
From brett@broadsoft.com  Mon Mar  1 05:40:30 2010
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From: Brett Tate <brett@broadsoft.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, SIPCORE <sipcore@ietf.org>
Date: Mon, 1 Mar 2010 05:35:04 -0800
Thread-Topic: [sipcore] OPTIONS: What does it mean?
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Subject: Re: [sipcore] OPTIONS: What does it mean?
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> > *If* the OPTIONS has a to-tag, and there is no matching=20
> > dialog there are several things that might happen:
> > - it might be rejected with a 481
> > - it might succeed because it is treated as if received=20
> >   outside a dialog

<snip>

> > - it might cause the dialog to be recreated, and then=20
> >   succeed within it
> > In the latter case, presumably the dialog would then end,=20
> > because there are no dialog usages.
>=20
> And, this will of course depend on whether the UAS still has=20
> any knowledge about the dialog.
>=20
> Personally I think this alternative is rather strange, and=20
> I've never heard about it being used. But, I agree that the=20
> text seems to allow it.

Regardless if justified by the above bullets 2 or 3, some venders do return=
 200 instead of 481.  For instance if OPTIONS is sent within dialog associa=
ted with a currently/formerly known subscription, the UAS wouldn't want to =
return 481 (because of the potential impacts) unless it actually verified t=
hat the subscription is no longer known.

Since it highlights why potentially relevant to get OPTIONS 481 ambiguity c=
larified, the following is snippet a 2008 thread.

"Just trying to understand what rfc3261 and rfc5057 indicate or tried to in=
dicate concerning 1) if OPTIONS 481 should ever be sent, and 2) the related=
 impacts upon the dialog usages if OPTIONS 481 is received, 3) OPTIONS' abi=
lity to temporality revive/recreate the dialog per UAS' desire to return 20=
0.

If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent requests/r=
esponses to terminate the dialog, I'm sure some would like to use it within=
 early dialogs, subscriptions, and elsewhere."


From christer.holmberg@ericsson.com  Mon Mar  1 10:39:32 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Brett Tate' <brett@broadsoft.com>, SIPCORE <sipcore@ietf.org>
Date: Mon, 1 Mar 2010 19:39:30 +0100
Thread-Topic: [sipcore] OPTIONS: What does it mean?
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Subject: Re: [sipcore] OPTIONS: What does it mean?
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Hi,

If we forget what the RFCs say (or, what we believe they say :), how would =
people like the UAS to handle OPTIONS in the case when it contains a to-tag=
 but the UAS currently has no state of a dialog associated with that to-tag=
 value?

Regards,

Christer

=20

-----Original Message-----
From: Brett Tate [mailto:brett@broadsoft.com]=20
Sent: 1. maaliskuuta 2010 15:35
To: Christer Holmberg; SIPCORE
Subject: RE: [sipcore] OPTIONS: What does it mean?

> > *If* the OPTIONS has a to-tag, and there is no matching dialog there=20
> > are several things that might happen:
> > - it might be rejected with a 481
> > - it might succeed because it is treated as if received=20
> >   outside a dialog

<snip>

> > - it might cause the dialog to be recreated, and then=20
> >   succeed within it
> > In the latter case, presumably the dialog would then end, because=20
> > there are no dialog usages.
>=20
> And, this will of course depend on whether the UAS still has any=20
> knowledge about the dialog.
>=20
> Personally I think this alternative is rather strange, and I've never=20
> heard about it being used. But, I agree that the text seems to allow=20
> it.

Regardless if justified by the above bullets 2 or 3, some venders do return=
 200 instead of 481.  For instance if OPTIONS is sent within dialog associa=
ted with a currently/formerly known subscription, the UAS wouldn't want to =
return 481 (because of the potential impacts) unless it actually verified t=
hat the subscription is no longer known.

Since it highlights why potentially relevant to get OPTIONS 481 ambiguity c=
larified, the following is snippet a 2008 thread.

"Just trying to understand what rfc3261 and rfc5057 indicate or tried to in=
dicate concerning 1) if OPTIONS 481 should ever be sent, and 2) the related=
 impacts upon the dialog usages if OPTIONS 481 is received, 3) OPTIONS' abi=
lity to temporality revive/recreate the dialog per UAS' desire to return 20=
0.

If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent requests/r=
esponses to terminate the dialog, I'm sure some would like to use it within=
 early dialogs, subscriptions, and elsewhere."


From pkyzivat@cisco.com  Mon Mar  1 10:58:52 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] OPTIONS: What does it mean?
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IMO, if the request has a to-tag, then the UAS should treat it as an 
in-dialog request not associated with any particular usage. If there is 
no matching dialog, then it should return a 481 response.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> 
> If we forget what the RFCs say (or, what we believe they say :), how would people like the UAS to handle OPTIONS in the case when it contains a to-tag but the UAS currently has no state of a dialog associated with that to-tag value?
> 
> Regards,
> 
> Christer
> 
>  
> 
> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com] 
> Sent: 1. maaliskuuta 2010 15:35
> To: Christer Holmberg; SIPCORE
> Subject: RE: [sipcore] OPTIONS: What does it mean?
> 
>>> *If* the OPTIONS has a to-tag, and there is no matching dialog there 
>>> are several things that might happen:
>>> - it might be rejected with a 481
>>> - it might succeed because it is treated as if received 
>>>   outside a dialog
> 
> <snip>
> 
>>> - it might cause the dialog to be recreated, and then 
>>>   succeed within it
>>> In the latter case, presumably the dialog would then end, because 
>>> there are no dialog usages.
>> And, this will of course depend on whether the UAS still has any 
>> knowledge about the dialog.
>>
>> Personally I think this alternative is rather strange, and I've never 
>> heard about it being used. But, I agree that the text seems to allow 
>> it.
> 
> Regardless if justified by the above bullets 2 or 3, some venders do return 200 instead of 481.  For instance if OPTIONS is sent within dialog associated with a currently/formerly known subscription, the UAS wouldn't want to return 481 (because of the potential impacts) unless it actually verified that the subscription is no longer known.
> 
> Since it highlights why potentially relevant to get OPTIONS 481 ambiguity clarified, the following is snippet a 2008 thread.
> 
> "Just trying to understand what rfc3261 and rfc5057 indicate or tried to indicate concerning 1) if OPTIONS 481 should ever be sent, and 2) the related impacts upon the dialog usages if OPTIONS 481 is received, 3) OPTIONS' ability to temporality revive/recreate the dialog per UAS' desire to return 200.
> 
> If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent requests/responses to terminate the dialog, I'm sure some would like to use it within early dialogs, subscriptions, and elsewhere."
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From brett@broadsoft.com  Mon Mar  1 11:40:26 2010
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From: Brett Tate <brett@broadsoft.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, SIPCORE <sipcore@ietf.org>
Date: Mon, 1 Mar 2010 11:39:49 -0800
Thread-Topic: [sipcore] OPTIONS: What does it mean?
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Subject: Re: [sipcore] OPTIONS: What does it mean?
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> If we forget what the RFCs say (or, what we believe they say :),
> how would people like the UAS to handle OPTIONS in the case when=20
> it contains a to-tag but the UAS currently has no state of a=20
> dialog associated with that to-tag value?

To avoid encouraging the use of OPTIONS to check dialog state (INVITE usage=
 and/or SUBSCRIBE usage), I'd prefer it not be mandatory to return 481.


> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: 1. maaliskuuta 2010 15:35
> To: Christer Holmberg; SIPCORE
> Subject: RE: [sipcore] OPTIONS: What does it mean?
>=20
> > > *If* the OPTIONS has a to-tag, and there is no matching dialog
> there
> > > are several things that might happen:
> > > - it might be rejected with a 481
> > > - it might succeed because it is treated as if received
> > >   outside a dialog
>=20
> <snip>
>=20
> > > - it might cause the dialog to be recreated, and then
> > >   succeed within it
> > > In the latter case, presumably the dialog would then end, because
> > > there are no dialog usages.
> >
> > And, this will of course depend on whether the UAS still has any
> > knowledge about the dialog.
> >
> > Personally I think this alternative is rather strange, and I've never
> > heard about it being used. But, I agree that the text seems to allow
> > it.
>=20
> Regardless if justified by the above bullets 2 or 3, some venders do
> return 200 instead of 481.  For instance if OPTIONS is sent within
> dialog associated with a currently/formerly known subscription, the UAS
> wouldn't want to return 481 (because of the potential impacts) unless
> it actually verified that the subscription is no longer known.
>=20
> Since it highlights why potentially relevant to get OPTIONS 481
> ambiguity clarified, the following is snippet a 2008 thread.
>=20
> "Just trying to understand what rfc3261 and rfc5057 indicate or tried
> to indicate concerning 1) if OPTIONS 481 should ever be sent, and 2)
> the related impacts upon the dialog usages if OPTIONS 481 is received,
> 3) OPTIONS' ability to temporality revive/recreate the dialog per UAS'
> desire to return 200.
>=20
> If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent
> requests/responses to terminate the dialog, I'm sure some would like to
> use it within early dialogs, subscriptions, and elsewhere."


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Date: Mon, 1 Mar 2010 20:44:39 +0100
Message-ID: <9ae56b1e1003011144g39987844u2671c6180333a7ab@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Content-Type: multipart/alternative; boundary=00163602611adbb5790480c27a86
Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] OPTIONS: What does it mean?
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Agree.

/Hans Erik van Elburg


On Mon, Mar 1, 2010 at 7:58 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

> IMO, if the request has a to-tag, then the UAS should treat it as an
> in-dialog request not associated with any particular usage. If there is no
> matching dialog, then it should return a 481 response.
>
>        Thanks,
>        Paul
>
>
> Christer Holmberg wrote:
>
>> Hi,
>>
>> If we forget what the RFCs say (or, what we believe they say :), how would
>> people like the UAS to handle OPTIONS in the case when it contains a to-tag
>> but the UAS currently has no state of a dialog associated with that to-tag
>> value?
>>
>> Regards,
>>
>> Christer
>>
>>
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com] Sent: 1. maaliskuuta 2010
>> 15:35
>> To: Christer Holmberg; SIPCORE
>> Subject: RE: [sipcore] OPTIONS: What does it mean?
>>
>>  *If* the OPTIONS has a to-tag, and there is no matching dialog there are
>>>> several things that might happen:
>>>> - it might be rejected with a 481
>>>> - it might succeed because it is treated as if received  outside a
>>>> dialog
>>>>
>>>
>> <snip>
>>
>>  - it might cause the dialog to be recreated, and then  succeed within it
>>>> In the latter case, presumably the dialog would then end, because there
>>>> are no dialog usages.
>>>>
>>> And, this will of course depend on whether the UAS still has any
>>> knowledge about the dialog.
>>>
>>> Personally I think this alternative is rather strange, and I've never
>>> heard about it being used. But, I agree that the text seems to allow it.
>>>
>>
>> Regardless if justified by the above bullets 2 or 3, some venders do
>> return 200 instead of 481.  For instance if OPTIONS is sent within dialog
>> associated with a currently/formerly known subscription, the UAS wouldn't
>> want to return 481 (because of the potential impacts) unless it actually
>> verified that the subscription is no longer known.
>>
>> Since it highlights why potentially relevant to get OPTIONS 481 ambiguity
>> clarified, the following is snippet a 2008 thread.
>>
>> "Just trying to understand what rfc3261 and rfc5057 indicate or tried to
>> indicate concerning 1) if OPTIONS 481 should ever be sent, and 2) the
>> related impacts upon the dialog usages if OPTIONS 481 is received, 3)
>> OPTIONS' ability to temporality revive/recreate the dialog per UAS' desire
>> to return 200.
>>
>> If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent
>> requests/responses to terminate the dialog, I'm sure some would like to use
>> it within early dialogs, subscriptions, and elsewhere."
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>>  _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

--00163602611adbb5790480c27a86
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Agree.<br><br clear=3D"all">/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Mon, Mar 1, 2010 at 7:58 PM, Paul Kyz=
ivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@cisco.com">pkyzivat@c=
isco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; p=
adding-left: 1ex;">
IMO, if the request has a to-tag, then the UAS should treat it as an in-dia=
log request not associated with any particular usage. If there is no matchi=
ng dialog, then it should return a 481 response.<br>
<br>
 =A0 =A0 =A0 =A0Thanks,<br><font color=3D"#888888">
 =A0 =A0 =A0 =A0Paul</font><div><div></div><div class=3D"h5"><br>
<br>
Christer Holmberg wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br>
<br>
If we forget what the RFCs say (or, what we believe they say :), how would =
people like the UAS to handle OPTIONS in the case when it contains a to-tag=
 but the UAS currently has no state of a dialog associated with that to-tag=
 value?<br>

<br>
Regards,<br>
<br>
Christer<br>
<br>
=A0<br>
-----Original Message-----<br>
From: Brett Tate [mailto:<a href=3D"mailto:brett@broadsoft.com" target=3D"_=
blank">brett@broadsoft.com</a>] Sent: 1. maaliskuuta 2010 15:35<br>
To: Christer Holmberg; SIPCORE<br>
Subject: RE: [sipcore] OPTIONS: What does it mean?<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><blockquote class=
=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin=
: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

*If* the OPTIONS has a to-tag, and there is no matching dialog there are se=
veral things that might happen:<br>
- it might be rejected with a 481<br>
- it might succeed because it is treated as if received  =A0outside a dialo=
g<br>
</blockquote></blockquote>
<br>
&lt;snip&gt;<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><blockquote class=
=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin=
: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

- it might cause the dialog to be recreated, and then  =A0succeed within it=
<br>
In the latter case, presumably the dialog would then end, because there are=
 no dialog usages.<br>
</blockquote>
And, this will of course depend on whether the UAS still has any knowledge =
about the dialog.<br>
<br>
Personally I think this alternative is rather strange, and I&#39;ve never h=
eard about it being used. But, I agree that the text seems to allow it.<br>
</blockquote>
<br>
Regardless if justified by the above bullets 2 or 3, some venders do return=
 200 instead of 481. =A0For instance if OPTIONS is sent within dialog assoc=
iated with a currently/formerly known subscription, the UAS wouldn&#39;t wa=
nt to return 481 (because of the potential impacts) unless it actually veri=
fied that the subscription is no longer known.<br>

<br>
Since it highlights why potentially relevant to get OPTIONS 481 ambiguity c=
larified, the following is snippet a 2008 thread.<br>
<br>
&quot;Just trying to understand what rfc3261 and rfc5057 indicate or tried =
to indicate concerning 1) if OPTIONS 481 should ever be sent, and 2) the re=
lated impacts upon the dialog usages if OPTIONS 481 is received, 3) OPTIONS=
&#39; ability to temporality revive/recreate the dialog per UAS&#39; desire=
 to return 200.<br>

<br>
If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent requests/r=
esponses to terminate the dialog, I&#39;m sure some would like to use it wi=
thin early dialogs, subscriptions, and elsewhere.&quot;<br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
<br>
</blockquote>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br>

--00163602611adbb5790480c27a86--

From pkyzivat@cisco.com  Mon Mar  1 12:24:02 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] OPTIONS: What does it mean?
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Brett,

I would prefer that what to respond in this case had always been 
unambiguous.

Given that is has been ambiguous, the question is whether its worthwhile 
to attempt to fix it. Practically speaking, the UAC will have to deal 
with UASes that were implemented long ago with the ambiguity. So it will 
continue to be improper to draw a conclusion about the status of the 
dialog from a 200 response.

	Thanks,
	Paul

Brett Tate wrote:
>> If we forget what the RFCs say (or, what we believe they say :),
>> how would people like the UAS to handle OPTIONS in the case when 
>> it contains a to-tag but the UAS currently has no state of a 
>> dialog associated with that to-tag value?
> 
> To avoid encouraging the use of OPTIONS to check dialog state (INVITE usage and/or SUBSCRIBE usage), I'd prefer it not be mandatory to return 481.
> 
> 
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com]
>> Sent: 1. maaliskuuta 2010 15:35
>> To: Christer Holmberg; SIPCORE
>> Subject: RE: [sipcore] OPTIONS: What does it mean?
>>
>>>> *If* the OPTIONS has a to-tag, and there is no matching dialog
>> there
>>>> are several things that might happen:
>>>> - it might be rejected with a 481
>>>> - it might succeed because it is treated as if received
>>>>   outside a dialog
>> <snip>
>>
>>>> - it might cause the dialog to be recreated, and then
>>>>   succeed within it
>>>> In the latter case, presumably the dialog would then end, because
>>>> there are no dialog usages.
>>> And, this will of course depend on whether the UAS still has any
>>> knowledge about the dialog.
>>>
>>> Personally I think this alternative is rather strange, and I've never
>>> heard about it being used. But, I agree that the text seems to allow
>>> it.
>> Regardless if justified by the above bullets 2 or 3, some venders do
>> return 200 instead of 481.  For instance if OPTIONS is sent within
>> dialog associated with a currently/formerly known subscription, the UAS
>> wouldn't want to return 481 (because of the potential impacts) unless
>> it actually verified that the subscription is no longer known.
>>
>> Since it highlights why potentially relevant to get OPTIONS 481
>> ambiguity clarified, the following is snippet a 2008 thread.
>>
>> "Just trying to understand what rfc3261 and rfc5057 indicate or tried
>> to indicate concerning 1) if OPTIONS 481 should ever be sent, and 2)
>> the related impacts upon the dialog usages if OPTIONS 481 is received,
>> 3) OPTIONS' ability to temporality revive/recreate the dialog per UAS'
>> desire to return 200.
>>
>> If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent
>> requests/responses to terminate the dialog, I'm sure some would like to
>> use it within early dialogs, subscriptions, and elsewhere."
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From christer.holmberg@ericsson.com  Mon Mar  1 13:53:54 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Brett Tate <brett@broadsoft.com>
Date: Mon, 1 Mar 2010 22:50:02 +0100
Thread-Topic: [sipcore] OPTIONS: What does it mean?
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Subject: Re: [sipcore] OPTIONS: What does it mean?
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Hi,

>I would prefer that what to respond in this case had always been
>unambiguous.
>
>Given that is has been ambiguous, the question is whether its worthwhile
>to attempt to fix it. Practically speaking, the UAC will have to deal
>with UASes that were implemented long ago with the ambiguity. So it will
>continue to be improper to draw a conclusion about the status of the
>dialog from a 200 response.

I agree. But, just because it has been ambiguos, it doesn't mean we can cla=
rify how it should be in future.

Regards,

Christer


Brett Tate wrote:
>> If we forget what the RFCs say (or, what we believe they say :),
>> how would people like the UAS to handle OPTIONS in the case when
>> it contains a to-tag but the UAS currently has no state of a
>> dialog associated with that to-tag value?
>
> To avoid encouraging the use of OPTIONS to check dialog state (INVITE usa=
ge and/or SUBSCRIBE usage), I'd prefer it not be mandatory to return 481.
>
>
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com]
>> Sent: 1. maaliskuuta 2010 15:35
>> To: Christer Holmberg; SIPCORE
>> Subject: RE: [sipcore] OPTIONS: What does it mean?
>>
>>>> *If* the OPTIONS has a to-tag, and there is no matching dialog
>> there
>>>> are several things that might happen:
>>>> - it might be rejected with a 481
>>>> - it might succeed because it is treated as if received
>>>>   outside a dialog
>> <snip>
>>
>>>> - it might cause the dialog to be recreated, and then
>>>>   succeed within it
>>>> In the latter case, presumably the dialog would then end, because
>>>> there are no dialog usages.
>>> And, this will of course depend on whether the UAS still has any
>>> knowledge about the dialog.
>>>
>>> Personally I think this alternative is rather strange, and I've never
>>> heard about it being used. But, I agree that the text seems to allow
>>> it.
>> Regardless if justified by the above bullets 2 or 3, some venders do
>> return 200 instead of 481.  For instance if OPTIONS is sent within
>> dialog associated with a currently/formerly known subscription, the UAS
>> wouldn't want to return 481 (because of the potential impacts) unless
>> it actually verified that the subscription is no longer known.
>>
>> Since it highlights why potentially relevant to get OPTIONS 481
>> ambiguity clarified, the following is snippet a 2008 thread.
>>
>> "Just trying to understand what rfc3261 and rfc5057 indicate or tried
>> to indicate concerning 1) if OPTIONS 481 should ever be sent, and 2)
>> the related impacts upon the dialog usages if OPTIONS 481 is received,
>> 3) OPTIONS' ability to temporality revive/recreate the dialog per UAS'
>> desire to return 200.
>>
>> If OPTIONS 481 really MUST be sent and SHOULD trigger subsequent
>> requests/responses to terminate the dialog, I'm sure some would like to
>> use it within early dialogs, subscriptions, and elsewhere."
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=

From christer.holmberg@ericsson.com  Mon Mar  1 14:09:21 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Mon, 1 Mar 2010 23:09:18 +0100
Thread-Topic: OPTIONS 3xx: To fork or not to fork?
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Subject: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,

Moving forward the the OPTIONS 3xx solution, one issue (raised by Paul, I t=
hink) is how a registrar, responsible for an AOR, makes a decission whether=
 to send a 3xx response to a received OPTIONS request addressed for that AO=
R, or whether to fork the request towards the registered UAS(s).

Alternative 1:    The decission is based on local policy (by operators and/=
or other SDOs)

Alternative 2:    The registrar always sends a 3xx response.

Alternative 3:    The decission is based on some information contained in t=
he OPTIONS request.

Again, we cannot do very much about existing deployments, but at least we w=
ould be able to provide clear guidelines for future implementations/specifi=
cations.

Regards,

Christer

NOTE: The issue only applies when the request is addressed to an AOR, and r=
eceived by a registrar responseible for that AOR. In all other cases I assu=
me the request will be forwarded by the proxy.=

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Subject: Re: [sipcore] OPTIONS: What does it mean?
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Paul Kyzivat wrote:
> IMO, if the request has a to-tag, then the UAS should treat it as an
> in-dialog request not associated with any particular usage. If there is
> no matching dialog, then it should return a 481 response.

+1 (based entirely on my desire to not have to special-case OPTIONS when
deciding whether a request is to be treated as in-dialog or not)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

From adam@nostrum.com  Mon Mar  1 15:06:26 2010
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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I would like to see #2 as a BCP. Not just for OPTIONS, but for all cases 
that a proxy/registrar might fork an inbound request.

/a

On 3/1/10 16:09, Mar 1, Christer Holmberg wrote:
> Hi,
>
> Moving forward the the OPTIONS 3xx solution, one issue (raised by Paul, I think) is how a registrar, responsible for an AOR, makes a decission whether to send a 3xx response to a received OPTIONS request addressed for that AOR, or whether to fork the request towards the registered UAS(s).
>
> Alternative 1:    The decission is based on local policy (by operators and/or other SDOs)
>
> Alternative 2:    The registrar always sends a 3xx response.
>
> Alternative 3:    The decission is based on some information contained in the OPTIONS request.
>
> Again, we cannot do very much about existing deployments, but at least we would be able to provide clear guidelines for future implementations/specifications.
>
> Regards,
>
> Christer
>
> NOTE: The issue only applies when the request is addressed to an AOR, and received by a registrar responseible for that AOR. In all other cases I assume the request will be forwarded by the proxy.
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>    


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Unless we make some radical change in philosophy, the "proxy" has 
ultimate discretion here, though I think we could produce a BCP on it.

There is already a header that can be used to indicate your preference:

Request-Disposition: proxy
or
Request-Disposition: redirect

But it is still up to the proxy to honor it or not.

My recommendation for a BCP would be:
- honor Request-Disposition if present. Else
- return 3xx if there is more than one contact
- either 3xx or forward is ok if there is only one contact

This has to be discretionary because some may not want to return the 
registered contact to the caller. (Certainly *some* contact will have to 
be returned to the caller for an INVITE to succeed, but it might not be 
the registered contact. It might be a temp gruu, or something from an 
SBC, ...) And of course no contact has to be returned if the called UA 
rejects the request. And none need be returned for an OPTIONS either.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> 
> Moving forward the the OPTIONS 3xx solution, one issue (raised by Paul, I think) is how a registrar, responsible for an AOR, makes a decission whether to send a 3xx response to a received OPTIONS request addressed for that AOR, or whether to fork the request towards the registered UAS(s).
> 
> Alternative 1:    The decission is based on local policy (by operators and/or other SDOs)
> 
> Alternative 2:    The registrar always sends a 3xx response.
> 
> Alternative 3:    The decission is based on some information contained in the OPTIONS request.
> 
> Again, we cannot do very much about existing deployments, but at least we would be able to provide clear guidelines for future implementations/specifications.
> 
> Regards,
> 
> Christer
> 
> NOTE: The issue only applies when the request is addressed to an AOR, and received by a registrar responseible for that AOR. In all other cases I assume the request will be forwarded by the proxy.
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From adam@nostrum.com  Mon Mar  1 16:08:50 2010
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-rfc3265bis-01.txt
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Thanks. This will be fixed in the -02 version.

