=20
> Thanks
> Shida
>=20
>=20
> Spencer Dawkins wrote:
> > Hi, Jonathan and Shida,
> >
> > And now that Peili jogged me to read the actual words in=20
> the charter,=20
> > isn't presence the sole province of SIMPLE?
> >
> > Thanks,
> >
> > Spencer
> >
> >
> >> After looking at the following sentences in the draft charger "The=20
> >> difference between SIPPING and BLISS, is that BLISS is=20
> focused on a=20
> >> particular type of SIP application - call features, and in=20
> >> particular, advanced call features requiring non-trivial call=20
> >> control. SIP applications such as configuration, presence=20
> and IM, and=20
> >> session policy are clearly out of scope for BLISS and=20
> remain the sole=20
> >> province of SIPPING. "
> >>
> >> It seems the WG will actually focus on features initiated=20
> by INVITE=20
> >> and these features are more or less related to traditional=20
> PSTN/ISDN=20
> >> supplementary services.
> >> If I am correct, what about session mode messaging service=20
> which is=20
> >> also a call initiated by INVITE?
> >>
> >> thanks
> >> Peili
> >
> >
> >
> > _______________________________________________
> > BLISS mailing list
> > BLISS@ietf.org
> > https://www1.ietf.org/mailman/listinfo/bliss
> >
> >
>=20
>=20
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
>=20

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From bliss-bounces@ietf.org Thu Feb 01 04:09:48 2007
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Subject: RE: [BLISS] Welcome! Main question - what services?
Date: Thu, 1 Feb 2007 10:09:00 +0100
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From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: <shida@ntt-at.com>, "Peili Xu" <xupeili@huawei.com>
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For all the services when used in the PSTN/ISDN the general intent has
to make these services more generally available to all services AS
APPROPRIATE.

Thus to take for example DND, on the PSTN, there it was primarily
applicable to voice, because voice was the main service used on the
PSTN. In fact it would be more correct to say when activated it applied
generally to all services, thus if you had a fax machine on the same
line, that was also affected  by invocation of the service.

With the introduction of the ISDN, and the ability to recognise the
requested service or communication type in the network and the terminal,
the service expanded to allow it to be invoked for specific services or
communication types, whereas for example DND would not apply on others.

I don't see why we cannot extend this argument to the work in BLISS.=20

If the service makes sense for any SIP capability, then define it for
that.=20

DND can be applied (if desired) for any request that is not in an
existing dialog.=20

Taking call park as an example, it obviously cannot be applied as
currently understood to capabilities supported by MESSAGE requests, due
to the transitory nature of such capabilities, but there is no reason
why it cannot be applied, either generically or for specific SDPs, to
any INVITE initiated dialog.

Regards

Keith



> -----Original Message-----
> From: Shida Schubert [mailto:shida@ntt-at.com]=20
> Sent: 31 January 2007 21:48
> To: Peili Xu
> Cc: bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
>=20
>=20
> Hi Peili;
>=20
> Session mode messaging service as I mentioned in the response=20
> to Spencer is not in scope of BLISS. I will modify the text=20
> that raised this question.
>=20
> Thanks.
> Shida
>=20
> Peili Xu wrote:
> > After looking at the following sentences in the draft charger "The=20
> > difference between SIPPING and BLISS, is that BLISS is focused on a=20
> > particular type of SIP application - call features, and in=20
> particular,=20
> > advanced call features requiring non-trivial call control. SIP=20
> > applications such as configuration, presence and IM, and session=20
> > policy are clearly out of scope for BLISS and remain the=20
> sole province of SIPPING. "
> >
> > It seems the WG will actually focus on features initiated by INVITE=20
> > and these features are more or less related to traditional=20
> PSTN/ISDN=20
> > supplementary services.
> > If I am correct, what about session mode messaging service which is=20
> > also a call initiated by INVITE?
> >
> > thanks
> > Peili
> >
> > -----Original Message-----
> > From: Shida Schubert [mailto:shida@ntt-at.com]
> > Sent: Saturday, January 27, 2007 2:29 AM
> > To: Jonathan Rosenberg
> > Cc: Miguel Garcia; bliss@ietf.org
> > Subject: Re: [BLISS] Welcome! Main question - what services?
> >
> >
> > I tried to incorporate some of the comments provided in the=20
> last week=20
> > including the one below to the charter.
> > Updated charter is available at
> >
> > http://www.bliss-ietf.org/charter.html
> >
> > diff is available at http://www.bliss-ietf.org/diff1.html
> >
> > Shida
> >
> >
> > Jonathan Rosenberg wrote:
> >  =20
> >> OK, I think this is a reasonable change to the charter.
> >>
> >> -Jonathan R.
> >>
> >> Miguel Garcia wrote:
> >>
> >>    =20
> >>> Jonathan Rosenberg wrote:
> >>>
> >>>      =20
> >>>> Miguel Garcia wrote:
> >>>>
> >>>>        =20
> >>>>> I think it is reasonable to work on the approach that=20
> you propose,=20
> >>>>> Jonathan. I have a couple of questions and comments about the=20
> >>>>> proposed charter.
> >>>>>
> >>>>> 1) When the charter speaks about heterogeneous=20
> networks, does it=20
> >>>>> refer only to SIP networks or does it consider interoperability=20
> >>>>> across, say, the PSTN. I am sure the TISPAN guys will ask this=20
> >>>>> question sooner or later, so better to write the=20
> answer, whatever=20
> >>>>> it is, in the charter.
> >>>>>          =20
> >>>> Its referring to SIP neworks. The heterogeneity refers to=20
> >>>> endpoints, which includes gateways to the PSTN. Thus,=20
> the idea was=20
> >>>> most definitely to take into considerating how a gateway might=20
> >>>> implement these services due to the fact that it is a=20
> gateway, and=20
> >>>> is trying to interwork to another network.
> >>>>
> >>>> This is captured in the charter already here:
> >>>>
> >>>> "The problem is exacerbated by the desire for these features to=20
> >>>> work across many types of SIP endpoints, including SIP=20
> hardphones,=20
> >>>> softphones, and gateways to the PSTN and other VoIP=20
> networks, and=20
> >>>> for the desire to work across domain boundaries."
> >>>>
> >>>> Can you suggest some additional text to clarify?
> >>>>        =20
> >>> I am satisfied with your explanation. When I read the charter I=20
> >>> interpreted "domain boundaries" as in "SIP domain=20
> boundaries", but I=20
> >>> understand now that you mean crossing to the PSTN, for=20
> example. If=20
> >>> you want, you could add make the sentence to read "...=20
> and for the=20
> >>> desire to work across domain boundaries and to seamless interwork=20
> >>> with the PSTN, when applicable".
> >>>
> >>> /Miguel
> >>>
> >>>
> >>>
> >>>      =20
> >
> >
> > _______________________________________________
> > BLISS mailing list
> > BLISS@ietf.org
> > https://www1.ietf.org/mailman/listinfo/bliss
> >
> >
> >
> >  =20
>=20
>=20
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
>=20

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From bliss-bounces@ietf.org Thu Feb 01 05:36:42 2007
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Hi Keith;

That is the intention.

I think the following text in the charter under
"guiding principles for this working group will include" covers your
comments.

" Where extensions to standards are required, bring the analysis and need
for new extensions to the appropriate working group (SIP or SIPPING). "

I guess SIMPLE is missing from the list.


Drage, Keith (Keith) wrote:
> I don't think this statement helps.
>
> What should happen is that if BLISS identifies the need for any of your
> list, it develops the requirements and passes the requirements to each
> of those groups you have mentioned.
>
> Keith 
>
>   
>> -----Original Message-----
>> From: Shida Schubert [mailto:shida@ntt-at.com] 
>> Sent: 31 January 2007 21:34
>> To: Spencer Dawkins
>> Cc: bliss@ietf.org
>> Subject: Re: Fw: [BLISS] Welcome! Main question - what services?
>>
>>
>> Hi Spencer;
>>
>> Yes you are right. I will change the text to say..
>>
>> "SIP applications such as configuration, presence and IM, and 
>> session policy are clearly out of scope for BLISS and remain 
>> the sole province of SIPPING and SIMPLE."
>>
>> Thanks
>> Shida
>>
>>
>> Spencer Dawkins wrote:
>>     
>>> Hi, Jonathan and Shida,
>>>
>>> And now that Peili jogged me to read the actual words in 
>>>       
>> the charter, 
>>     
>>> isn't presence the sole province of SIMPLE?
>>>
>>> Thanks,
>>>
>>> Spencer
>>>
>>>
>>>       
>>>> After looking at the following sentences in the draft charger "The 
>>>> difference between SIPPING and BLISS, is that BLISS is 
>>>>         
>> focused on a 
>>     
>>>> particular type of SIP application - call features, and in 
>>>> particular, advanced call features requiring non-trivial call 
>>>> control. SIP applications such as configuration, presence 
>>>>         
>> and IM, and 
>>     
>>>> session policy are clearly out of scope for BLISS and 
>>>>         
>> remain the sole 
>>     
>>>> province of SIPPING. "
>>>>
>>>> It seems the WG will actually focus on features initiated 
>>>>         
>> by INVITE 
>>     
>>>> and these features are more or less related to traditional 
>>>>         
>> PSTN/ISDN 
>>     
>>>> supplementary services.
>>>> If I am correct, what about session mode messaging service 
>>>>         
>> which is 
>>     
>>>> also a call initiated by INVITE?
>>>>
>>>> thanks
>>>> Peili
>>>>         
>>>
>>> _______________________________________________
>>> BLISS mailing list
>>> BLISS@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/bliss
>>>
>>>
>>>       
>> _______________________________________________
>> BLISS mailing list
>> BLISS@ietf.org
>> https://www1.ietf.org/mailman/listinfo/bliss
>>
>>     
>
>
>   


_______________________________________________
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I do think the argument can work, I believe optimally BLISS should always
keep various capabilities in mind when conducting the analysis and
documenting
the variations. We should also make sure we get the best out of what SIP
has
to offer, and not limit the possibilities just because the pre-existing
service is
offered in a certain way. (I do want to stress though that I don't want to
hinder the work for "nice to have" or "may be nice to have" feature that
never gets implemented.)

If the variations include different media such as MSRP or the call
feature is
best served by using presence, I do think we should look into them and ask
appropriate workgroup for an extensions if one is needed. I have no
problems
with that and rather supportive of that..

But defining services solely used by session based messaging and
presence has
a home called SIMPLE at this present moment, and I see no reasons why BLISS
should take on them as long as SIMPLE exists.

Thought?

Regards
Shida




Drage, Keith (Keith) wrote:
> For all the services when used in the PSTN/ISDN the general intent has
> to make these services more generally available to all services AS
> APPROPRIATE.
>
> Thus to take for example DND, on the PSTN, there it was primarily
> applicable to voice, because voice was the main service used on the
> PSTN. In fact it would be more correct to say when activated it applied
> generally to all services, thus if you had a fax machine on the same
> line, that was also affected  by invocation of the service.
>
> With the introduction of the ISDN, and the ability to recognise the
> requested service or communication type in the network and the terminal,
> the service expanded to allow it to be invoked for specific services or
> communication types, whereas for example DND would not apply on others.
>
> I don't see why we cannot extend this argument to the work in BLISS. 
>
> If the service makes sense for any SIP capability, then define it for
> that. 
>
> DND can be applied (if desired) for any request that is not in an
> existing dialog. 
>
> Taking call park as an example, it obviously cannot be applied as
> currently understood to capabilities supported by MESSAGE requests, due
> to the transitory nature of such capabilities, but there is no reason
> why it cannot be applied, either generically or for specific SDPs, to
> any INVITE initiated dialog.
>
> Regards
>
> Keith
>
>
>
>   
>> -----Original Message-----
>> From: Shida Schubert [mailto:shida@ntt-at.com] 
>> Sent: 31 January 2007 21:48
>> To: Peili Xu
>> Cc: bliss@ietf.org
>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>
>>
>> Hi Peili;
>>
>> Session mode messaging service as I mentioned in the response 
>> to Spencer is not in scope of BLISS. I will modify the text 
>> that raised this question.
>>
>> Thanks.
>> Shida
>>
>> Peili Xu wrote:
>>     
>>> After looking at the following sentences in the draft charger "The 
>>> difference between SIPPING and BLISS, is that BLISS is focused on a 
>>> particular type of SIP application - call features, and in 
>>>       
>> particular, 
>>     
>>> advanced call features requiring non-trivial call control. SIP 
>>> applications such as configuration, presence and IM, and session 
>>> policy are clearly out of scope for BLISS and remain the 
>>>       
>> sole province of SIPPING. "
>>     
>>> It seems the WG will actually focus on features initiated by INVITE 
>>> and these features are more or less related to traditional 
>>>       
>> PSTN/ISDN 
>>     
>>> supplementary services.
>>> If I am correct, what about session mode messaging service which is 
>>> also a call initiated by INVITE?
>>>
>>> thanks
>>> Peili
>>>
>>> -----Original Message-----
>>> From: Shida Schubert [mailto:shida@ntt-at.com]
>>> Sent: Saturday, January 27, 2007 2:29 AM
>>> To: Jonathan Rosenberg
>>> Cc: Miguel Garcia; bliss@ietf.org
>>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>>
>>>
>>> I tried to incorporate some of the comments provided in the 
>>>       
>> last week 
>>     
>>> including the one below to the charter.
>>> Updated charter is available at
>>>
>>> http://www.bliss-ietf.org/charter.html
>>>
>>> diff is available at http://www.bliss-ietf.org/diff1.html
>>>
>>> Shida
>>>
>>>
>>> Jonathan Rosenberg wrote:
>>>   
>>>       
>>>> OK, I think this is a reasonable change to the charter.
>>>>
>>>> -Jonathan R.
>>>>
>>>> Miguel Garcia wrote:
>>>>
>>>>     
>>>>         
>>>>> Jonathan Rosenberg wrote:
>>>>>
>>>>>       
>>>>>           
>>>>>> Miguel Garcia wrote:
>>>>>>
>>>>>>         
>>>>>>             
>>>>>>> I think it is reasonable to work on the approach that 
>>>>>>>               
>> you propose, 
>>     
>>>>>>> Jonathan. I have a couple of questions and comments about the 
>>>>>>> proposed charter.
>>>>>>>
>>>>>>> 1) When the charter speaks about heterogeneous 
>>>>>>>               
>> networks, does it 
>>     
>>>>>>> refer only to SIP networks or does it consider interoperability 
>>>>>>> across, say, the PSTN. I am sure the TISPAN guys will ask this 
>>>>>>> question sooner or later, so better to write the 
>>>>>>>               
>> answer, whatever 
>>     
>>>>>>> it is, in the charter.
>>>>>>>           
>>>>>>>               
>>>>>> Its referring to SIP neworks. The heterogeneity refers to 
>>>>>> endpoints, which includes gateways to the PSTN. Thus, 
>>>>>>             
>> the idea was 
>>     
>>>>>> most definitely to take into considerating how a gateway might 
>>>>>> implement these services due to the fact that it is a 
>>>>>>             
>> gateway, and 
>>     
>>>>>> is trying to interwork to another network.
>>>>>>
>>>>>> This is captured in the charter already here:
>>>>>>
>>>>>> "The problem is exacerbated by the desire for these features to 
>>>>>> work across many types of SIP endpoints, including SIP 
>>>>>>             
>> hardphones, 
>>     
>>>>>> softphones, and gateways to the PSTN and other VoIP 
>>>>>>             
>> networks, and 
>>     
>>>>>> for the desire to work across domain boundaries."
>>>>>>
>>>>>> Can you suggest some additional text to clarify?
>>>>>>         
>>>>>>             
>>>>> I am satisfied with your explanation. When I read the charter I 
>>>>> interpreted "domain boundaries" as in "SIP domain 
>>>>>           
>> boundaries", but I 
>>     
>>>>> understand now that you mean crossing to the PSTN, for 
>>>>>           
>> example. If 
>>     
>>>>> you want, you could add make the sentence to read "... 
>>>>>           
>> and for the 
>>     
>>>>> desire to work across domain boundaries and to seamless interwork 
>>>>> with the PSTN, when applicable".
>>>>>
>>>>> /Miguel
>>>>>
>>>>>
>>>>>
>>>>>       
>>>>>           
>>> _______________________________________________
>>> BLISS mailing list
>>> BLISS@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/bliss
>>>
>>>
>>>
>>>   
>>>       
>> _______________________________________________
>> BLISS mailing list
>> BLISS@ietf.org
>> https://www1.ietf.org/mailman/listinfo/bliss
>>
>>     
>
>
>   


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From: Paul Kyzivat <pkyzivat@cisco.com>
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I have been inovlved in an exercise of trying to map traditional PSTN 
services to SIP, and make them apply more generally to the broader set 
of things that can happen in SIP.