/a

On 2/22/10 04:27, Feb 22, Elwell, John wrote:
> Adam,
> The newly added reference RFC 4528 is wrong - should be RFC 4538.
> John
>
>     ------------------------------------------------------------------------
>     *From:* sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org]
>     *On Behalf Of *Adam Roach
>     *Sent:* 19 February 2010 20:10
>     *To:* sipcore@ietf.org
>     *Subject:* Re: [sipcore] I-D
>     Action:draft-ietf-sipcore-rfc3265bis-01.txt
>
>     [as participant]
>
>     There are only a handful of changes in this version of the
>     document, none of them pertaining to the open issues. I will be
>     posting the open issues to the mailing list ahead of the Anaheim
>     meeting to hopefully get closure on them, so that we don't need to
>     spend much time discussing them at the face-to-face meeting.
>
>     As always, comments are welcome -- especially on the identified
>     open issues.
>
>     /a
>
>     On 2/19/10 2:00 PM, Internet-Drafts@ietf.org wrote:
>>     A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>     This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.
>>
>>
>>     	Title           : SIP-Specific Event Notification
>>     	Author(s)       : A. Roach
>>     	Filename        : draft-ietf-sipcore-rfc3265bis-01.txt
>>     	Pages           : 52
>>     	Date            : 2010-02-19
>>
>>     This document describes an extension to the Session Initiation
>>     Protocol (SIP).  The purpose of this extension is to provide an
>>     extensible framework by which SIP nodes can request notification from
>>     remote nodes indicating that certain events have occurred.
>>
>>     Note that the event notification mechanisms defined herein are NOT
>>     intended to be a general-purpose infrastructure for all classes of
>>     event subscription and notification.
>>
>>     A URL for this Internet-Draft is:
>>     http://www.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-01.txt
>>
>>     Internet-Drafts are also available by anonymous FTP at:
>>     ftp://ftp.ietf.org/internet-drafts/
>>
>>     Below is the data which will enable a MIME compliant mail reader
>>     implementation to automatically retrieve the ASCII version of the
>>     Internet-Draft.
>>        
>>
>>
>>     _______________________________________________
>>     sipcore mailing list
>>     sipcore@ietf.org
>>     https://www.ietf.org/mailman/listinfo/sipcore
>>        
>


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</head>
<body bgcolor="#ffffff" text="#000000">
Thanks. This will be fixed in the -02 version.<br>
<br>
/a<br>
<br>
On 2/22/10 04:27, Feb 22, Elwell, John wrote:
<blockquote
 cite="mid:A444A0F8084434499206E78C106220CAB93E109D@MCHP058A.global-ad.net"
 type="cite">
  <meta http-equiv="Content-Type"
 content="text/html; charset=ISO-8859-1">
  <meta content="MSHTML 6.00.2900.5921" name="GENERATOR">
  <div dir="ltr" align="left"><span class="816492510-22022010"><font
 color="#0000ff" face="Arial" size="2">Adam,</font></span></div>
  <div dir="ltr" align="left"><span class="816492510-22022010"></span>&nbsp;</div>
  <div dir="ltr" align="left"><span class="816492510-22022010"><font
 color="#0000ff" face="Arial" size="2">The newly added reference RFC
4528 is wrong - should be RFC 4538.</font></span></div>
  <div dir="ltr" align="left"><span class="816492510-22022010"></span>&nbsp;</div>
  <div dir="ltr" align="left"><span class="816492510-22022010"><font
 color="#0000ff" face="Arial" size="2">John</font></span></div>
  <div dir="ltr" align="left"><span class="816492510-22022010"></span>&nbsp;</div>
  <br>
  <blockquote
 style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px; margin-right: 0px;">
    <div class="OutlookMessageHeader" dir="ltr" align="left"
 lang="en-us">
    <hr tabindex="-1"> <font face="Tahoma" size="2"><b>From:</b>
<a class="moz-txt-link-abbreviated" href="mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounces@ietf.org</a>] <b>On
Behalf Of </b>Adam Roach<br>
    <b>Sent:</b> 19 February 2010 20:10<br>
    <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
    <b>Subject:</b> Re: [sipcore] I-D
Action:draft-ietf-sipcore-rfc3265bis-01.txt<br>
    </font><br>
    </div>
[as participant]<br>
    <br>
There are only a handful of changes in this version of the document,
none of them pertaining to the open issues. I will be posting the open
issues to the mailing list ahead of the Anaheim meeting to hopefully
get closure on them, so that we don't need to spend much time
discussing them at the face-to-face meeting.<br>
    <br>
As always, comments are welcome -- especially on the identified open
issues.<br>
    <br>
/a<br>
    <br>
On 2/19/10 2:00 PM, <a moz-do-not-send="true"
 class="moz-txt-link-abbreviated" href="mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a>
wrote:
    <blockquote cite="mid:20100219200001.9598F28C0F8@core3.amsl.com"
 type="cite">
      <pre wrap="">A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.


	Title           : SIP-Specific Event Notification
	Author(s)       : A. Roach
	Filename        : draft-ietf-sipcore-rfc3265bis-01.txt
	Pages           : 52
	Date            : 2010-02-19

This document describes an extension to the Session Initiation
Protocol (SIP).  The purpose of this extension is to provide an
extensible framework by which SIP nodes can request notification from
remote nodes indicating that certain events have occurred.

Note that the event notification mechanisms defined herein are NOT
intended to be a general-purpose infrastructure for all classes of
event subscription and notification.

A URL for this Internet-Draft is:
<a moz-do-not-send="true" class="moz-txt-link-freetext"
 href="http://www.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-01.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-01.txt</a>

Internet-Drafts are also available by anonymous FTP at:
<a moz-do-not-send="true" class="moz-txt-link-freetext"
 href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a>

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.
  </pre>
      <pre wrap=""><fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
sipcore mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated"
 href="mailto:sipcore@ietf.org">sipcore@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext"
 href="https://www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.org/mailman/listinfo/sipcore</a>
  </pre>
    </blockquote>
    <br>
  </blockquote>
</blockquote>
<br>
</body>
</html>

--------------000902040300090401010403--

From christer.holmberg@ericsson.com  Mon Mar  1 21:52:12 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Date: Tue, 2 Mar 2010 06:52:07 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,=20

>I would like to see #2 as a BCP. Not just for OPTIONS, but for all cases t=
hat a proxy/registrar might fork an inbound request.

Just to clarify, I guess you don't include dialog establsihment requests (I=
NVITE, SUBSCRIBE)? They should always be forwarded/forked even if addressed=
 to the AOR for which the registrar is responseible.

Regards,

Christer


> On 3/1/10 16:09, Mar 1, Christer Holmberg wrote:
> > Hi,
> >
> > Moving forward the the OPTIONS 3xx solution, one issue=20
> (raised by Paul, I think) is how a registrar, responsible for=20
> an AOR, makes a decission whether to send a 3xx response to a=20
> received OPTIONS request addressed for that AOR, or whether=20
> to fork the request towards the registered UAS(s).
> >
> > Alternative 1:    The decission is based on local policy=20
> (by operators and/or other SDOs)
> >
> > Alternative 2:    The registrar always sends a 3xx response.
> >
> > Alternative 3:    The decission is based on some=20
> information contained in the OPTIONS request.
> >
> > Again, we cannot do very much about existing deployments,=20
> but at least we would be able to provide clear guidelines for=20
> future implementations/specifications.
> >
> > Regards,
> >
> > Christer
> >
> > NOTE: The issue only applies when the request is addressed=20
> to an AOR, and received by a registrar responseible for that=20
> AOR. In all other cases I assume the request will be=20
> forwarded by the proxy.
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >   =20
>=20
> =

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 2 Mar 2010 06:57:31 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,=20

>Unless we make some radical change in philosophy, the "proxy"=20
>has ultimate discretion here, though I think we could produce=20
>a BCP on it.
>=20
>There is already a header that can be used to indicate your=20
>preference:
>=20
>Request-Disposition: proxy
>or
>Request-Disposition: redirect
>=20
>But it is still up to the proxy to honor it or not.

True. But, having clear guidelines will at least ensure (hopefully) that al=
l have a common understanding on how to do it - whether they choose to impl=
ement/honor it or not.

>My recommendation for a BCP would be:
>- honor Request-Disposition if present. Else
>- return 3xx if there is more than one contact
>- either 3xx or forward is ok if there is only one contact
>=20
>This has to be discretionary because some may not want to=20
>return the registered contact to the caller. (Certainly=20
>*some* contact will have to be returned to the caller for an=20
>INVITE to succeed, but it might not be the registered=20
>contact. It might be a temp gruu, or something from an SBC,=20
>...)
>
>And of course no contact has to be returned if the=20
>called UA rejects the request. And none need be returned for=20
>an OPTIONS either.

If the proxy does not want to return a contact (other than its own perhaps)=
, I guess it should not send a 3xx in the first place. In that case a 200 O=
K might be better?

Regards,

Christer




> Christer Holmberg wrote:
> > Hi,
> >=20
> > Moving forward the the OPTIONS 3xx solution, one issue=20
> (raised by Paul, I think) is how a registrar, responsible for=20
> an AOR, makes a decission whether to send a 3xx response to a=20
> received OPTIONS request addressed for that AOR, or whether=20
> to fork the request towards the registered UAS(s).
> >=20
> > Alternative 1:    The decission is based on local policy=20
> (by operators and/or other SDOs)
> >=20
> > Alternative 2:    The registrar always sends a 3xx response.
> >=20
> > Alternative 3:    The decission is based on some=20
> information contained in the OPTIONS request.
> >=20
> > Again, we cannot do very much about existing deployments,=20
> but at least we would be able to provide clear guidelines for=20
> future implementations/specifications.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> > NOTE: The issue only applies when the request is addressed=20
> to an AOR, and received by a registrar responseible for that=20
> AOR. In all other cases I assume the request will be=20
> forwarded by the proxy.
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> =

From gonzalo.camarillo@ericsson.com  Tue Mar  2 00:16:23 2010
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Date: Tue, 02 Mar 2010 10:16:19 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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Hi,

just to stress Adam's point, we need to understand if doing this is
worth the effort. If this is causing interoperability problems and
confusion about implementers, then we should go ahead fix it. If we are
doing this just for completeness so that our table 2 is up to date, then
we'd better employ our cycles on something else.

Note that I am not arguing one way or the other. My point is that given
the amount of cycles we have available (clearly a limited resource) we
need to be careful when choosing how to use them.

Cheers,

Gonzalo


Adam Roach wrote:
> Re-sending with a new subject line, since this has almost nothing to do
> with 3265bis any longer. Sorry for the additional noise.
> 
> [as a participant]
> 
> On 2/22/10 07:21, Feb 22, Paul Kyzivat wrote:
>>
>>
>> Dean Willis wrote:
>>
>>>> We had this discussion a while ago, and it may be good to start it
>>>> again. I think at least Dean mentioned abandoning Table 2. I talked
>>>> about having a website instead, which would be easier to keep up to
>>>> date, but I think the IETF proceures didn't allow that.
>>>
>>> Right. for things that one might use a website for, we have IANA
>>> registries.
>>>
>>> One could replace Table 2 with a registry, and have new RFCs update
>>> that registry.
>>
>> I support this approach.
> 
> Okay. If we go down this path -- and I think it's a little bit crazy --
> then we'll need to ferret out the correct values for the current header
> fields and methods to set up the initial registry.
> 
> Because of the haphazard way we've handled this, the table contains
> substantial gaps. I wrote a script to parse out all the formal Table 2
> expansions from the RFCs that define SIP header fields and SIP methods,
> and merge them into a single view of what has been defined to date. With
> the caveat that this is a lot of data, and something might have been
> missed, here's what it looks like:
> 
> Header Field 	Where 	Proxy 	ACK 	BYE 	CAN 	INF 	INV 	MES 	NOT 	OPT
> PRA 	PUB 	REF 	REG 	SUB 	UPD
> Accept 	2xx 	
> 	- 	- 	- 	? 	o 	- 	- 	m* 	- 	- 	- 	o 	- 	o
> Accept 	415 	
> 	- 	c 	- 	? 	c 	m* 	o 	c 	c 	m* 	c 	c 	o 	c
> Accept 	R 	
> 	- 	o 	- 	o 	o 	- 	o 	m* 	o 	o 	o 	o 	o 	o
> Accept-Contact 	R 	ar 	o 	o 	o 	o 	o 	o 	o 	o 	o 	? 	o 	- 	o 	o
> Accept-Encoding 	2xx 	
> 	- 	- 	- 	? 	o 	- 	- 	m* 	- 	- 	- 	o 	- 	o
> Accept-Encoding 	415 	
> 	- 	c 	- 	? 	c 	m* 	o 	c 	c 	m* 	c 	c 	o 	c
> Accept-Encoding 	R 	
> 	- 	o 	- 	o 	o 	- 	o 	o 	o 	o 	o 	o 	o 	o
> Accept-Language 	2xx 	
> 	- 	- 	- 	? 	o 	- 	- 	m* 	- 	- 	- 	o 	- 	o
> Accept-Language 	415 	
> 	- 	c 	- 	? 	c 	m* 	o 	c 	c 	m* 	c 	c 	o 	c
> Accept-Language 	R 	
> 	- 	o 	- 	o 	o 	- 	o 	o 	o 	o 	o 	o 	o 	o
> Accept-Resource-Priority 	200 	amdr 	- 	o 	o 	o 	o 	o 	o 	o 	o
> o 	o 	o 	o 	o
> Accept-Resource-Priority 	417 	amdr 	- 	o 	o 	o 	o 	o 	o 	o 	o
> o 	o 	o 	o 	o
> Alert-Info 	180 	
> 	- 	- 	- 	? 	o 	- 	- 	- 	- 	? 	? 	- 	- 	?
> Alert-Info 	R 	
> 	- 	- 	- 	? 	o 	- 	- 	- 	- 	- 	- 	- 	- 	-
> Alert-Info 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	- 	- 	? 	? 	-
> Allow 	2xx 	
> 	- 	o 	- 	? 	m* 	o 	o 	m* 	o 	? 	? 	o 	o 	o
> Allow 	405 	
> 	- 	m 	- 	o 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> Allow 	R 	
> 	- 	o 	- 	? 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o
> Allow 	r 	
> 	- 	o 	- 	? 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o
> Allow-Events 	2xx 	
> 	- 	o 	- 	? 	o 	? 	o 	o 	o 	? 	? 	o 	o 	?
> Allow-Events 	489 	
> 	- 	- 	- 	? 	- 	? 	m 	- 	- 	m 	? 	- 	m 	?
> Allow-Events 	R 	
> 	o 	o 	- 	? 	o 	? 	o 	o 	o 	o 	? 	o 	o 	-
> Allow-Events 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	-
> Answer-Mode 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Authentication-Info 	2xx 	
> 	- 	o 	- 	? 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Authorization 	R 	
> 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Call-ID 	c 	r 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> Call-Info 	R 	ar 	- 	- 	- 	? 	o 	o 	? 	o 	- 	o 	- 	o 	? 	o
> Call-Info 	r 	ar 	- 	- 	- 	? 	o 	o 	? 	o 	- 	o 	- 	o 	? 	o
> Contact 	1xx 	
> 	- 	- 	- 	- 	o 	- 	o 	- 	- 	- 	- 	- 	o 	o
> Contact 	2xx 	
> 	- 	- 	- 	- 	m 	- 	o 	o 	- 	- 	m 	o 	m 	m
> Contact 	3xx 	
> 	- 	o 	- 	- 	o 	o 	m 	o 	o 	o 	? 	o 	m 	o
> Contact 	485 	
> 	- 	o 	- 	- 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o
> Contact 	R 	
> 	o 	- 	- 	o 	m 	- 	m 	o 	- 	- 	m 	o 	m 	m
> Content-Disposition 	R 	
> 	o 	o 	- 	? 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Content-Disposition 	r 	
> 	o 	o 	- 	? 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Content-Encoding 	R 	
> 	o 	o 	- 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Content-Encoding 	r 	
> 	o 	o 	- 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Content-Language 	R 	
> 	o 	o 	- 	? 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Content-Language 	r 	
> 	o 	o 	- 	? 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Content-Length 	R 	ar 	t 	t 	t 	o 	t 	t 	t 	t 	t 	t 	o 	t 	t 	t
> Content-Length 	r 	ar 	t 	t 	t 	o 	t 	t 	t 	t 	t 	t 	o 	t 	t 	t
> Content-Type 	R 	
> 	* 	* 	- 	* 	* 	* 	* 	* 	* 	* 	* 	* 	* 	*
> Content-Type 	r 	
> 	* 	* 	- 	* 	* 	* 	* 	* 	* 	* 	* 	* 	* 	*
> CSeq 	c 	r 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> Date 	R 	a 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Date 	r 	a 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Encryption 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Error-Info 	300-699 	a 	- 	o 	o 	? 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o
> Event 	R 	
> 	- 	- 	- 	? 	- 	? 	m 	- 	- 	m 	? 	- 	m 	-
> Event 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	-
> Expires 	R 	
> 	- 	- 	- 	o 	o 	o 	- 	- 	- 	o 	o 	o 	o 	-
> Expires 	r 	
> 	- 	- 	- 	o 	o 	o 	- 	- 	- 	o 	? 	o 	o 	-
> Flow-Timer 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> From 	c 	r 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> History-Info 	R 	amdr 	- 	- 	- 	- 	o 	o 	o 	o 	- 	o 	o 	o 	o 	-
> History-Info 	r 	amdr 	- 	- 	- 	- 	o 	o 	o 	o 	- 	o 	o 	o 	o 	-
> Identity 	R 	a 	o 	o 	- 	? 	o 	? 	? 	o 	? 	? 	? 	o 	? 	?
> Identity-Info 	R 	a 	o 	o 	- 	? 	o 	? 	? 	o 	? 	? 	? 	o 	? 	?
> In-Reply-To 	R 	
> 	- 	- 	- 	? 	o 	o 	- 	- 	- 	- 	- 	- 	- 	-
> In-Reply-To 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	- 	? 	? 	-
> Join 	R 	
> 	- 	- 	- 	- 	o 	- 	- 	- 	- 	- 	- 	- 	- 	-
> Max-Breadth 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Max-Forwards 	R 	amr 	m 	m 	m 	o 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> MIME-Version 	R 	
> 	o 	o 	- 	? 	o 	? 	o 	o 	o 	o 	o 	o 	o 	o
> MIME-Version 	r 	
> 	o 	o 	- 	? 	o 	? 	o 	o 	o 	o 	o 	o 	o 	o
> Min-Expires 	423 	
> 	- 	- 	- 	? 	- 	? 	- 	- 	- 	m 	? 	m 	m 	?
> Min-Expires 	R 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	- 	? 	? 	-
> Min-Expires 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	- 	? 	? 	-
> Min-SE 	422 	
> 	- 	- 	- 	? 	m 	? 	- 	- 	- 	? 	? 	- 	- 	m
> Min-SE 	R 	amr 	- 	- 	- 	? 	o 	? 	- 	- 	- 	? 	? 	- 	- 	o
> Organization 	R 	ar 	- 	- 	- 	o 	o 	o 	- 	o 	- 	o 	o 	o 	o 	o
> Organization 	r 	ar 	- 	- 	- 	o 	o 	o 	- 	o 	- 	o 	o 	o 	o 	o
> P-Access-Network-Info 	R 	dr 	- 	o 	- 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o 	o
> P-Access-Network-Info 	r 	dr 	- 	o 	- 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o 	o
> P-Answer-State 	18x 	ar 	- 	- 	- 	? 	o 	? 	? 	- 	? 	? 	? 	- 	- 	?
> P-Answer-State 	2xx 	ar 	- 	- 	- 	? 	o 	? 	? 	- 	? 	? 	? 	- 	- 	?
> P-Asserted-Identity 	R 	adr 	- 	o 	- 	? 	o 	? 	? 	o 	? 	? 	? 	- 	? 	?
> P-Asserted-Identity 	r 	adr 	- 	o 	- 	? 	o 	? 	? 	o 	? 	? 	? 	- 	? 	?
> P-Associated-URI 	2xx 	
> 	- 	- 	- 	? 	- 	? 	? 	- 	? 	? 	? 	o 	? 	?
> P-Called-Party-ID 	R 	amr 	- 	- 	- 	- 	o 	o 	- 	o 	- 	? 	o 	- 	o 	-
> P-Charging-Function-Addresses 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> ? 	? 	? 	? 	?
> P-Charging-Vector 	R 	admr 	- 	o 	- 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o 	o
> P-Charging-Vector 	r 	admr 	- 	o 	- 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o 	o
> P-DCS-Trace-Party-ID 	R 	dmr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-DCS-OSPS 	R 	dr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-DCS-Billing-Info 	R 	admr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-DCS-Billing-Info 	r 	admr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-DCS-LAES 	R 	adr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-DCS-LAES 	r 	adr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-DCS-Redirect 	R 	adr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-DCS-Redirect 	r 	adr 	- 	- 	- 	? 	o 	? 	? 	- 	? 	- 	? 	- 	? 	?
> P-Early-Media 	18x 	amr 	- 	- 	- 	? 	o 	? 	? 	- 	- 	? 	? 	- 	? 	-
> P-Early-Media 	2xx 	amr 	- 	- 	- 	? 	- 	? 	? 	- 	o 	? 	? 	- 	? 	o
> P-Early-Media 	R 	amr 	- 	- 	- 	? 	o 	? 	? 	- 	o 	? 	? 	- 	? 	o
> P-Media-Authorization 	101-199 	ad 	- 	- 	- 	? 	o 	? 	? 	- 	? 	?
> ? 	- 	? 	?
> P-Media-Authorization 	2xx 	ad 	- 	- 	- 	- 	o 	? 	- 	- 	o 	? 	? 	- 	- 	o
> P-Media-Authorization 	R 	ad 	o 	- 	- 	- 	o 	? 	- 	- 	o 	? 	? 	- 	- 	o
> P-Preferred-Identity 	R 	adr 	- 	o 	- 	? 	o 	? 	? 	o 	? 	? 	? 	- 	? 	?
> P-Preferred-Identity 	r 	adr 	- 	o 	- 	? 	o 	? 	? 	o 	? 	? 	? 	- 	? 	?
> P-Profile-Key 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> P-Refused-URI-List 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> P-Served-User 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> P-User-Database 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> P-Visited-Network-ID 	R 	ad 	- 	- 	- 	- 	o 	o 	- 	o 	- 	? 	o 	o 	o 	-
> Path 	2xx 	- 	- 	- 	- 	? 	- 	? 	? 	- 	? 	? 	? 	o 	? 	?
> Path 	R 	ar 	- 	- 	- 	? 	- 	? 	? 	- 	? 	? 	? 	o 	? 	?
> Permission-Missing 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Priority 	R 	ar 	- 	- 	- 	o 	o 	o 	- 	- 	- 	o 	- 	- 	o 	-
> Priority 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	-
> Priv-Answer-Mode 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Privacy 	R 	amrd 	o 	o 	o 	? 	o 	o 	o 	o 	? 	? 	? 	o 	o 	o
> Privacy 	r 	amrd 	o 	o 	o 	? 	o 	o 	o 	o 	? 	? 	? 	o 	o 	o
> Proxy-Authenticate 	401 	ar 	- 	o 	o 	? 	o 	o 	? 	o 	o 	o 	o 	o 	? 	o
> Proxy-Authenticate 	407 	ar 	- 	m 	- 	o 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> Proxy-Authorization 	R 	dr 	o 	o 	- 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> RAck 	R 	
> 	- 	- 	- 	? 	- 	? 	- 	- 	m 	? 	? 	- 	- 	-
> Reason 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Record-Route 	18x 	mr 	- 	o 	o 	? 	o 	? 	? 	o 	o 	? 	o 	- 	? 	o
> Record-Route 	2xx 	mr 	- 	o 	o 	o 	o 	? 	o 	o 	o 	? 	o 	- 	o 	o
> Record-Route 	R 	ar 	o 	o 	o 	o 	o 	- 	o 	o 	o 	- 	o 	- 	o 	o
> Record-Route 	r 	ar 	? 	? 	? 	? 	? 	- 	? 	? 	? 	- 	? 	? 	? 	?
> Refer-Sub 	2xx 	
> 	- 	- 	- 	- 	- 	- 	- 	- 	- 	- 	o 	- 	- 	-
> Refer-Sub 	R 	
> 	- 	- 	- 	- 	- 	- 	- 	- 	- 	- 	o 	- 	- 	-
> Refer-To 	R 	
> 	- 	- 	- 	? 	- 	? 	? 	- 	? 	? 	? 	- 	? 	?
> Referred-By 	R 	
> 	- 	o 	- 	? 	o 	? 	? 	o 	? 	? 	? 	o 	? 	?
> Reject-Contact 	R 	ar 	o 	o 	o 	o 	o 	o 	o 	o 	o 	? 	o 	- 	o 	o
> Replaces 	R 	
> 	- 	- 	- 	- 	o 	- 	- 	- 	- 	- 	- 	- 	- 	-
> Reply-To 	R 	
> 	- 	- 	- 	? 	o 	o 	- 	- 	- 	- 	- 	- 	- 	-
> Reply-To 	r 	
> 	- 	- 	- 	? 	o 	o 	- 	- 	- 	- 	- 	- 	- 	-
> Request-Disposition 	R 	ar 	o 	o 	o 	o 	o 	o 	o 	o 	o 	? 	o 	o 	o 	o
> Require 	R 	ar 	- 	c 	- 	o 	c 	c 	o 	c 	c 	o 	c 	c 	o 	c
> Require 	r 	ar 	- 	c 	- 	? 	c 	c 	o 	c 	c 	o 	c 	c 	o 	c
> Resource-Priority 	R 	amdr 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Response-Key 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Retry-After 	404 	
> 	- 	o 	o 	o 	o 	? 	o 	o 	o 	o 	o 	o 	o 	o
> Retry-After 	413 	
> 	- 	o 	o 	? 	o 	? 	o 	o 	o 	o 	o 	o 	o 	o
> Retry-After 	480 	
> 	- 	o 	o 	o 	o 	? 	o 	o 	o 	o 	o 	o 	o 	o
> Retry-After 	486 	
> 	- 	o 	o 	o 	o 	? 	o 	o 	o 	o 	o 	o 	o 	o
> Retry-After 	4xx 	
> 	? 	? 	? 	? 	? 	o 	? 	? 	? 	? 	? 	? 	? 	?
> Route 	R 	adr 	c 	c 	c 	o 	c 	o 	c 	c 	c 	c 	c 	c 	c 	c
> RSeq 	1xx 	
> 	- 	- 	- 	? 	o 	? 	o 	- 	- 	? 	? 	- 	o 	?
> RSeq 	R 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	-
> RSeq 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	-
> Security-Client 	R 	ard 	- 	o 	- 	? 	o 	o 	o 	o 	? 	? 	? 	o 	o 	o
> Security-Server 	421 	
> 	- 	o 	- 	? 	o 	o 	o 	o 	? 	? 	? 	o 	o 	o
> Security-Server 	494 	
> 	- 	o 	- 	? 	o 	o 	o 	o 	? 	? 	? 	o 	o 	o
> Security-Verify 	R 	ard 	- 	o 	- 	? 	o 	o 	o 	o 	? 	? 	? 	o 	o 	o
> Server 	r 	
> 	- 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Service-Route 	2xx 	ar 	- 	- 	- 	? 	- 	? 	? 	- 	- 	? 	? 	o 	? 	?
> Session-Expires 	2xx 	ar 	- 	- 	- 	? 	o 	? 	- 	- 	- 	? 	? 	- 	- 	o
> Session-Expires 	R 	amr 	- 	- 	- 	? 	o 	? 	- 	- 	- 	? 	? 	- 	- 	o
> SIP-ETag 	2xx 	
> 	- 	- 	- 	- 	- 	- 	- 	- 	- 	m 	- 	- 	- 	-
> SIP-If-Match 	R 	
> 	- 	- 	- 	- 	- 	- 	- 	- 	- 	o 	- 	- 	- 	-
> Subject 	R 	
> 	- 	- 	- 	o 	o 	o 	- 	- 	- 	o 	- 	- 	- 	-
> Subject 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	-
> Subscription-State 	R 	
> 	- 	- 	- 	? 	- 	? 	m 	- 	- 	? 	? 	- 	- 	-
> Subscription-State 	r 	
> 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	-
> Supported 	2xx 	
> 	- 	o 	o 	? 	m* 	? 	o 	m* 	o 	o 	o 	o 	o 	o
> Supported 	R 	
> 	- 	o 	o 	? 	m* 	? 	o 	o 	o 	o 	o 	o 	o 	o
> Target-Dialog 	R 	- 	- 	- 	- 	- 	o 	- 	- 	- 	- 	- 	o 	- 	o 	-
> Timestamp 	R 	
> 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Timestamp 	r 	
> 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> To 	c 	r 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> Trigger-Consent 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	? 	?
> Unsupported 	420 	
> 	- 	m 	- 	o 	m 	o 	o 	m 	m 	o 	o 	m 	o 	m
> User-Agent 	R 	
> 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> User-Agent 	r 	
> 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> Via 	R 	amr 	m 	m 	m 	? 	m 	m 	? 	m 	? 	m 	? 	m 	? 	m
> Via 	rc 	dr 	m 	m 	m 	? 	m 	m 	? 	m 	? 	m 	? 	m 	? 	m
> Warning 	r 	
> 	- 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o 	o
> WWW-Authenticate 	401 	ar 	- 	m 	- 	o 	m 	m 	m 	m 	m 	m 	m 	m 	m 	m
> WWW-Authenticate 	407 	ar 	- 	o 	- 	? 	o 	o 	? 	o 	? 	o 	o 	o 	? 	o
> 
> 
> 
> See all those question marks? Someone needs to come up with actual
> values for each of those cells. Some are going to be pretty easy to
> answer; others will require substantial research and/or debate.
> 
> It seems like a lot of effort to me. Are we certain the return on our
> investment of time will be worth it?
> 
> /a
> 


From christer.holmberg@ericsson.com  Tue Mar  2 00:28:34 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, Adam Roach <adam@nostrum.com>
Date: Tue, 2 Mar 2010 09:28:29 +0100
Thread-Topic: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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Subject: Re: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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Hi,

I agree that we should be careful, and try to ensure that we have the neede=
d commitment from people.

But, note that it is not only about interoperability. If we do not do this,=
 I still think we need some guideline on how to deal with the header/method=
 tables whenever we define a new method and/or header field. Do we list all=
 methods/headers that have currently been specified at that moment, or?

Regards,

Christer

> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> Sent: 2. maaliskuuta 2010 10:16
> To: Adam Roach
> Cc: SIPCORE
> Subject: Re: [sipcore] SIP Table 2 (was Re: 3265bis and Call-Info)
>
> Hi,
>
> just to stress Adam's point, we need to understand if doing
> this is worth the effort. If this is causing interoperability
> problems and confusion about implementers, then we should go
> ahead fix it. If we are doing this just for completeness so
> that our table 2 is up to date, then we'd better employ our
> cycles on something else.
>
> Note that I am not arguing one way or the other. My point is
> that given the amount of cycles we have available (clearly a
> limited resource) we need to be careful when choosing how to use them.
>
> Cheers,
>
> Gonzalo
>
>
> Adam Roach wrote:
> > Re-sending with a new subject line, since this has almost
> nothing to
> > do with 3265bis any longer. Sorry for the additional noise.
> >
> > [as a participant]
> >
> > On 2/22/10 07:21, Feb 22, Paul Kyzivat wrote:
> >>
> >>
> >> Dean Willis wrote:
> >>
> >>>> We had this discussion a while ago, and it may be good
> to start it
> >>>> again. I think at least Dean mentioned abandoning Table
> 2. I talked
> >>>> about having a website instead, which would be easier to
> keep up to
> >>>> date, but I think the IETF proceures didn't allow that.
> >>>
> >>> Right. for things that one might use a website for, we have IANA
> >>> registries.
> >>>
> >>> One could replace Table 2 with a registry, and have new
> RFCs update
> >>> that registry.
> >>
> >> I support this approach.
> >
> > Okay. If we go down this path -- and I think it's a little
> bit crazy
> > -- then we'll need to ferret out the correct values for the current
> > header fields and methods to set up the initial registry.
> >
> > Because of the haphazard way we've handled this, the table contains
> > substantial gaps. I wrote a script to parse out all the
> formal Table 2
> > expansions from the RFCs that define SIP header fields and SIP
> > methods, and merge them into a single view of what has been
> defined to
> > date. With the caveat that this is a lot of data, and
> something might
> > have been missed, here's what it looks like:
> >
> > Header Field        Where   Proxy   ACK     BYE     CAN
> INF   INV     MES     NOT     OPT
> > PRA         PUB     REF     REG     SUB     UPD
> > Accept      2xx
> >     -       -       -       ?       o       -       -
> m*    -       -       -       o       -       o
> > Accept      415
> >     -       c       -       ?       c       m*      o
> c     c       m*      c       c       o       c
> > Accept      R
> >     -       o       -       o       o       -       o
> m*    o       o       o       o       o       o
> > Accept-Contact      R       ar      o       o       o
> o     o       o       o       o       o       ?       o
> -     o       o
> > Accept-Encoding     2xx
> >     -       -       -       ?       o       -       -
> m*    -       -       -       o       -       o
> > Accept-Encoding     415
> >     -       c       -       ?       c       m*      o
> c     c       m*      c       c       o       c
> > Accept-Encoding     R
> >     -       o       -       o       o       -       o
> o     o       o       o       o       o       o
> > Accept-Language     2xx
> >     -       -       -       ?       o       -       -
> m*    -       -       -       o       -       o
> > Accept-Language     415
> >     -       c       -       ?       c       m*      o
> c     c       m*      c       c       o       c
> > Accept-Language     R
> >     -       o       -       o       o       -       o
> o     o       o       o       o       o       o
> > Accept-Resource-Priority    200     amdr    -       o
> o     o       o       o       o       o       o
> > o   o       o       o       o
> > Accept-Resource-Priority    417     amdr    -       o
> o     o       o       o       o       o       o
> > o   o       o       o       o
> > Alert-Info  180
> >     -       -       -       ?       o       -       -
> -     -       ?       ?       -       -       ?
> > Alert-Info  R
> >     -       -       -       ?       o       -       -
> -     -       -       -       -       -       -
> > Alert-Info  r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       -       -       ?       ?       -
> > Allow       2xx
> >     -       o       -       ?       m*      o       o
> m*    o       ?       ?       o       o       o
> > Allow       405
> >     -       m       -       o       m       m       m
> m     m       m       m       m       m       m
> > Allow       R
> >     -       o       -       ?       o       o       o
> o     o       o       ?       o       o       o
> > Allow       r
> >     -       o       -       ?       o       o       o
> o     o       o       ?       o       o       o
> > Allow-Events        2xx
> >     -       o       -       ?       o       ?       o
> o     o       ?       ?       o       o       ?
> > Allow-Events        489
> >     -       -       -       ?       -       ?       m
> -     -       m       ?       -       m       ?
> > Allow-Events        R
> >     o       o       -       ?       o       ?       o
> o     o       o       ?       o       o       -
> > Allow-Events        r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       -
> > Answer-Mode         ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > Authentication-Info         2xx
> >     -       o       -       ?       o       o       o
> o     o       o       o       o       o       o
> > Authorization       R
> >     o       o       o       o       o       o       o
> o     o       o       o       o       o       o
> > Call-ID     c       r       m       m       m       m
> m     m       m       m       m       m       m       m
> m     m
> > Call-Info   R       ar      -       -       -       ?
> o     o       ?       o       -       o       -       o
> ?     o
> > Call-Info   r       ar      -       -       -       ?
> o     o       ?       o       -       o       -       o
> ?     o
> > Contact     1xx
> >     -       -       -       -       o       -       o
> -     -       -       -       -       o       o
> > Contact     2xx
> >     -       -       -       -       m       -       o
> o     -       -       m       o       m       m
> > Contact     3xx
> >     -       o       -       -       o       o       m
> o     o       o       ?       o       m       o
> > Contact     485
> >     -       o       -       -       o       o       o
> o     o       o       ?       o       o       o
> > Contact     R
> >     o       -       -       o       m       -       m
> o     -       -       m       o       m       m
> > Content-Disposition         R
> >     o       o       -       ?       o       o       o
> o     o       o       o       o       o       o
> > Content-Disposition         r
> >     o       o       -       ?       o       o       o
> o     o       o       o       o       o       o
> > Content-Encoding    R
> >     o       o       -       o       o       o       o
> o     o       o       o       o       o       o
> > Content-Encoding    r
> >     o       o       -       o       o       o       o
> o     o       o       o       o       o       o
> > Content-Language    R
> >     o       o       -       ?       o       o       o
> o     o       o       o       o       o       o
> > Content-Language    r
> >     o       o       -       ?       o       o       o
> o     o       o       o       o       o       o
> > Content-Length      R       ar      t       t       t
> o     t       t       t       t       t       t       o
> t     t       t
> > Content-Length      r       ar      t       t       t
> o     t       t       t       t       t       t       o
> t     t       t
> > Content-Type        R
> >     *       *       -       *       *       *       *
> *     *       *       *       *       *       *
> > Content-Type        r
> >     *       *       -       *       *       *       *
> *     *       *       *       *       *       *
> > CSeq        c       r       m       m       m       m
> m     m       m       m       m       m       m       m
> m     m
> > Date        R       a       o       o       o       o
> o     o       o       o       o       o       o       o
> o     o
> > Date        r       a       o       o       o       o
> o     o       o       o       o       o       o       o
> o     o
> > Encryption  ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?
> > Error-Info  300-699         a       -       o       o
> ?     o       o       o       o       o       o       ?
> o     o       o
> > Event       R
> >     -       -       -       ?       -       ?       m
> -     -       m       ?       -       m       -
> > Event       r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       -
> > Expires     R
> >     -       -       -       o       o       o       -
> -     -       o       o       o       o       -
> > Expires     r
> >     -       -       -       o       o       o       -
> -     -       o       ?       o       o       -
> > Flow-Timer  ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?
> > From        c       r       m       m       m       m
> m     m       m       m       m       m       m       m
> m     m
> > History-Info        R       amdr    -       -       -
> -     o       o       o       o       -       o       o
> o     o       -
> > History-Info        r       amdr    -       -       -
> -     o       o       o       o       -       o       o
> o     o       -
> > Identity    R       a       o       o       -       ?
> o     ?       ?       o       ?       ?       ?       o
> ?     ?
> > Identity-Info       R       a       o       o       -
> ?     o       ?       ?       o       ?       ?       ?
> o     ?       ?
> > In-Reply-To         R
> >     -       -       -       ?       o       o       -
> -     -       -       -       -       -       -
> > In-Reply-To         r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       -       ?       ?       -
> > Join        R
> >     -       -       -       -       o       -       -
> -     -       -       -       -       -       -
> > Max-Breadth         ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > Max-Forwards        R       amr     m       m       m
> o     m       m       m       m       m       m       m
> m     m       m
> > MIME-Version        R
> >     o       o       -       ?       o       ?       o
> o     o       o       o       o       o       o
> > MIME-Version        r
> >     o       o       -       ?       o       ?       o
> o     o       o       o       o       o       o
> > Min-Expires         423
> >     -       -       -       ?       -       ?       -
> -     -       m       ?       m       m       ?
> > Min-Expires         R
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       -       ?       ?       -
> > Min-Expires         r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       -       ?       ?       -
> > Min-SE      422
> >     -       -       -       ?       m       ?       -
> -     -       ?       ?       -       -       m
> > Min-SE      R       amr     -       -       -       ?
> o     ?       -       -       -       ?       ?       -
> -     o
> > Organization        R       ar      -       -       -
> o     o       o       -       o       -       o       o
> o     o       o
> > Organization        r       ar      -       -       -
> o     o       o       -       o       -       o       o
> o     o       o
> > P-Access-Network-Info       R       dr      -       o
> -     o       o       o       o       o       o       ?
> o     o       o       o
> > P-Access-Network-Info       r       dr      -       o
> -     o       o       o       o       o       o       ?
> o     o       o       o
> > P-Answer-State      18x     ar      -       -       -
> ?     o       ?       ?       -       ?       ?       ?
> -     -       ?
> > P-Answer-State      2xx     ar      -       -       -
> ?     o       ?       ?       -       ?       ?       ?
> -     -       ?
> > P-Asserted-Identity         R       adr     -       o
> -     ?       o       ?       ?       o       ?       ?
> ?     -       ?       ?
> > P-Asserted-Identity         r       adr     -       o
> -     ?       o       ?       ?       o       ?       ?
> ?     -       ?       ?
> > P-Associated-URI    2xx
> >     -       -       -       ?       -       ?       ?
> -     ?       ?       ?       o       ?       ?
> > P-Called-Party-ID   R       amr     -       -       -
> -     o       o       -       o       -       ?       o
> -     o       -
> > P-Charging-Function-Addresses       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> > ?   ?       ?       ?       ?
> > P-Charging-Vector   R       admr    -       o       -
> o     o       o       o       o       o       ?       o
> o     o       o
> > P-Charging-Vector   r       admr    -       o       -
> o     o       o       o       o       o       ?       o
> o     o       o
> > P-DCS-Trace-Party-ID        R       dmr     -       -
> -     ?       o       ?       ?       -       ?       -
> ?     -       ?       ?
> > P-DCS-OSPS  R       dr      -       -       -       ?
> o     ?       ?       -       ?       -       ?       -
> ?     ?
> > P-DCS-Billing-Info  R       admr    -       -       -
> ?     o       ?       ?       -       ?       -       ?
> -     ?       ?
> > P-DCS-Billing-Info  r       admr    -       -       -
> ?     o       ?       ?       -       ?       -       ?
> -     ?       ?
> > P-DCS-LAES  R       adr     -       -       -       ?
> o     ?       ?       -       ?       -       ?       -
> ?     ?
> > P-DCS-LAES  r       adr     -       -       -       ?
> o     ?       ?       -       ?       -       ?       -
> ?     ?
> > P-DCS-Redirect      R       adr     -       -       -
> ?     o       ?       ?       -       ?       -       ?
> -     ?       ?
> > P-DCS-Redirect      r       adr     -       -       -
> ?     o       ?       ?       -       ?       -       ?
> -     ?       ?
> > P-Early-Media       18x     amr     -       -       -
> ?     o       ?       ?       -       -       ?       ?
> -     ?       -
> > P-Early-Media       2xx     amr     -       -       -
> ?     -       ?       ?       -       o       ?       ?
> -     ?       o
> > P-Early-Media       R       amr     -       -       -
> ?     o       ?       ?       -       o       ?       ?
> -     ?       o
> > P-Media-Authorization       101-199         ad      -
> -     -       ?       o       ?       ?       -       ?       ?
> > ?   -       ?       ?
> > P-Media-Authorization       2xx     ad      -       -
> -     -       o       ?       -       -       o       ?
> ?     -       -       o
> > P-Media-Authorization       R       ad      o       -
> -     -       o       ?       -       -       o       ?
> ?     -       -       o
> > P-Preferred-Identity        R       adr     -       o
> -     ?       o       ?       ?       o       ?       ?
> ?     -       ?       ?
> > P-Preferred-Identity        r       adr     -       o
> -     ?       o       ?       ?       o       ?       ?
> ?     -       ?       ?
> > P-Profile-Key       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > P-Refused-URI-List  ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > P-Served-User       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > P-User-Database     ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > P-Visited-Network-ID        R       ad      -       -
> -     -       o       o       -       o       -       ?
> o     o       o       -
> > Path        2xx     -       -       -       -       ?
> -     ?       ?       -       ?       ?       ?       o
> ?     ?
> > Path        R       ar      -       -       -       ?
> -     ?       ?       -       ?       ?       ?       o
> ?     ?
> > Permission-Missing  ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > Priority    R       ar      -       -       -       o
> o     o       -       -       -       o       -       -
> o     -
> > Priority    r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       -
> > Priv-Answer-Mode    ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > Privacy     R       amrd    o       o       o       ?
> o     o       o       o       ?       ?       ?       o
> o     o
> > Privacy     r       amrd    o       o       o       ?
> o     o       o       o       ?       ?       ?       o
> o     o
> > Proxy-Authenticate  401     ar      -       o       o
> ?     o       o       ?       o       o       o       o
> o     ?       o
> > Proxy-Authenticate  407     ar      -       m       -
> o     m       m       m       m       m       m       m
> m     m       m
> > Proxy-Authorization         R       dr      o       o
> -     o       o       o       o       o       o       o
> o     o       o       o
> > RAck        R
> >     -       -       -       ?       -       ?       -
> -     m       ?       ?       -       -       -
> > Reason      ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?
> > Record-Route        18x     mr      -       o       o
> ?     o       ?       ?       o       o       ?       o
> -     ?       o
> > Record-Route        2xx     mr      -       o       o
> o     o       ?       o       o       o       ?       o
> -     o       o
> > Record-Route        R       ar      o       o       o
> o     o       -       o       o       o       -       o
> -     o       o
> > Record-Route        r       ar      ?       ?       ?
> ?     ?       -       ?       ?       ?       -       ?
> ?     ?       ?
> > Refer-Sub   2xx
> >     -       -       -       -       -       -       -
> -     -       -       o       -       -       -
> > Refer-Sub   R
> >     -       -       -       -       -       -       -
> -     -       -       o       -       -       -
> > Refer-To    R
> >     -       -       -       ?       -       ?       ?
> -     ?       ?       ?       -       ?       ?
> > Referred-By         R
> >     -       o       -       ?       o       ?       ?
> o     ?       ?       ?       o       ?       ?
> > Reject-Contact      R       ar      o       o       o
> o     o       o       o       o       o       ?       o
> -     o       o
> > Replaces    R
> >     -       -       -       -       o       -       -
> -     -       -       -       -       -       -
> > Reply-To    R
> >     -       -       -       ?       o       o       -
> -     -       -       -       -       -       -
> > Reply-To    r
> >     -       -       -       ?       o       o       -
> -     -       -       -       -       -       -
> > Request-Disposition         R       ar      o       o
> o     o       o       o       o       o       o       ?
> o     o       o       o
> > Require     R       ar      -       c       -       o
> c     c       o       c       c       o       c       c
> o     c
> > Require     r       ar      -       c       -       ?
> c     c       o       c       c       o       c       c
> o     c
> > Resource-Priority   R       amdr    o       o       o
> o     o       o       o       o       o       o       o
> o     o       o
> > Response-Key        ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > Retry-After         404
> >     -       o       o       o       o       ?       o
> o     o       o       o       o       o       o
> > Retry-After         413
> >     -       o       o       ?       o       ?       o
> o     o       o       o       o       o       o
> > Retry-After         480
> >     -       o       o       o       o       ?       o
> o     o       o       o       o       o       o
> > Retry-After         486
> >     -       o       o       o       o       ?       o
> o     o       o       o       o       o       o
> > Retry-After         4xx
> >     ?       ?       ?       ?       ?       o       ?
> ?     ?       ?       ?       ?       ?       ?
> > Route       R       adr     c       c       c       o
> c     o       c       c       c       c       c       c
> c     c
> > RSeq        1xx
> >     -       -       -       ?       o       ?       o
> -     -       ?       ?       -       o       ?
> > RSeq        R
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       -
> > RSeq        r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       -
> > Security-Client     R       ard     -       o       -
> ?     o       o       o       o       ?       ?       ?
> o     o       o
> > Security-Server     421
> >     -       o       -       ?       o       o       o
> o     ?       ?       ?       o       o       o
> > Security-Server     494
> >     -       o       -       ?       o       o       o
> o     ?       ?       ?       o       o       o
> > Security-Verify     R       ard     -       o       -
> ?     o       o       o       o       ?       ?       ?
> o     o       o
> > Server      r
> >     -       o       o       o       o       o       o
> o     o       o       o       o       o       o
> > Service-Route       2xx     ar      -       -       -
> ?     -       ?       ?       -       -       ?       ?
> o     ?       ?
> > Session-Expires     2xx     ar      -       -       -
> ?     o       ?       -       -       -       ?       ?
> -     -       o
> > Session-Expires     R       amr     -       -       -
> ?     o       ?       -       -       -       ?       ?
> -     -       o
> > SIP-ETag    2xx
> >     -       -       -       -       -       -       -
> -     -       m       -       -       -       -
> > SIP-If-Match        R
> >     -       -       -       -       -       -       -
> -     -       o       -       -       -       -
> > Subject     R
> >     -       -       -       o       o       o       -
> -     -       o       -       -       -       -
> > Subject     r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       -
> > Subscription-State  R
> >     -       -       -       ?       -       ?       m
> -     -       ?       ?       -       -       -
> > Subscription-State  r
> >     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       -
> > Supported   2xx
> >     -       o       o       ?       m*      ?       o
> m*    o       o       o       o       o       o
> > Supported   R
> >     -       o       o       ?       m*      ?       o
> o     o       o       o       o       o       o
> > Target-Dialog       R       -       -       -       -
> -     o       -       -       -       -       -       o
> -     o       -
> > Timestamp   R
> >     o       o       o       o       o       o       o
> o     o       o       o       o       o       o
> > Timestamp   r
> >     o       o       o       o       o       o       o
> o     o       o       o       o       o       o
> > To  c       r       m       m       m       m       m
> m     m       m       m       m       m       m       m       m
> > Trigger-Consent     ?       ?       ?       ?       ?
> ?     ?       ?       ?       ?       ?       ?       ?
> ?     ?       ?
> > Unsupported         420
> >     -       m       -       o       m       o       o
> m     m       o       o       m       o       m
> > User-Agent  R
> >     o       o       o       o       o       o       o
> o     o       o       o       o       o       o
> > User-Agent  r
> >     o       o       o       o       o       o       o
> o     o       o       o       o       o       o
> > Via         R       amr     m       m       m       ?
> m     m       ?       m       ?       m       ?       m
> ?     m
> > Via         rc      dr      m       m       m       ?
> m     m       ?       m       ?       m       ?       m
> ?     m
> > Warning     r
> >     -       o       o       o       o       o       o
> o     o       o       o       o       o       o
> > WWW-Authenticate    401     ar      -       m       -
> o     m       m       m       m       m       m       m
> m     m       m
> > WWW-Authenticate    407     ar      -       o       -
> ?     o       o       ?       o       ?       o       o
> o     ?       o
> >
> >
> >
> > See all those question marks? Someone needs to come up with actual
> > values for each of those cells. Some are going to be pretty easy to
> > answer; others will require substantial research and/or debate.
> >
> > It seems like a lot of effort to me. Are we certain the
> return on our
> > investment of time will be worth it?
> >
> > /a
> >
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

From gonzalo.camarillo@ericsson.com  Tue Mar  2 00:54:14 2010
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Date: Tue, 02 Mar 2010 10:54:11 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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Hi Christer,

> I agree that we should be careful, and try to ensure that we have the
> needed commitment from people.

in addition to the commitment bit, which is obviously essential, I was
talking about the usefulness of the exercise. I want to make sure people
commit to something useful. So, discussing the usefulness of the whole
thing would be, well, useful :o)

> But, note that it is not only about interoperability. If we do not do
> this, I still think we need some guideline on how to deal with the
> header/method tables whenever we define a new method and/or header
> field. Do we list all methods/headers that have currently been
> specified at that moment, or?

Providing guidance would be easy. Fixing the whole table would require
more work... we need to have a good ROI in terms of cycles committed and
usefulness to the community.

Cheers,

Gonzalo

From christer.holmberg@ericsson.com  Tue Mar  2 01:06:13 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Date: Tue, 2 Mar 2010 10:06:09 +0100
Thread-Topic: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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Hi,=20

>>I agree that we should be careful, and try to ensure that we have the=20
>>needed commitment from people.
>=20
>in addition to the commitment bit, which is obviously=20
>essential, I was talking about the usefulness of the=20
>exercise. I want to make sure people commit to something=20
>useful. So, discussing the usefulness of the whole thing=20
>would be, well, useful :o)
>=20
>>But, note that it is not only about interoperability. If we=20
>>do not do this, I still think we need some guideline on how to deal with =
the=20
>>header/method tables whenever we define a new method and/or header=20
>>field. Do we list all methods/headers that have currently been=20
>>specified at that moment, or?
>=20
>Providing guidance would be easy. Fixing the whole table=20
>would require more work... we need to have a good ROI in=20
>terms of cycles committed and usefulness to the community.

Sure.

Adam, are you planning to request agenda time in order to discuss this in A=
naheim?

Regards,

Christer

From christer.holmberg@ericsson.com  Tue Mar  2 03:45:58 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Date: Tue, 2 Mar 2010 12:45:43 +0100
Thread-Topic: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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Hi,=20

>>I agree that we should be careful, and try to ensure that we have the=20
>>needed commitment from people.
>=20
>in addition to the commitment bit, which is obviously=20
>essential, I was talking about the usefulness of the=20
>exercise. I want to make sure people commit to something=20
>useful. So, discussing the usefulness of the whole thing=20
>would be, well, useful :o)

Yes, having useful discussions is normally useful :)

Based on the comments on this issue so far, I DO think people consider it u=
seful. But, more input is welcome.

>>But, note that it is not only about interoperability. If we=20
>>do not do this, I still think we need some guideline on how to deal with =
the=20
>>header/method tables whenever we define a new method and/or header=20
>>field. Do we list all methods/headers that have currently been=20
>>specified at that moment, or?
>=20
>Providing guidance would be easy.

Do you have any ideas on what such guidance would say?

Regards,

Christer

From adam@nostrum.com  Tue Mar  2 06:25:43 2010
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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On 3/1/10 23:52, Mar 1, Christer Holmberg wrote:
> Hi,
>
>    
>> I would like to see #2 as a BCP. Not just for OPTIONS, but for all cases that a proxy/registrar might fork an inbound request.
>>      
> Just to clarify, I guess you don't include dialog establsihment requests (INVITE, SUBSCRIBE)? They should always be forwarded/forked even if addressed to the AOR for which the registrar is responseible.
>    

Nope -- I mean INVITE and SUBSCRIBE as well. It's somewhat orthogonal to 
the discussion at hand, but it solves the HERFP problem in the general 
case, and it solves it perfectly.

/a

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Subject: Re: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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On 3/2/10 03:06, Mar 2, Christer Holmberg wrote:
> Adam, are you planning to request agenda time in order to discuss this 
> in Anaheim?

I hadn't really planned on it, but if people want to spend face-to-face 
time on the topic, I'll put something together.

/a

From dean.willis@softarmor.com  Tue Mar  2 06:58:55 2010
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------- Original message -------
> From: Paul Kyzivat <pkyzivat@cisco.com>
> My recommendation for a BCP would be:
> - honor Request-Disposition if present. Else
> - return 3xx if there is more than one contact
> - either 3xx or forward is ok if there is only one contact
>
  What hapened to a 3xx with temp-gruus instead of contacts?

--
Dean 

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Date: Tue, 2 Mar 2010 16:01:51 +0100
Message-ID: <9ae56b1e1003020701t731ca950s1c6968256399ebc3@mail.gmail.com>
From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Adam Roach <adam@nostrum.com>
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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You mean bypassing all the RFC3261 forking logic in the proxy responsible
for the domain and pushing back that burden to the UAC?