Some services seem to apply to an AOR, others to a UA, and some 
potentially to a particular media stream. And in some cases there are 
multiple meaningful ways to apply the service. For instance:

- DND can meaningfully apply at either the AOR level or the UA level.
   You could consider these to be different services.

- HOLD can apply to a whole dialog, or to a particular media stream
   within a dialog.

Its important to consider these things.

	Paul

Drage, Keith (Keith) wrote:
> For all the services when used in the PSTN/ISDN the general intent has
> to make these services more generally available to all services AS
> APPROPRIATE.
> 
> Thus to take for example DND, on the PSTN, there it was primarily
> applicable to voice, because voice was the main service used on the
> PSTN. In fact it would be more correct to say when activated it applied
> generally to all services, thus if you had a fax machine on the same
> line, that was also affected  by invocation of the service.
> 
> With the introduction of the ISDN, and the ability to recognise the
> requested service or communication type in the network and the terminal,
> the service expanded to allow it to be invoked for specific services or
> communication types, whereas for example DND would not apply on others.
> 
> I don't see why we cannot extend this argument to the work in BLISS. 
> 
> If the service makes sense for any SIP capability, then define it for
> that. 
> 
> DND can be applied (if desired) for any request that is not in an
> existing dialog. 
> 
> Taking call park as an example, it obviously cannot be applied as
> currently understood to capabilities supported by MESSAGE requests, due
> to the transitory nature of such capabilities, but there is no reason
> why it cannot be applied, either generically or for specific SDPs, to
> any INVITE initiated dialog.
> 
> Regards
> 
> Keith
> 
> 
> 
>> -----Original Message-----
>> From: Shida Schubert [mailto:shida@ntt-at.com] 
>> Sent: 31 January 2007 21:48
>> To: Peili Xu
>> Cc: bliss@ietf.org
>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>
>>
>> Hi Peili;
>>
>> Session mode messaging service as I mentioned in the response 
>> to Spencer is not in scope of BLISS. I will modify the text 
>> that raised this question.
>>
>> Thanks.
>> Shida
>>
>> Peili Xu wrote:
>>> After looking at the following sentences in the draft charger "The 
>>> difference between SIPPING and BLISS, is that BLISS is focused on a 
>>> particular type of SIP application - call features, and in 
>> particular, 
>>> advanced call features requiring non-trivial call control. SIP 
>>> applications such as configuration, presence and IM, and session 
>>> policy are clearly out of scope for BLISS and remain the 
>> sole province of SIPPING. "
>>> It seems the WG will actually focus on features initiated by INVITE 
>>> and these features are more or less related to traditional 
>> PSTN/ISDN 
>>> supplementary services.
>>> If I am correct, what about session mode messaging service which is 
>>> also a call initiated by INVITE?
>>>
>>> thanks
>>> Peili
>>>
>>> -----Original Message-----
>>> From: Shida Schubert [mailto:shida@ntt-at.com]
>>> Sent: Saturday, January 27, 2007 2:29 AM
>>> To: Jonathan Rosenberg
>>> Cc: Miguel Garcia; bliss@ietf.org
>>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>>
>>>
>>> I tried to incorporate some of the comments provided in the 
>> last week 
>>> including the one below to the charter.
>>> Updated charter is available at
>>>
>>> http://www.bliss-ietf.org/charter.html
>>>
>>> diff is available at http://www.bliss-ietf.org/diff1.html
>>>
>>> Shida
>>>
>>>
>>> Jonathan Rosenberg wrote:
>>>   
>>>> OK, I think this is a reasonable change to the charter.
>>>>
>>>> -Jonathan R.
>>>>
>>>> Miguel Garcia wrote:
>>>>
>>>>     
>>>>> Jonathan Rosenberg wrote:
>>>>>
>>>>>       
>>>>>> Miguel Garcia wrote:
>>>>>>
>>>>>>         
>>>>>>> I think it is reasonable to work on the approach that 
>> you propose, 
>>>>>>> Jonathan. I have a couple of questions and comments about the 
>>>>>>> proposed charter.
>>>>>>>
>>>>>>> 1) When the charter speaks about heterogeneous 
>> networks, does it 
>>>>>>> refer only to SIP networks or does it consider interoperability 
>>>>>>> across, say, the PSTN. I am sure the TISPAN guys will ask this 
>>>>>>> question sooner or later, so better to write the 
>> answer, whatever 
>>>>>>> it is, in the charter.
>>>>>>>           
>>>>>> Its referring to SIP neworks. The heterogeneity refers to 
>>>>>> endpoints, which includes gateways to the PSTN. Thus, 
>> the idea was 
>>>>>> most definitely to take into considerating how a gateway might 
>>>>>> implement these services due to the fact that it is a 
>> gateway, and 
>>>>>> is trying to interwork to another network.
>>>>>>
>>>>>> This is captured in the charter already here:
>>>>>>
>>>>>> "The problem is exacerbated by the desire for these features to 
>>>>>> work across many types of SIP endpoints, including SIP 
>> hardphones, 
>>>>>> softphones, and gateways to the PSTN and other VoIP 
>> networks, and 
>>>>>> for the desire to work across domain boundaries."
>>>>>>
>>>>>> Can you suggest some additional text to clarify?
>>>>>>         
>>>>> I am satisfied with your explanation. When I read the charter I 
>>>>> interpreted "domain boundaries" as in "SIP domain 
>> boundaries", but I 
>>>>> understand now that you mean crossing to the PSTN, for 
>> example. If 
>>>>> you want, you could add make the sentence to read "... 
>> and for the 
>>>>> desire to work across domain boundaries and to seamless interwork 
>>>>> with the PSTN, when applicable".
>>>>>
>>>>> /Miguel
>>>>>
>>>>>
>>>>>
>>>>>       
>>>
>>> _______________________________________________
>>> BLISS mailing list
>>> BLISS@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/bliss
>>>
>>>
>>>
>>>   
>>
>> _______________________________________________
>> BLISS mailing list
>> BLISS@ietf.org
>> https://www1.ietf.org/mailman/listinfo/bliss
>>
> 
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

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   From: Shida Schubert <shida@ntt-at.com>

   But defining services solely used by session based messaging and
   presence has a home called SIMPLE at this present moment, and I see
   no reasons why BLISS should take on them as long as SIMPLE exists.

I don't know about presence.  But within the SIP philosophy,
session-based messaging is "just another media type" when compared to
audio calling.  So it seems to me that if we discern an apparent
difference in "calling features" between session-based messaging and
audio calling, we are probably misunderstanding the situation.

Dale

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DND can also apply to a particular media stream. In particular, if I am 
in a meeting, I may want to activate the DND for audio and video, but 
not for messaging.

I guess in general, you need to consider all the different media streams 
that can apply to service. It would be a big mistake just to restrict to 
audio or to "things that are initiated with an INVITE" because this is 
what the PSTN did.

/Miguel

Paul Kyzivat wrote:
> I have been inovlved in an exercise of trying to map traditional PSTN 
> services to SIP, and make them apply more generally to the broader set 
> of things that can happen in SIP.
> 
> Some services seem to apply to an AOR, others to a UA, and some 
> potentially to a particular media stream. And in some cases there are 
> multiple meaningful ways to apply the service. For instance:
> 
> - DND can meaningfully apply at either the AOR level or the UA level.
>   You could consider these to be different services.
> 
> - HOLD can apply to a whole dialog, or to a particular media stream
>   within a dialog.
> 
> Its important to consider these things.
> 
>     Paul
> 


-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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Subject: RE: [BLISS] Welcome! Main question - what services?
Date: Thu, 1 Feb 2007 16:08:34 +0100
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	<45C1F4A6.9040704@cisco.com> <45C200DF.2020908@nokia.com>
From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>,
	"Paul Kyzivat" <pkyzivat@cisco.com>
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I assume that you mean you wish to be DND to a request with a SIP offer
of a particular media stream, rather than just DND to a particular media
stream.

Regards

Keith=20

> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
> Sent: 01 February 2007 15:02
> To: Paul Kyzivat
> Cc: Drage, Keith (Keith); bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
>=20
> DND can also apply to a particular media stream. In=20
> particular, if I am in a meeting, I may want to activate the=20
> DND for audio and video, but not for messaging.
>=20
> I guess in general, you need to consider all the different=20
> media streams that can apply to service. It would be a big=20
> mistake just to restrict to audio or to "things that are=20
> initiated with an INVITE" because this is what the PSTN did.
>=20
> /Miguel
>=20
> Paul Kyzivat wrote:
> > I have been inovlved in an exercise of trying to map=20
> traditional PSTN=20
> > services to SIP, and make them apply more generally to the=20
> broader set=20
> > of things that can happen in SIP.
> >=20
> > Some services seem to apply to an AOR, others to a UA, and some=20
> > potentially to a particular media stream. And in some cases=20
> there are=20
> > multiple meaningful ways to apply the service. For instance:
> >=20
> > - DND can meaningfully apply at either the AOR level or the=20
> UA level.
> >   You could consider these to be different services.
> >=20
> > - HOLD can apply to a whole dialog, or to a particular media stream
> >   within a dialog.
> >=20
> > Its important to consider these things.
> >=20
> >     Paul
> >=20
>=20
>=20
> --=20
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.garcia@neonsite.net
> Nokia Research Center      Helsinki, Finland
>=20

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Subject: Re: [BLISS] Welcome! Main question - what services?
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Right... the capability of indicating a DND for SIP offers that contain 
certain media streams. Similarly with SIP methods.

/Miguel

Drage, Keith (Keith) wrote:
> I assume that you mean you wish to be DND to a request with a SIP offer
> of a particular media stream, rather than just DND to a particular media
> stream.
> 
> Regards
> 
> Keith 
> 
>> -----Original Message-----
>> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
>> Sent: 01 February 2007 15:02
>> To: Paul Kyzivat
>> Cc: Drage, Keith (Keith); bliss@ietf.org
>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>
>> DND can also apply to a particular media stream. In 
>> particular, if I am in a meeting, I may want to activate the 
>> DND for audio and video, but not for messaging.
>>
>> I guess in general, you need to consider all the different 
>> media streams that can apply to service. It would be a big 
>> mistake just to restrict to audio or to "things that are 
>> initiated with an INVITE" because this is what the PSTN did.
>>
>> /Miguel
>>
>> Paul Kyzivat wrote:
>>> I have been inovlved in an exercise of trying to map 
>> traditional PSTN 
>>> services to SIP, and make them apply more generally to the 
>> broader set 
>>> of things that can happen in SIP.
>>>
>>> Some services seem to apply to an AOR, others to a UA, and some 
>>> potentially to a particular media stream. And in some cases 
>> there are 
>>> multiple meaningful ways to apply the service. For instance:
>>>
>>> - DND can meaningfully apply at either the AOR level or the 
>> UA level.
>>>   You could consider these to be different services.
>>>
>>> - HOLD can apply to a whole dialog, or to a particular media stream
>>>   within a dialog.
>>>
>>> Its important to consider these things.
>>>
>>>     Paul
>>>
>>
>> -- 
>> Miguel A. Garcia           tel:+358-50-4804586
>> sip:miguel.garcia@neonsite.net
>> Nokia Research Center      Helsinki, Finland
>>

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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From bliss-bounces@ietf.org Thu Feb 01 10:21:57 2007
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From: xupeili 18916 <xupeili@huawei.com>
Subject: Re: [BLISS] Welcome! Main question - what services?
To: Paul Kyzivat <pkyzivat@cisco.com>
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I fully agree with Paul.

When we are talking about service, we are looking at  the things from user's point of view.

I would rather treat the services listed in the Charter as kinds of "Service Pattern", e.g DND means rejecting sth when the user don't want to accept it.

Here, the "sth" can be a voice call, a message or whatever (of course implemented by SIP).

we can always find several key points for a "Service Pattern", e.g the entity, the time, the media......

We may get more general solution if we consider the pattern instead of a specific service instance.

Peili






----- Original Message -----
From: Paul Kyzivat <pkyzivat@cisco.com>
Date: Thursday, February 1, 2007 10:09 pm
Subject: Re: [BLISS] Welcome! Main question - what services?

> I have been inovlved in an exercise of trying to map traditional 
> PSTN 
> services to SIP, and make them apply more generally to the broader 
> set 
> of things that can happen in SIP.
> 
> Some services seem to apply to an AOR, others to a UA, and some 
> potentially to a particular media stream. And in some cases there 
> are 
> multiple meaningful ways to apply the service. For instance:
> 
> - DND can meaningfully apply at either the AOR level or the UA level.
>   You could consider these to be different services.
> 
> - HOLD can apply to a whole dialog, or to a particular media stream
>   within a dialog.
> 
> Its important to consider these things.
> 
> 	Paul
> 
> Drage, Keith (Keith) wrote:
> > For all the services when used in the PSTN/ISDN the general 
> intent has
> > to make these services more generally available to all services AS
> > APPROPRIATE.
> > 
> > Thus to take for example DND, on the PSTN, there it was primarily
> > applicable to voice, because voice was the main service used on the
> > PSTN. In fact it would be more correct to say when activated it 
> applied> generally to all services, thus if you had a fax machine 
> on the same
> > line, that was also affected  by invocation of the service.
> > 
> > With the introduction of the ISDN, and the ability to recognise the
> > requested service or communication type in the network and the 
> terminal,> the service expanded to allow it to be invoked for 
> specific services or
> > communication types, whereas for example DND would not apply on 
> others.> 
> > I don't see why we cannot extend this argument to the work in 
> BLISS. 
> > 
> > If the service makes sense for any SIP capability, then define it 
> for> that. 
> > 
> > DND can be applied (if desired) for any request that is not in an
> > existing dialog. 
> > 
> > Taking call park as an example, it obviously cannot be applied as
> > currently understood to capabilities supported by MESSAGE 
> requests, due
> > to the transitory nature of such capabilities, but there is no 
> reason> why it cannot be applied, either generically or for 
> specific SDPs, to
> > any INVITE initiated dialog.
> > 
> > Regards
> > 
> > Keith
> > 
> > 
> > 
> >> -----Original Message-----
> >> From: Shida Schubert [shida@ntt-at.com] 
> >> Sent: 31 January 2007 21:48
> >> To: Peili Xu
> >> Cc: bliss@ietf.org
> >> Subject: Re: [BLISS] Welcome! Main question - what services?
> >>
> >>
> >> Hi Peili;
> >>
> >> Session mode messaging service as I mentioned in the response 
> >> to Spencer is not in scope of BLISS. I will modify the text 
> >> that raised this question.
> >>
> >> Thanks.
> >> Shida
> >>
> >> Peili Xu wrote:
> >>> After looking at the following sentences in the draft charger 
> "The 
> >>> difference between SIPPING and BLISS, is that BLISS is focused 
> on a 
> >>> particular type of SIP application - call features, and in 
> >> particular, 
> >>> advanced call features requiring non-trivial call control. SIP 
> >>> applications such as configuration, presence and IM, and 
> session 
> >>> policy are clearly out of scope for BLISS and remain the 
> >> sole province of SIPPING. "
> >>> It seems the WG will actually focus on features initiated by 
> INVITE 
> >>> and these features are more or less related to traditional 
> >> PSTN/ISDN 
> >>> supplementary services.
> >>> If I am correct, what about session mode messaging service 
> which is 
> >>> also a call initiated by INVITE?
> >>>
> >>> thanks
> >>> Peili
> >>>
> >>> -----Original Message-----
> >>> From: Shida Schubert [shida@ntt-at.com]
> >>> Sent: Saturday, January 27, 2007 2:29 AM
> >>> To: Jonathan Rosenberg
> >>> Cc: Miguel Garcia; bliss@ietf.org
> >>> Subject: Re: [BLISS] Welcome! Main question - what services?
> >>>
> >>>
> >>> I tried to incorporate some of the comments provided in the 
> >> last week 
> >>> including the one below to the charter.
> >>> Updated charter is available at
> >>>
> >>> http://www.bliss-ietf.org/charter.html
> >>>
> >>> diff is available at http://www.bliss-ietf.org/diff1.html
> >>>
> >>> Shida
> >>>
> >>>
> >>> Jonathan Rosenberg wrote:
> >>>   
> >>>> OK, I think this is a reasonable change to the charter.
> >>>>
> >>>> -Jonathan R.
> >>>>
> >>>> Miguel Garcia wrote:
> >>>>
> >>>>     
> >>>>> Jonathan Rosenberg wrote:
> >>>>>
> >>>>>       
> >>>>>> Miguel Garcia wrote:
> >>>>>>
> >>>>>>         
> >>>>>>> I think it is reasonable to work on the approach that 
> >> you propose, 
> >>>>>>> Jonathan. I have a couple of questions and comments about 
> the 
> >>>>>>> proposed charter.
> >>>>>>>
> >>>>>>> 1) When the charter speaks about heterogeneous 
> >> networks, does it 
> >>>>>>> refer only to SIP networks or does it consider 
> interoperability 
> >>>>>>> across, say, the PSTN. I am sure the TISPAN guys will ask 
> this 
> >>>>>>> question sooner or later, so better to write the 
> >> answer, whatever 
> >>>>>>> it is, in the charter.
> >>>>>>>           
> >>>>>> Its referring to SIP neworks. The heterogeneity refers to 
> >>>>>> endpoints, which includes gateways to the PSTN. Thus, 
> >> the idea was 
> >>>>>> most definitely to take into considerating how a gateway 
> might 
> >>>>>> implement these services due to the fact that it is a 
> >> gateway, and 
> >>>>>> is trying to interwork to another network.
> >>>>>>
> >>>>>> This is captured in the charter already here:
> >>>>>>
> >>>>>> "The problem is exacerbated by the desire for these features 
> to 
> >>>>>> work across many types of SIP endpoints, including SIP 
> >> hardphones, 
> >>>>>> softphones, and gateways to the PSTN and other VoIP 
> >> networks, and 
> >>>>>> for the desire to work across domain boundaries."
> >>>>>>
> >>>>>> Can you suggest some additional text to clarify?
> >>>>>>         
> >>>>> I am satisfied with your explanation. When I read the charter 
> I 
> >>>>> interpreted "domain boundaries" as in "SIP domain 
> >> boundaries", but I 
> >>>>> understand now that you mean crossing to the PSTN, for 
> >> example. If 
> >>>>> you want, you could add make the sentence to read "... 
> >> and for the 
> >>>>> desire to work across domain boundaries and to seamless 
> interwork 
> >>>>> with the PSTN, when applicable".
> >>>>>
> >>>>> /Miguel
> >>>>>
> >>>>>
> >>>>>
> >>>>>       
> >>>
> >>> _______________________________________________
> >>> BLISS mailing list
> >>> BLISS@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/bliss
> >>>
> >>>
> >>>
> >>>   
> >>
> >> _______________________________________________
> >> BLISS mailing list
> >> BLISS@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/bliss
> >>
> > 
> > _______________________________________________
> > BLISS mailing list
> > BLISS@ietf.org
> > https://www1.ietf.org/mailman/listinfo/bliss
> > 
> 


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   From: Miguel Garcia <Miguel.An.Garcia@nokia.com>

   DND can also apply to a particular media stream. In particular, if I am 
   in a meeting, I may want to activate the DND for audio and video, but 
   not for messaging.

I assume you mean "media type" here -- a media stream is what is
established by an m line in the SDP of an existing dialog.  But you
couldn't apply DND to a media *stream* until it exists, and at that
point, it's too late.

Dale

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Subject: RE: [BLISS] Welcome! Main question - what services?
Date: Thu, 1 Feb 2007 16:30:44 +0100
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From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Agreed.

It is these type of considerations that allow to actually know whether
the service makes sense in its new environment. They do need to form
part of any requirements analysis in the proposed BLISS.

Keith=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: 01 February 2007 14:10
> To: Drage, Keith (Keith)
> Cc: shida@ntt-at.com; Peili Xu; bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
>=20
> I have been inovlved in an exercise of trying to map=20
> traditional PSTN services to SIP, and make them apply more=20
> generally to the broader set of things that can happen in SIP.
>=20
> Some services seem to apply to an AOR, others to a UA, and=20
> some potentially to a particular media stream. And in some=20
> cases there are multiple meaningful ways to apply the=20
> service. For instance:
>=20
> - DND can meaningfully apply at either the AOR level or the UA level.
>    You could consider these to be different services.
>=20
> - HOLD can apply to a whole dialog, or to a particular media stream
>    within a dialog.
>=20
> Its important to consider these things.
>=20
> 	Paul
>=20
> Drage, Keith (Keith) wrote:
> > For all the services when used in the PSTN/ISDN the general=20
> intent has=20
> > to make these services more generally available to all services AS=20
> > APPROPRIATE.
> >=20
> > Thus to take for example DND, on the PSTN, there it was primarily=20
> > applicable to voice, because voice was the main service used on the=20
> > PSTN. In fact it would be more correct to say when activated it=20
> > applied generally to all services, thus if you had a fax machine on=20
> > the same line, that was also affected  by invocation of the service.
> >=20
> > With the introduction of the ISDN, and the ability to recognise the=20
> > requested service or communication type in the network and the=20
> > terminal, the service expanded to allow it to be invoked=20
> for specific=20
> > services or communication types, whereas for example DND=20
> would not apply on others.
> >=20
> > I don't see why we cannot extend this argument to the work=20
> in BLISS.=20
> >=20
> > If the service makes sense for any SIP capability, then=20
> define it for=20
> > that.
> >=20
> > DND can be applied (if desired) for any request that is not in an=20
> > existing dialog.
> >=20
> > Taking call park as an example, it obviously cannot be applied as=20
> > currently understood to capabilities supported by MESSAGE requests,=20
> > due to the transitory nature of such capabilities, but there is no=20
> > reason why it cannot be applied, either generically or for specific=20
> > SDPs, to any INVITE initiated dialog.
> >=20
> > Regards
> >=20
> > Keith
> >=20
> >=20
> >=20
> >> -----Original Message-----
> >> From: Shida Schubert [mailto:shida@ntt-at.com]
> >> Sent: 31 January 2007 21:48
> >> To: Peili Xu
> >> Cc: bliss@ietf.org
> >> Subject: Re: [BLISS] Welcome! Main question - what services?
> >>
> >>
> >> Hi Peili;
> >>
> >> Session mode messaging service as I mentioned in the response to=20
> >> Spencer is not in scope of BLISS. I will modify the text=20
> that raised=20
> >> this question.
> >>
> >> Thanks.
> >> Shida
> >>
> >> Peili Xu wrote:
> >>> After looking at the following sentences in the draft=20
> charger "The=20
> >>> difference between SIPPING and BLISS, is that BLISS is=20
> focused on a=20
> >>> particular type of SIP application - call features, and in
> >> particular,
> >>> advanced call features requiring non-trivial call control. SIP=20
> >>> applications such as configuration, presence and IM, and session=20
> >>> policy are clearly out of scope for BLISS and remain the
> >> sole province of SIPPING. "
> >>> It seems the WG will actually focus on features initiated=20
> by INVITE=20
> >>> and these features are more or less related to traditional
> >> PSTN/ISDN
> >>> supplementary services.
> >>> If I am correct, what about session mode messaging=20
> service which is=20
> >>> also a call initiated by INVITE?
> >>>
> >>> thanks
> >>> Peili
> >>>
> >>> -----Original Message-----
> >>> From: Shida Schubert [mailto:shida@ntt-at.com]
> >>> Sent: Saturday, January 27, 2007 2:29 AM
> >>> To: Jonathan Rosenberg
> >>> Cc: Miguel Garcia; bliss@ietf.org
> >>> Subject: Re: [BLISS] Welcome! Main question - what services?
> >>>
> >>>
> >>> I tried to incorporate some of the comments provided in the
> >> last week
> >>> including the one below to the charter.
> >>> Updated charter is available at
> >>>
> >>> http://www.bliss-ietf.org/charter.html
> >>>
> >>> diff is available at http://www.bliss-ietf.org/diff1.html
> >>>
> >>> Shida
> >>>
> >>>
> >>> Jonathan Rosenberg wrote:
> >>>  =20
> >>>> OK, I think this is a reasonable change to the charter.
> >>>>
> >>>> -Jonathan R.
> >>>>
> >>>> Miguel Garcia wrote:
> >>>>
> >>>>    =20
> >>>>> Jonathan Rosenberg wrote:
> >>>>>
> >>>>>      =20
> >>>>>> Miguel Garcia wrote:
> >>>>>>
> >>>>>>        =20
> >>>>>>> I think it is reasonable to work on the approach that
> >> you propose,
> >>>>>>> Jonathan. I have a couple of questions and comments about the=20
> >>>>>>> proposed charter.
> >>>>>>>
> >>>>>>> 1) When the charter speaks about heterogeneous
> >> networks, does it
> >>>>>>> refer only to SIP networks or does it consider=20
> interoperability=20
> >>>>>>> across, say, the PSTN. I am sure the TISPAN guys will=20
> ask this=20
> >>>>>>> question sooner or later, so better to write the
> >> answer, whatever
> >>>>>>> it is, in the charter.
> >>>>>>>          =20
> >>>>>> Its referring to SIP neworks. The heterogeneity refers to=20
> >>>>>> endpoints, which includes gateways to the PSTN. Thus,
> >> the idea was
> >>>>>> most definitely to take into considerating how a gateway might=20
> >>>>>> implement these services due to the fact that it is a
> >> gateway, and
> >>>>>> is trying to interwork to another network.
> >>>>>>
> >>>>>> This is captured in the charter already here:
> >>>>>>
> >>>>>> "The problem is exacerbated by the desire for these=20
> features to=20
> >>>>>> work across many types of SIP endpoints, including SIP
> >> hardphones,
> >>>>>> softphones, and gateways to the PSTN and other VoIP
> >> networks, and
> >>>>>> for the desire to work across domain boundaries."
> >>>>>>
> >>>>>> Can you suggest some additional text to clarify?
> >>>>>>        =20
> >>>>> I am satisfied with your explanation. When I read the charter I=20
> >>>>> interpreted "domain boundaries" as in "SIP domain
> >> boundaries", but I
> >>>>> understand now that you mean crossing to the PSTN, for
> >> example. If
> >>>>> you want, you could add make the sentence to read "...=20
> >> and for the
> >>>>> desire to work across domain boundaries and to seamless=20
> interwork=20
> >>>>> with the PSTN, when applicable".
> >>>>>
> >>>>> /Miguel
> >>>>>
> >>>>>
> >>>>>
> >>>>>      =20
> >>>
> >>> _______________________________________________
> >>> BLISS mailing list
> >>> BLISS@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/bliss
> >>>
> >>>
> >>>
> >>>  =20
> >>
> >> _______________________________________________
> >> BLISS mailing list
> >> BLISS@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/bliss
> >>
> >=20
> > _______________________________________________
> > BLISS mailing list
> > BLISS@ietf.org
> > https://www1.ietf.org/mailman/listinfo/bliss
> >=20
>=20

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Correct, s/media stream/media type

/Miguel

bliss-bounces@ietf.org wrote:
>    From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
> 
>    DND can also apply to a particular media stream. In particular, if I am 
>    in a meeting, I may want to activate the DND for audio and video, but 
>    not for messaging.
> 
> I assume you mean "media type" here -- a media stream is what is
> established by an m line in the SDP of an existing dialog.  But you
> couldn't apply DND to a media *stream* until it exists, and at that
> point, it's too late.
> 
> Dale
> 
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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Date: Thu, 01 Feb 2007 09:58:38 -0600
From: Spencer Dawkins <spencer@mcsr-labs.org>
Subject: Re: [BLISS] Welcome! Main question - what services?
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I agree with Keith, Peili, and Paul here.

I am sympathetic to the need to get something out quickly, but also see the 
need for requirements analysis, so we don't codify patterns that don't work 
in situations that we weren't thinking about.

Spencer

From: "Drage, Keith (Keith)" <drage@alcatel-lucent.com>

Sent: Thursday, February 01, 2007 9:30 AM
Subject: RE: [BLISS] Welcome! Main question - what services?


Agreed.

It is these type of considerations that allow to actually know whether
the service makes sense in its new environment. They do need to form
part of any requirements analysis in the proposed BLISS.