/Hans Erik van Elburg


On Tue, Mar 2, 2010 at 3:25 PM, Adam Roach <adam@nostrum.com> wrote:

> On 3/1/10 23:52, Mar 1, Christer Holmberg wrote:
>
>> Hi,
>>
>>
>>
>>> I would like to see #2 as a BCP. Not just for OPTIONS, but for all cases
>>> that a proxy/registrar might fork an inbound request.
>>>
>>>
>> Just to clarify, I guess you don't include dialog establsihment requests
>> (INVITE, SUBSCRIBE)? They should always be forwarded/forked even if
>> addressed to the AOR for which the registrar is responseible.
>>
>>
>
> Nope -- I mean INVITE and SUBSCRIBE as well. It's somewhat orthogonal to
> the discussion at hand, but it solves the HERFP problem in the general case,
> and it solves it perfectly.
>
> /a
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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You mean bypassing all the RFC3261 forking logic in the proxy responsible f=
or the domain and pushing back that burden to the UAC?<br><br clear=3D"all"=
>/Hans Erik van Elburg<br>
<br><br><div class=3D"gmail_quote">On Tue, Mar 2, 2010 at 3:25 PM, Adam Roa=
ch <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com">adam@nostrum.c=
om</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"bord=
er-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-l=
eft: 1ex;">
<div class=3D"im">On 3/1/10 23:52, Mar 1, Christer Holmberg wrote:<br>
</div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb=
(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br>
<br>
 =A0 <br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I would like to see #2 as a BCP. Not just for OPTIONS, but for all cases th=
at a proxy/registrar might fork an inbound request.<br>
 =A0 =A0 <br>
</blockquote><div class=3D"im">
Just to clarify, I guess you don&#39;t include dialog establsihment request=
s (INVITE, SUBSCRIBE)? They should always be forwarded/forked even if addre=
ssed to the AOR for which the registrar is responseible.<br>
 =A0 <br>
</div></blockquote>
<br>
Nope -- I mean INVITE and SUBSCRIBE as well. It&#39;s somewhat orthogonal t=
o the discussion at hand, but it solves the HERFP problem in the general ca=
se, and it solves it perfectly.<br><font color=3D"#888888">
<br>
/a</font><div><div></div><div class=3D"h5"><br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br>

--00163613756a5283bd0480d2a5bb--

From dean.willis@softarmor.com  Tue Mar  2 07:09:57 2010
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Date: Tue, 02 Mar 2010 09:09:43 -0600
From: "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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------- Original message -------
> From: Christer Holmberg <christer.holmberg@ericsson.com>
>
>>I would like to see #2 as a BCP. Not just for OPTIONS, but for all cases 
>> that a proxy/registrar might fork an inbound request.
>
> Just to clarify, I guess you don't include dialog establsihment requests 
> (INVITE, SUBSCRIBE)? They should always be forwarded/forked even if 
> addressed to the AOR for which the registrar is responseible.

I think he really meant ALL REQUESTS. Replace forked-up behavior with more 
predictable redirection. When you really need downstream-node forking, you 
probably need to do it in a B28UA to handle all the possible responses, 
including media-path responses. Look at how Google Voice handles forking 
for an example. We have known for years that forking does not play well 
with PSTN gateways and early media, but have persistently failed to deal 
with that problem.

--
Dean


From pkyzivat@cisco.com  Tue Mar  2 07:27:05 2010
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Dean Willis wrote:
> 
> 
> ------- Original message -------
>> From: Paul Kyzivat <pkyzivat@cisco.com>
>> My recommendation for a BCP would be:
>> - honor Request-Disposition if present. Else
>> - return 3xx if there is more than one contact
>> - either 3xx or forward is ok if there is only one contact
>>
>  What hapened to a 3xx with temp-gruus instead of contacts?

AFAIK it would be inappropriate for the proxy to select and return 
gruus, especially temp gruus in a 3xx. I think the UAS would have to do 
that. While the proxy could choose the perm gruu, its a policy decision 
to use the perm/temp gruu, and normally I think it would be the UAS that 
would make that decision.

	Thanks,
	Paul

From adam@nostrum.com  Tue Mar  2 07:40:26 2010
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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On 3/2/10 09:26, Mar 2, Paul Kyzivat wrote:
> AFAIK it would be inappropriate for the proxy to select and return 
> gruus, especially temp gruus in a 3xx. I think the UAS would have to 
> do that. While the proxy could choose the perm gruu, its a policy 
> decision to use the perm/temp gruu, and normally I think it would be 
> the UAS that would make that decision.

I'm not sure why -- in this context, it's just an opaque cookie that the 
proxy/registrar minted for this one-time use and handed out.

I don't think people are proposing that this be a temp-gruu that the 
proxy actually issued to the terminal, so the terminal's policy about 
what to do with the temp-gruus it receives is maintained.

/a

From pkyzivat@cisco.com  Tue Mar  2 08:00:30 2010
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Adam Roach wrote:
> On 3/2/10 09:26, Mar 2, Paul Kyzivat wrote:
>> AFAIK it would be inappropriate for the proxy to select and return 
>> gruus, especially temp gruus in a 3xx. I think the UAS would have to 
>> do that. While the proxy could choose the perm gruu, its a policy 
>> decision to use the perm/temp gruu, and normally I think it would be 
>> the UAS that would make that decision.
> 
> I'm not sure why -- in this context, it's just an opaque cookie that the 
> proxy/registrar minted for this one-time use and handed out.
> 
> I don't think people are proposing that this be a temp-gruu that the 
> proxy actually issued to the terminal, so the terminal's policy about 
> what to do with the temp-gruus it receives is maintained.

Oh, then you are postulating a new mechanism, similar to the temp gruu 
mechanism, but with (presumably) different value.

Certainly such is possible. We just haven't defined any such mechanism 
yet. I was going to say that it wouldn't even need to be standardized. 
But gruus tend to break a lot of other things unless special account is 
taken of them. So it probably does need to be specified.

This is likely a long road to go down, so we should only start down it 
if we are prepared for such a long journey.

	Thanks,
	Paul

From kpfleming@digium.com  Tue Mar  2 09:08:32 2010
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Editorial comments:

Section 1:

"There has been some confusion among implentors regarding how a UAS"

... should be 'implementors'.

Section 3.7:

SDP4 is improperly formatted.

Content comments:

Section 3.3:

"A change to the session state is considered to have been executed if
  an offer/answer without preconditions [RFC4032] for the stream has
  completed successfully or the UAs have exchanged media using the new
  parameters.  Connection establishment messages (e.g., TCP SYN),
  connectivity checks (e.g., when using ICE [I-D.ietf-mmusic-ice]), and
  any other messages used in the process of meeting the preconditions
  for a stream are not considered media."

This is the issue I spoke briefly with Gonzalo about in Hiroshima, that
we've seen in the field.

The scenario is that a UAC sends an INVITE with only an audio stream;
the UAS accepts it with 200 OK and the ACK is sent by the UAC. Later,
one party (does not matter which) sends a re-INVITE requesting that that
single stream be changed from audio to image/t38, using the *same port
number* to receive the media. Because the port number did not change,
and there is likely already audio media in RTP packets in-flight towards
that endpoint, as soon as any of those RTP packets are received by that
endpoint, it will consider that change to have been executed by the
party that received the re-INVITE, even before receiving any SIP
response to the re-INVITE itself. If the party receiving the re-INVITE
does not want to switch to T.38, it can send a 488 Not Acceptable Here,
but from the point of view of the party that sent the re-INVITE, this is
too late... the change has been executed. This will occur any time the
re-INVITE changes an existing media stream in a way that cannot easily
be detected via packet inspection and does *not* change port numbers.

Given that, I see only two possibilities to improve the interoperability
situation:

1) We include a normative statement in this document that specifies that
a re-INVITE that changes the content-type of a media stream SHOULD NOT
use the same port number as was previously used for that media stream,
so that the offerer can explicitly determine whether the answerer is
executing the change

or

2) We include a set of statements in this document telling implementors
that if they receive a re-INVITE that changes the content type of an
existing media stream and does not change port numbers, that they will
not be effectively able to send a rejection of that re-INVITE if the
previous stream state was such that media was already flowing, and that
they should either use UPDATE within the re-INVITE transaction (as
documented in this draft) to attempt to revert to the pre-re-INVITE
state, or they should accept the re-INVITE and then issue one of their
own to revert to the pre-re-INVITE state.

While section 3.4 attempts to address this situation, my concern is that
the unless the UAS takes the lack of port number change into account, it
may send an error response to the re-INVITE under the (valid) assumption
that none of the requested changes have been executed... but the UAC
thinks that they have been executed. Does everyone else on the list
think that section 3.4's language adequately addresses this situation?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

From pkyzivat@cisco.com  Tue Mar  2 09:27:49 2010
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Subject: Re: [sipcore] Comments on draft-ieft-sipcore-reinvite-01
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Kevin,

I don't understand your scenario. IIUC,

- A & B negotiate a voice stream
- A is receiving voice media
- A sends reinvite to B with offer of t.38
- A continues to receive voice media in accord with
   session in effect prior to the reinvite
- B rejects the reinvite with 488
- A and B continue to use voice media

I don't see where there was a problem there. The alternate scenario is

- A & B negotiate a voice stream
- A is receiving voice media
- A sends reinvite to B with offer of t.38
- A continues to receive voice media in accord with
   session in effect prior to the reinvite
- B sends 2xx accepting the change
- B sends t.38 media
- A receives t.38 media and infers change was accepted
- A receives the 2xx
- session continues with t.38.

What is significant above is that A must be prepared to receive *either* 
voice or t.38 on that port for some period of time. If it can't tolerate 
that, then it had better use a different port.

What am I missing?

	Thanks,
	Paul


Kevin P. Fleming wrote:
> Editorial comments:
> 
> Section 1:
> 
> "There has been some confusion among implentors regarding how a UAS"
> 
> ... should be 'implementors'.
> 
> Section 3.7:
> 
> SDP4 is improperly formatted.
> 
> Content comments:
> 
> Section 3.3:
> 
> "A change to the session state is considered to have been executed if
>   an offer/answer without preconditions [RFC4032] for the stream has
>   completed successfully or the UAs have exchanged media using the new
>   parameters.  Connection establishment messages (e.g., TCP SYN),
>   connectivity checks (e.g., when using ICE [I-D.ietf-mmusic-ice]), and
>   any other messages used in the process of meeting the preconditions
>   for a stream are not considered media."
> 
> This is the issue I spoke briefly with Gonzalo about in Hiroshima, that
> we've seen in the field.
> 
> The scenario is that a UAC sends an INVITE with only an audio stream;
> the UAS accepts it with 200 OK and the ACK is sent by the UAC. Later,
> one party (does not matter which) sends a re-INVITE requesting that that
> single stream be changed from audio to image/t38, using the *same port
> number* to receive the media. Because the port number did not change,
> and there is likely already audio media in RTP packets in-flight towards
> that endpoint, as soon as any of those RTP packets are received by that
> endpoint, it will consider that change to have been executed by the
> party that received the re-INVITE, even before receiving any SIP
> response to the re-INVITE itself. If the party receiving the re-INVITE
> does not want to switch to T.38, it can send a 488 Not Acceptable Here,
> but from the point of view of the party that sent the re-INVITE, this is
> too late... the change has been executed. This will occur any time the
> re-INVITE changes an existing media stream in a way that cannot easily
> be detected via packet inspection and does *not* change port numbers.
> 
> Given that, I see only two possibilities to improve the interoperability
> situation:
> 
> 1) We include a normative statement in this document that specifies that
> a re-INVITE that changes the content-type of a media stream SHOULD NOT
> use the same port number as was previously used for that media stream,
> so that the offerer can explicitly determine whether the answerer is
> executing the change
> 
> or
> 
> 2) We include a set of statements in this document telling implementors
> that if they receive a re-INVITE that changes the content type of an
> existing media stream and does not change port numbers, that they will
> not be effectively able to send a rejection of that re-INVITE if the
> previous stream state was such that media was already flowing, and that
> they should either use UPDATE within the re-INVITE transaction (as
> documented in this draft) to attempt to revert to the pre-re-INVITE
> state, or they should accept the re-INVITE and then issue one of their
> own to revert to the pre-re-INVITE state.
> 
> While section 3.4 attempts to address this situation, my concern is that
> the unless the UAS takes the lack of port number change into account, it
> may send an error response to the re-INVITE under the (valid) assumption
> that none of the requested changes have been executed... but the UAC
> thinks that they have been executed. Does everyone else on the list
> think that section 3.4's language adequately addresses this situation?
> 

From kpfleming@digium.com  Tue Mar  2 09:36:34 2010
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Paul Kyzivat wrote:

> I don't see where there was a problem there. The alternate scenario is
> 
> - A & B negotiate a voice stream
> - A is receiving voice media
> - A sends reinvite to B with offer of t.38
> - A continues to receive voice media in accord with
>   session in effect prior to the reinvite
> - B sends 2xx accepting the change
> - B sends t.38 media
> - A receives t.38 media and infers change was accepted
> - A receives the 2xx
> - session continues with t.38.
> 
> What is significant above is that A must be prepared to receive *either*
> voice or t.38 on that port for some period of time. If it can't tolerate
> that, then it had better use a different port.
> 
> What am I missing?

That is in fact the issue, and the problem we see in the field is that
because there is no such thing as a protocol number at the UDP level,
and T.38 doesn't use RTP for transport, there is no reliable way to
discriminate an incoming packet to determine whether it is RTP carrying
G.711 PCMU audio or UDPTL carrying a T.38 IFP. At best you can attempt
to completely decode it as both forms, and then try to decide what you
think it is, but it could very easily be the case that the first 'n'
bytes of a UDPTL frame when interpreted as an RTP header contain a valid
payload number for the existing audio session in the right location, and
then there's really no way the endpoint can know for sure.

This really falls into the 'best practices' area I suppose, and we'll
certainly consider including a recommendation of this type in the SIP
Forum work we are doing, but it seems that given the RFC3261 language
about a change being 'executed' as soon as the offerer receives media
using the 'new parameters', implementors should be cautioned (if not
strongly warned) about sending a re-INVITE that would not allow them to
determine whether the new parameters are being used or not. If the UAC
and UAS can't explicitly indicate to each other that the new parameters
are in use because of this problem, then implicit behavior will result,
and the endpoints will get out of sync. This can be completely avoided
with very little effort on the part of the implementor, though, which
was my goal in raising this issue.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

From john.elwell@siemens-enterprise.com  Tue Mar  2 09:45:11 2010
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I read this document again and didn't find anything to prevent it progressi=
ng (although I would not claim to have done a thorough review).

Minor comments and nits:

- Section 1:
"A UAC willing to
   act the offerer"
Change "act" to "act as".

- Section 2: Where "UAC" and "UAS" expanded, it would probably be worth sta=
ting that the terms are used with respect to INVITE and re-INVITE transacti=
ons only. Other transactions within an INVITE or re-INVITE transaction can =
be in the reverse direction and the roles are reversed.

- Section 3.1:
"SDP4:
            m=3Daudio 31000 RTP/AVP 0
            c=3DIN IP4 192.0.2.5
            m=3Dvideo 0 RTP/AVP 31
            c=3DIN IP4 192.0.2.2"=20
It seems unreasonable to include in an answer, below a rejected m-line, a c=
-line echoing the c-line in the offer.

- In the last example in 3.1, it should perhaps be pointed out that SDP5 (i=
n the UPDATE request) is an offer. Also the contents of SDP6 (SDP answer) s=
hould perhaps also be shown for completeness.

- In section 3.3:
"During a session establishment, the UAS can wait for
   using the media parameters until "
Change "wait for" to "wait before" (or alternatively "delay").

- In section 3.7:
"The proxy relays the UPDATE request (9) to UA1.  The UPDATE request
   (9) arrives at UA1 before the 4xx response (7) that had been
   previously sent.  UA2 accepts the changes in the UPDATE request and
   returns a 200 (OK) response (10) to it ."
I think the last sentence should begin "UA1 accepts the changes..."

- "   SDP4: m=3Daudio 20000 RTP/AVP 0 a=3Dsendonly m=3Dvideo 30002 RTP/AVP =
31
   a=3Dinactive"
Line breaks missing (c.f., the other examples).

- In section 4.4:
"In this document, reliable provisional responses are
      those that use the mechanism defined in RFC 3262 [RFC3262] on any
      other SIP-based mechanism that may be specified in the future."
Change "on" to "or".

- Change "If instead sending" to "If instead of sending"


John


From pkyzivat@cisco.com  Tue Mar  2 09:58:57 2010
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Subject: Re: [sipcore] Comments on draft-ieft-sipcore-reinvite-01
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OK. I agree this falls into the category of a "logical necessity" that 
could be well handled by a BCP.

	Thanks,
	Paul

Kevin P. Fleming wrote:
> Paul Kyzivat wrote:
> 
>> I don't see where there was a problem there. The alternate scenario is
>>
>> - A & B negotiate a voice stream
>> - A is receiving voice media
>> - A sends reinvite to B with offer of t.38
>> - A continues to receive voice media in accord with
>>   session in effect prior to the reinvite
>> - B sends 2xx accepting the change
>> - B sends t.38 media
>> - A receives t.38 media and infers change was accepted
>> - A receives the 2xx
>> - session continues with t.38.
>>
>> What is significant above is that A must be prepared to receive *either*
>> voice or t.38 on that port for some period of time. If it can't tolerate
>> that, then it had better use a different port.
>>
>> What am I missing?
> 
> That is in fact the issue, and the problem we see in the field is that
> because there is no such thing as a protocol number at the UDP level,
> and T.38 doesn't use RTP for transport, there is no reliable way to
> discriminate an incoming packet to determine whether it is RTP carrying
> G.711 PCMU audio or UDPTL carrying a T.38 IFP. At best you can attempt
> to completely decode it as both forms, and then try to decide what you
> think it is, but it could very easily be the case that the first 'n'
> bytes of a UDPTL frame when interpreted as an RTP header contain a valid
> payload number for the existing audio session in the right location, and
> then there's really no way the endpoint can know for sure.
> 
> This really falls into the 'best practices' area I suppose, and we'll
> certainly consider including a recommendation of this type in the SIP
> Forum work we are doing, but it seems that given the RFC3261 language
> about a change being 'executed' as soon as the offerer receives media
> using the 'new parameters', implementors should be cautioned (if not
> strongly warned) about sending a re-INVITE that would not allow them to
> determine whether the new parameters are being used or not. If the UAC
> and UAS can't explicitly indicate to each other that the new parameters
> are in use because of this problem, then implicit behavior will result,
> and the endpoints will get out of sync. This can be completely avoided
> with very little effort on the part of the implementor, though, which
> was my goal in raising this issue.
> 

From adam@nostrum.com  Tue Mar  2 11:47:42 2010
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
> You mean bypassing all the RFC3261 forking logic in the proxy 
> responsible for the domain and pushing back that burden to the UAC?

Burden?

The client already has logic to handle this. Servers can throw 300-class 
responses back for INVITEs whenever they want. So the development cost 
is already built into every SIP terminal in the world. There is no 
additional software burden.

In terms of CPU burden... An average large network server has to handle 
several thousand or several tens of thousands of transactions per 
second. Your average terminal probably has to handle several dozen a 
day. Let's aim high and ballpark it around 0.001 transactions per second.

You honestly think that moving some small portion of processing from the 
thing doing 10,000 TPS to the thing doing 0.001 TPS is going in the 
wrong direction?

Or it could be that I completely missed your point, since both of these 
interpretations of "burden" make your assertion nonsense. Would you like 
to clarify your statement?

/a

From christer.holmberg@ericsson.com  Tue Mar  2 12:40:39 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>, Adam Roach <adam@nostrum.com>
Date: Tue, 2 Mar 2010 21:40:37 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,=20

>>>AFAIK it would be inappropriate for the proxy to select and return=20
>>>gruus, especially temp gruus in a 3xx. I think the UAS would have to=20
>>>do that. While the proxy could choose the perm gruu, its a policy=20
>>>decision to use the perm/temp gruu, and normally I think it would be=20
>>>the UAS that would make that decision.
>>>=20
>>I'm not sure why -- in this context, it's just an opaque cookie that=20
>>the proxy/registrar minted for this one-time use and handed out.
>>=20
>>I don't think people are proposing that this be a temp-gruu that the=20
>>proxy actually issued to the terminal, so the terminal's policy about=20
>>what to do with the temp-gruus it receives is maintained.
>
>Oh, then you are postulating a new mechanism, similar to the temp gruu mec=
hanism, but with (presumably) different value.
>
>Certainly such is possible. We just haven't defined any such mechanism yet=
. I was going to say that it wouldn't even need to be standardized.=20
>But gruus tend to break a lot of other things unless special account is ta=
ken of them. So it probably does need to be specified.
>
>This is likely a long road to go down, so we should only start down it if =
we are prepared for such a long journey.

I think we shall start from the fact that the registrar returns a Contact h=
eader field for the UAS(s) in the 3xx response.

Then, if the registrar includes the actual UAS contact information, or some=
thing else, could be a policy decission. It is not really bound to the 3xx =
mechanism itself - as long as the UAC can use whatever was returned to cont=
act the UAS(s).

Regarding the usage of gruus, I don't see any problem/risk in returning the=
 perm-gruu, but agin, that could be a policy decission.

Regards,

Christer


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Subject: Re: [sipcore] SIP Table 2 (was Re:  3265bis and Call-Info)
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At 08:27 AM 3/2/2010, Adam Roach wrote:
>On 3/2/10 03:06, Mar 2, Christer Holmberg wrote:
>>Adam, are you planning to request agenda time in order to discuss 
>>this in Anaheim?
>
>I hadn't really planned on it, but if people want to spend 
>face-to-face time on the topic, I'll put something together.

Having at least some meeting time is a good way to have a high 
bandwidth exchange about the pros and cons of this effort, and 
whether or not it is worth the cycles.

bluntly - I think there are many individuals that don't or aren't 
right now writing SIPCORE IDs, and do few of the reviews -- yet are 
able (i.e., have the necessary clue about SIP) and would be willing 
to spend cycles doing this effort. This would therefore (if true) not 
necessarily drain from existing efforts to get this Table 2 effort done.

This isn't ignoring the ultimate review by the WG on any such effort, 
and doesn't ignore the impact on the chairs.

That said -- this would fix an unsettled item this WG has had since 
it's inception, as was the case for the SIP WG once the first 
extension to SIP occurred after 3261 was published and Table 2 became 
"not accurate".

my 1.89 cents (before deflation)

james


>/a
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Christer,

I agree - its mostly policy.

If somebody referenced the AoR, then it seems like it ought to be ok to 
return the public gruu, since that doesn't disclose anything the caller 
didn't already know. However that may not be true if the call was 
forwarded. E.g.

Alice calls Bob, who forwards the call to Carol.
However, Bob may not want Alice to know it was forwarded to Carol, or 
Carol might be willing to take the call for Bob, but doesn't want her 
AoR disclosed to Alice.

If this were an INVITE reaching Carol's phone, it could use a temp gruu 
to hide identity. The proxy for Carol could perhaps do that on her 
behalf based on policy, but there might need to be new gruu-like 
mechanisms to do so.

Nevertheless, I think we can simply specify that the 3xx contain 
contacts that reference each candidate UA.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi, 
> 
>>>> AFAIK it would be inappropriate for the proxy to select and return 
>>>> gruus, especially temp gruus in a 3xx. I think the UAS would have to 
>>>> do that. While the proxy could choose the perm gruu, its a policy 
>>>> decision to use the perm/temp gruu, and normally I think it would be 
>>>> the UAS that would make that decision.
>>>>
>>> I'm not sure why -- in this context, it's just an opaque cookie that 
>>> the proxy/registrar minted for this one-time use and handed out.
>>>
>>> I don't think people are proposing that this be a temp-gruu that the 
>>> proxy actually issued to the terminal, so the terminal's policy about 
>>> what to do with the temp-gruus it receives is maintained.
>> Oh, then you are postulating a new mechanism, similar to the temp gruu mechanism, but with (presumably) different value.
>>
>> Certainly such is possible. We just haven't defined any such mechanism yet. I was going to say that it wouldn't even need to be standardized. 
>> But gruus tend to break a lot of other things unless special account is taken of them. So it probably does need to be specified.
>>
>> This is likely a long road to go down, so we should only start down it if we are prepared for such a long journey.
> 
> I think we shall start from the fact that the registrar returns a Contact header field for the UAS(s) in the 3xx response.
> 
> Then, if the registrar includes the actual UAS contact information, or something else, could be a policy decission. It is not really bound to the 3xx mechanism itself - as long as the UAC can use whatever was returned to contact the UAS(s).
> 
> Regarding the usage of gruus, I don't see any problem/risk in returning the perm-gruu, but agin, that could be a policy decission.
> 
> Regards,
> 
> Christer
> 
> 

From christer.holmberg@ericsson.com  Tue Mar  2 12:57:30 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Dean Willis' <dean.willis@softarmor.com>
Date: Tue, 2 Mar 2010 21:57:28 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,=20

>>>I would like to see #2 as a BCP. Not just for OPTIONS, but for all=20
>>>cases  that a proxy/registrar might fork an inbound request.
>>
>>Just to clarify, I guess you don't include dialog establsihment=20
>>requests (INVITE, SUBSCRIBE)? They should always be forwarded/forked=20
>>even if addressed to the AOR for which the registrar is responseible.
>
>I think he really meant ALL REQUESTS. Replace forked-up behavior with more=
 predictable redirection. When you really need downstream-node forking, you=
 probably need to do it in a B28UA to handle all the=20
>possible responses, including media-path responses. Look at how Google Voi=
ce handles forking for an example. We have known for years that forking doe=
s not play well with PSTN gateways and early media,=20
>but have persistently failed to deal with that problem.

I think serial forking can normally be handled pretty well, when it comes t=
o early media etc.

The problem is generally parallel forking.

Regards,

Christer



From christer.holmberg@ericsson.com  Tue Mar  2 13:10:28 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Adam Roach' <adam@nostrum.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Tue, 2 Mar 2010 22:10:26 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,

It sure is an interesting idea. But, apart from the fact that it would give=
 the UAC total control on which UAS(s) to contact, when and in which order =
to contact them, would it bring much advantage from using parallel forking,=
 perhaps together with feature tags in order to "control" the forking?=20

Regards,

Christer


-----Original Message-----
From: Adam Roach [mailto:adam@nostrum.com]=20
Sent: 2. maaliskuuta 2010 21:48
To: Hans Erik van Elburg
Cc: Christer Holmberg; SIPCORE
Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?

On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
> You mean bypassing all the RFC3261 forking logic in the proxy=20
> responsible for the domain and pushing back that burden to the UAC?

Burden?

The client already has logic to handle this. Servers can throw 300-class re=
sponses back for INVITEs whenever they want. So the development cost is alr=
eady built into every SIP terminal in the world. There is no additional sof=
tware burden.