Keith

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 01 February 2007 14:10
> To: Drage, Keith (Keith)
> Cc: shida@ntt-at.com; Peili Xu; bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
>
> I have been inovlved in an exercise of trying to map
> traditional PSTN services to SIP, and make them apply more
> generally to the broader set of things that can happen in SIP.
>
> Some services seem to apply to an AOR, others to a UA, and
> some potentially to a particular media stream. And in some
> cases there are multiple meaningful ways to apply the
> service. For instance:
>
> - DND can meaningfully apply at either the AOR level or the UA level.
>    You could consider these to be different services.
>
> - HOLD can apply to a whole dialog, or to a particular media stream
>    within a dialog.
>
> Its important to consider these things.
>
> Paul
>
> Drage, Keith (Keith) wrote:
> > For all the services when used in the PSTN/ISDN the general
> intent has
> > to make these services more generally available to all services AS
> > APPROPRIATE.
> >
> > Thus to take for example DND, on the PSTN, there it was primarily
> > applicable to voice, because voice was the main service used on the
> > PSTN. In fact it would be more correct to say when activated it
> > applied generally to all services, thus if you had a fax machine on
> > the same line, that was also affected  by invocation of the service.
> >
> > With the introduction of the ISDN, and the ability to recognise the
> > requested service or communication type in the network and the
> > terminal, the service expanded to allow it to be invoked
> for specific
> > services or communication types, whereas for example DND
> would not apply on others.
> >
> > I don't see why we cannot extend this argument to the work
> in BLISS.
> >
> > If the service makes sense for any SIP capability, then
> define it for
> > that.
> >
> > DND can be applied (if desired) for any request that is not in an
> > existing dialog.
> >
> > Taking call park as an example, it obviously cannot be applied as
> > currently understood to capabilities supported by MESSAGE requests,
> > due to the transitory nature of such capabilities, but there is no
> > reason why it cannot be applied, either generically or for specific
> > SDPs, to any INVITE initiated dialog.
> >
> > Regards
> >
> > Keith 



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I don't think I buy this flavor of DND. I think it will cause interop 
problems.

IMO the key aspect of DND is about alerting. Which media that are 
eligible to be used may also be relevant, but I think it is an 
orthogonal issue.

For instance, if you are willing to use MSRP, but not audio or video, 
then your UA can reject offered audio and video media streams while 
accepting msrp streams. If rejecting audio and video means that no 
offered streams are being accepted, then the UA may indeed want to 
reject the call without alerting. But I don't think that is DND.

AFAIK DND only applies during session establishment. Willingness to use 
media streams applies to mid-call changes as well.

I think there is a very interesting subject of how media in a call are 
negotiated and changed. This is relatively clear from an O/A 
perspective, but when you start to consider user preferences at each end 
and how they can change mid call it gets much less clear. (E.g. when you 
have audio/video capable UAs you often will want to start with video 
disabled, and then maybe enable it later. But both ends need to enable 
before it can be used. Requires a little more that existing O/A to 
support a plausible UI.) While interesting, I think this probably falls 
into the category of "advanced" features.

	Paul

Miguel Garcia wrote:
> Correct, s/media stream/media type
> 
> /Miguel
> 
> bliss-bounces@ietf.org wrote:
>>    From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
>>
>>    DND can also apply to a particular media stream. In particular, if 
>> I am    in a meeting, I may want to activate the DND for audio and 
>> video, but    not for messaging.
>>
>> I assume you mean "media type" here -- a media stream is what is
>> established by an m line in the SDP of an existing dialog.  But you
>> couldn't apply DND to a media *stream* until it exists, and at that
>> point, it's too late.
>>
>> Dale
>>
>> _______________________________________________
>> BLISS mailing list
>> BLISS@ietf.org
>> https://www1.ietf.org/mailman/listinfo/bliss
>>
> 

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From bliss-bounces@ietf.org Thu Feb 01 14:43:48 2007
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Hi Dale;

I think we are on the same page.

As I mentioned in another e-mail, I see session based
messaging as just another media, and I don't think special
attention is needed for the session based messaging as it's
just another media, the service invocation would doubtfully
be different..

 What I was trying to state below is if there is a service
needed to be defined which is to be utilized only by
session based messaging(a specific media), I don't think
BLISS is the place to do it(chat based on session based
messaging for example).

Regards
Shida

Dale.Worley@comcast.net wrote:
>    From: Shida Schubert <shida@ntt-at.com>
>
>    But defining services solely used by session based messaging and
>    presence has a home called SIMPLE at this present moment, and I see
>    no reasons why BLISS should take on them as long as SIMPLE exists.
>
> I don't know about presence.  But within the SIP philosophy,
> session-based messaging is "just another media type" when compared to
> audio calling.  So it seems to me that if we discern an apparent
> difference in "calling features" between session-based messaging and
> audio calling, we are probably misunderstanding the situation.
>
> Dale
>
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
>
>
>   



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From bliss-bounces@ietf.org Fri Feb 02 03:34:18 2007
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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [BLISS] Welcome! Main question - what services?
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	"Drage, Keith (Keith)" <drage@alcatel-lucent.com>
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Miguel,

An entity providing a DND service should be able filter based on any number
of criteria. One criterion might be the set of media requested (e.g., DND to
any offer involving audio). This filtering doesn't need to be standardised,
unless some sort of reason (e.g., can't accept audio) needs to be returned
to the caller.

John 

> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
> Sent: 01 February 2007 15:13
> To: Drage, Keith (Keith)
> Cc: bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
> 
> Right... the capability of indicating a DND for SIP offers 
> that contain 
> certain media streams. Similarly with SIP methods.
> 
> /Miguel
> 
> Drage, Keith (Keith) wrote:
> > I assume that you mean you wish to be DND to a request with 
> a SIP offer
> > of a particular media stream, rather than just DND to a 
> particular media
> > stream.
> > 
> > Regards
> > 
> > Keith 
> > 
> >> -----Original Message-----
> >> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
> >> Sent: 01 February 2007 15:02
> >> To: Paul Kyzivat
> >> Cc: Drage, Keith (Keith); bliss@ietf.org
> >> Subject: Re: [BLISS] Welcome! Main question - what services?
> >>
> >> DND can also apply to a particular media stream. In 
> >> particular, if I am in a meeting, I may want to activate the 
> >> DND for audio and video, but not for messaging.
> >>
> >> I guess in general, you need to consider all the different 
> >> media streams that can apply to service. It would be a big 
> >> mistake just to restrict to audio or to "things that are 
> >> initiated with an INVITE" because this is what the PSTN did.
> >>
> >> /Miguel
> >>
> >> Paul Kyzivat wrote:
> >>> I have been inovlved in an exercise of trying to map 
> >> traditional PSTN 
> >>> services to SIP, and make them apply more generally to the 
> >> broader set 
> >>> of things that can happen in SIP.
> >>>
> >>> Some services seem to apply to an AOR, others to a UA, and some 
> >>> potentially to a particular media stream. And in some cases 
> >> there are 
> >>> multiple meaningful ways to apply the service. For instance:
> >>>
> >>> - DND can meaningfully apply at either the AOR level or the 
> >> UA level.
> >>>   You could consider these to be different services.
> >>>
> >>> - HOLD can apply to a whole dialog, or to a particular 
> media stream
> >>>   within a dialog.
> >>>
> >>> Its important to consider these things.
> >>>
> >>>     Paul
> >>>
> >>
> >> -- 
> >> Miguel A. Garcia           tel:+358-50-4804586
> >> sip:miguel.garcia@neonsite.net
> >> Nokia Research Center      Helsinki, Finland
> >>
> 
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.garcia@neonsite.net
> Nokia Research Center      Helsinki, Finland
> 
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

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Inline discussion.

Paul Kyzivat wrote:
> I don't think I buy this flavor of DND. I think it will cause interop 
> problems.

This is exactly what BLISS has to solve, avoid interop problems.

> 
> IMO the key aspect of DND is about alerting. Which media that are 
> eligible to be used may also be relevant, but I think it is an 
> orthogonal issue.

I agree that DND is about alerting. The conditions to indicate such 
alerting cases are driven by the user, and the user should be able to 
set such conditions based on whatever parameters he wants. Perhaps based 
on time (connected to calendar), based on present media types in the 
Offer as we discussed earlier, etc.

> 
> For instance, if you are willing to use MSRP, but not audio or video, 
> then your UA can reject offered audio and video media streams while 
> accepting msrp streams. If rejecting audio and video means that no 
> offered streams are being accepted, then the UA may indeed want to 
> reject the call without alerting. But I don't think that is DND.

I agree that rejecting media is not DND, and I agree that DND is about 
alerting. So I don't think we shouldn't be discussing the rejecting 
aspects, which are not DND. We should be discussing when the alert is 
delivered to the end user, and I am saying that this not a binary 
operation, but they are many inputs to that decision.

> 
> AFAIK DND only applies during session establishment. Willingness to use 
> media streams applies to mid-call changes as well.

Agreed too.

> 
> I think there is a very interesting subject of how media in a call are 
> negotiated and changed. This is relatively clear from an O/A 
> perspective, but when you start to consider user preferences at each end 
> and how they can change mid call it gets much less clear. (E.g. when you 
> have audio/video capable UAs you often will want to start with video 
> disabled, and then maybe enable it later. But both ends need to enable 
> before it can be used. Requires a little more that existing O/A to 
> support a plausible UI.) While interesting, I think this probably falls 
> into the category of "advanced" features.

Agreed too.

/Miguel

> 
>     Paul
> 
> Miguel Garcia wrote:
>> Correct, s/media stream/media type
>>
>> /Miguel
>>
>> bliss-bounces@ietf.org wrote:
>>>    From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
>>>
>>>    DND can also apply to a particular media stream. In particular, if 
>>> I am    in a meeting, I may want to activate the DND for audio and 
>>> video, but    not for messaging.
>>>
>>> I assume you mean "media type" here -- a media stream is what is
>>> established by an m line in the SDP of an existing dialog.  But you
>>> couldn't apply DND to a media *stream* until it exists, and at that
>>> point, it's too late.
>>>
>>> Dale
>>>
>>> _______________________________________________
>>> BLISS mailing list
>>> BLISS@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/bliss
>>>
>>

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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From bliss-bounces@ietf.org Fri Feb 02 08:03:36 2007
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From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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John:

We agree that one possible criterion to DND is the media type(s) 
requested in the offer. While this does not need to be standardized 
itself, perhaps we need to standardize the interface from the endpoint 
to the application server that executes the DND server. This depends 
basically on how the service is implemented, and in particular, on 
whether the service is network provided or not. But I think it is worth 
having a discussion on how the service can work in such circumstances.

/Miguel

Elwell, John wrote:
> Miguel,
> 
> An entity providing a DND service should be able filter based on any number
> of criteria. One criterion might be the set of media requested (e.g., DND to
> any offer involving audio). This filtering doesn't need to be standardised,
> unless some sort of reason (e.g., can't accept audio) needs to be returned
> to the caller.
> 
> John 
> 
>> -----Original Message-----
>> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
>> Sent: 01 February 2007 15:13
>> To: Drage, Keith (Keith)
>> Cc: bliss@ietf.org
>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>
>> Right... the capability of indicating a DND for SIP offers 
>> that contain 
>> certain media streams. Similarly with SIP methods.
>>
>> /Miguel
>>
>> Drage, Keith (Keith) wrote:
>>> I assume that you mean you wish to be DND to a request with 
>> a SIP offer
>>> of a particular media stream, rather than just DND to a 
>> particular media
>>> stream.
>>>
>>> Regards
>>>
>>> Keith 
>>>
>>>> -----Original Message-----
>>>> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
>>>> Sent: 01 February 2007 15:02
>>>> To: Paul Kyzivat
>>>> Cc: Drage, Keith (Keith); bliss@ietf.org
>>>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>>>
>>>> DND can also apply to a particular media stream. In 
>>>> particular, if I am in a meeting, I may want to activate the 
>>>> DND for audio and video, but not for messaging.
>>>>
>>>> I guess in general, you need to consider all the different 
>>>> media streams that can apply to service. It would be a big 
>>>> mistake just to restrict to audio or to "things that are 
>>>> initiated with an INVITE" because this is what the PSTN did.
>>>>
>>>> /Miguel
>>>>
>>>> Paul Kyzivat wrote:
>>>>> I have been inovlved in an exercise of trying to map 
>>>> traditional PSTN 
>>>>> services to SIP, and make them apply more generally to the 
>>>> broader set 
>>>>> of things that can happen in SIP.
>>>>>
>>>>> Some services seem to apply to an AOR, others to a UA, and some 
>>>>> potentially to a particular media stream. And in some cases 
>>>> there are 
>>>>> multiple meaningful ways to apply the service. For instance:
>>>>>
>>>>> - DND can meaningfully apply at either the AOR level or the 
>>>> UA level.
>>>>>   You could consider these to be different services.
>>>>>
>>>>> - HOLD can apply to a whole dialog, or to a particular 
>> media stream
>>>>>   within a dialog.
>>>>>
>>>>> Its important to consider these things.
>>>>>
>>>>>     Paul
>>>>>
>>>> -- 
>>>> Miguel A. Garcia           tel:+358-50-4804586
>>>> sip:miguel.garcia@neonsite.net
>>>> Nokia Research Center      Helsinki, Finland
>>>>
>> -- 
>> Miguel A. Garcia           tel:+358-50-4804586
>> sip:miguel.garcia@neonsite.net
>> Nokia Research Center      Helsinki, Finland
>>
>> _______________________________________________
>> BLISS mailing list
>> BLISS@ietf.org
>> https://www1.ietf.org/mailman/listinfo/bliss
>>

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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From bliss-bounces@ietf.org Fri Feb 02 10:56:33 2007
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   From: "Elwell, John" <john.elwell@siemens.com>

   An entity providing a DND service should be able filter based on
   any number of criteria. One criterion might be the set of media
   requested (e.g., DND to any offer involving audio).

I don't quite see what the dispute is here -- Certainly, it's possible
to put a UA into a state where it doesn't accept a media type that it
normally would.  Whether we want to call this a form of "Do Not
Disturb" seems to be entirely a matter of definition.  And since all
the machinery needed to implement it already exists, I see no need for
further standardization.

   This filtering doesn't need to be standardised, unless some sort of
   reason (e.g., can't accept audio) needs to be returned to the
   caller.

I'm not up on the offer/answer machinery, but isn't there a way to
indicate this, by declining every offered m-line in the INVITE's SDP
in a 200 response?  I know that there is a standard indication for
"all of the codecs/media you have offered are unacceptable to me".

My belief is that 488 is restricted to meaning "I can't process
application/sdp [or whatever content-type was in the request]", but I
might very well be wrong about that.

Dale

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From bliss-bounces@ietf.org Fri Feb 02 11:05:15 2007
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   From: Paul Kyzivat <pkyzivat@cisco.com>

   I think there is a very interesting subject of how media in a call are 
   negotiated and changed. This is relatively clear from an O/A 
   perspective, but when you start to consider user preferences at each end 
   and how they can change mid call it gets much less clear. (E.g. when you 
   have audio/video capable UAs you often will want to start with video 
   disabled, and then maybe enable it later. But both ends need to enable 
   before it can be used. Requires a little more that existing O/A to 
   support a plausible UI.) While interesting, I think this probably falls 
   into the category of "advanced" features.

I don't see this as troublesome.  Whenever either end changes its
"allow video" status, it initiates a re-INVITE cycle.  Video will only
beenabled when both ends have "allow video" set.

For example:


	A				B

	(no video)			(no video)

	INVITE (audio) -->
					<-- 200 (audio)

			only audio passes

	(user A enables video)

	reINVITE (audio, video) -->
					<-- 200 (audio)

			only audio passes

					(user B enables video)

					<-- reINVITE (audio, video)
	200 (audio, video) -->

			audio and video pass

	(user A disables video)

	reINVITE (audio) -->
					<-- 200 (audio)

			only audio passes

Dale

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Dale,

what you outline will work. But when A enabled video he presumably was 
interested in having it. But B (the human) doesn't know that. So it 
requires some other jog for B to enable video.

There has been some discussion off and on that when the reinvite comes 
with a video offer, B should alert its user and ask for confirmation. 
But I think that will cause problems. An alternative is for the device 
to note that it has been offered something that it could support but 
that is currently disabled, and to display an indicator to that effect. 
This gives the user the hint that he might want to enable his end.

I'm not suggesting that we want to standardize these kinds of UI 
behaviors. And I think we have all the signaling mechanisms needed to 
make this work. But I think we have the potential for poor interop 
unless some use cases are worked through, and some best practices 
identified. The most obvious problem is if some devices start alerting 
on reinvite with new media while others depend on reinvites completing 
immediately without alerting.

Another interesting case is whether a device should ever accept a call 
while rejecting all the media offered in the call. If it does, it might 
prevent the call from being answered by a different device that supports 
at least one of the offered media. But in some cases there might be 
value in answering. (E.g. if you have your audio+im phone set to im-only 
and a voice-only call comes in, maybe you want it to let you answer 
without the audio, offer IM back, and if that fails let you know so you 
can go someplace else and enable the audio.)

I expect there are many of these mixed mode use cases, intimately tied 
to the UI on the device. We need to "let a 1000 flowers bloom" here, but 
at the same time maximize the interoperability of them all. (I certainly 
don't want to see every one identified as a separate "service" that can 
only talk to another device that supports the same service.)

	Paul

Dale.Worley@comcast.net wrote:
>    From: Paul Kyzivat <pkyzivat@cisco.com>
> 
>    I think there is a very interesting subject of how media in a call are 
>    negotiated and changed. This is relatively clear from an O/A 
>    perspective, but when you start to consider user preferences at each end 
>    and how they can change mid call it gets much less clear. (E.g. when you 
>    have audio/video capable UAs you often will want to start with video 
>    disabled, and then maybe enable it later. But both ends need to enable 
>    before it can be used. Requires a little more that existing O/A to 
>    support a plausible UI.) While interesting, I think this probably falls 
>    into the category of "advanced" features.
> 
> I don't see this as troublesome.  Whenever either end changes its
> "allow video" status, it initiates a re-INVITE cycle.  Video will only
> beenabled when both ends have "allow video" set.
> 
> For example:
> 
> 
> 	A				B
> 
> 	(no video)			(no video)
> 
> 	INVITE (audio) -->
> 					<-- 200 (audio)
> 
> 			only audio passes
> 
> 	(user A enables video)
> 
> 	reINVITE (audio, video) -->
> 					<-- 200 (audio)
> 
> 			only audio passes
> 
> 					(user B enables video)
> 
> 					<-- reINVITE (audio, video)
> 	200 (audio, video) -->
> 
> 			audio and video pass
> 
> 	(user A disables video)
> 
> 	reINVITE (audio) -->
> 					<-- 200 (audio)
> 
> 			only audio passes
> 
> Dale
> 
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

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From bliss-bounces@ietf.org Fri Feb 02 16:08:13 2007
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Subject: Re: [BLISS] Welcome! Main question - what services?
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   From: Paul Kyzivat <pkyzivat@cisco.com>

   But I think we have the potential for poor interop unless some use
   cases are worked through, and some best practices identified.

Ah, yes, a subtle and interesting point -- we have standardized all
the mechanisms that we need, but unless we *restrict* them with good
BCPs, interoperability suffers.  I believe that I didn't notice this
because I tend to have strong opinions about how mechanisms should be
used, and didn't realize the interop problems if someone else does
things differently.

   Another interesting case is whether a device should ever accept a call 
   while rejecting all the media offered in the call. If it does, it might 
   prevent the call from being answered by a different device that supports 
   at least one of the offered media. But in some cases there might be 
   value in answering. (E.g. if you have your audio+im phone set to im-only 
   and a voice-only call comes in, maybe you want it to let you answer 
   without the audio, offer IM back, and if that fails let you know so you 
   can go someplace else and enable the audio.)

Ugh.  It seems like rejecting the call will almost always be best (but
that assumes that the caller proposes all media that he would be
willing to accept under any circumstances).

BTW, what error should a UA use in this case?  It seems that 488 is
correct, but the writeup in 3261 almost makes that sound improper.
(Or is this coverd in the new offer/answer I-D?)

   (I certainly don't want to see every one identified as a separate
   "service" that can only talk to another device that supports the
   same service.)

Yes, the automatic fallback to the
highest-shared-level-of-functionality is one of the strengths of the
Internet design philosophy.  And this is commercially important, too
-- I've seen presentations given to telecomm operators by SIP
equipment vendors, and a major selling point is "no profiles are
needed", that when you interconnect two vendors' SIP equipment, they
figure out what they have in common with no specific configuration.
It seems with traditional telecomm equipment, someone sits down and
figures out the lowest common denominator, and then configures
everything to know that they can only use those features.

Which is why the "service identifier" work seems strange to me -- it's
clear that the service identifiers are intended to be Requires
features, but in most cases that assumes that both endpoints have to
know that they both support the same feature before they can
interconnect at all.

Dale

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From bliss-bounces@ietf.org Fri Feb 02 17:55:53 2007
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Subject: RE: [BLISS] Welcome! Main question - what services?
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From: =?iso-8859-1?Q?G=F6ran_Eriksson_AP_=28KI/EAB=29?=
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To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi Paul,

   From: Paul Kyzivat <pkyzivat@cisco.com>

		I'm not suggesting that we want to standardize these kinds of UI =
behaviors.=20
		And I think we have all the signaling mechanisms needed to make this =
work.=20
		But I think we have the potential for poor interop unless some use =
cases are=20
		worked through, and some best practices identified.

You mean defining best practice in the UI design (human-device/system =
interaction) for a particular type of "service", right?

Assuming You have that in mind, You're saying don't standardise the UI =
behaviour but describe best practice. How does that solve
the problem (in any way different than standardisation)? Is it that we =
will only "standardise", with BCP, a part of the
service behaviour (as seen from the end user)?

Regards
G=F6ran

PS.=20
=20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: den 2 februari 2007 19:01
To: Dale.Worley@comcast.net
Cc: bliss@ietf.org
Subject: Re: [BLISS] Welcome! Main question - what services?

Dale,

what you outline will work. But when A enabled video he presumably was =
interested in having it. But B (the human) doesn't know that. So it =
requires some other jog for B to enable video.

There has been some discussion off and on that when the reinvite comes =
with a video offer, B should alert its user and ask for confirmation.=20
But I think that will cause problems. An alternative is for the device =
to note that it has been offered something that it could support but =
that is currently disabled, and to display an indicator to that effect.=20
This gives the user the hint that he might want to enable his end.

I'm not suggesting that we want to standardize these kinds of UI =
behaviors. And I think we have all the signaling mechanisms needed to =
make this work. But I think we have the potential for poor interop =
unless some use cases are worked through, and some best practices =
identified. The most obvious problem is if some devices start alerting =
on reinvite with new media while others depend on reinvites completing =
immediately without alerting.

Another interesting case is whether a device should ever accept a call =
while rejecting all the media offered in the call. If it does, it might =
prevent the call from being answered by a different device that supports =
at least one of the offered media. But in some cases there might be =
value in answering. (E.g. if you have your audio+im phone set to im-only =
and a voice-only call comes in, maybe you want it to let you answer =
without the audio, offer IM back, and if that fails let you know so you =
can go someplace else and enable the audio.)

I expect there are many of these mixed mode use cases, intimately tied =
to the UI on the device. We need to "let a 1000 flowers bloom" here, but =
at the same time maximize the interoperability of them all. (I certainly =
don't want to see every one identified as a separate "service" that can =
only talk to another device that supports the same service.)

	Paul

Dale.Worley@comcast.net wrote:
>    From: Paul Kyzivat <pkyzivat@cisco.com>
>=20
>    I think there is a very interesting subject of how media in a call =
are=20
>    negotiated and changed. This is relatively clear from an O/A=20
>    perspective, but when you start to consider user preferences at =
each end=20
>    and how they can change mid call it gets much less clear. (E.g. =
when you=20
>    have audio/video capable UAs you often will want to start with =
video=20
>    disabled, and then maybe enable it later. But both ends need to =
enable=20
>    before it can be used. Requires a little more that existing O/A to=20
>    support a plausible UI.) While interesting, I think this probably =
falls=20
>    into the category of "advanced" features.
>=20
> I don't see this as troublesome.  Whenever either end changes its=20
> "allow video" status, it initiates a re-INVITE cycle.  Video will only =

> beenabled when both ends have "allow video" set.
>=20
> For example:
>=20
>=20
> 	A				B
>=20
> 	(no video)			(no video)
>=20
> 	INVITE (audio) -->
> 					<-- 200 (audio)
>=20
> 			only audio passes
>=20
> 	(user A enables video)
>=20
> 	reINVITE (audio, video) -->
> 					<-- 200 (audio)
>=20
> 			only audio passes
>=20
> 					(user B enables video)
>=20
> 					<-- reINVITE (audio, video)
> 	200 (audio, video) -->
>=20
> 			audio and video pass
>=20
> 	(user A disables video)
>=20
> 	reINVITE (audio) -->
> 					<-- 200 (audio)
>=20
> 			only audio passes
>=20
> Dale
>=20
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
>=20

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

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Göran Eriksson AP (KI/EAB) wrote:
> Hi Paul,
> 
>    From: Paul Kyzivat <pkyzivat@cisco.com>
> 
> 		I'm not suggesting that we want to standardize these kinds of UI behaviors. 
> 		And I think we have all the signaling mechanisms needed to make this work. 
> 		But I think we have the potential for poor interop unless some use cases are 
> 		worked through, and some best practices identified.
> 
> You mean defining best practice in the UI design (human-device/system interaction) for a particular type of "service", right?

No. I mean defining best practice in sip (and sdp o/a) usage so that the 
various scenarios that people come up with can work and interoperate 
reasonably when the two ends don't have the same UI.

> Assuming You have that in mind, You're saying don't standardise the UI behaviour but describe best practice. How does that solve
> the problem (in any way different than standardisation)? Is it that we will only "standardise", with BCP, a part of the
> service behaviour (as seen from the end user)?
> 
> Regards
> Göran
> 
> PS. 
>  
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: den 2 februari 2007 19:01
> To: Dale.Worley@comcast.net
> Cc: bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
> 
> Dale,
> 
> what you outline will work. But when A enabled video he presumably was interested in having it. But B (the human) doesn't know that. So it requires some other jog for B to enable video.
> 
> There has been some discussion off and on that when the reinvite comes with a video offer, B should alert its user and ask for confirmation. 
> But I think that will cause problems. An alternative is for the device to note that it has been offered something that it could support but that is currently disabled, and to display an indicator to that effect. 
> This gives the user the hint that he might want to enable his end.
> 
> I'm not suggesting that we want to standardize these kinds of UI behaviors. And I think we have all the signaling mechanisms needed to make this work. But I think we have the potential for poor interop unless some use cases are worked through, and some best practices identified. The most obvious problem is if some devices start alerting on reinvite with new media while others depend on reinvites completing immediately without alerting.
> 
> Another interesting case is whether a device should ever accept a call while rejecting all the media offered in the call. If it does, it might prevent the call from being answered by a different device that supports at least one of the offered media. But in some cases there might be value in answering. (E.g. if you have your audio+im phone set to im-only and a voice-only call comes in, maybe you want it to let you answer without the audio, offer IM back, and if that fails let you know so you can go someplace else and enable the audio.)
> 
> I expect there are many of these mixed mode use cases, intimately tied to the UI on the device. We need to "let a 1000 flowers bloom" here, but at the same time maximize the interoperability of them all. (I certainly don't want to see every one identified as a separate "service" that can only talk to another device that supports the same service.)
> 
> 	Paul
> 
> Dale.Worley@comcast.net wrote:
>>    From: Paul Kyzivat <pkyzivat@cisco.com>
>>
>>    I think there is a very interesting subject of how media in a call are 
>>    negotiated and changed. This is relatively clear from an O/A 
>>    perspective, but when you start to consider user preferences at each end 
>>    and how they can change mid call it gets much less clear. (E.g. when you 
>>    have audio/video capable UAs you often will want to start with video 
>>    disabled, and then maybe enable it later. But both ends need to enable 
>>    before it can be used. Requires a little more that existing O/A to 
>>    support a plausible UI.) While interesting, I think this probably falls 
>>    into the category of "advanced" features.
>>
>> I don't see this as troublesome.  Whenever either end changes its 
>> "allow video" status, it initiates a re-INVITE cycle.  Video will only 
>> beenabled when both ends have "allow video" set.
>>
>> For example:
>>
>>
>> 	A				B
>>
>> 	(no video)			(no video)
>>
>> 	INVITE (audio) -->
>> 					<-- 200 (audio)
>>
>> 			only audio passes
>>
>> 	(user A enables video)
>>
>> 	reINVITE (audio, video) -->
>> 					<-- 200 (audio)
>>
>> 			only audio passes
>>
>> 					(user B enables video)
>>
>> 					<-- reINVITE (audio, video)
>> 	200 (audio, video) -->
>>
>> 			audio and video pass
>>
>> 	(user A disables video)
>>
>> 	reINVITE (audio) -->
>> 					<-- 200 (audio)
>>
>> 			only audio passes
>>
>> Dale
>>
>> _______________________________________________
>> BLISS mailing list
>> BLISS@ietf.org
>> https://www1.ietf.org/mailman/listinfo/bliss
>>
> 
> _______________________________________________
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> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

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From bliss-bounces@ietf.org Fri Feb 02 18:26:36 2007
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Subject: RE: [BLISS] Welcome! Main question - what services?
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From: =?iso-8859-1?Q?G=F6ran_Eriksson_AP_=28KI/EAB=29?=
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	interoperate reasonably=20

Sorry for being picky or perhaps just a bit slow this late in the =
evening in the dark Nordic night, but interoperate from which =
perspective?

OK, not the way interaction design details like buttons color, etc, but =
at least interaction design in terms
of what the user expects from the "system", read device, and how he/she =
interacts with it to achieve some result, right?

Let's not call it UI design but it is then getting us into defining the =
user-system interaction, or rather describing
a best practice for it?

BR
G=F6ran

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: den 3 februari 2007 00:20
To: G=F6ran Eriksson AP (KI/EAB)
Cc: bliss@ietf.org; Dale.Worley@comcast.net
Subject: Re: [BLISS] Welcome! Main question - what services?



G=F6ran Eriksson AP (KI/EAB) wrote:
> Hi Paul,
>=20
>    From: Paul Kyzivat <pkyzivat@cisco.com>
>=20
> 		I'm not suggesting that we want to standardize these kinds of UI =
behaviors.=20
> 		And I think we have all the signaling mechanisms needed to make this =
work.=20
> 		But I think we have the potential for poor interop unless some use =
cases are=20
> 		worked through, and some best practices identified.
>=20
> You mean defining best practice in the UI design (human-device/system =
interaction) for a particular type of "service", right?

No. I mean defining best practice in sip (and sdp o/a) usage so that the =
various scenarios that people come up with can work and interoperate =
reasonably when the two ends don't have the same UI.