In terms of CPU burden... An average large network server has to handle sev=
eral thousand or several tens of thousands of transactions per second. Your=
 average terminal probably has to handle several dozen a day. Let's aim hig=
h and ballpark it around 0.001 transactions per second.

You honestly think that moving some small portion of processing from the th=
ing doing 10,000 TPS to the thing doing 0.001 TPS is going in the wrong dir=
ection?

Or it could be that I completely missed your point, since both of these int=
erpretations of "burden" make your assertion nonsense. Would you like to cl=
arify your statement?

/a

From pkyzivat@cisco.com  Tue Mar  2 13:29:40 2010
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Christer Holmberg wrote:
> Hi,
> 
> It sure is an interesting idea. But, apart from the fact that it would give the UAC total control on which UAS(s) to contact, when and in which order to contact them, would it bring much advantage from using parallel forking, perhaps together with feature tags in order to "control" the forking? 

It provides the UAC with the opportunity for a richer UI with more 
control for the end user. And it provides a way to do parallel forking 
without the early media issues.

In a way its also a "poor man's presence".

But as long as it isn't mandatory, UACs will still have to deal with all 
the troublesome forking issues too.

It might be better to really use presence instead. The big problem I see 
with that is that typically people will only provide presence data to a 
limited number of their buddies, whereas you potentially want this for 
all callers. That could perhaps be dealt with by giving the presence 
server a "public presence" policy for what you are willing to disclose 
to anybody, and perhaps couple that with a restriction to just polling 
rather than a long term subscription.

	Thanks,
	Paul

> Regards,
> 
> Christer
> 
> 
> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com] 
> Sent: 2. maaliskuuta 2010 21:48
> To: Hans Erik van Elburg
> Cc: Christer Holmberg; SIPCORE
> Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
> 
> On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
>> You mean bypassing all the RFC3261 forking logic in the proxy 
>> responsible for the domain and pushing back that burden to the UAC?
> 
> Burden?
> 
> The client already has logic to handle this. Servers can throw 300-class responses back for INVITEs whenever they want. So the development cost is already built into every SIP terminal in the world. There is no additional software burden.
> 
> In terms of CPU burden... An average large network server has to handle several thousand or several tens of thousands of transactions per second. Your average terminal probably has to handle several dozen a day. Let's aim high and ballpark it around 0.001 transactions per second.
> 
> You honestly think that moving some small portion of processing from the thing doing 10,000 TPS to the thing doing 0.001 TPS is going in the wrong direction?
> 
> Or it could be that I completely missed your point, since both of these interpretations of "burden" make your assertion nonsense. Would you like to clarify your statement?
> 
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From christer.holmberg@ericsson.com  Tue Mar  2 13:29:59 2010
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Hi,

If you use different ports, and you use ICE, I guess that means that you ne=
ed to perform new ICE procedures for that port before you can start using i=
t.

Regards,

Christer=20

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Paul Kyzivat
Sent: 2. maaliskuuta 2010 19:59
To: Kevin P. Fleming
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Comments on draft-ieft-sipcore-reinvite-01

OK. I agree this falls into the category of a "logical necessity" that coul=
d be well handled by a BCP.

	Thanks,
	Paul

Kevin P. Fleming wrote:
> Paul Kyzivat wrote:
>=20
>> I don't see where there was a problem there. The alternate scenario=20
>> is
>>
>> - A & B negotiate a voice stream
>> - A is receiving voice media
>> - A sends reinvite to B with offer of t.38
>> - A continues to receive voice media in accord with
>>   session in effect prior to the reinvite
>> - B sends 2xx accepting the change
>> - B sends t.38 media
>> - A receives t.38 media and infers change was accepted
>> - A receives the 2xx
>> - session continues with t.38.
>>
>> What is significant above is that A must be prepared to receive=20
>> *either* voice or t.38 on that port for some period of time. If it=20
>> can't tolerate that, then it had better use a different port.
>>
>> What am I missing?
>=20
> That is in fact the issue, and the problem we see in the field is that=20
> because there is no such thing as a protocol number at the UDP level,=20
> and T.38 doesn't use RTP for transport, there is no reliable way to=20
> discriminate an incoming packet to determine whether it is RTP=20
> carrying
> G.711 PCMU audio or UDPTL carrying a T.38 IFP. At best you can attempt=20
> to completely decode it as both forms, and then try to decide what you=20
> think it is, but it could very easily be the case that the first 'n'
> bytes of a UDPTL frame when interpreted as an RTP header contain a=20
> valid payload number for the existing audio session in the right=20
> location, and then there's really no way the endpoint can know for sure.
>=20
> This really falls into the 'best practices' area I suppose, and we'll=20
> certainly consider including a recommendation of this type in the SIP=20
> Forum work we are doing, but it seems that given the RFC3261 language=20
> about a change being 'executed' as soon as the offerer receives media=20
> using the 'new parameters', implementors should be cautioned (if not=20
> strongly warned) about sending a re-INVITE that would not allow them=20
> to determine whether the new parameters are being used or not. If the=20
> UAC and UAS can't explicitly indicate to each other that the new=20
> parameters are in use because of this problem, then implicit behavior=20
> will result, and the endpoints will get out of sync. This can be=20
> completely avoided with very little effort on the part of the=20
> implementor, though, which was my goal in raising this issue.
>=20
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

From christer.holmberg@ericsson.com  Tue Mar  2 13:34:28 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Tue, 2 Mar 2010 22:34:26 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,=20

>>It sure is an interesting idea. But, apart from the fact that it would gi=
ve the UAC total control on which UAS(s) to contact, when and in which orde=
r to contact them, would it bring much advantage from=20
>>using parallel forking, perhaps together with feature tags in order to "c=
ontrol" the forking?=20
>
>It provides the UAC with the opportunity for a richer UI with more control=
 for the end user. And it provides a way to do parallel forking without the=
 early media issues.

Perhaps, but then you will also need to take charging/serivce aspects etc i=
nto account. Without a correlation mechanism, I guess all these "parallel f=
orks" from the UAC would be seen as separate calls.

>In a way its also a "poor man's presence".
>
>But as long as it isn't mandatory, UACs will still have to deal with all t=
he troublesome forking issues too.

Again, I think using serial forking would solve lots of problems :)

>It might be better to really use presence instead. The big problem I see w=
ith that is that typically people will only provide presence data to a limi=
ted number of their buddies, whereas you potentially=20
>want this for all callers. That could perhaps be dealt with by giving the =
presence server a "public presence" policy for what you are willing to disc=
lose to anybody, and perhaps couple that with a=20
>restriction to just polling rather than a long term subscription.

Would presence work with PSTN gateways?

Regards,

Christer





> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: 2. maaliskuuta 2010 21:48
> To: Hans Erik van Elburg
> Cc: Christer Holmberg; SIPCORE
> Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
>=20
> On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
>> You mean bypassing all the RFC3261 forking logic in the proxy=20
>> responsible for the domain and pushing back that burden to the UAC?
>=20
> Burden?
>=20
> The client already has logic to handle this. Servers can throw 300-class =
responses back for INVITEs whenever they want. So the development cost is a=
lready built into every SIP terminal in the world. There is no additional s=
oftware burden.
>=20
> In terms of CPU burden... An average large network server has to handle s=
everal thousand or several tens of thousands of transactions per second. Yo=
ur average terminal probably has to handle several dozen a day. Let's aim h=
igh and ballpark it around 0.001 transactions per second.
>=20
> You honestly think that moving some small portion of processing from the =
thing doing 10,000 TPS to the thing doing 0.001 TPS is going in the wrong d=
irection?
>=20
> Or it could be that I completely missed your point, since both of these i=
nterpretations of "burden" make your assertion nonsense. Would you like to =
clarify your statement?
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From pkyzivat@cisco.com  Tue Mar  2 14:00:52 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Christer Holmberg wrote:
> Hi, 
> 
>>> It sure is an interesting idea. But, apart from the fact that it would give the UAC total control on which UAS(s) to contact, when and in which order to contact them, would it bring much advantage from 
>>> using parallel forking, perhaps together with feature tags in order to "control" the forking? 
>> It provides the UAC with the opportunity for a richer UI with more control for the end user. And it provides a way to do parallel forking without the early media issues.
> 
> Perhaps, but then you will also need to take charging/serivce aspects etc into account. Without a correlation mechanism, I guess all these "parallel forks" from the UAC would be seen as separate calls.

Yes. But it doesn't change the reality of the situation. Assuming only 
one of the forks answers, the others are just "failed" call attempts 
that AFAIK are not normally charged.

>> In a way its also a "poor man's presence".
>>
>> But as long as it isn't mandatory, UACs will still have to deal with all the troublesome forking issues too.
> 
> Again, I think using serial forking would solve lots of problems :)

Sure, but its not always what you want.
Its quite reasonable that you would want your office phone and your cell 
phone to ring in parallel so you are perceived as being equally 
responsive regardless of where you are.

>> It might be better to really use presence instead. The big problem I see with that is that typically people will only provide presence data to a limited number of their buddies, whereas you potentially 
>> want this for all callers. That could perhaps be dealt with by giving the presence server a "public presence" policy for what you are willing to disclose to anybody, and perhaps couple that with a 
>> restriction to just polling rather than a long term subscription.
> 
> Would presence work with PSTN gateways?

In principle a PSTN gateway could support presence subscriptions (easier 
if they are just polling calls). But in the end it doesn't have much 
information to offer. Typically there won't be multiple contacts.

So failing to support presence in this case just means falling back to a 
basic call.

	Thanks,
	Paul

> Regards,
> 
> Christer
> 
> 
> 
> 
> 
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: 2. maaliskuuta 2010 21:48
>> To: Hans Erik van Elburg
>> Cc: Christer Holmberg; SIPCORE
>> Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
>>
>> On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
>>> You mean bypassing all the RFC3261 forking logic in the proxy 
>>> responsible for the domain and pushing back that burden to the UAC?
>> Burden?
>>
>> The client already has logic to handle this. Servers can throw 300-class responses back for INVITEs whenever they want. So the development cost is already built into every SIP terminal in the world. There is no additional software burden.
>>
>> In terms of CPU burden... An average large network server has to handle several thousand or several tens of thousands of transactions per second. Your average terminal probably has to handle several dozen a day. Let's aim high and ballpark it around 0.001 transactions per second.
>>
>> You honestly think that moving some small portion of processing from the thing doing 10,000 TPS to the thing doing 0.001 TPS is going in the wrong direction?
>>
>> Or it could be that I completely missed your point, since both of these interpretations of "burden" make your assertion nonsense. Would you like to clarify your statement?
>>
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> 

From ietf.hanserik@gmail.com  Tue Mar  2 14:47:50 2010
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Date: Tue, 2 Mar 2010 23:47:45 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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--001485e9a5ad7cd75e0480d92756
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Burden as in handling the forking process. Didn't ment to overload that with
lots of negative meaning.

I thought the first part was more interesting. Where every forking proxy
becomes a redirect server.

There are also other consequences, where youre home proxy will always have
the same way of handling youre contacts q-values. UAC's come in a lot of
different flavors, they will all behave slightly different and most probably
some will misbehave. So the UAS's owner loose some predictability about how
incoming calls will be handled. No consistent experience, hard to explain.

Also users will want to have services or policies be executed on their
behalf on their home proxy, how does such approach impact that.

Does such model mean anything to call setup delays, considering lets say
scenarios where there are several intermediary proxies interconnecting some
corporate voip clouds.

Where the 3xx approach is very elegant in some cases, to say as a general
BCP that any forking proxy should send 3xx also for INVITEs in any forking
case seems in that respect over the top.

/Hans Erik van Elburg

On Tue, Mar 2, 2010 at 8:47 PM, Adam Roach <adam@nostrum.com> wrote:

> On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
>
>> You mean bypassing all the RFC3261 forking logic in the proxy responsible
>> for the domain and pushing back that burden to the UAC?
>>
>
> Burden?
>
> The client already has logic to handle this. Servers can throw 300-class
> responses back for INVITEs whenever they want. So the development cost is
> already built into every SIP terminal in the world. There is no additional
> software burden.
>
> In terms of CPU burden... An average large network server has to handle
> several thousand or several tens of thousands of transactions per second.
> Your average terminal probably has to handle several dozen a day. Let's aim
> high and ballpark it around 0.001 transactions per second.
>
> You honestly think that moving some small portion of processing from the
> thing doing 10,000 TPS to the thing doing 0.001 TPS is going in the wrong
> direction?
>
> Or it could be that I completely missed your point, since both of these
> interpretations of "burden" make your assertion nonsense. Would you like to
> clarify your statement?
>
> /a
>

--001485e9a5ad7cd75e0480d92756
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Burden as in handling the forking process. Didn&#39;t ment to overload that=
 with lots of negative meaning.<br><br>I thought the first part was more in=
teresting. Where every forking proxy becomes a redirect server. <br><br>
There are also other consequences, where youre home proxy will always have =
the same way of handling youre contacts q-values. UAC&#39;s come in a lot o=
f different flavors, they will all behave slightly different and most proba=
bly some will misbehave. So the UAS&#39;s owner loose some predictability a=
bout how incoming calls will be handled. No consistent experience, hard to =
explain.<br>
<br>Also users will want to have services or policies be executed on their =
behalf on their home proxy, how does such approach impact that.<br><br>Does=
 such model mean anything to call setup delays, considering lets say scenar=
ios where there are several intermediary proxies interconnecting some corpo=
rate voip clouds.<br>
<br>Where the 3xx approach is very elegant in some cases, to say as a gener=
al BCP that any forking proxy should send 3xx also for INVITEs in any forki=
ng case seems in that respect over the top.=A0 <br><br clear=3D"all">/Hans =
Erik van Elburg<br>

<br><div class=3D"gmail_quote">On Tue, Mar 2, 2010 at 8:47 PM, Adam Roach <=
span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com">adam@nostrum.com</=
a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"border-l=
eft: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left:=
 1ex;">
<div class=3D"im">On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
You mean bypassing all the RFC3261 forking logic in the proxy responsible f=
or the domain and pushing back that burden to the UAC?<br>
</blockquote>
<br></div>
Burden?<br>
<br>
The client already has logic to handle this. Servers can throw 300-class re=
sponses back for INVITEs whenever they want. So the development cost is alr=
eady built into every SIP terminal in the world. There is no additional sof=
tware burden.<br>

<br>
In terms of CPU burden... An average large network server has to handle sev=
eral thousand or several tens of thousands of transactions per second. Your=
 average terminal probably has to handle several dozen a day. Let&#39;s aim=
 high and ballpark it around 0.001 transactions per second.<br>

<br>
You honestly think that moving some small portion of processing from the th=
ing doing 10,000 TPS to the thing doing 0.001 TPS is going in the wrong dir=
ection?<br>
<br>
Or it could be that I completely missed your point, since both of these int=
erpretations of &quot;burden&quot; make your assertion nonsense. Would you =
like to clarify your statement?<br><font color=3D"#888888">
<br>
/a<br>
</font></blockquote></div><br>

--001485e9a5ad7cd75e0480d92756--

From adam@nostrum.com  Tue Mar  2 14:54:24 2010
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Subject: [sipcore] 3265bis Open Issue: Allow-Events
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[as a participant]

In Stockholm, we had a discussion of the open issues in the 3265bis One 
of the topics that we discussed was how we handle advertising support 
for specific templates. The high-level exchange is captured in the 
meeting notes:

http://www.ietf.org/proceedings/75/minutes/sipcore.html#Open%20Issue:%20Allow-Events%20and%20Templates

The high level conclusion from that conversation -- really, a dialog 
between me and Robert -- is that the relatively slow pace of template 
package development (only one in 8 years) makes support for template 
event packages amenable to probing (as opposed to advertising).

This message is a call for consensus around this point. If you agree 
with the proposal, please chime in and say so. Of course, if you think 
we need to do something different, send an email with a counter-proposal 
as well.

Thanks.

/a

From adam@nostrum.com  Tue Mar  2 15:22:57 2010
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Subject: [sipcore] 3265 Open Issue: 202 Response Code
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[as a participant]

In Stockholm, we had a discussion of the open issues in the 3265bis One 
of the topics that we discussed was the utility of the 202 response code 
in RFC 3265. A summary of the conversation can be found in the minutes:

http://www.ietf.org/proceedings/75/minutes/sipcore.html#Open%20Issue:%20The%20Fate%20of%20202

There was no clear agreement in that conversation, and several 
participants seemed to hold very nuanced opinions on the topic.

In order to drive conversation on the topic, I will propose the 
following text as a starting point.

The key change is in Section 8.3.1, which I propose to be updated as 
follows:

    For historical purposes, the 202 (Accepted) response code is
    added to the "Success" header field definition.

    This document does not specify the use of the 202 response code
    in conjunction with the SUBSCRIBE or NOTIFY methods. Previous
    versions of the SIP Events Framework assigned specific semantics
    to the 202 response code. Implementations conformant with this
    specification MUST treat an incoming 202 response as identical
    to a 200 response, and MUST NOT generate 202 response codes to
    SUBSCRIBE or NOTIFY messages.

That is the change in policy that I am proposing. Several other changes 
in the document follow from this change; I list them below.

In section 4.1.2.1, the third paragraph will be truncated after the word 
"immedately." In its revised form, it will simply read:

    This SUBSCRIBE request will be confirmed with a final response.
    200- class responses indicate that the subscription has been
    accepted, and that a NOTIFY will be sent immediately.

In section 4.2.1.1, the 6th and 7th paragraphs (both starting wht "If 
the notifier...") will be combined into a single paragraph reading:

    Once the notifier determines that it has enough information to
    create the subscription (i.e., it understands the event package,
    the subscription pertains to a known resource, and there are no
    other barriers to creating the subscription), it creates the
    subscription and a dialog, and returns a 200 (OK) response.

In section 4.2.1.3, the fifth paragraph will replace "202 Accept"
with "200 OK", with the final result being:

    If the notifier owner is interactively queried to determine
    whether a subscription is allowed, a 200 (OK) response is returned
    immediately.  Note that a NOTIFY message is still formed and
    sent under these circumstances, as described in the previous
    section.

In section 5.4.6, the final sentance of the final paragraph will be 
removed, resulting in the following text:

    Information in this section includes details of how to authenticate
    subscribers and authorization issues for the package.

The first paragraph of section 6.2 will be updated to read:

    The mere act of returning certain 4xx and 6xx responses to
    SUBSCRIBE requests may, under certain circumstances, create
    privacy concerns by revealing sensitive policy information.  In
    these cases, the notifier SHOULD always return a 200 (OK) response.
    While the subsequent NOTIFY message may not convey true state,
    it MUST appear to contain a potentially correct piece of data
    from the point of view of the subscriber, indistinguishable from
    a valid response.  Information about whether a user is authorized
    to subscribe to the requested state is never conveyed back to
    the original user under these circumstances.


/a

From adam@nostrum.com  Tue Mar  2 15:47:36 2010
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Subject: [sipcore] Agenda Requests?
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[as chair]

If you wish to request agenda time for the SIPCORE face-to-face meeting, 
please let Oscar by this upcoming Friday, March 5th. Please be certain 
to copy me and Gonzalo on your requests. So far, we have one request (no 
associated draft) to discuss issues related to the OPTIONS method.

/a

From keith.drage@alcatel-lucent.com  Tue Mar  2 18:01:26 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 3 Mar 2010 03:01:19 +0100
Thread-Topic: [sipcore] 3265 Open Issue: 202 Response Code
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Rather than change for change sake, what problem are we actually trying to =
fix here. The meeting notes refer to HERFP but that is hardly enlightening.

regards

Keith=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: Tuesday, March 02, 2010 11:23 PM
> To: SIPCORE
> Subject: [sipcore] 3265 Open Issue: 202 Response Code
>=20
> [as a participant]
>=20
> In Stockholm, we had a discussion of the open issues in the=20
> 3265bis One of the topics that we discussed was the utility=20
> of the 202 response code in RFC 3265. A summary of the=20
> conversation can be found in the minutes:
>=20
> http://www.ietf.org/proceedings/75/minutes/sipcore.html#Open%2
> 0Issue:%20The%20Fate%20of%20202
>=20
> There was no clear agreement in that conversation, and=20
> several participants seemed to hold very nuanced opinions on=20
> the topic.
>=20
> In order to drive conversation on the topic, I will propose=20
> the following text as a starting point.
>=20
> The key change is in Section 8.3.1, which I propose to be updated as
> follows:
>=20
>     For historical purposes, the 202 (Accepted) response code is
>     added to the "Success" header field definition.
>=20
>     This document does not specify the use of the 202 response code
>     in conjunction with the SUBSCRIBE or NOTIFY methods. Previous
>     versions of the SIP Events Framework assigned specific semantics
>     to the 202 response code. Implementations conformant with this
>     specification MUST treat an incoming 202 response as identical
>     to a 200 response, and MUST NOT generate 202 response codes to
>     SUBSCRIBE or NOTIFY messages.
>=20
> That is the change in policy that I am proposing. Several=20
> other changes in the document follow from this change; I list=20
> them below.
>=20
> In section 4.1.2.1, the third paragraph will be truncated=20
> after the word "immedately." In its revised form, it will simply read:
>=20
>     This SUBSCRIBE request will be confirmed with a final response.
>     200- class responses indicate that the subscription has been
>     accepted, and that a NOTIFY will be sent immediately.
>=20
> In section 4.2.1.1, the 6th and 7th paragraphs (both starting=20
> wht "If the notifier...") will be combined into a single=20
> paragraph reading:
>=20
>     Once the notifier determines that it has enough information to
>     create the subscription (i.e., it understands the event package,
>     the subscription pertains to a known resource, and there are no
>     other barriers to creating the subscription), it creates the
>     subscription and a dialog, and returns a 200 (OK) response.
>=20
> In section 4.2.1.3, the fifth paragraph will replace "202 Accept"
> with "200 OK", with the final result being:
>=20
>     If the notifier owner is interactively queried to determine
>     whether a subscription is allowed, a 200 (OK) response is returned
>     immediately.  Note that a NOTIFY message is still formed and
>     sent under these circumstances, as described in the previous
>     section.
>=20
> In section 5.4.6, the final sentance of the final paragraph=20
> will be removed, resulting in the following text:
>=20
>     Information in this section includes details of how to=20
> authenticate
>     subscribers and authorization issues for the package.
>=20
> The first paragraph of section 6.2 will be updated to read:
>=20
>     The mere act of returning certain 4xx and 6xx responses to
>     SUBSCRIBE requests may, under certain circumstances, create
>     privacy concerns by revealing sensitive policy information.  In
>     these cases, the notifier SHOULD always return a 200 (OK)=20
> response.
>     While the subsequent NOTIFY message may not convey true state,
>     it MUST appear to contain a potentially correct piece of data
>     from the point of view of the subscriber, indistinguishable from
>     a valid response.  Information about whether a user is authorized
>     to subscribe to the requested state is never conveyed back to
>     the original user under these circumstances.
>=20
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Tue Mar  2 21:42:28 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 3 Mar 2010 06:42:24 +0100
Thread-Topic: [sipcore] 3265 Open Issue: 202 Response Code
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Hi,

I think this is a little related to one issue/question a raised a while ago=
, related to the new meaning of 200/202 for SUBSCRIBE.

I assume the 200/202 response would cease the re-transmission of the SUBSCR=
IBE request, but what is the *state* after 200/202 has been received and be=
fore the first NOTIFY has been received?=20

If I remember correctly, dialogs won't be established until the NOTIFY requ=
ests arrive, so is there some "pre-dialog" state between the SUB 200/202 an=
d the first NOTIFY? Or, is there a subscription dialog usage already after =
200/202, eventhough it's not "used" for anything until the NOTIFY comes?

And, for how long will the UAC wait for the first NOTIFY, in case it for wh=
atever reason does not arrive immedialtely?=20

So, if the SUB 200/202 response doesn't do anything else than cease the SUB=
 re-transmission, maybe 202 would be more appropriate then 200? It's not a =
suggestion at this point - just some loud thinking :)

Regards,

Christer




________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Adam=
 Roach [adam@nostrum.com]
Sent: Wednesday, March 03, 2010 1:22 AM
To: SIPCORE
Subject: [sipcore] 3265 Open Issue: 202 Response Code

[as a participant]

In Stockholm, we had a discussion of the open issues in the 3265bis One
of the topics that we discussed was the utility of the 202 response code
in RFC 3265. A summary of the conversation can be found in the minutes:

http://www.ietf.org/proceedings/75/minutes/sipcore.html#Open%20Issue:%20The=
%20Fate%20of%20202

There was no clear agreement in that conversation, and several
participants seemed to hold very nuanced opinions on the topic.

In order to drive conversation on the topic, I will propose the
following text as a starting point.

The key change is in Section 8.3.1, which I propose to be updated as
follows:

    For historical purposes, the 202 (Accepted) response code is
    added to the "Success" header field definition.

    This document does not specify the use of the 202 response code
    in conjunction with the SUBSCRIBE or NOTIFY methods. Previous
    versions of the SIP Events Framework assigned specific semantics
    to the 202 response code. Implementations conformant with this
    specification MUST treat an incoming 202 response as identical
    to a 200 response, and MUST NOT generate 202 response codes to
    SUBSCRIBE or NOTIFY messages.

That is the change in policy that I am proposing. Several other changes
in the document follow from this change; I list them below.

In section 4.1.2.1, the third paragraph will be truncated after the word
"immedately." In its revised form, it will simply read:

    This SUBSCRIBE request will be confirmed with a final response.
    200- class responses indicate that the subscription has been
    accepted, and that a NOTIFY will be sent immediately.

In section 4.2.1.1, the 6th and 7th paragraphs (both starting wht "If
the notifier...") will be combined into a single paragraph reading:

    Once the notifier determines that it has enough information to
    create the subscription (i.e., it understands the event package,
    the subscription pertains to a known resource, and there are no
    other barriers to creating the subscription), it creates the
    subscription and a dialog, and returns a 200 (OK) response.

In section 4.2.1.3, the fifth paragraph will replace "202 Accept"
with "200 OK", with the final result being:

    If the notifier owner is interactively queried to determine
    whether a subscription is allowed, a 200 (OK) response is returned
    immediately.  Note that a NOTIFY message is still formed and
    sent under these circumstances, as described in the previous
    section.

In section 5.4.6, the final sentance of the final paragraph will be
removed, resulting in the following text:

    Information in this section includes details of how to authenticate
    subscribers and authorization issues for the package.

The first paragraph of section 6.2 will be updated to read:

    The mere act of returning certain 4xx and 6xx responses to
    SUBSCRIBE requests may, under certain circumstances, create
    privacy concerns by revealing sensitive policy information.  In
    these cases, the notifier SHOULD always return a 200 (OK) response.
    While the subsequent NOTIFY message may not convey true state,
    it MUST appear to contain a potentially correct piece of data
    from the point of view of the subscriber, indistinguishable from
    a valid response.  Information about whether a user is authorized
    to subscribe to the requested state is never conveyed back to
    the original user under these circumstances.


/a
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

From christer.holmberg@ericsson.com  Tue Mar  2 21:52:09 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 3 Mar 2010 06:52:07 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Hi,

>>>> It sure is an interesting idea. But, apart from the fact that it would=
 give the UAC total control on which UAS(s) to contact, when and in which o=
rder to contact them, would it bring much advantage from
>>>> using parallel forking, perhaps together with feature tags in order to=
 "control" the forking?
>>>It provides the UAC with the opportunity for a richer UI with more contr=
ol for the end user. And it provides a way to do parallel forking without t=
he early media issues.
>>
>>Perhaps, but then you will also need to take charging/serivce aspects etc=
 into account. Without a correlation mechanism, I guess all these "parallel=
 forks" from the UAC would be seen as separate calls.
>
>Yes. But it doesn't change the reality of the situation. Assuming only one=
 of the forks answers, the others are just "failed" call attempt that AFAIK=
 are not normally charged.

True. But, there could be network services etc which are triggered for each=
 INVITE, which might not be the desired behavior, etc etc etc. I am just sa=
ying the potential impacts would need to be studied.

One could say that all this is already allowed, but I doubt many (if any) U=
ACs would start sending multiple new INVITEs if it received a 3xx response =
to the previous INVITE.

>>>In a way its also a "poor man's presence".
>>>
>>>But as long as it isn't mandatory, UACs will still have to deal with all=
 the troublesome forking issues too.
>>
>>Again, I think using serial forking would solve lots of problems :)
>
>Sure, but its not always what you want.
>Its quite reasonable that you would want your office phone and your cell
>phone to ring in parallel so you are perceived as being equally
>responsive regardless of where you are.

Sure. It would be good if the forking proxy, in the case of parallel forkin=
g, could tell the UAS(s) to not send early media.

Regards,

Christer




>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: 2. maaliskuuta 2010 21:48
>> To: Hans Erik van Elburg
>> Cc: Christer Holmberg; SIPCORE
>> Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
>>
>> On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
>>> You mean bypassing all the RFC3261 forking logic in the proxy
>>> responsible for the domain and pushing back that burden to the UAC?
>> Burden?
>>
>> The client already has logic to handle this. Servers can throw 300-class=
 responses back for INVITEs whenever they want. So the development cost is =
already built into every SIP terminal in the world. There is no additional =
software burden.
>>
>> In terms of CPU burden... An average large network server has to handle =
several thousand or several tens of thousands of transactions per second. Y=
our average terminal probably has to handle several dozen a day. Let's aim =
high and ballpark it around 0.001 transactions per second.
>>
>> You honestly think that moving some small portion of processing from the=
 thing doing 10,000 TPS to the thing doing 0.001 TPS is going in the wrong =
direction?
>>
>> Or it could be that I completely missed your point, since both of these =
interpretations of "burden" make your assertion nonsense. Would you like to=
 clarify your statement?
>>
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>=

From dean.willis@softarmor.com  Tue Mar  2 23:11:12 2010
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On Mar 2, 2010, at 1:47 PM, Adam Roach wrote:

> On 3/2/10 09:01, Mar 2, Hans Erik van Elburg wrote:
>> You mean bypassing all the RFC3261 forking logic in the proxy  
>> responsible for the domain and pushing back that burden to the UAC?
>
> Burden?
>
> The client already has logic to handle this. Servers can throw 300- 
> class responses back for INVITEs whenever they want. So the  
> development cost is already built into every SIP terminal in the  
> world. There is no additional software burden.
>
> In terms of CPU burden... An average large network server has to  
> handle several thousand or several tens of thousands of transactions  
> per second. Your average terminal probably has to handle several  
> dozen a day. Let's aim high and ballpark it around 0.001  
> transactions per second.
>
> You honestly think that moving some small portion of processing from  
> the thing doing 10,000 TPS to the thing doing 0.001 TPS is going in  
> the wrong direction?
>
> Or it could be that I completely missed your point, since both of  
> these interpretations of "burden" make your assertion nonsense.  
> Would you like to clarify your statement?
>

Well, somebody has to deal with the burden of resolving all those  
early-media streams and figuring out which one to render to the user.  
This is a hard problem, so whoever gets doing it is going to have to  
devote some resources to doing it right. On the other hand, it's  
arguably impossible for a proxy to do it right, so giving the  
additional state to the UAC needed for it to come up with a reasonable  
solution might not be an unreasonable burden.

--
Dean


From dean.willis@softarmor.com  Tue Mar  2 23:16:41 2010
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On Mar 2, 2010, at 3:10 PM, Christer Holmberg wrote:

>
> Hi,
>
> It sure is an interesting idea. But, apart from the fact that it  
> would give the UAC total control on which UAS(s) to contact, when  
> and in which order to contact them, would it bring much advantage  
> from using parallel forking, perhaps together with feature tags in  
> order to "control" the forking?

Since it gives the UAC a separate O/A cycle for each media set, it at  
least makes it possible for the UA to correlate received media with a  
specific UAS. This would appear to make automated handling of the  
media channels in the UAC much more feasible; for example, the UAC  
respond with media saying "Press 1 if you're REALLY answering this  
call", then only connect the media flow to the user if a "1" is heard  
in response. This is what everything that parallel-forks through PSTN  
gateways pretty much has to do to work right.

--
Dean


From dean.willis@softarmor.com  Tue Mar  2 23:19:20 2010
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On Mar 2, 2010, at 3:34 PM, Christer Holmberg wrote:

>
> Hi,
>
>>> It sure is an interesting idea. But, apart from the fact that it  
>>> would give the UAC total control on which UAS(s) to contact, when  
>>> and in which order to contact them, would it bring much advantage  
>>> from
>>> using parallel forking, perhaps together with feature tags in  
>>> order to "control" the forking?
>>
>> It provides the UAC with the opportunity for a richer UI with more  
>> control for the end user. And it provides a way to do parallel  
>> forking without the early media issues.
>
> Perhaps, but then you will also need to take charging/serivce  
> aspects etc into account. Without a correlation mechanism, I guess  
> all these "parallel forks" from the UAC would be seen as separate  
> calls.

They are, aren't they? If the carrier chooses not to bill for them  
separately, that's up to the carrier.

>
>> In a way its also a "poor man's presence".
>>
>> But as long as it isn't mandatory, UACs will still have to deal  
>> with all the troublesome forking issues too.
>
> Again, I think using serial forking would solve lots of problems :)
>

Explain how serial forking resolves the "PSTN-fork that went to voice  
mail" scenario?


>> It might be better to really use presence instead. The big problem  
>> I see with that is that typically people will only provide presence  
>> data to a limited number of their buddies, whereas you potentially
>> want this for all callers. That could perhaps be dealt with by  
>> giving the presence server a "public presence" policy for what you  
>> are willing to disclose to anybody, and perhaps couple that with a
>> restriction to just polling rather than a long term subscription.
>
> Would presence work with PSTN gateways?

Nope.

--
Dean


From dean.willis@softarmor.com  Tue Mar  2 23:22:39 2010
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On Mar 2, 2010, at 4:00 PM, Paul Kyzivat wrote:

>
>
> Christer Holmberg wrote:
>> Hi,
>>>> It sure is an interesting idea. But, apart from the fact that it  
>>>> would give the UAC total control on which UAS(s) to contact, when  
>>>> and in which order to contact them, would it bring much advantage  
>>>> from using parallel forking, perhaps together with feature tags  
>>>> in order to "control" the forking?
>>> It provides the UAC with the opportunity for a richer UI with more  
>>> control for the end user. And it provides a way to do parallel  
>>> forking without the early media issues.
>> Perhaps, but then you will also need to take charging/serivce  
>> aspects etc into account. Without a correlation mechanism, I guess  
>> all these "parallel forks" from the UAC would be seen as separate  
>> calls.
>
> Yes. But it doesn't change the reality of the situation. Assuming  
> only one of the forks answers, the others are just "failed" call  
> attempts that AFAIK are not normally charged.

But what happens when multiple of them are "answered", either with  
early media or a real session?


> In principle a PSTN gateway could support presence subscriptions  
> (easier if they are just polling calls). But in the end it doesn't  
> have much information to offer. Typically there won't be multiple  
> contacts.

One of my inbound phone numbers forks to as many as 6 different phone  
numbers. All of those phone numbers have their own voice mail system,  
but if the incoming call is not answered by "a human", the system is  
smart enough to route the call to a special voice-mail system for the  
DID number. There have been systems with this functionality for years;  
it just doesn't work with proxy-forking in SIP.

--
Dean

From dean.willis@softarmor.com  Tue Mar  2 23:25:08 2010
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On Mar 2, 2010, at 4:47 PM, Hans Erik van Elburg wrote:

> Burden as in handling the forking process. Didn't ment to overload  
> that with lots of negative meaning.
>
> I thought the first part was more interesting. Where every forking  
> proxy becomes a redirect server.
>
> There are also other consequences, where youre home proxy will  
> always have the same way of handling youre contacts q-values. UAC's  
> come in a lot of different flavors, they will all behave slightly  
> different and most probably some will misbehave. So the UAS's owner  
> loose some predictability about how incoming calls will be handled.  
> No consistent experience, hard to explain.
>
> Also users will want to have services or policies be executed on  
> their behalf on their home proxy, how does such approach impact that.

Then the home proxy has to act as  a B2BUA and do the expansion into  
multiple connected calls itself.

>
> Does such model mean anything to call setup delays, considering lets  
> say scenarios where there are several intermediary proxies  
> interconnecting some corporate voip clouds.
>
> Where the 3xx approach is very elegant in some cases, to say as a  
> general BCP that any forking proxy should send 3xx also for INVITEs  
> in any forking case seems in that respect over the top.

There are non-gateway scenarios where parallel forking actually works.  
We don't find them very often.

--
Dean

From christer.holmberg@ericsson.com  Wed Mar  3 00:05:33 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Date: Wed, 3 Mar 2010 09:05:30 +0100
Thread-Topic: INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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Subject: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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=20
Hi,

I've changed the subject.

>>It sure is an interesting idea. But, apart from the fact that it would=20
>>give the UAC total control on which UAS(s) to contact, when and in=20
>>which order to contact them, would it bring much advantage=20
>>from using parallel forking, perhaps together with feature tags in order =
to=20
>>"control" the forking?
>=20
>Since it gives the UAC a separate O/A cycle for each media=20
>set, it at least makes it possible for the UA to correlate=20
>received media with a specific UAS. This would appear to make=20
>automated handling of the media channels in the UAC much more=20
>feasible; for example, the UAC respond with media saying=20
>"Press 1 if you're REALLY answering this call", then only=20
>connect the media flow to the user if a "1" is heard in=20
>response. This is what everything that parallel-forks through=20
>PSTN gateways pretty much has to do to work right.

I think that such call selection can also be done by an B2BUA in the networ=
k.

Keep in mind that even if only one specific media stream is played to the u=
ser behind the UAC, the fact that multiple media streams are sent towards t=
he UAC can cause resource issues. So, in my opinion the best thing is to tr=
y to only send the media stream towards the UAC that will be played to the =
user.

>>>>It sure is an interesting idea. But, apart from the fact that it=20
>>>>would give the UAC total control on which UAS(s) to contact, when=20
>>>>and in which order to contact them, would it bring much advantage=20
>>>>from using parallel forking, perhaps together with feature tags in=20
>>>>order to "control" the forking?
>>>
>>>It provides the UAC with the opportunity for a richer UI with more=20
>>>control for the end user. And it provides a way to do parallel=20
>>>forking without the early media issues.
>>
>>Perhaps, but then you will also need to take charging/serivce aspects=20
>>etc into account. Without a correlation mechanism, I guess all these=20
>>"parallel forks" from the UAC would be seen as separate calls.
>
>They are, aren't they? If the carrier chooses not to bill for them separat=
ely, that's up to the carrier.

Yes, but how does the carrier do that?

Also, applying the same service (e.g. some kind of anouncement) to all thos=
e "calls" is not very optimized usage of service resources.

>>>In a way its also a "poor man's presence".
>>>
>>>But as long as it isn't mandatory, UACs will still have to deal =20
>>>with all the troublesome forking issues too.
>>
>>Again, I think using serial forking would solve lots of problems :)
>
>
>Explain how serial forking resolves the "PSTN-fork that went to voice mail=
" scenario?

Could you explain that scenario? :)

One would also need to take into account the trust issue related with INVIT=
E 3xx. What if the new INVITE is sent to some expensive service number? Yes=
, in some phones you may allow the user to actually decided whether to trig=
ger a new INVITE, but it won't always work.

Regards,

Christer=

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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Content-Type: multipart/alternative; boundary=001636c5971ffcb6f90480e1779f
Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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--001636c5971ffcb6f90480e1779f
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Forking in essence  is a service provided by a downstream proxy (serving
domein B.com) to a user that has multiple devices registered to this proxy.
This service (served to him by his own domain) will allow him to receive
calls at any of his devices.

To push the responsibility for providing  this service of a B.com user to
another domain lets say A.com, is to say the least problematic. Even if it
solves some issues like those related to early media and HERFP, it
introduces lots of other issues.

/Hans Erik van Elburg

--001636c5971ffcb6f90480e1779f
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Forking in essence=A0 is a service provided by a downstream proxy (serving =
domein B.com) to a user that has multiple devices registered to this proxy.=
 This service (served to him by his own domain) will allow him to receive c=
alls at any of his devices.<br>
<br>To push the responsibility for providing=A0 this service of a B.com use=
r to another domain lets say A.com, is to say the least problematic. Even i=
f it solves some issues like those related to early media and HERFP, it int=
roduces lots of other issues. <br>
<br clear=3D"all">/Hans Erik van Elburg<br>
<br>

--001636c5971ffcb6f90480e1779f--

From john.elwell@siemens-enterprise.com  Wed Mar  3 01:37:10 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, 'Dean Willis' <dean.willis@softarmor.com>
Date: Wed, 3 Mar 2010 10:37:08 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Serial forking with early media depends on support for PRACK and UPDATE, wh=
ich are not mandatory items for UAs and therefore will not always be availa=
ble.

John

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 02 March 2010 20:57
> To: 'Dean Willis'
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
>=20
>=20
> Hi,=20
>=20
> >>>I would like to see #2 as a BCP. Not just for OPTIONS, but for all=20
> >>>cases  that a proxy/registrar might fork an inbound request.
> >>
> >>Just to clarify, I guess you don't include dialog establsihment=20
> >>requests (INVITE, SUBSCRIBE)? They should always be=20
> forwarded/forked=20
> >>even if addressed to the AOR for which the registrar is=20
> responseible.
> >
> >I think he really meant ALL REQUESTS. Replace forked-up=20
> behavior with more predictable redirection. When you really=20
> need downstream-node forking, you probably need to do it in a=20
> B28UA to handle all the=20
> >possible responses, including media-path responses. Look at=20
> how Google Voice handles forking for an example. We have=20
> known for years that forking does not play well with PSTN=20
> gateways and early media,=20
> >but have persistently failed to deal with that problem.
>=20
> I think serial forking can normally be handled pretty well,=20
> when it comes to early media etc.
>=20
> The problem is generally parallel forking.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From john.elwell@siemens-enterprise.com  Wed Mar  3 01:40:52 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Dean Willis <dean.willis@softarmor.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 3 Mar 2010 10:40:51 +0100
Thread-Topic: [sipcore] OPTIONS 3xx: To fork or not to fork?
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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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I would like to expand on this issue a little. Suppose I have at my home pr=
oxy a policy whereby calls go to B1 and B2 in parallel, and after T seconds=
 of no answer calls instead go to B3 (voicemail, perhaps). There is no way =
that the proxy could convey that policy in a 3xx response to the UAC. To ma=
ke this policy (or similar policies) work, the proxy would need to manufact=
urer temporary contact URIs that map to B1 and B2 and send these back in a =
302. It would then need to hope that the UAC does indeed establish new requ=
ests in parallel to the two temporary contacts for B1 and B2. The proxy wou=
ld need to treat the two requests as related and if both are unanswered aft=
er T seconds, reject one with a 4xx and the other with a 302 (this time wit=
h a third temporary contact URI mapped to B3). So policy CAN be met by the =
proxy, although with dependence on the UAC doing the right thing.

Elsewhere in this thread, somebody mentioned charging. The fact is that cal=
ls might cross a domain that charges (e.g., for international calls via a s=
erver provider into an enterprise that has a proxy that makes use of 302 re=
sponses to achieve forking). Since only one of the calls should normally be=
 answered (or if a second is answered it should be cleared very quickly), c=
harging based on duration after answer should not be a big issue. However, =
if a domain charges for call establishment attempts, it could be a problem.=
 In this example, the domain handling the international leg would not be aw=
are that the multiple calls are really part of the same call. I don't know =
whether this is an issue in practice or not.

John

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
> Sent: 03 March 2010 07:25
> To: Hans Erik van Elburg
> Cc: SIPCORE
> Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
>=20
>=20
> On Mar 2, 2010, at 4:47 PM, Hans Erik van Elburg wrote:
>=20
> > Burden as in handling the forking process. Didn't ment to overload =20
> > that with lots of negative meaning.
> >
> > I thought the first part was more interesting. Where every forking =20
> > proxy becomes a redirect server.
> >
> > There are also other consequences, where youre home proxy will =20
> > always have the same way of handling youre contacts=20
> q-values. UAC's =20
> > come in a lot of different flavors, they will all behave slightly =20
> > different and most probably some will misbehave. So the=20
> UAS's owner =20
> > loose some predictability about how incoming calls will be=20
> handled. =20
> > No consistent experience, hard to explain.
> >
> > Also users will want to have services or policies be executed on =20
> > their behalf on their home proxy, how does such approach=20
> impact that.
>=20
> Then the home proxy has to act as  a B2BUA and do the expansion into =20
> multiple connected calls itself.
>=20
> >
> > Does such model mean anything to call setup delays,=20
> considering lets =20
> > say scenarios where there are several intermediary proxies =20
> > interconnecting some corporate voip clouds.
> >
> > Where the 3xx approach is very elegant in some cases, to say as a =20
> > general BCP that any forking proxy should send 3xx also for=20
> INVITEs =20
> > in any forking case seems in that respect over the top.
>=20
> There are non-gateway scenarios where parallel forking=20
> actually works. =20
> We don't find them very often.
>=20
> --
> Dean
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Wed Mar  3 02:51:10 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, 'Dean Willis' <dean.willis@softarmor.com>
Date: Wed, 3 Mar 2010 11:51:08 +0100
Thread-Topic: INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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=20
Hi,

>Serial forking with early media depends on support for PRACK=20
>and UPDATE, which are not mandatory items for UAs and=20
>therefore will not always be available.

Well, that depends on how it is implemented, and if you want to give the UA=
C control of the early media.

But, even if the UAC is not able to control early media, serial forking wil=
l at least ensure a single early media stream at a certain time.

Regards,

Christer



> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> > Sent: 02 March 2010 20:57
> > To: 'Dean Willis'
> > Cc: sipcore@ietf.org
> > Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
> >=20
> >=20
> > Hi,
> >=20
> > >>>I would like to see #2 as a BCP. Not just for OPTIONS,=20
> but for all=20
> > >>>cases  that a proxy/registrar might fork an inbound request.
> > >>
> > >>Just to clarify, I guess you don't include dialog establsihment=20
> > >>requests (INVITE, SUBSCRIBE)? They should always be
> > forwarded/forked
> > >>even if addressed to the AOR for which the registrar is
> > responseible.
> > >
> > >I think he really meant ALL REQUESTS. Replace forked-up
> > behavior with more predictable redirection. When you really need=20
> > downstream-node forking, you probably need to do it in a B28UA to=20
> > handle all the
> > >possible responses, including media-path responses. Look at
> > how Google Voice handles forking for an example. We have known for=20
> > years that forking does not play well with PSTN gateways and early=20
> > media,
> > >but have persistently failed to deal with that problem.
> >=20
> > I think serial forking can normally be handled pretty well, when it=20
> > comes to early media etc.
> >=20
> > The problem is generally parallel forking.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> > =

From christer.holmberg@ericsson.com  Wed Mar  3 02:57:10 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Dean Willis <dean.willis@softarmor.com>, Hans Erik van Elburg <ietf.hanserik@gmail.com>
Date: Wed, 3 Mar 2010 11:56:37 +0100
Thread-Topic: INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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Subject: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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=20
Hi,

>I would like to expand on this issue a little. Suppose I have=20
>at my home proxy a policy whereby calls go to B1 and B2 in=20
>parallel, and after T seconds of no answer calls instead go=20
>to B3 (voicemail, perhaps). There is no way that the proxy=20
>could convey that policy in a 3xx response to the UAC. To=20
>make this policy (or similar policies) work, the proxy would=20
>need to manufacturer temporary contact URIs that map to B1=20
>and B2 and send these back in a 302. It would then need to=20
>hope that the UAC does indeed establish new requests in=20
>parallel to the two temporary contacts for B1 and B2. The=20
>proxy would need to treat the two requests as related and if=20
>both are unanswered after T seconds, reject one with a 4xx=20
>and the other with a 302 (this time with a third temporary=20
>contact URI mapped to B3). So policy CAN be met by the proxy,=20
>although with dependence on the UAC doing the right thing.
>=20
>Elsewhere in this thread, somebody mentioned charging. The=20
>fact is that calls might cross a domain that charges (e.g.,=20
>for international calls via a server provider into an=20
>enterprise that has a proxy that makes use of 302 responses=20
>to achieve forking). Since only one of the calls should=20
>normally be answered (or if a second is answered it should be=20
>cleared very quickly), charging based on duration after=20
>answer should not be a big issue. However, if a domain=20
>charges for call establishment attempts, it could be a=20
>problem. In this example, the domain handling the=20
>international leg would not be aware that the multiple calls=20
>are really part of the same call. I don't know whether this=20
>is an issue in practice or not.

I am the one who mentioned charging.

Maybe charging as such is not a big problem. I am more concerned about how =
network services are affected.

In your example above, where B1 sends 4xx and B2 sends 302, the forking pro=
xy needs to know that the calls are related. Maybe some correlation header =
field can be used for that, or maybe it can somehow be embedded into the te=
mporary contact URI, but the correlation still needs to be there.

Regards,

Christer


> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Dean Willis
> > Sent: 03 March 2010 07:25
> > To: Hans Erik van Elburg
> > Cc: SIPCORE
> > Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
> >=20
> >=20
> > On Mar 2, 2010, at 4:47 PM, Hans Erik van Elburg wrote:
> >=20
> > > Burden as in handling the forking process. Didn't ment to=20
> overload=20
> > > that with lots of negative meaning.
> > >
> > > I thought the first part was more interesting. Where=20
> every forking=20
> > > proxy becomes a redirect server.
> > >
> > > There are also other consequences, where youre home proxy will=20
> > > always have the same way of handling youre contacts
> > q-values. UAC's
> > > come in a lot of different flavors, they will all behave slightly=20
> > > different and most probably some will misbehave. So the
> > UAS's owner
> > > loose some predictability about how incoming calls will be
> > handled. =20
> > > No consistent experience, hard to explain.
> > >
> > > Also users will want to have services or policies be executed on=20
> > > their behalf on their home proxy, how does such approach
> > impact that.
> >=20
> > Then the home proxy has to act as  a B2BUA and do the=20
> expansion into=20
> > multiple connected calls itself.
> >=20
> > >
> > > Does such model mean anything to call setup delays,
> > considering lets
> > > say scenarios where there are several intermediary proxies=20
> > > interconnecting some corporate voip clouds.
> > >
> > > Where the 3xx approach is very elegant in some cases, to say as a=20
> > > general BCP that any forking proxy should send 3xx also for
> > INVITEs
> > > in any forking case seems in that respect over the top.
> >=20
> > There are non-gateway scenarios where parallel forking=20
> actually works.
> > We don't find them very often.
> >=20
> > --
> > Dean
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From kpfleming@digium.com  Wed Mar  3 04:25:08 2010
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Christer Holmberg wrote:

> If you use different ports, and you use ICE, I guess that means that you need to perform new ICE procedures for that port before you can start using it.

That would be true, yes. On the SIP Forum mailing list when I brought
this up there were comments about 'using the same port will keep the NAT
mapping open if there is one', but that's only true if both endpoints
use the same port for the replaced media stream.

I suppose that the real cause of this issue is UDPTL itself, and the
fact that it's not possible to easily discriminate between UDPTL and
RTP. If the T.38 media stream was in audio/t38 mode (and thus using RTP
for transport), as long as the payload number chosen was not one of the
possible options for the previous media stream, discrimination of the
incoming packets would be trivial.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

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Christer Holmberg wrote:

> If you use different ports, and you use ICE, I guess that means that you need to perform new ICE procedures for that port before you can start using it.

While hashing over this I came up with another scenario where the 'media
exchanged using new parameters' trigger is not sufficient to actually
know that an offerer's changes have been executed, so I'd like to
propose an alternative way of describing this problem that may also make
it more appropriate for inclusion in this draft.

As a degenerate case of the 'port number did not change' scenario,
imagine that an endpoint is currently exchanging audio media using the
iLBC codec over RTP, which requires usage of a dynamic payload number
since iLBC does not have an assigned payload number. Later, it wants to
switch to Speex, which also does not have an assigned payload number, so
it sends a re-INVITE with an offer that uses the same port number *and*
payload number, but provides an rtpmap from that payload number to Speex
 instead of iLBC. Again in this scenario the offerer cannot safely use
'media exchanged using the new parameters' as a trigger for 'changes
have been executed' because in-flight RTP packets will have that payload
number already. I understand this is a contrived scenario, but it does
not seem that far outside the bounds of what implementors tend to do :-)

So, I'd like to propose this sort of language instead:

An offerer in a re-INVITE transaction is allowed by RFC3261 to treat its
offered changes as 'executed' based on *either* receiving a non-error
response to its offer *or* determining that media has been exchanged
using the new session parameters. However, if the offer's session
parameters do not differ sufficiently from the previous session state to
allow this determination to be made conclusively, the offerer should not
use media exchange as the threshold for execution of the changes, and
should instead use only the reception of a non-error response as the
trigger. This affords the answerer the opportunity to provide an error
response to the offer if desired without the two endpoints getting out
of sync in their understanding of the session state.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

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Subject: [sipcore] Fwd: request for help in developing a tool that may be helpful to WG chairs
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[as chair]

SIPCORE participants:

Please consider filling out the survey cited below. Thanks.

/a

-------- Original Message --------
Subject: 	request for help in developing a tool that may be helpful to 
WG chairs
Date: 	Wed, 3 Mar 2010 10:36:19 -0500 (EST)
From: 	sob@harvard.edu (Scott O. Bradner)
To: 	wgchairs@ietf.org



IETF working group chairs;

We are developing an open-source tool for monitoring the status and
progress of conflicts in on-line working groups (WG).  The tool works by
analyzing the WG mailing list.  When developed, this tool should be
helpful to WG chairs trying to understand the status of WG discussions
(how close to consensus is the WG, what is the distribution of
participation, etc).