> Assuming You have that in mind, You're saying don't standardise the UI =

> behaviour but describe best practice. How does that solve the problem=20
> (in any way different than standardisation)? Is it that we will only =
"standardise", with BCP, a part of the service behaviour (as seen from =
the end user)?
>=20
> Regards
> G=F6ran
>=20
> PS.=20
> =20
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: den 2 februari 2007 19:01
> To: Dale.Worley@comcast.net
> Cc: bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
>=20
> Dale,
>=20
> what you outline will work. But when A enabled video he presumably was =
interested in having it. But B (the human) doesn't know that. So it =
requires some other jog for B to enable video.
>=20
> There has been some discussion off and on that when the reinvite comes =
with a video offer, B should alert its user and ask for confirmation.=20
> But I think that will cause problems. An alternative is for the device =
to note that it has been offered something that it could support but =
that is currently disabled, and to display an indicator to that effect.=20
> This gives the user the hint that he might want to enable his end.
>=20
> I'm not suggesting that we want to standardize these kinds of UI =
behaviors. And I think we have all the signaling mechanisms needed to =
make this work. But I think we have the potential for poor interop =
unless some use cases are worked through, and some best practices =
identified. The most obvious problem is if some devices start alerting =
on reinvite with new media while others depend on reinvites completing =
immediately without alerting.
>=20
> Another interesting case is whether a device should ever accept a call =

> while rejecting all the media offered in the call. If it does, it=20
> might prevent the call from being answered by a different device that=20
> supports at least one of the offered media. But in some cases there=20
> might be value in answering. (E.g. if you have your audio+im phone set =

> to im-only and a voice-only call comes in, maybe you want it to let=20
> you answer without the audio, offer IM back, and if that fails let you =

> know so you can go someplace else and enable the audio.)
>=20
> I expect there are many of these mixed mode use cases, intimately tied =

> to the UI on the device. We need to "let a 1000 flowers bloom" here,=20
> but at the same time maximize the interoperability of them all. (I=20
> certainly don't want to see every one identified as a separate=20
> "service" that can only talk to another device that supports the same=20
> service.)
>=20
> 	Paul
>=20
> Dale.Worley@comcast.net wrote:
>>    From: Paul Kyzivat <pkyzivat@cisco.com>
>>
>>    I think there is a very interesting subject of how media in a call =
are=20
>>    negotiated and changed. This is relatively clear from an O/A=20
>>    perspective, but when you start to consider user preferences at =
each end=20
>>    and how they can change mid call it gets much less clear. (E.g. =
when you=20
>>    have audio/video capable UAs you often will want to start with =
video=20
>>    disabled, and then maybe enable it later. But both ends need to =
enable=20
>>    before it can be used. Requires a little more that existing O/A to =

>>    support a plausible UI.) While interesting, I think this probably =
falls=20
>>    into the category of "advanced" features.
>>
>> I don't see this as troublesome.  Whenever either end changes its=20
>> "allow video" status, it initiates a re-INVITE cycle.  Video will=20
>> only beenabled when both ends have "allow video" set.
>>
>> For example:
>>
>>
>> 	A				B
>>
>> 	(no video)			(no video)
>>
>> 	INVITE (audio) -->
>> 					<-- 200 (audio)
>>
>> 			only audio passes
>>
>> 	(user A enables video)
>>
>> 	reINVITE (audio, video) -->
>> 					<-- 200 (audio)
>>
>> 			only audio passes
>>
>> 					(user B enables video)
>>
>> 					<-- reINVITE (audio, video)
>> 	200 (audio, video) -->
>>
>> 			audio and video pass
>>
>> 	(user A disables video)
>>
>> 	reINVITE (audio) -->
>> 					<-- 200 (audio)
>>
>> 			only audio passes
>>
>> Dale
>>
>> _______________________________________________
>> BLISS mailing list
>> BLISS@ietf.org
>> https://www1.ietf.org/mailman/listinfo/bliss
>>
>=20
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
>=20

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This BOF is approved.

Cullen <with my AD hat on>



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From bliss-bounces@ietf.org Sun Feb 04 11:38:21 2007
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Hi;
Can we split the discussion to several threads ?
We are talking about different issues but all of them under the same 
Subject.
It is very difficult to track posts  and response when everything 
appears in the same thread.

Thanks
Diego B


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Excellent! Thank you Cullen.

-Jonathan R.

Cullen Jennings wrote:

> 
> This BOF is approved.
> 
> Cullen <with my AD hat on>
> 
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Subject: Re: [BLISS] Welcome! Main question - what services?
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There is an issue here, and it has to do with 'invocation' - how does 
the user signal that they'd like this desired behavior?

If DND is being done in the network, and we want a user on a phone to be 
able to hit the 'DND for Video' button, then we do need a standardized 
way of communicating from the phone to the network element providing the 
feature (say, its proxy or IP PBX or CSCF or whatever) that this is the 
desired intent.

We have this kind of problem even if the phone implements the DND 
processing. In every real system I've seen, there are always multiple 
ways of setting this DND status - from a user profile web page, from 
phone keys, from an admin web page, and so on. So even if the processing 
is implemented in the phone, we'd need a way to communicate the desired 
criteria from the web server/provisioning system or whatever, to the 
phone. Indeed, its exactly the same information we'd need to communicate 
from the phone to the 'network' when the actual call processing is 
implemented in the network.

This (fancy conditions for defining when to block a call) may be an area 
that we decide to leave alone; that is, BLISS is about *basic* 
interoperability. We need to define the basic framework needed for this 
to interoperate, and make sure we leave the hooks in place for vendors 
to do special stuff when its their phones ontop of their servers, such 
as a fancier DND service.

In the particular case of DND for example, let us say we specify that 
DND call processing is done in the network, and that the service is 
invoked by including a body in a REGISTER message that indicates the 
criteria and desired action (just for arguments sake). We could define a 
basic format for this object which defines no conditions (i.e., DND is 
either on or off) and defines three actions (reject, send-to-voicemail, 
ring-indefinitely). We do a nice job in the XML and leave it extensible 
for other conditions and actions. We define XML namespace techniques for 
adding new conditions and actions. We support negotiation of supported 
namespaces during registration time, so the phone knows whether the 
server supports the extensions it knows about. Then, it can gray out 
those buttons on its UI, or not show the menu options for them, etc, 
when the phone finds its connected to a server that only supports the 
baseline format.

-Jonathan R.


Elwell, John wrote:
> Miguel,
> 
> An entity providing a DND service should be able filter based on any number
> of criteria. One criterion might be the set of media requested (e.g., DND to
> any offer involving audio). This filtering doesn't need to be standardised,
> unless some sort of reason (e.g., can't accept audio) needs to be returned
> to the caller.
> 
> John 
> 
> 
>>-----Original Message-----
>>From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
>>Sent: 01 February 2007 15:13
>>To: Drage, Keith (Keith)
>>Cc: bliss@ietf.org
>>Subject: Re: [BLISS] Welcome! Main question - what services?
>>
>>Right... the capability of indicating a DND for SIP offers 
>>that contain 
>>certain media streams. Similarly with SIP methods.
>>
>>/Miguel
>>
>>Drage, Keith (Keith) wrote:
>>
>>>I assume that you mean you wish to be DND to a request with 
>>
>>a SIP offer
>>
>>>of a particular media stream, rather than just DND to a 
>>
>>particular media
>>
>>>stream.
>>>
>>>Regards
>>>
>>>Keith 
>>>
>>>
>>>>-----Original Message-----
>>>>From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
>>>>Sent: 01 February 2007 15:02
>>>>To: Paul Kyzivat
>>>>Cc: Drage, Keith (Keith); bliss@ietf.org
>>>>Subject: Re: [BLISS] Welcome! Main question - what services?
>>>>
>>>>DND can also apply to a particular media stream. In 
>>>>particular, if I am in a meeting, I may want to activate the 
>>>>DND for audio and video, but not for messaging.
>>>>
>>>>I guess in general, you need to consider all the different 
>>>>media streams that can apply to service. It would be a big 
>>>>mistake just to restrict to audio or to "things that are 
>>>>initiated with an INVITE" because this is what the PSTN did.
>>>>
>>>>/Miguel
>>>>
>>>>Paul Kyzivat wrote:
>>>>
>>>>>I have been inovlved in an exercise of trying to map 
>>>>
>>>>traditional PSTN 
>>>>
>>>>>services to SIP, and make them apply more generally to the 
>>>>
>>>>broader set 
>>>>
>>>>>of things that can happen in SIP.
>>>>>
>>>>>Some services seem to apply to an AOR, others to a UA, and some 
>>>>>potentially to a particular media stream. And in some cases 
>>>>
>>>>there are 
>>>>
>>>>>multiple meaningful ways to apply the service. For instance:
>>>>>
>>>>>- DND can meaningfully apply at either the AOR level or the 
>>>>
>>>>UA level.
>>>>
>>>>>  You could consider these to be different services.
>>>>>
>>>>>- HOLD can apply to a whole dialog, or to a particular 
>>
>>media stream
>>
>>>>>  within a dialog.
>>>>>
>>>>>Its important to consider these things.
>>>>>
>>>>>    Paul
>>>>>
>>>>
>>>>-- 
>>>>Miguel A. Garcia           tel:+358-50-4804586
>>>>sip:miguel.garcia@neonsite.net
>>>>Nokia Research Center      Helsinki, Finland
>>>>
>>
>>-- 
>>Miguel A. Garcia           tel:+358-50-4804586
>>sip:miguel.garcia@neonsite.net
>>Nokia Research Center      Helsinki, Finland
>>
>>_______________________________________________
>>BLISS mailing list
>>BLISS@ietf.org
>>https://www1.ietf.org/mailman/listinfo/bliss
>>
> 
> 
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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inline.

Dale.Worley@comcast.net wrote:
>    From: Jonathan Rosenberg <jdrosen@cisco.com>
> 
>    I don't want to have a spec that rigorously defines all 8 of the above, 
>    with specific call flows for each and a name for each. I want vendors to 
>    be able to implement any one of these 8 and for it to work. Indeed, any 
>    of the above 8 would work as long as we define how the invocation works, 
>    since that IMHO is the primary interop problem around this feature. If 
>    one phone expects the network to do the DND processing, and the 
>    corresponding IP PBX expects the phone to do it, we have a serious 
>    interop failure.
> 
> Yes, I think that at this point in the discussion, we want to avoid
> trying to be too exact about what "services" are.  Really, exploring
> the meaning of any given named "service" is part of our charter.

Yes - with the caveat as I've stated that the "B" in BLISS is important. 
I don't think its within our scope to spec out how to do the most 
uber-complicated variant on DND we are able to think of. I think we need 
to make the basic stuff work and leave hooks that allow extensibility 
and vendor proprietary tricks, with all of the capabilitiy negotiation 
and discovery necessary to make that work.

> Ideally, the endpoint will be a clear understanding of what services
> can be provided, and what the building blocks of the services are, and
> how those building blocks interact to provide services, and
> consequently, what users need to do to obtain a service.
> 
> As an example, the sipX call pick-up operation involves 3 phones:
> 
>     A phone A makes a call to a phone B.  From a phone C, intercept the
>     ringing call and talk to A.
> 
> To do this requires:
> 
>     phone A implements INVITE-with-Replaces,
> 
>     phone B implements the dialog event package, and
> 
>     either:
> 
> 	phone C implements the pick-up operation natively, or
> 
> 	the proxy implements a "call pick-up agent" (like sipX does) to
> 	perform the operation

Right. So here are some other variations that you might see, which can 
illustrate the interop failures we might see:

1. instead of having phone B implement the dialog package, phone B uses 
PUBLISH to push its dialog state into the network, so that the SUBSCRIBE 
terminates on a state server in the network.

2. instead of that, in another implementation, phone B does nothing 
since it assumes that its IP-PBX/proxy/call agent is watching the SIP 
messages and knows the dialog state by inspection.

3. instead of PUBLISH with the dialog event package, the phone publishes 
presence status which includes a service describing its call status.

If you add to that the two variants that Dale describes for phone C, we 
have a total of 8 different combinations of how this can be done, and 
most will not interoperate with each other.

So what I'd expect in the 'park and pickup' document is thta it would 
list exactly these different cases, showing some call flows on how it 
can fail, and then make a concrete recommendation. For example, we could 
say something like:

1. All user agents MUST support the receiving subscriptions to the 
dialog event package
2. All user agents MUST be capable of invoking pickup by sending an 
INVITE with Replaces, and all user agents MUST be capable of subscrining 
to the dialog package for their own AOR.

Whats interesting is that this approach still allows for flexibility on 
what you do both in the phone and in the network. For example, vendor X 
decides to still use the publish mechanism with an event server in the 
IP-PBX.  So, its phones do PUBLISH and its server handles the 
subscriptions. However, to be be compliant to our spec, the phone still 
supports receipt of SUBSCRIBE. Now, when vendor X's phones and IP-PBX 
are paired up, the phone would never even receive the SUBSCRIBE request 
and it would be processed in the network. However, if vendor X's phone 
is paired with IP-PBX from vendor Y, and vendor Y assumed the endpoint 
handled subscriptions, things would still work. Vendor Y's IP PBX would 
reject the PUBLISH operations, and would proxy dialog subscriptions to 
the phone. The phone would be able to process them since its compliant 
to our spec, and things still work.


> 
> But it was the result of a lot of analysis to discern that that is
> what is needed for call pick-up.  And that has left out a number of
> interesting questions in regard to operation across SIP administrative
> domains.

Right. I think we want to look at that kind of stuff too.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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Yes, lets add SIMPLE to the list.

Does that address your concern Keith?

-Jonathan R.