As part of the development process we have been using a prototype tool
to analyze IETF WG mailing list archives to determine the amount of
conflict and how effective this conflict is being (has been) resolved.
As the first step, we need to understand the relationship between the
conflicts in a working group and the structure of the communication
network in that group. While having conflicts is not necessarily a bad
thing for a working group effort, some conflicts can escalate into
disasters. We are interested in finding the communication patterns
related to the evolution of group conflicts. Results from this study
will provide the base for the development of the tool that helps working
group chairs to decide when to intervene with an internal conflict
before it becomes irreversibly negative as well as being a tool that may
help determine where there is consensus on a particular topic.

We would like your help in understanding the level of conflicts within
your working groups and how the conflicts affect productivity and group
members' perception on the working group. It will be greatly appreciated
if you could ask your WG members to anonymously fill a short survey at

https://spreadsheets.google.com/viewform?hl=en&formkey=dExTbEU5QmRncnhFbjhQUVR4bzBGMEE6MA

Thank you!

Best Regards,

Bin Zhu, Mark Gaynor, Scott Bradner, and Jialun Qin



--------------060701090909020007020308
Content-Type: text/html; charset=ISO-8859-1
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>

<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body bgcolor="#ffffff" text="#000000">
[as chair]<br>
<br>
SIPCORE participants:<br>
<br>
Please consider filling out the survey cited below. Thanks.<br>
<br>
/a<br>
<br>
-------- Original Message --------
<table class="moz-email-headers-table" border="0" cellpadding="0"
 cellspacing="0">
  <tbody>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
      <td>request for help in developing a tool that may be helpful to
WG chairs</td>
    </tr>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
      <td>Wed, 3 Mar 2010 10:36:19 -0500 (EST)</td>
    </tr>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
      <td><a class="moz-txt-link-abbreviated" href="mailto:sob@harvard.edu">sob@harvard.edu</a> (Scott O. Bradner)</td>
    </tr>
    <tr>
      <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
      <td><a class="moz-txt-link-abbreviated" href="mailto:wgchairs@ietf.org">wgchairs@ietf.org</a></td>
    </tr>
  </tbody>
</table>
<br>
<br>
<pre>IETF working group chairs;

We are developing an open-source tool for monitoring the status and
progress of conflicts in on-line working groups (WG).  The tool works by
analyzing the WG mailing list.  When developed, this tool should be
helpful to WG chairs trying to understand the status of WG discussions
(how close to consensus is the WG, what is the distribution of
participation, etc).

As part of the development process we have been using a prototype tool
to analyze IETF WG mailing list archives to determine the amount of
conflict and how effective this conflict is being (has been) resolved.
As the first step, we need to understand the relationship between the
conflicts in a working group and the structure of the communication
network in that group. While having conflicts is not necessarily a bad
thing for a working group effort, some conflicts can escalate into
disasters. We are interested in finding the communication patterns
related to the evolution of group conflicts. Results from this study
will provide the base for the development of the tool that helps working
group chairs to decide when to intervene with an internal conflict
before it becomes irreversibly negative as well as being a tool that may
help determine where there is consensus on a particular topic.

We would like your help in understanding the level of conflicts within
your working groups and how the conflicts affect productivity and group
members' perception on the working group. It will be greatly appreciated
if you could ask your WG members to anonymously fill a short survey at 

<a class="moz-txt-link-freetext" href="https://spreadsheets.google.com/viewform?hl=en&amp;formkey=dExTbEU5QmRncnhFbjhQUVR4bzBGMEE6MA">https://spreadsheets.google.com/viewform?hl=en&amp;formkey=dExTbEU5QmRncnhFbjhQUVR4bzBGMEE6MA</a>

Thank you!

Best Regards, 

Bin Zhu, Mark Gaynor, Scott Bradner, and Jialun Qin

</pre>
</body>
</html>

--------------060701090909020007020308--

From pkyzivat@cisco.com  Wed Mar  3 09:41:56 2010
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Date: Wed, 03 Mar 2010 12:41:52 -0500
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Subject: Re: [sipcore] Comments on draft-ieft-sipcore-reinvite-01
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Kevin,

Once a payload number has been bound to a codec in a session, it isn't 
supposed to be used for anything else in that session. (I'll have to 
hunt to find the text that says that, but its somewhere.)

However it turns out that this is another one of those things that can 
be broken by 3pcc. So a robust implementation should be prepared to cope 
with this if it happens - to the extent that there is sufficient 
information to cope. E.g. if a payload number that was once mentioned 
has not been in use for awhile, a robust implementation ought to be 
capable of supporting its subsequent use for a different codec.

Anyway, when this is just UA-UA, a conforming implementation will not 
get into the situation you describe.

	Thanks,
	Paul

Kevin P. Fleming wrote:
> Christer Holmberg wrote:
> 
>> If you use different ports, and you use ICE, I guess that means that you need to perform new ICE procedures for that port before you can start using it.
> 
> While hashing over this I came up with another scenario where the 'media
> exchanged using new parameters' trigger is not sufficient to actually
> know that an offerer's changes have been executed, so I'd like to
> propose an alternative way of describing this problem that may also make
> it more appropriate for inclusion in this draft.
> 
> As a degenerate case of the 'port number did not change' scenario,
> imagine that an endpoint is currently exchanging audio media using the
> iLBC codec over RTP, which requires usage of a dynamic payload number
> since iLBC does not have an assigned payload number. Later, it wants to
> switch to Speex, which also does not have an assigned payload number, so
> it sends a re-INVITE with an offer that uses the same port number *and*
> payload number, but provides an rtpmap from that payload number to Speex
>  instead of iLBC. Again in this scenario the offerer cannot safely use
> 'media exchanged using the new parameters' as a trigger for 'changes
> have been executed' because in-flight RTP packets will have that payload
> number already. I understand this is a contrived scenario, but it does
> not seem that far outside the bounds of what implementors tend to do :-)
> 
> So, I'd like to propose this sort of language instead:
> 
> An offerer in a re-INVITE transaction is allowed by RFC3261 to treat its
> offered changes as 'executed' based on *either* receiving a non-error
> response to its offer *or* determining that media has been exchanged
> using the new session parameters. However, if the offer's session
> parameters do not differ sufficiently from the previous session state to
> allow this determination to be made conclusively, the offerer should not
> use media exchange as the threshold for execution of the changes, and
> should instead use only the reception of a non-error response as the
> trigger. This affords the answerer the opportunity to provide an error
> response to the offer if desired without the two endpoints getting out
> of sync in their understanding of the session state.
> 

From pkyzivat@cisco.com  Wed Mar  3 09:50:01 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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Hans Erik van Elburg wrote:
> Forking in essence  is a service provided by a downstream proxy (serving 
> domein B.com) to a user that has multiple devices registered to this 
> proxy. This service (served to him by his own domain) will allow him to 
> receive calls at any of his devices.
> 
> To push the responsibility for providing  this service of a B.com user 
> to another domain lets say A.com, is to say the least problematic. Even 
> if it solves some issues like those related to early media and HERFP, it 
> introduces lots of other issues.

Forking downstream doesn't relieve the upstream device of burden, it 
simply makes it be a different burden.

If the downstream server wants to take this on, a B2BUA could be a 
better choice.

And regardless of whether user B views this as a service to him, user A 
may also regard it as a service to him to see the multiple contacts and 
be able to choose among them.

	Thanks,
	Paul

From ietf.hanserik@gmail.com  Wed Mar  3 10:06:16 2010
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Date: Wed, 3 Mar 2010 19:06:11 +0100
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From: Hans Erik van Elburg <ietf.hanserik@gmail.com>
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Subject: Re: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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--00504502c6e55b1c9d0480e95677
Content-Type: text/plain; charset=ISO-8859-1

Demonstrating the point that a simple BCP saying "though shalt always send
3xx in all INVITE forking cases" is not appropriate.

More inline...

/Hans Erik van Elburg

On Wed, Mar 3, 2010 at 6:49 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

>
>
> Hans Erik van Elburg wrote:
>
>> Forking in essence  is a service provided by a downstream proxy (serving
>> domein B.com) to a user that has multiple devices registered to this proxy.
>> This service (served to him by his own domain) will allow him to receive
>> calls at any of his devices.
>>
>> To push the responsibility for providing  this service of a B.com user to
>> another domain lets say A.com, is to say the least problematic. Even if it
>> solves some issues like those related to early media and HERFP, it
>> introduces lots of other issues.
>>
>
> Forking downstream doesn't relieve the upstream device of burden, it simply
> makes it be a different burden.
>
> Agree.


> If the downstream server wants to take this on, a B2BUA could be a better
> choice.
>
> Maybe.


> And regardless of whether user B views this as a service to him, user A may
> also regard it as a service to him to see the multiple contacts and be able
> to choose among them.
>
> Well it is hard to see how B's forking proxy's behaviour for INVITEs can be
seen as a service to A. Also INVITE does not express A's wish to see B's
contacts, but a wish of A's UA to establish a session with whatever UA that
B registered.

Of course A may want to know B's multiple contacts, but if you phrase it
like that it is indeed more of a presence use case and you would not use
INVITE for that.

--00504502c6e55b1c9d0480e95677
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Demonstrating the point that a simple BCP saying &quot;though shalt always =
send 3xx in all INVITE forking cases&quot; is not appropriate.<br><br>More =
inline...<br><br clear=3D"all">/Hans Erik van Elburg<br>
<br><div class=3D"gmail_quote">On Wed, Mar 3, 2010 at 6:49 PM, Paul Kyzivat=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@cisco.com">pkyzivat@cisco=
.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"bo=
rder-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding=
-left: 1ex;">
<div class=3D"im"><br>
<br>
Hans Erik van Elburg wrote:<br>
</div><div class=3D"im"><blockquote class=3D"gmail_quote" style=3D"border-l=
eft: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left:=
 1ex;">
Forking in essence =A0is a service provided by a downstream proxy (serving =
domein B.com) to a user that has multiple devices registered to this proxy.=
 This service (served to him by his own domain) will allow him to receive c=
alls at any of his devices.<br>

<br>
To push the responsibility for providing =A0this service of a B.com user to=
 another domain lets say A.com, is to say the least problematic. Even if it=
 solves some issues like those related to early media and HERFP, it introdu=
ces lots of other issues.<br>

</blockquote>
<br></div>
Forking downstream doesn&#39;t relieve the upstream device of burden, it si=
mply makes it be a different burden.<br>
<br></blockquote><div>Agree.<br>=A0</div><blockquote class=3D"gmail_quote" =
style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8=
ex; padding-left: 1ex;">
If the downstream server wants to take this on, a B2BUA could be a better c=
hoice.<br>
<br></blockquote><div>Maybe.<br>=A0</div><blockquote class=3D"gmail_quote" =
style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8=
ex; padding-left: 1ex;">
And regardless of whether user B views this as a service to him, user A may=
 also regard it as a service to him to see the multiple contacts and be abl=
e to choose among them.<br>
<br></blockquote><div>Well it is hard to see how B&#39;s forking proxy&#39;=
s behaviour for INVITEs can be seen as a service to A. Also INVITE does not=
 express A&#39;s wish to see B&#39;s contacts, but a wish of A&#39;s UA to =
establish a session with whatever UA that B registered.<br>
<br>Of course A may want to know B&#39;s multiple contacts, but if you phra=
se it like that it is indeed more of a presence use case and you would not =
use INVITE for that.<br>=A0</div><br></div>

--00504502c6e55b1c9d0480e95677--

From christer.holmberg@ericsson.com  Wed Mar  3 12:03:18 2010
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To: 'Hans Erik van Elburg' <ietf.hanserik@gmail.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 3 Mar 2010 21:03:00 +0100
Thread-Topic: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

I am not sure a BCP can say "SHALL" about anything...

Also, I don't think we could mandate such behavior. Too many things would g=
et messed up.

Do I see "SIP 3.0" on Dean's lips? ;)

Regards,

Christer



________________________________
From: Hans Erik van Elburg [mailto:ietf.hanserik@gmail.com]
Sent: 3. maaliskuuta 2010 20:06
To: Paul Kyzivat
Cc: Christer Holmberg; SIPCORE
Subject: Re: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?=
]

Demonstrating the point that a simple BCP saying "though shalt always send =
3xx in all INVITE forking cases" is not appropriate.

More inline...

/Hans Erik van Elburg

On Wed, Mar 3, 2010 at 6:49 PM, Paul Kyzivat <pkyzivat@cisco.com<mailto:pky=
zivat@cisco.com>> wrote:


Hans Erik van Elburg wrote:
Forking in essence  is a service provided by a downstream proxy (serving do=
mein B.com) to a user that has multiple devices registered to this proxy. T=
his service (served to him by his own domain) will allow him to receive cal=
ls at any of his devices.

To push the responsibility for providing  this service of a B.com user to a=
nother domain lets say A.com, is to say the least problematic. Even if it s=
olves some issues like those related to early media and HERFP, it introduce=
s lots of other issues.

Forking downstream doesn't relieve the upstream device of burden, it simply=
 makes it be a different burden.

Agree.

If the downstream server wants to take this on, a B2BUA could be a better c=
hoice.

Maybe.

And regardless of whether user B views this as a service to him, user A may=
 also regard it as a service to him to see the multiple contacts and be abl=
e to choose among them.

Well it is hard to see how B's forking proxy's behaviour for INVITEs can be=
 seen as a service to A. Also INVITE does not express A's wish to see B's c=
ontacts, but a wish of A's UA to establish a session with whatever UA that =
B registered.

Of course A may want to know B's multiple contacts, but if you phrase it li=
ke that it is indeed more of a presence use case and you would not use INVI=
TE for that.



--_000_FF84A09F50A6DC48ACB6714F4666CC745C1102CCC4ESESSCMS0354e_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.3790.4639" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010>Hi,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010>I am not sure a BCP can say "SHALL" about anythi=
ng...=20
</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010>Also, I don't think we could mandate such behavi=
or. Too=20
many things would get messed up.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010>Do I see&nbsp;"SIP 3.0" on Dean's lips?=20
;)</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D483385919-03032010>Christer</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Hans Erik van Elburg=20
[mailto:ietf.hanserik@gmail.com] <BR><B>Sent:</B> 3. maaliskuuta 2010=20
20:06<BR><B>To:</B> Paul Kyzivat<BR><B>Cc:</B> Christer Holmberg;=20
SIPCORE<BR><B>Subject:</B> Re: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fo=
rk or=20
not to fork?]<BR></FONT><BR></DIV>
<DIV></DIV>Demonstrating the point that a simple BCP saying "though shalt a=
lways=20
send 3xx in all INVITE forking cases" is not appropriate.<BR><BR>More=20
inline...<BR><BR clear=3Dall>/Hans Erik van Elburg<BR><BR>
<DIV class=3Dgmail_quote>On Wed, Mar 3, 2010 at 6:49 PM, Paul Kyzivat <SPAN=
=20
dir=3Dltr>&lt;<A=20
href=3D"mailto:pkyzivat@cisco.com">pkyzivat@cisco.com</A>&gt;</SPAN> wrote:=
<BR>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204=
,204,204) 1px solid">
  <DIV class=3Dim><BR><BR>Hans Erik van Elburg wrote:<BR></DIV>
  <DIV class=3Dim>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(2=
04,204,204) 1px solid">Forking=20
    in essence &nbsp;is a service provided by a downstream proxy (serving d=
omein=20
    B.com) to a user that has multiple devices registered to this proxy. Th=
is=20
    service (served to him by his own domain) will allow him to receive cal=
ls at=20
    any of his devices.<BR><BR>To push the responsibility for providing=20
    &nbsp;this service of a B.com user to another domain lets say A.com, is=
 to=20
    say the least problematic. Even if it solves some issues like those rel=
ated=20
    to early media and HERFP, it introduces lots of other=20
  issues.<BR></BLOCKQUOTE><BR></DIV>Forking downstream doesn't relieve the=
=20
  upstream device of burden, it simply makes it be a different=20
burden.<BR><BR></BLOCKQUOTE>
<DIV>Agree.<BR>&nbsp;</DIV>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204=
,204,204) 1px solid">If=20
  the downstream server wants to take this on, a B2BUA could be a better=20
  choice.<BR><BR></BLOCKQUOTE>
<DIV>Maybe.<BR>&nbsp;</DIV>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204=
,204,204) 1px solid">And=20
  regardless of whether user B views this as a service to him, user A may a=
lso=20
  regard it as a service to him to see the multiple contacts and be able to=
=20
  choose among them.<BR><BR></BLOCKQUOTE>
<DIV>Well it is hard to see how B's forking proxy's behaviour for INVITEs c=
an be=20
seen as a service to A. Also INVITE does not express A's wish to see B's=20
contacts, but a wish of A's UA to establish a session with whatever UA that=
 B=20
registered.<BR><BR>Of course A may want to know B's multiple contacts, but =
if=20
you phrase it like that it is indeed more of a presence use case and you wo=
uld=20
not use INVITE for that.<BR>&nbsp;</DIV><BR></DIV></BODY></HTML>

--_000_FF84A09F50A6DC48ACB6714F4666CC745C1102CCC4ESESSCMS0354e_--

From kpfleming@digium.com  Wed Mar  3 12:54:12 2010
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Paul Kyzivat wrote:

> Once a payload number has been bound to a codec in a session, it isn't
> supposed to be used for anything else in that session. (I'll have to
> hunt to find the text that says that, but its somewhere.)

OK, that makes complete sense.

> However it turns out that this is another one of those things that can
> be broken by 3pcc. So a robust implementation should be prepared to cope
> with this if it happens - to the extent that there is sufficient
> information to cope. E.g. if a payload number that was once mentioned
> has not been in use for awhile, a robust implementation ought to be
> capable of supporting its subsequent use for a different codec.
> 
> Anyway, when this is just UA-UA, a conforming implementation will not
> get into the situation you describe.

That does help then. Thanks for the information.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

From gonzalo.camarillo@ericsson.com  Wed Mar  3 23:49:41 2010
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Hi Kevin,

> Editorial comments:
> 
> Section 1:
> 
> "There has been some confusion among implentors regarding how a UAS"
> 
> ... should be 'implementors'.

Fixed.

> Section 3.7:
> 
> SDP4 is improperly formatted.

Fixed.

Thanks,

Gonzalo

From gonzalo.camarillo@ericsson.com  Thu Mar  4 00:01:41 2010
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Hi,

yes, what Paul indicates is documented in Section 8.3.2 of RFC 3264:

http://tools.ietf.org/html/rfc3264#section-8.3.2

"However, in the case of RTP, the mapping from a particular dynamic
payload type number to a particular codec within that media stream MUST
NOT change for the duration of a session."

Cheers,

Gonzalo


Kevin P. Fleming wrote:
> Paul Kyzivat wrote:
> 
>> Once a payload number has been bound to a codec in a session, it isn't
>> supposed to be used for anything else in that session. (I'll have to
>> hunt to find the text that says that, but its somewhere.)
> 
> OK, that makes complete sense.
> 
>> However it turns out that this is another one of those things that can
>> be broken by 3pcc. So a robust implementation should be prepared to cope
>> with this if it happens - to the extent that there is sufficient
>> information to cope. E.g. if a payload number that was once mentioned
>> has not been in use for awhile, a robust implementation ought to be
>> capable of supporting its subsequent use for a different codec.
>>
>> Anyway, when this is just UA-UA, a conforming implementation will not
>> get into the situation you describe.
> 
> That does help then. Thanks for the information.
> 


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Hi Kevin,

Section 8.3.3 of RFC 3264 describes when it makes sense to reuse an m
line when changing the media type:

http://tools.ietf.org/html/rfc3264#section-8.3.3

  "It
   is RECOMMENDED that the media type be changed (as opposed to adding a
   new stream), when the same logical data is being conveyed, but just
   in a different media format."

In any case, what you describe are scenarios where the UA will not
notice it has received media using the new parameters until it actually
decodes the media in order to play it out. That is fine. When the UA
finally notices the new parameters, it will consider the changes to have
been executed.

Note that the text in the draft talks about new media "having been"
exchanged. It does not talk about UAs "assuming" new media has been
exchanged. If the UA does not know for sure new media has been
exchanged, it does not do anything.

In order to address your comment and make this process clearer, I will
add the following paragraph after the third paragraph of Section 3.3

  Normally, a UA receiving media can easily detect when the new
  parameters for the media stream are used (e.g,. media is received on a
  new port). However, in some scenarios the UA will have to process
  incoming media packets in order to detect whether they use the old or
  the new parameters.


Thanks for your comments,

Gonzalo


Kevin P. Fleming wrote:
> Christer Holmberg wrote:
> 
>> If you use different ports, and you use ICE, I guess that means
>> that you need to perform new ICE procedures for that port before
>> you can start using it.
> 
> While hashing over this I came up with another scenario where the
> 'media exchanged using new parameters' trigger is not sufficient to
> actually know that an offerer's changes have been executed, so I'd
> like to propose an alternative way of describing this problem that
> may also make it more appropriate for inclusion in this draft.
> 
> As a degenerate case of the 'port number did not change' scenario, 
> imagine that an endpoint is currently exchanging audio media using
> the iLBC codec over RTP, which requires usage of a dynamic payload
> number since iLBC does not have an assigned payload number. Later, it
> wants to switch to Speex, which also does not have an assigned
> payload number, so it sends a re-INVITE with an offer that uses the
> same port number *and* payload number, but provides an rtpmap from
> that payload number to Speex instead of iLBC. Again in this scenario
> the offerer cannot safely use 'media exchanged using the new
> parameters' as a trigger for 'changes have been executed' because
> in-flight RTP packets will have that payload number already. I
> understand this is a contrived scenario, but it does not seem that
> far outside the bounds of what implementors tend to do :-)
> 
> So, I'd like to propose this sort of language instead:
> 
> An offerer in a re-INVITE transaction is allowed by RFC3261 to treat
> its offered changes as 'executed' based on *either* receiving a
> non-error response to its offer *or* determining that media has been
> exchanged using the new session parameters. However, if the offer's
> session parameters do not differ sufficiently from the previous
> session state to allow this determination to be made conclusively,
> the offerer should not use media exchange as the threshold for
> execution of the changes, and should instead use only the reception
> of a non-error response as the trigger. This affords the answerer the
> opportunity to provide an error response to the offer if desired
> without the two endpoints getting out of sync in their understanding
> of the session state.
> 


From gonzalo.camarillo@ericsson.com  Thu Mar  4 01:00:47 2010
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Hi John,

thanks for your comments. Answers inline:

Elwell, John wrote:
> I read this document again and didn't find anything to prevent it progressing (although I would not claim to have done a thorough review).
> 
> Minor comments and nits:
> 
> - Section 1:
> "A UAC willing to
>    act the offerer"
> Change "act" to "act as".

Fixed.

> - Section 2: Where "UAC" and "UAS" expanded, it would probably be worth stating that the terms are used with respect to INVITE and re-INVITE transactions only. Other transactions within an INVITE or re-INVITE transaction can be in the reverse direction and the roles are reversed.

I have added the following paragraph:

"Note that the terms UAC and UAS apply to a transaction, not to a
dialog. For example, the UAC of a re-INVITE transaction can be the UAS
of an UPDATE transaction within the re-INVITE."

> 
> - Section 3.1:
> "SDP4:
>             m=audio 31000 RTP/AVP 0
>             c=IN IP4 192.0.2.5
>             m=video 0 RTP/AVP 31
>             c=IN IP4 192.0.2.2" 
> It seems unreasonable to include in an answer, below a rejected m-line, a c-line echoing the c-line in the offer.

Fixed.

> - In the last example in 3.1, it should perhaps be pointed out that SDP5 (in the UPDATE request) is an offer. Also the contents of SDP6 (SDP answer) should perhaps also be shown for completeness.

I have clarified SDP5 is an offer and added SDP6.

> 
> - In section 3.3:
> "During a session establishment, the UAS can wait for
>    using the media parameters until "
> Change "wait for" to "wait before" (or alternatively "delay").

I have used "wait before".

> - In section 3.7:
> "The proxy relays the UPDATE request (9) to UA1.  The UPDATE request
>    (9) arrives at UA1 before the 4xx response (7) that had been
>    previously sent.  UA2 accepts the changes in the UPDATE request and
>    returns a 200 (OK) response (10) to it ."
> I think the last sentence should begin "UA1 accepts the changes..."

Fixed.

> - "   SDP4: m=audio 20000 RTP/AVP 0 a=sendonly m=video 30002 RTP/AVP 31
>    a=inactive"
> Line breaks missing (c.f., the other examples).

Fixed.

> 
> - In section 4.4:
> "In this document, reliable provisional responses are
>       those that use the mechanism defined in RFC 3262 [RFC3262] on any
>       other SIP-based mechanism that may be specified in the future."
> Change "on" to "or".

Fixed.

> 
> - Change "If instead sending" to "If instead of sending"

Fixed.

Thanks!

Gonzalo


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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Gonzalo,

Thanks - I still have a concern, see below.=20

> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
> Sent: 04 March 2010 09:01
> To: Elwell, John
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] WGLC comments on draft-ietf-sipcore-reinvite-01
>=20
> Hi John,
>=20
> thanks for your comments. Answers inline:
>=20
> Elwell, John wrote:
> > I read this document again and didn't find anything to=20
> prevent it progressing (although I would not claim to have=20
> done a thorough review).
> >=20
> > Minor comments and nits:
> >=20
> > - Section 1:
> > "A UAC willing to
> >    act the offerer"
> > Change "act" to "act as".
>=20
> Fixed.
>=20
> > - Section 2: Where "UAC" and "UAS" expanded, it would=20
> probably be worth stating that the terms are used with=20
> respect to INVITE and re-INVITE transactions only. Other=20
> transactions within an INVITE or re-INVITE transaction can be=20
> in the reverse direction and the roles are reversed.
>=20
> I have added the following paragraph:
>=20
> "Note that the terms UAC and UAS apply to a transaction, not to a
> dialog. For example, the UAC of a re-INVITE transaction can be the UAS
> of an UPDATE transaction within the re-INVITE."
[JRE] I got the impression that the terms UAC and UAS were always used with=
 respect to the re-INVITE transaction, even though there might also be an U=
PDATE transaction in the reverse direction. For example, if I look at figur=
e 3, the entire left hand column is labelled "UAC", even though that entity=
 is the UAS for the received UPDATE request ((6). I was looking for text mo=
re along the lines of:
"Note that the terms UAC and UAS are used with respect to an INVITE or re-I=
NVITE transaction, and do not necessarily reflect the role of the UA concer=
ned with respect to any other transaction, such as an UPDATE transaction oc=
curring during the INVITE transaction."