Shida Schubert wrote:
> Hi Keith;
> 
> That is the intention.
> 
> I think the following text in the charter under
> "guiding principles for this working group will include" covers your
> comments.
> 
> " Where extensions to standards are required, bring the analysis and need
> for new extensions to the appropriate working group (SIP or SIPPING). "
> 
> I guess SIMPLE is missing from the list.
> 
> 
> Drage, Keith (Keith) wrote:
> 
>>I don't think this statement helps.
>>
>>What should happen is that if BLISS identifies the need for any of your
>>list, it develops the requirements and passes the requirements to each
>>of those groups you have mentioned.
>>
>>Keith 
>>
>>  
>>
>>>-----Original Message-----
>>>From: Shida Schubert [mailto:shida@ntt-at.com] 
>>>Sent: 31 January 2007 21:34
>>>To: Spencer Dawkins
>>>Cc: bliss@ietf.org
>>>Subject: Re: Fw: [BLISS] Welcome! Main question - what services?
>>>
>>>
>>>Hi Spencer;
>>>
>>>Yes you are right. I will change the text to say..
>>>
>>>"SIP applications such as configuration, presence and IM, and 
>>>session policy are clearly out of scope for BLISS and remain 
>>>the sole province of SIPPING and SIMPLE."
>>>
>>>Thanks
>>>Shida
>>>
>>>
>>>Spencer Dawkins wrote:
>>>    
>>>
>>>>Hi, Jonathan and Shida,
>>>>
>>>>And now that Peili jogged me to read the actual words in 
>>>>      
>>>
>>>the charter, 
>>>    
>>>
>>>>isn't presence the sole province of SIMPLE?
>>>>
>>>>Thanks,
>>>>
>>>>Spencer
>>>>
>>>>
>>>>      
>>>>
>>>>>After looking at the following sentences in the draft charger "The 
>>>>>difference between SIPPING and BLISS, is that BLISS is 
>>>>>        
>>>
>>>focused on a 
>>>    
>>>
>>>>>particular type of SIP application - call features, and in 
>>>>>particular, advanced call features requiring non-trivial call 
>>>>>control. SIP applications such as configuration, presence 
>>>>>        
>>>
>>>and IM, and 
>>>    
>>>
>>>>>session policy are clearly out of scope for BLISS and 
>>>>>        
>>>
>>>remain the sole 
>>>    
>>>
>>>>>province of SIPPING. "
>>>>>
>>>>>It seems the WG will actually focus on features initiated 
>>>>>        
>>>
>>>by INVITE 
>>>    
>>>
>>>>>and these features are more or less related to traditional 
>>>>>        
>>>
>>>PSTN/ISDN 
>>>    
>>>
>>>>>supplementary services.
>>>>>If I am correct, what about session mode messaging service 
>>>>>        
>>>
>>>which is 
>>>    
>>>
>>>>>also a call initiated by INVITE?
>>>>>
>>>>>thanks
>>>>>Peili
>>>>>        
>>>>
>>>>_______________________________________________
>>>>BLISS mailing list
>>>>BLISS@ietf.org
>>>>https://www1.ietf.org/mailman/listinfo/bliss
>>>>
>>>>
>>>>      
>>>
>>>_______________________________________________
>>>BLISS mailing list
>>>BLISS@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/bliss
>>>
>>>    
>>
>>
>>  
> 
> 
> 
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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To summarize this discussion, I think its fair to say that *anything* 
that BLISS does will consider multimedia (audio, video, messaging, and 
of course - text). I also think we should add that explicitly to the 
charter to be clear. As Paul and others have said, in some cases the 
flows may really not care at all what the media types are; for others it 
may matter, or we at least need to think about whether they matter.

-Jonathan R.

Dale.Worley@comcast.net wrote:

>    From: Shida Schubert <shida@ntt-at.com>
> 
>    But defining services solely used by session based messaging and
>    presence has a home called SIMPLE at this present moment, and I see
>    no reasons why BLISS should take on them as long as SIMPLE exists.
> 
> I don't know about presence.  But within the SIP philosophy,
> session-based messaging is "just another media type" when compared to
> audio calling.  So it seems to me that if we discern an apparent
> difference in "calling features" between session-based messaging and
> audio calling, we are probably misunderstanding the situation.
> 
> Dale
> 
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>, <shida@ntt-at.com>
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Yes that would solve it.=20

It may also be appropriate to mention RFC 3427 somewhere around here in
regard to the SIP extension.

Keith=20

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: 16 February 2007 21:30
> To: shida@ntt-at.com
> Cc: Drage, Keith (Keith); bliss@ietf.org
> Subject: Re: Fw: [BLISS] Welcome! Main question - what services?
>=20
> Yes, lets add SIMPLE to the list.
>=20
> Does that address your concern Keith?
>=20
> -Jonathan R.
>=20
> Shida Schubert wrote:
> > Hi Keith;
> >=20
> > That is the intention.
> >=20
> > I think the following text in the charter under "guiding principles=20
> > for this working group will include" covers your comments.
> >=20
> > " Where extensions to standards are required, bring the=20
> analysis and=20
> > need for new extensions to the appropriate working group=20
> (SIP or SIPPING). "
> >=20
> > I guess SIMPLE is missing from the list.
> >=20
> >=20
> > Drage, Keith (Keith) wrote:
> >=20
> >>I don't think this statement helps.
> >>
> >>What should happen is that if BLISS identifies the need for any of=20
> >>your list, it develops the requirements and passes the=20
> requirements to=20
> >>each of those groups you have mentioned.
> >>
> >>Keith
> >>
> >> =20
> >>
> >>>-----Original Message-----
> >>>From: Shida Schubert [mailto:shida@ntt-at.com]
> >>>Sent: 31 January 2007 21:34
> >>>To: Spencer Dawkins
> >>>Cc: bliss@ietf.org
> >>>Subject: Re: Fw: [BLISS] Welcome! Main question - what services?
> >>>
> >>>
> >>>Hi Spencer;
> >>>
> >>>Yes you are right. I will change the text to say..
> >>>
> >>>"SIP applications such as configuration, presence and IM,=20
> and session=20
> >>>policy are clearly out of scope for BLISS and remain the sole=20
> >>>province of SIPPING and SIMPLE."
> >>>
> >>>Thanks
> >>>Shida
> >>>
> >>>
> >>>Spencer Dawkins wrote:
> >>>   =20
> >>>
> >>>>Hi, Jonathan and Shida,
> >>>>
> >>>>And now that Peili jogged me to read the actual words in
> >>>>     =20
> >>>
> >>>the charter,
> >>>   =20
> >>>
> >>>>isn't presence the sole province of SIMPLE?
> >>>>
> >>>>Thanks,
> >>>>
> >>>>Spencer
> >>>>
> >>>>
> >>>>     =20
> >>>>
> >>>>>After looking at the following sentences in the draft=20
> charger "The=20
> >>>>>difference between SIPPING and BLISS, is that BLISS is
> >>>>>       =20
> >>>
> >>>focused on a
> >>>   =20
> >>>
> >>>>>particular type of SIP application - call features, and in=20
> >>>>>particular, advanced call features requiring non-trivial call=20
> >>>>>control. SIP applications such as configuration, presence
> >>>>>       =20
> >>>
> >>>and IM, and
> >>>   =20
> >>>
> >>>>>session policy are clearly out of scope for BLISS and
> >>>>>       =20
> >>>
> >>>remain the sole
> >>>   =20
> >>>
> >>>>>province of SIPPING. "
> >>>>>
> >>>>>It seems the WG will actually focus on features initiated
> >>>>>       =20
> >>>
> >>>by INVITE
> >>>   =20
> >>>
> >>>>>and these features are more or less related to traditional
> >>>>>       =20
> >>>
> >>>PSTN/ISDN
> >>>   =20
> >>>
> >>>>>supplementary services.
> >>>>>If I am correct, what about session mode messaging service
> >>>>>       =20
> >>>
> >>>which is
> >>>   =20
> >>>
> >>>>>also a call initiated by INVITE?
> >>>>>
> >>>>>thanks
> >>>>>Peili
> >>>>>       =20
> >>>>
> >>>>_______________________________________________
> >>>>BLISS mailing list
> >>>>BLISS@ietf.org
> >>>>https://www1.ietf.org/mailman/listinfo/bliss
> >>>>
> >>>>
> >>>>     =20
> >>>
> >>>_______________________________________________
> >>>BLISS mailing list
> >>>BLISS@ietf.org
> >>>https://www1.ietf.org/mailman/listinfo/bliss
> >>>
> >>>   =20
> >>
> >>
> >> =20
> >=20
> >=20
> >=20
> > _______________________________________________
> > BLISS mailing list
> > BLISS@ietf.org
> > https://www1.ietf.org/mailman/listinfo/bliss
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20

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From bliss-bounces@ietf.org Mon Feb 19 02:22:40 2007
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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [BLISS] Welcome! Main question - what services?
To: Jonathan Rosenberg <jdrosen@cisco.com>
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Jonathan,

I agree with your point about defining a basic but extensible mechanism.
What is not clear is whether mechanisms for "invocation" of features like
DND should be within scope or not. On the one hand, these can be generally
done by web pages and other non-SIP means. The question is whether we need
to define a SIP means too or simply leave it out of scope.

John

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
> Sent: 16 February 2007 21:40
> To: Elwell, John
> Cc: Miguel Garcia; Drage, Keith (Keith); bliss@ietf.org
> Subject: Re: [BLISS] Welcome! Main question - what services?
> 
> There is an issue here, and it has to do with 'invocation' - how does 
> the user signal that they'd like this desired behavior?
> 
> If DND is being done in the network, and we want a user on a 
> phone to be 
> able to hit the 'DND for Video' button, then we do need a 
> standardized 
> way of communicating from the phone to the network element 
> providing the 
> feature (say, its proxy or IP PBX or CSCF or whatever) that 
> this is the 
> desired intent.
> 
> We have this kind of problem even if the phone implements the DND 
> processing. In every real system I've seen, there are always multiple 
> ways of setting this DND status - from a user profile web page, from 
> phone keys, from an admin web page, and so on. So even if the 
> processing 
> is implemented in the phone, we'd need a way to communicate 
> the desired 
> criteria from the web server/provisioning system or whatever, to the 
> phone. Indeed, its exactly the same information we'd need to 
> communicate 
> from the phone to the 'network' when the actual call processing is 
> implemented in the network.
> 
> This (fancy conditions for defining when to block a call) may 
> be an area 
> that we decide to leave alone; that is, BLISS is about *basic* 
> interoperability. We need to define the basic framework 
> needed for this 
> to interoperate, and make sure we leave the hooks in place 
> for vendors 
> to do special stuff when its their phones ontop of their 
> servers, such 
> as a fancier DND service.
> 
> In the particular case of DND for example, let us say we specify that 
> DND call processing is done in the network, and that the service is 
> invoked by including a body in a REGISTER message that indicates the 
> criteria and desired action (just for arguments sake). We 
> could define a 
> basic format for this object which defines no conditions 
> (i.e., DND is 
> either on or off) and defines three actions (reject, 
> send-to-voicemail, 
> ring-indefinitely). We do a nice job in the XML and leave it 
> extensible 
> for other conditions and actions. We define XML namespace 
> techniques for 
> adding new conditions and actions. We support negotiation of 
> supported 
> namespaces during registration time, so the phone knows whether the 
> server supports the extensions it knows about. Then, it can gray out 
> those buttons on its UI, or not show the menu options for them, etc, 
> when the phone finds its connected to a server that only supports the 
> baseline format.
> 
> -Jonathan R.
> 

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   In the particular case of DND for example, let us say we specify that 
   DND call processing is done in the network, and that the service is 
   invoked by including a body in a REGISTER message that indicates the 
   criteria and desired action (just for arguments sake).

Um, if I might be so bold to ask, *why* would be want to have DND call
processing done in the network?  It's true that if we want to have DND
done in the network, we need to define some moderately complex
mechanism for having the UA report to the network its DND status.  But
that's really an argument that DND should be done in the phone, and
activated by a button (like it is on every SIP phone).

If we're really going to try to define such a mechanism, we first need
to review that there really is such a requirement, that such an
architecture is really the best solution to the DND "problem" in a
reasonable fraction of real-world cases.  That this is a significant
deviation from the end-to-end principle is a warning sign...

Dale

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   From: Jonathan Rosenberg <jdrosen@cisco.com>

   As Paul and others have said, in some cases the flows may really
   not care at all what the media types are; for others it may matter,
   or we at least need to think about whether they matter.

If the media matters, it's likely we've made a mistake.

Dale

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From bliss-bounces@ietf.org Mon Feb 19 18:17:49 2007
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John;

 I personally think it would depend on the feature/service we look into.

 After exploring the current deployments/implementations, we may realize
that certain features are best served with defining a SIP means for 
"invocation".

 If that's the case defining a mechanism for "invocation" is likely to 
be in scope
for that specific feature we explore.

 In contrary, if it is realized that defining SIP means to invocate the 
feature/service
is not appropriate/unnecessary/overkill then text mentioning this 
approach may
be noted that it does exist, but it would not be the recommended approach.

 Regards
  Shida

 

Elwell, John wrote:
> Jonathan,
>
> I agree with your point about defining a basic but extensible mechanism.
> What is not clear is whether mechanisms for "invocation" of features like
> DND should be within scope or not. On the one hand, these can be generally
> done by web pages and other non-SIP means. The question is whether we need
> to define a SIP means too or simply leave it out of scope.
>
> John
>
>   
>> -----Original Message-----
>> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
>> Sent: 16 February 2007 21:40
>> To: Elwell, John
>> Cc: Miguel Garcia; Drage, Keith (Keith); bliss@ietf.org
>> Subject: Re: [BLISS] Welcome! Main question - what services?
>>
>> There is an issue here, and it has to do with 'invocation' - how does 
>> the user signal that they'd like this desired behavior?
>>
>> If DND is being done in the network, and we want a user on a 
>> phone to be 
>> able to hit the 'DND for Video' button, then we do need a 
>> standardized 
>> way of communicating from the phone to the network element 
>> providing the 
>> feature (say, its proxy or IP PBX or CSCF or whatever) that 
>> this is the 
>> desired intent.
>>
>> We have this kind of problem even if the phone implements the DND 
>> processing. In every real system I've seen, there are always multiple 
>> ways of setting this DND status - from a user profile web page, from 
>> phone keys, from an admin web page, and so on. So even if the 
>> processing 
>> is implemented in the phone, we'd need a way to communicate 
>> the desired 
>> criteria from the web server/provisioning system or whatever, to the 
>> phone. Indeed, its exactly the same information we'd need to 
>> communicate 
>> from the phone to the 'network' when the actual call processing is 
>> implemented in the network.
>>
>> This (fancy conditions for defining when to block a call) may 
>> be an area 
>> that we decide to leave alone; that is, BLISS is about *basic* 
>> interoperability. We need to define the basic framework 
>> needed for this 
>> to interoperate, and make sure we leave the hooks in place 
>> for vendors 
>> to do special stuff when its their phones ontop of their 
>> servers, such 
>> as a fancier DND service.
>>
>> In the particular case of DND for example, let us say we specify that 
>> DND call processing is done in the network, and that the service is 
>> invoked by including a body in a REGISTER message that indicates the 
>> criteria and desired action (just for arguments sake). We 
>> could define a 
>> basic format for this object which defines no conditions 
>> (i.e., DND is 
>> either on or off) and defines three actions (reject, 
>> send-to-voicemail, 
>> ring-indefinitely). We do a nice job in the XML and leave it 
>> extensible 
>> for other conditions and actions. We define XML namespace 
>> techniques for 
>> adding new conditions and actions. We support negotiation of 
>> supported 
>> namespaces during registration time, so the phone knows whether the 
>> server supports the extensions it knows about. Then, it can gray out 
>> those buttons on its UI, or not show the menu options for them, etc, 
>> when the phone finds its connected to a server that only supports the 
>> baseline format.
>>
>> -Jonathan R.
>>
>>     
>
> _______________________________________________
> BLISS mailing list
> BLISS@ietf.org
> https://www1.ietf.org/mailman/listinfo/bliss
>
>
>   


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From bliss-bounces@ietf.org Mon Feb 19 22:43:31 2007
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Date: Tue, 20 Feb 2007 09:12:36 +0530
From: Jaimin Pancholi <jaiminp@huawei.com>
Subject: RE: [BLISS] Welcome! Main question - what services?
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Hi ,
	I may be a junior in this area but as per my understanding
the problem that the circuit switched network has is the centralized logic
for
almost everything. The end terminal does not have any intelligence at all,
making 
deployment of the service / application a big issue !

In general, SIP network (VoIP) network provides a advantage i.e. it has
decentralized the
logic and had proposed some intelligence in the end terminal.

I think, I would agree with Dale that intelligence in end terminal is good
for basic
services like DND, Call forward

Cheers
Jaimin

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>-----Original Message-----
>From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net] 
>Sent: Monday, February 19, 2007 10:31 PM
>To: bliss@ietf.org
>Subject: Re: [BLISS] Welcome! Main question - what services?
>
>   In the particular case of DND for example, let us say we 
>specify that 
>   DND call processing is done in the network, and that the service is 
>   invoked by including a body in a REGISTER message that 
>indicates the 
>   criteria and desired action (just for arguments sake).
>
>Um, if I might be so bold to ask, *why* would be want to have 
>DND call processing done in the network?  It's true that if we 
>want to have DND done in the network, we need to define some 
>moderately complex mechanism for having the UA report to the 
>network its DND status.  But that's really an argument that 
>DND should be done in the phone, and activated by a button 
>(like it is on every SIP phone).
>
>If we're really going to try to define such a mechanism, we 
>first need to review that there really is such a requirement, 
>that such an architecture is really the best solution to the 
>DND "problem" in a reasonable fraction of real-world cases.  
>That this is a significant deviation from the end-to-end 
>principle is a warning sign...
>
>Dale
>
>_______________________________________________
>BLISS mailing list
>BLISS@ietf.org
>https://www1.ietf.org/mailman/listinfo/bliss
>



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Dale.Worley@comcast.net wrote:
>    In the particular case of DND for example, let us say we specify that 
>    DND call processing is done in the network, and that the service is 
>    invoked by including a body in a REGISTER message that indicates the 
>    criteria and desired action (just for arguments sake).
> 
> Um, if I might be so bold to ask, *why* would be want to have DND call
> processing done in the network?  It's true that if we want to have DND
> done in the network, we need to define some moderately complex
> mechanism for having the UA report to the network its DND status.  But
> that's really an argument that DND should be done in the phone, and
> activated by a button (like it is on every SIP phone).
> 
> If we're really going to try to define such a mechanism, we first need
> to review that there really is such a requirement, that such an
> architecture is really the best solution to the DND "problem" in a
> reasonable fraction of real-world cases.  That this is a significant
> deviation from the end-to-end principle is a warning sign...

If I have multiple phones all registered to the same AOR, then I may 
want the DND to prevent them all from ringing. A DND implemented in the 
phone on my desk won't prevent the phones in the living room and bedroom 
and kitchen from ringing. A network based DND will.

Of course there are other ways to skin this cat. For instance, I could 
send a DND feature activation notice to all the other phones. That would 
only work for phones that were on at the time, so I would have to 
monitor for other phones registering, and send notifications to them 
when they do. And if a phone reboots I might have to send it again. (Not 
sure how I would detect that.)

When there are common policies that should be applied to any and all 
registered devices then there is some potential value in implementing 
that policy at the common point.

Of course it may be that I don't want my DND applied to the other 
registered phones. Specifically, if I am in my hotel room in Prague and 
have registered a phone to my home phone number, then I probably don't 
want DND selected by someone at home to affect me.

IMO this says that DND isn't just one feature. Instead, there is 
DND-device, and DND-AOR. They both have value in different cases. And 
both can be implemented in different ways.

	Paul

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   From: Paul Kyzivat <pkyzivat@cisco.com>

   Of course it may be that I don't want my DND applied to the other 
   registered phones. Specifically, if I am in my hotel room in Prague and 
   have registered a phone to my home phone number, then I probably don't 
   want DND selected by someone at home to affect me.

   IMO this says that DND isn't just one feature. Instead, there is 
   DND-device, and DND-AOR. They both have value in different cases. And 
   both can be implemented in different ways.

So there are various "scopes" to which the DND might apply, which
superficially seem to be the same AOR.  I've seen this phenomenon in
other places, particularly SUBSCRIBEs to obtain status information.
Often, the scope of forking that I want is not quite the same as all
the contacts for an AOR.

Dale

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From bliss-bounces@ietf.org Tue Feb 20 21:22:42 2007
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Subject: [BLISS] I-D on implementing GRUU
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I've updated an Internet-Draft giving condensed instructions for UA
implementors on how to implement GRUU support.  (This is updated for
version -11 of the GRUU I-D, but it only discusses "public GRUUs",
which is what we need for our environment.)  In our product's
environment (a "true SIP", proxy-based PBX), we've found that to do
call control well, the UAs need to implement (public) GRUU.  This I-D
summarizes what UAs need to do

   Guidelines for Implementing the GRUU Support in User Agents
   draft-worley-sip-gruu-implement-02

   http://www.ietf.org/internet-drafts/draft-worley-sip-gruu-implement-02.txt

   Abstract

   Several applications of the Session Initiation Protocol (SIP) require
   a user agent (UA) to construct and distribute a URI which can be used
   by anyone on the Internet to route a request to that specific UA
   instance.  A URI which routes to a specific UA instance is called a
   Globally Routable UA URI (GRUU).  An Internet-Draft is progressing
   toward standardization that specifies procedures for proxies and UAs
   by which proxies can construct and delver GRUUs to UAs that request
   them.  This document summarizes for UA implementors how to obtain
   "public" GRUUs and gives guidance on effectively using public GRUUs.

Feel free to contact me with any comments.

Dale

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From bliss-bounces@ietf.org Tue Feb 20 22:52:02 2007
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I've submitted an Internet-Draft giving guidelines for UAs that want
to implement the dialog event package.  In our product's environment
(a "true SIP", proxy-based PBX), we've found that to do call control
well, the UAs need to implement the dialog event package well.  This
I-D describes the needed requirements.

(If you want to test your UA against the guidelines, you can use our
interoperability server at http://interop.pingtel.com.)
   
   Guidelines for Implementing the Dialog Event Package in User Agents
   draft-worley-sip-dialog-03

   http://www.ietf.org/internet-drafts/draft-worley-sip-dialog-03.txt

   Abstract

   This document sets out guidelines for how Session Initiation Protocol
   (SIP) user agents (UAs) should implement the dialog event package in
   order to be usable in systems that implement end-point controlled
   call-control (EPCC) features.  The guidelines are in addition to the
   basic requirements for dialog event package implementation in RFC
   3265 and RFC 4235.

Feel free to contact me with any comments.

Dale

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From bliss-bounces@ietf.org Mon Feb 26 06:03:27 2007
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I have just submitted an Internet-Draft concerning call park/retrieve.
It is actually a draft I wrote almost 3 years ago, but which sank
without trace.  Given the BLISS charter, I think that it may be time to
revisit it.

Until it appears, you can see the original one at:
http://tools.ietf.org/html/draft-procter-sipping-call-park-extension-00
(The new version only updates the dates and references and the like).

The draft documents a real problem associated with parking calls in a
multi-vendor environment.  The solution is based upon the
service-examples document and is workable, although refinements are
clearly possible.

I'd be interested to hear comments on the draft, both the scope of the
problem and the approach taken.  I'd also be interested in how other
people have solved this problem, or even if they have avoided it
entirely.

Regards,

Michael Procter

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From bliss-bounces@ietf.org Mon Feb 26 17:18:00 2007
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Michael,

The Park to orbit (aka. park to server or park to system) whereby the
User must input a orbit number is old feature operation which may result
in the user entering an already used orbit. This form of Park to server
is somewhat limited.

Enhanced feature Park to Server operation should allow the user to
request the server to park a call and the server should allocate the
first available park orbit.  The response could be given to the user in
two forms: announcement and additional information i.e.., park orbit
number given back to the end user.=20

Both Old and Enhanced may require a bit different implementation e.g.,
if park orbit not given then selects the first orbit or if park orbit
given, attempt to park in the requested orbit.=20

For Call Retrieve the orbit must be specified.

I did not see a Recall from park, whereby the Park exceeds a given time
and will attempt to notify or redistribute the parked party back to the
UA that parked the call to the server. In the case of redistributing the
parked party back to the UA, it would be nice for the UA to know the
call coming back was a parked call and not a new incoming call.  Is
recall from park in the plans?


The old park to server feature whereby the user enters a park
destination is more in tune with the Park to Subscriber feature.  In
this case one UA parks the call to another UI by indicating park to
subscriber and the destination to be parked-to (not orbit).




Best Regards,
=20
Dennis L. Kucmerowski=20

561-923-1703
Systems Engineering
=20

-----Original Message-----
From: Michael Procter [mailto:michael.procter@citel.com]=20
Sent: Monday, February 26, 2007 6:03 AM
To: bliss@ietf.org
Subject: [BLISS] Call Park/Retrieve

I have just submitted an Internet-Draft concerning call park/retrieve.
It is actually a draft I wrote almost 3 years ago, but which sank
without trace.  Given the BLISS charter, I think that it may be time to
revisit it.

Until it appears, you can see the original one at:
http://tools.ietf.org/html/draft-procter-sipping-call-park-extension-00
(The new version only updates the dates and references and the like).

The draft documents a real problem associated with parking calls in a
multi-vendor environment.  The solution is based upon the
service-examples document and is workable, although refinements are
clearly possible.

I'd be interested to hear comments on the draft, both the scope of the
problem and the approach taken.  I'd also be interested in how other
people have solved this problem, or even if they have avoided it
entirely.

Regards,

Michael Procter

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

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Subject: Re: [BLISS] Call Park/Retrieve
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   From: "Kucmerowski, Dennis \(Com US\)" <dennis.kucmerowski@siemens.com>

   Enhanced feature Park to Server operation should allow the user to
   request the server to park a call and the server should allocate the
   first available park orbit.  The response could be given to the user in
   two forms: announcement and additional information i.e.., park orbit
   number given back to the end user. 

Do you know of any implementations of this scheme?

Dale

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From bliss-bounces@ietf.org Tue Feb 27 11:28:11 2007
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Subject: RE: [BLISS] Call Park/Retrieve
Date: Tue, 27 Feb 2007 08:28:04 -0800
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From: "Michael Procter" <michael.procter@citel.com>
To: "Kucmerowski, Dennis \(Com US\)" <dennis.kucmerowski@siemens.com>,
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Dennis,

Thanks for your comments.  My responses are inline.

Kucmerowski, Dennis (Com US) wrote:=20
> Michael,
>=20
> The Park to orbit (aka. park to server or park to system) whereby the
> User must input a orbit number is old feature operation which may
result
> in the user entering an already used orbit. This form of Park to
server
> is somewhat limited.

I agree that the model is not new.  This isn't necessarily bad, in my
opinion.  And whilst the selected orbit might be in use, this is not
necessarily a limitation.  A Park Server may choose to accept multiple
calls and form a queue.  This is an implementation issue and not a
protocol issue though.

> Enhanced feature Park to Server operation should allow the user to
> request the server to park a call and the server should allocate the
> first available park orbit.  The response could be given to the user
in
> two forms: announcement and additional information i.e.., park orbit
> number given back to the end user.

Yes, this is the other normal model.  I discuss this briefly in section
4, except I don't mention announcements.  Further elaboration could be
added.

> Both Old and Enhanced may require a bit different implementation e.g.,
> if park orbit not given then selects the first orbit or if park orbit
> given, attempt to park in the requested orbit.

Both these approaches are possible, and could be implemented by
combining the approaches taken in sections 2 and 4.  In this scenario,
the initial orbit given in the REFER request-uri might be considered
'advisory' until a confirmed one was returned by the Park Server.

> For Call Retrieve the orbit must be specified.
>=20
> I did not see a Recall from park, whereby the Park exceeds a given
time
> and will attempt to notify or redistribute the parked party back to
the
> UA that parked the call to the server. In the case of redistributing
the
> parked party back to the UA, it would be nice for the UA to know the
> call coming back was a parked call and not a new incoming call.  Is
> recall from park in the plans?

My draft initially began as a way to resolve the apparent (in my
opinion) weakness in addressing associated with the service-examples
park/retrieve approach.  There are plenty of other areas surrounding
park/retrieve that could be addressed with a view to standardising them
somewhat, and I think that this forum is probably the best place to
discuss what needs attention and what can be left as implementation
defined.

> The old park to server feature whereby the user enters a park
> destination is more in tune with the Park to Subscriber feature.  In
> this case one UA parks the call to another UI by indicating park to
> subscriber and the destination to be parked-to (not orbit).

That is certainly one implementation that I have seen.  Parking against
a subscriber is not always appropriate though, especially when the
ultimate destination UA is not known when parking.

One thing that is clear is that people have different expectations of a
park/retrieve service.  Any attempt to codify one set of implementation
choices is likely to be counter-productive.  Instead, we need to
identify the missing building blocks that can be used in multiple ways
whilst still offering some level of interoperability between vendors.

Regards,

Michael Procter

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From bliss-bounces@ietf.org Tue Feb 27 13:37:19 2007
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Subject: RE: [BLISS] Call Park/Retrieve
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Dale,

In the non-SIP world yes.  Siemens supports this form of operation for
Park to Server.  The user requests to park the call via a park request.
The user at the endpoint receives indication of what park slot the call
was parked in.


Best Regards,
=20
Dennis L. Kucmerowski=20

561-923-1703
Systems Engineering
=20

-----Original Message-----
From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]=20
Sent: Tuesday, February 27, 2007 10:57 AM
To: bliss@ietf.org
Subject: Re: [BLISS] Call Park/Retrieve

   From: "Kucmerowski, Dennis \(Com US\)"
<dennis.kucmerowski@siemens.com>

   Enhanced feature Park to Server operation should allow the user to
   request the server to park a call and the server should allocate the
   first available park orbit.  The response could be given to the user
in
   two forms: announcement and additional information i.e.., park orbit
   number given back to the end user.=20

Do you know of any implementations of this scheme?

Dale

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From bliss-bounces@ietf.org Tue Feb 27 17:09:46 2007
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Subject: [BLISS] Fwd: I-D ACTION:draft-johnston-bliss-mla-req-00.txt
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All,

The authors of draft-anil-sipping-bla (which will be revised as -04 next 
week) have put together this draft discussing requirements and 
implementation options for set of features known as Bridged Line 
Appearance (BLA), Multiple Line Appearance (MLA), Single Line Extension, 
etc.

Comments are most welcome.

Thanks,
Alan

-------- Original Message --------
Subject: 	I-D ACTION:draft-johnston-bliss-mla-req-00.txt
Date: 	Tue, 27 Feb 2007 15:50:03 -0500
From: 	Internet-Drafts@ietf.org
Reply-To: 	internet-drafts@ietf.org
To: 	i-d-announce@ietf.org



A New Internet-Draft is available from the on-line Internet-Drafts 
directories.


	Title		: Requirements and Implementation Options for
                          the Multiple Line Appearance Feature using 
                          the Session Initiation Protocol (SIP)
	Author(s)	: A. Johnston, et al.
	Filename	: draft-johnston-bliss-mla-req-00.txt
	Pages		: 16
	Date		: 2007-2-27
	
   This document describes the requirements for implementation of a
   group telephony feature commonly known as Bridged Line Appearance
   (BLA) or Multiple Line Appearance, or Shared Call/Line Appearance
   (SCA).  When implemented using the Session Initiation Protocol (SIP),
   it is referred to as Multiple Appearances.  This feature is commonly
   offered in the IP Centrex services and IP-PBX offerings and is likely
   to be implemented on SIP IP telephones and SIP feature servers used
   in a business environment.  This document lists requirements and
   compares implementation options for this feature.



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