John

From gonzalo.camarillo@ericsson.com  Thu Mar  4 02:34:59 2010
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [sipcore] WGLC comments on draft-ietf-sipcore-reinvite-01
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Hi John,

I have added your text (I replaced "during" with "within").

Thanks,

Gonzalo

Elwell, John wrote:
> Gonzalo,
> 
> Thanks - I still have a concern, see below. 
> 
>> -----Original Message-----
>> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
>> Sent: 04 March 2010 09:01
>> To: Elwell, John
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] WGLC comments on draft-ietf-sipcore-reinvite-01
>>
>> Hi John,
>>
>> thanks for your comments. Answers inline:
>>
>> Elwell, John wrote:
>>> I read this document again and didn't find anything to 
>> prevent it progressing (although I would not claim to have 
>> done a thorough review).
>>> Minor comments and nits:
>>>
>>> - Section 1:
>>> "A UAC willing to
>>>    act the offerer"
>>> Change "act" to "act as".
>> Fixed.
>>
>>> - Section 2: Where "UAC" and "UAS" expanded, it would 
>> probably be worth stating that the terms are used with 
>> respect to INVITE and re-INVITE transactions only. Other 
>> transactions within an INVITE or re-INVITE transaction can be 
>> in the reverse direction and the roles are reversed.
>>
>> I have added the following paragraph:
>>
>> "Note that the terms UAC and UAS apply to a transaction, not to a
>> dialog. For example, the UAC of a re-INVITE transaction can be the UAS
>> of an UPDATE transaction within the re-INVITE."
> [JRE] I got the impression that the terms UAC and UAS were always used with respect to the re-INVITE transaction, even though there might also be an UPDATE transaction in the reverse direction. For example, if I look at figure 3, the entire left hand column is labelled "UAC", even though that entity is the UAS for the received UPDATE request ((6). I was looking for text more along the lines of:
> "Note that the terms UAC and UAS are used with respect to an INVITE or re-INVITE transaction, and do not necessarily reflect the role of the UA concerned with respect to any other transaction, such as an UPDATE transaction occurring during the INVITE transaction."
> 
> John


From gonzalo.camarillo@ericsson.com  Thu Mar  4 02:42:55 2010
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Date: Thu, 04 Mar 2010 12:42:54 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [sipcore] New version of draft-ietf-sipcore-reinvite-02.txt
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Folks,

I have just submitted a new revision of the following draft:

http://www.ietf.org/id/draft-ietf-sipcore-reinvite-02.txt

This revision includes all the WGLC comments received so far. Note that
the WGLC will end on Monday (March 8th).

Cheers,

Gonzalo

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.


	Title           : Re-INVITE and Target-refresh Request Handling in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo, et al.
	Filename        : draft-ietf-sipcore-reinvite-02.txt
	Pages           : 24
	Date            : 2010-03-04

In this document, we clarify the handling of re-INVITEs in SIP.  We
clarify in which situations a UAS (User Agent Server) should generate
a success response and in which situations a UAS should generate an
error response to a re-INVITE.  Additionally, we clarify issues
related to target-refresh requests.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-02.txt

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From gonzalo.camarillo@ericsson.com  Thu Mar  4 07:13:27 2010
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Hi,

John noticed a markup-related error in 02. Thanks for the follow up, John!

I have fixed it in a new revision of the draft:

http://www.ietf.org/id/draft-ietf-sipcore-reinvite-03.txt

Sorry for the confusion.

Cheers,

Gonzalo


Gonzalo Camarillo wrote:
> Folks,
> 
> I have just submitted a new revision of the following draft:
> 
> http://www.ietf.org/id/draft-ietf-sipcore-reinvite-02.txt
> 
> This revision includes all the WGLC comments received so far. Note that
> the WGLC will end on Monday (March 8th).
> 
> Cheers,
> 
> Gonzalo
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.


	Title           : Re-INVITE and Target-refresh Request Handling in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo, et al.
	Filename        : draft-ietf-sipcore-reinvite-03.txt
	Pages           : 24
	Date            : 2010-03-04

In this document, we clarify the handling of re-INVITEs in SIP.  We
clarify in which situations a UAS (User Agent Server) should generate
a success response and in which situations a UAS should generate an
error response to a re-INVITE.  Additionally, we clarify issues
related to target-refresh requests.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-03.txt

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From dean.willis@softarmor.com  Thu Mar  4 08:51:51 2010
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Subject: Re: [sipcore] INVITE 3xx [was OPTIONS 3xx: To fork or not to fork?]
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On Mar 3, 2010, at 2:42 AM, Hans Erik van Elburg wrote:

> Forking in essence  is a service provided by a downstream proxy  
> (serving domein B.com) to a user that has multiple devices  
> registered to this proxy. This service (served to him by his own  
> domain) will allow him to receive calls at any of his devices.

That's how it works in theory. In practice, it doesn't work all that  
well, as some of user B's devices are going to be either PSTN phones,  
or nodes that implement their own voicemail service. In other words,  
they're going to send back a media flow, either before or after a 200  
OK. A forking proxy cannot reconcile this; it requires a UA, either a  
B2BUA as Christer suggests, or all the way back to the originating  
user. And if the media streams are a result of a forking proxy, then  
the UA lacks sufficient information for sorting it out.

In other words, either fork in a B2BUA and be prepared to deal with  
the media, or send a 3XX.

>
> To push the responsibility for providing  this service of a B.com  
> user to another domain lets say A.com, is to say the least  
> problematic. Even if it solves some issues like those related to  
> early media and HERFP, it introduces lots of other issues.

However, those other issues are arguably tractable, in that at least  
the UA now has the information needed to do something intelligent  
rather than just failing. And "just failing" is what happens now to  
forked calls where multiple legs have early or late media.

--
Dean


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On Mar 3, 2010, at 3:37 AM, Elwell, John wrote:

> Serial forking with early media depends on support for PRACK and  
> UPDATE, which are not mandatory items for UAs and therefore will not  
> always be available.
>

As always, I think early media was just a really bad idea. But we  
always dance away from trying to fix the problem.

--
Dean

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Subject: Re: [sipcore] OPTIONS 3xx: To fork or not to fork?
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On Mar 3, 2010, at 3:40 AM, Elwell, John wrote:

> I would like to expand on this issue a little. Suppose I have at my  
> home proxy a policy whereby calls go to B1 and B2 in parallel, and  
> after T seconds of no answer calls instead go to B3 (voicemail,  
> perhaps). There is no way that the proxy could convey that policy in  
> a 3xx response to the UAC. To make this policy (or similar policies)  
> work, the proxy would need to manufacturer temporary contact URIs  
> that map to B1 and B2 and send these back in a 302. It would then  
> need to hope that the UAC does indeed establish new requests in  
> parallel to the two temporary contacts for B1 and B2. The proxy  
> would need to treat the two requests as related and if both are  
> unanswered after T seconds, reject one with a 4xx and the other with  
> a 302 (this time with a third temporary contact URI mapped to B3).  
> So policy CAN be met by the proxy, although with dependence on the  
> UAC doing the right thing.

So here, we either need your home proxy to be a home B2BUA (After all,  
what happens when B1 and B2 both send back early media, or both get  
answered by a bot?), OR we need a richer 302 response that conveys  
"Try these two destinations in parallel, and if neither replies in X  
seconds, try this third destination).

Either is tractable. Admittedly, the second requires extending the  
specification of a 302 response, but we have the capacity to do this.  
A few standardized parameters would work fine, with no backward  
compatibility issues for  older nodes that don't understand the  
parameters. Much could be done with q-values.


>
> Elsewhere in this thread, somebody mentioned charging. The fact is  
> that calls might cross a domain that charges (e.g., for  
> international calls via a server provider into an enterprise that  
> has a proxy that makes use of 302 responses to achieve forking).  
> Since only one of the calls should normally be answered (or if a  
> second is answered it should be cleared very quickly), charging  
> based on duration after answer should not be a big issue. However,  
> if a domain charges for call establishment attempts, it could be a  
> problem. In this example, the domain handling the international leg  
> would not be aware that the multiple calls are really part of the  
> same call. I don't know whether this is an issue in practice or not.

The real question is "who gets charged, and are they surprised by it?"

If the "home proxy" is forwarding the call to a toll destination, then  
either the "home proxy" needs to get charged for the toll (making it  
more B2BUA is), or the originating user needs to get charged for it,  
in which case the originating user needs to know about the toll  
destination (requiring a 302).

Having your home proxy do conventional SIP forking just doesn't work  
here.

--
Dean

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emphatic +1 to Dean's wise comment

G

On Mar 4, 2010, at 11:52 AM, Dean Willis wrote:

>=20
> On Mar 3, 2010, at 3:37 AM, Elwell, John wrote:
>=20
>> Serial forking with early media depends on support for PRACK and =
UPDATE, which are not mandatory items for UAs and therefore will not =
always be available.
>>=20
>=20
> As always, I think early media was just a really bad idea. But we =
always dance away from trying to fix the problem.
>=20
> --
> Dean
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] OPTIONS: What does it mean?
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On Mar 1, 2010, at 12:58 PM, Paul Kyzivat wrote:

> IMO, if the request has a to-tag, then the UAS should treat it as an  
> in-dialog request not associated with any particular usage. If there  
> is no matching dialog, then it should return a 481 response.
>

I haven't thought this thru: Is there any chance of this being used to  
fish for existing dialogs? If not, then your position seems reasonable.

--
Dean

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inline

Dean Willis wrote:
> 
> On Mar 3, 2010, at 3:40 AM, Elwell, John wrote:
> 
>> I would like to expand on this issue a little. Suppose I have at my 
>> home proxy a policy whereby calls go to B1 and B2 in parallel, and 
>> after T seconds of no answer calls instead go to B3 (voicemail, 
>> perhaps). There is no way that the proxy could convey that policy in a 
>> 3xx response to the UAC. To make this policy (or similar policies) 
>> work, the proxy would need to manufacturer temporary contact URIs that 
>> map to B1 and B2 and send these back in a 302. It would then need to 
>> hope that the UAC does indeed establish new requests in parallel to 
>> the two temporary contacts for B1 and B2. The proxy would need to 
>> treat the two requests as related and if both are unanswered after T 
>> seconds, reject one with a 4xx and the other with a 302 (this time 
>> with a third temporary contact URI mapped to B3). So policy CAN be met 
>> by the proxy, although with dependence on the UAC doing the right thing.
> 
> So here, we either need your home proxy to be a home B2BUA (After all, 
> what happens when B1 and B2 both send back early media, or both get 
> answered by a bot?), OR we need a richer 302 response that conveys "Try 
> these two destinations in parallel, and if neither replies in X seconds, 
> try this third destination).
> 
> Either is tractable. Admittedly, the second requires extending the 
> specification of a 302 response, but we have the capacity to do this. A 
> few standardized parameters would work fine, with no backward 
> compatibility issues for  older nodes that don't understand the 
> parameters. Much could be done with q-values.

This capability is (arguably) already there:

provide two contacts with equal q-values, and the third with a smaller 
q-value.

	Thanks,
	Paul

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] OPTIONS: What does it mean?
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Dean Willis wrote:
> 
> On Mar 1, 2010, at 12:58 PM, Paul Kyzivat wrote:
> 
>> IMO, if the request has a to-tag, then the UAS should treat it as an 
>> in-dialog request not associated with any particular usage. If there 
>> is no matching dialog, then it should return a 481 response.
>>
> 
> I haven't thought this thru: Is there any chance of this being used to 
> fish for existing dialogs? If not, then your position seems reasonable.

There is a *possibility* of using it that way.
But I don't think that is a very practical approach to anything.

If you don't already have access to the dialog info, but suspect there 
is a dialog and want to discover it, the namespace of dialog ids is 
*very* large. Trying to guess and verify this way would be very expensive.

And, if you thought this was useful, and we blocked use of OPTIONS for 
this, you could simply switch to some other message - e.g. UPDATE.

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Thu Mar  4 12:58:01 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Thu, 4 Mar 2010 21:57:59 +0100
Thread-Topic: INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

I thought we already had the annual thread on solving early media, but I gu=
ess it doesn't hurt to discuss it a little more often ;)

>>Serial forking with early media depends on support for PRACK and  UPDATE,=
 which are not mandatory items for UAs and therefore will not always be ava=
ilable.
>
>As always, I think early media was just a really bad idea. But we always d=
ance away from trying to fix the problem.
>
>>I would like to expand on this issue a little. Suppose I have at my
>>home proxy a policy whereby calls go to B1 and B2 in parallel, and
>>after T seconds of no answer calls instead go to B3 (voicemail,
>>perhaps). There is no way that the proxy could convey that policy in
>>a 3xx response to the UAC. To make this policy (or similar policies)
>>work, the proxy would need to manufacturer temporary contact URIs
>>that map to B1 and B2 and send these back in a 302. It would then
>>need to hope that the UAC does indeed establish new requests in
>>parallel to the two temporary contacts for B1 and B2. The proxy
>>would need to treat the two requests as related and if both are
>>unanswered after T seconds, reject one with a 4xx and the other with
>>a 302 (this time with a third temporary contact URI mapped to B3).
>>So policy CAN be met by the proxy, although with dependence on the
>>UAC doing the right thing.
>
>So here, we either need your home proxy to be a home B2BUA (After all,
>what happens when B1 and B2 both send back early media, or both get
>answered by a bot?), OR we need a richer 302 response that conveys
>"Try these two destinations in parallel, and if neither replies in X
>seconds, try this third destination).

But, the UAC still doesn't know whether B1 and B2 would send early media.

>Either is tractable. Admittedly, the second requires extending the
>specification of a 302 response, but we have the capacity to do this.
>A few standardized parameters would work fine, with no backward
>compatibility issues for  older nodes that don't understand the
>parameters. Much could be done with q-values.

Much could also have been done by alloing P-Early-Media:inactive in the INV=
ITE request, and then
indicating to the UAC for which early dialog early media has been "allowed"=
. The UAC can then change
that by sending a PRACK/UPDATE

Also, there could be a feature tag which UAS(s) use during registration to =
indicate whether they support sending of early media or not.

I don't think that SIP UAs will send early media that often. PSTN gateways =
can do it.=20

Appplication servers can also do it, but they are most likely located "in f=
ront of" the forking proxy. And, if they would send a 3xx response, the 3xx=
 from the forking proxy would most likely be rejected by the UAC (since it =
already received a final response from the AS). So, the application servers=
 could instead create a separate early dialog with the UAC, but that early =
dialog (and possible announcements etc that goes with it) would then be ter=
minated in an unwanted manner by the 3xx from the forking proxy.

>> Elsewhere in this thread, somebody mentioned charging. The fact is
>> that calls might cross a domain that charges (e.g., for
>> international calls via a server provider into an enterprise that
>> has a proxy that makes use of 302 responses to achieve forking).
>> Since only one of the calls should normally be answered (or if a
>> second is answered it should be cleared very quickly), charging
>> based on duration after answer should not be a big issue. However,
>> if a domain charges for call establishment attempts, it could be a
>> problem. In this example, the domain handling the international leg
>> would not be aware that the multiple calls are really part of the
>> same call. I don't know whether this is an issue in practice or not.
>
>The real question is "who gets charged, and are they surprised by it?"
>
>If the "home proxy" is forwarding the call to a toll destination, then
>either the "home proxy" needs to get charged for the toll (making it
>more B2BUA is), or the originating user needs to get charged for it,
>in which case the originating user needs to know about the toll
>destination (requiring a 302).

That assumes that the forking proxy has information about whether a contact=
 is associated with a toll number, and that the calling device is able to p=
resent that information to the caller.

There is a service called Advice Of Charge. I believe it contains a feature=
 to not allow certain calls to take place if the calling user can't be info=
rmed about it, but I don't remember the details.

Regards,

Christer=

From christer.holmberg@ericsson.com  Fri Mar  5 00:00:57 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Fri, 5 Mar 2010 09:00:56 +0100
Thread-Topic: OPTIONS 3xx: Contact filtering?
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Subject: [sipcore] OPTIONS 3xx: Contact filtering?
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Hi,

Paul suggested that a UAC would use the Request-Disposition header to indic=
ate whether forking proxy should proxy (using the "proxy" R-D value) the OP=
TIONS request, or send a 3xx response (using the "redirect" R-D value).

Now, assuming "redirect" is used, the question is:

If the request also contains an Accept-Contact/Reject-Contact header field,=
 would the proxy in the 3xx response only include contacts which matches th=
e A-C/R-C criterias (in the same way as the proxy would only fork towards U=
AS(s) that matches the criterias)?

I can't find any text in 3841 whichs indicates that A-C is not applicable w=
hen "R-D:redirect" is used.

I think that would be a good feature.

Regards,

Christer

From pkyzivat@cisco.com  Fri Mar  5 05:11:45 2010
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Yes, I would certainly expect that A-C would apply to selecting the 
contacts returned in the 3xx response.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> 
> Paul suggested that a UAC would use the Request-Disposition header to indicate whether forking proxy should proxy (using the "proxy" R-D value) the OPTIONS request, or send a 3xx response (using the "redirect" R-D value).
> 
> Now, assuming "redirect" is used, the question is:
> 
> If the request also contains an Accept-Contact/Reject-Contact header field, would the proxy in the 3xx response only include contacts which matches the A-C/R-C criterias (in the same way as the proxy would only fork towards UAS(s) that matches the criterias)?
> 
> I can't find any text in 3841 whichs indicates that A-C is not applicable when "R-D:redirect" is used.
> 
> I think that would be a good feature.
> 
> Regards,
> 
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From dean.willis@softarmor.com  Fri Mar  5 13:19:42 2010
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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On Mar 4, 2010, at 2:57 PM, Christer Holmberg wrote:
>>
>>> I would like to expand on this issue a little. Suppose I have at my
>>> home proxy a policy whereby calls go to B1 and B2 in parallel, and
>>> after T seconds of no answer calls instead go to B3 (voicemail,
>>> perhaps). There is no way that the proxy could convey that policy in
>>> a 3xx response to the UAC. To make this policy (or similar policies)
>>> work, the proxy would need to manufacturer temporary contact URIs
>>> that map to B1 and B2 and send these back in a 302. It would then
>>> need to hope that the UAC does indeed establish new requests in
>>> parallel to the two temporary contacts for B1 and B2. The proxy
>>> would need to treat the two requests as related and if both are
>>> unanswered after T seconds, reject one with a 4xx and the other with
>>> a 302 (this time with a third temporary contact URI mapped to B3).
>>> So policy CAN be met by the proxy, although with dependence on the
>>> UAC doing the right thing.
>>
>> So here, we either need your home proxy to be a home B2BUA (After  
>> all,
>> what happens when B1 and B2 both send back early media, or both get
>> answered by a bot?), OR we need a richer 302 response that conveys
>> "Try these two destinations in parallel, and if neither replies in X
>> seconds, try this third destination).
>
> But, the UAC still doesn't know whether B1 and B2 would send early  
> media.

By offering different ports to B1 and B2 (especially with the media-to- 
signaling coupling extensions that have been proposed) UAC at least  
knows which one of B1 and B2 each stream is coming from.


--
dean
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 4, 2010, =
at 2:57 PM, Christer Holmberg wrote:</div><blockquote =
type=3D"cite"><div><blockquote type=3D"cite"><font =
class=3D"Apple-style-span" =
color=3D"#000000"><br></font></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">I would like to expand on this =
issue a little. Suppose I have at =
my<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">home proxy a policy whereby calls go to B1 and B2 in =
parallel, and<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">after T seconds of no answer =
calls instead go to B3 =
(voicemail,<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">perhaps). There is no way that =
the proxy could convey that policy =
in<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">a 3xx response to the UAC. To make this policy (or similar =
policies)<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">work, the proxy would need to =
manufacturer temporary contact =
URIs<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">that map to B1 and B2 and send these back in a 302. It =
would then<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">need to hope that the UAC does =
indeed establish new requests =
in<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">parallel to the two temporary contacts for B1 and B2. The =
proxy<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">would need to treat the two requests as related and if =
both are<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">unanswered after T seconds, =
reject one with a 4xx and the other =
with<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">a 302 (this time with a third temporary contact URI mapped =
to B3).<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote=
 type=3D"cite">So policy CAN be met by the proxy, although with =
dependence on the<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">UAC doing the right =
thing.<br></blockquote></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">So here, we =
either need your home proxy to be a home B2BUA (After =
all,<br></blockquote><blockquote type=3D"cite">what happens when B1 and =
B2 both send back early media, or both get<br></blockquote><blockquote =
type=3D"cite">answered by a bot?), OR we need a richer 302 response that =
conveys<br></blockquote><blockquote type=3D"cite">"Try these two =
destinations in parallel, and if neither replies in =
X<br></blockquote><blockquote type=3D"cite">seconds, try this third =
destination).<br></blockquote><br>But, the UAC still doesn't know =
whether B1 and B2 would send early =
media.<br></div></blockquote><div><br></div><div>By offering different =
ports to B1 and B2 (especially with the media-to-signaling coupling =
extensions that have been proposed) UAC at least knows which one of B1 =
and B2 each stream is coming =
from.</div><div><br></div></div><div><br></div><div>--</div><div>dean</div=
></body></html>=

--Apple-Mail-11--393518732--

From christer.holmberg@ericsson.com  Fri Mar  5 14:36:05 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Dean Willis' <dean.willis@softarmor.com>
Date: Fri, 5 Mar 2010 23:36:02 +0100
Thread-Topic: INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,
	=09

>			I would like to expand on this issue a little. Suppose I have at my		=09
>			home proxy a policy whereby calls go to B1 and B2 in parallel, and
>			after T seconds of no answer calls instead go to B3 (voicemail,
>			perhaps). There is no way that the proxy could convey that policy in
>			a 3xx response to the UAC. To make this policy (or similar policies)
>			work, the proxy would need to manufacturer temporary contact URIs
>			that map to B1 and B2 and send these back in a 302. It would then
>			need to hope that the UAC does indeed establish new requests in
>			parallel to the two temporary contacts for B1 and B2. The proxy
>			would need to treat the two requests as related and if both are
>			unanswered after T seconds, reject one with a 4xx and the other with
>			a 302 (this time with a third temporary contact URI mapped to B3).
>			So policy CAN be met by the proxy, although with dependence on the
>			UAC doing the right thing.
>		=09
>
>		So here, we either need your home proxy to be a home B2BUA (After all,
>		what happens when B1 and B2 both send back early media, or both get
>		answered by a bot?), OR we need a richer 302 response that conveys
>		"Try these two destinations in parallel, and if neither replies in X
>		seconds, try this third destination).
>	=09
>	But, the UAC still doesn't know whether B1 and B2 would send early media.
>=09
>By offering different ports to B1 and B2 (especially with the media-to-sig=
naling coupling extensions that have been proposed) UAC at least knows whic=
h one of B1 and B2 each stream is coming from.

Yes. But, there are other mechanisms to solve that, and it still doesn't so=
lve the fundamental problem: the UAC can receive multiple media streams ass=
ociated with multiple early dialogs.

Also, in bandwidth sensitive access networks there is only limited bandwidt=
h, normally calculated based on the need for media associated with one earl=
y dialog. Even if the UAC is able to select and play only one stream to the=
 user, the voice quality will suffer due to too little bandwidth.

Regards,

Christer

From pkyzivat@cisco.com  Fri Mar  5 15:04:13 2010
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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If you want parallel forking, even if done by the UAC, there is no good 
solution to this problem. The parallel early media is going to be 
generated and something must happen to it. The only question is which of 
the bad solutions you prefer.

For instance, you may have one stream asking you to enter an access 
code, another playing an out-of-service tone, and a third sending no 
media which will eventually pick up with a real person. What would you 
*like* for the caller to hear?

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> 		
> 
>> 			I would like to expand on this issue a little. Suppose I have at my			
>> 			home proxy a policy whereby calls go to B1 and B2 in parallel, and
>> 			after T seconds of no answer calls instead go to B3 (voicemail,
>> 			perhaps). There is no way that the proxy could convey that policy in
>> 			a 3xx response to the UAC. To make this policy (or similar policies)
>> 			work, the proxy would need to manufacturer temporary contact URIs
>> 			that map to B1 and B2 and send these back in a 302. It would then
>> 			need to hope that the UAC does indeed establish new requests in
>> 			parallel to the two temporary contacts for B1 and B2. The proxy
>> 			would need to treat the two requests as related and if both are
>> 			unanswered after T seconds, reject one with a 4xx and the other with
>> 			a 302 (this time with a third temporary contact URI mapped to B3).
>> 			So policy CAN be met by the proxy, although with dependence on the
>> 			UAC doing the right thing.
>> 			
>>
>> 		So here, we either need your home proxy to be a home B2BUA (After all,
>> 		what happens when B1 and B2 both send back early media, or both get
>> 		answered by a bot?), OR we need a richer 302 response that conveys
>> 		"Try these two destinations in parallel, and if neither replies in X
>> 		seconds, try this third destination).
>> 		
>> 	But, the UAC still doesn't know whether B1 and B2 would send early media.
>> 	
>> By offering different ports to B1 and B2 (especially with the media-to-signaling coupling extensions that have been proposed) UAC at least knows which one of B1 and B2 each stream is coming from.
> 
> Yes. But, there are other mechanisms to solve that, and it still doesn't solve the fundamental problem: the UAC can receive multiple media streams associated with multiple early dialogs.
> 
> Also, in bandwidth sensitive access networks there is only limited bandwidth, normally calculated based on the need for media associated with one early dialog. Even if the UAC is able to select and play only one stream to the user, the voice quality will suffer due to too little bandwidth.
> 
> Regards,
> 
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From christer.holmberg@ericsson.com  Fri Mar  5 15:15:48 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Sat, 6 Mar 2010 00:15:47 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>If you want parallel forking, even if done by the UAC, there is no good so=
lution to this problem. The parallel early media is going to be generated a=
nd something must happen to it. The only question is=20
>which of the bad solutions you prefer.
>
>For instance, you may have one stream asking you to enter an access code, =
another playing an out-of-service tone, and a third sending no media which =
will eventually pick up with a real person. What would=20
>you *like* for the caller to hear?

For this reasons you have B2BUAs, application servers, gating nodes etc, th=
at at least TRY to deal with the problem, to ensure that whatever the user =
eventually hears, at least the quality will be good.

Regards,

Christer



Christer Holmberg wrote:
> Hi,
> 	=09
>=20
>> 			I would like to expand on this issue a little. Suppose I have at my		=
=09
>> 			home proxy a policy whereby calls go to B1 and B2 in parallel, and
>> 			after T seconds of no answer calls instead go to B3 (voicemail,
>> 			perhaps). There is no way that the proxy could convey that policy in
>> 			a 3xx response to the UAC. To make this policy (or similar policies)
>> 			work, the proxy would need to manufacturer temporary contact URIs
>> 			that map to B1 and B2 and send these back in a 302. It would then
>> 			need to hope that the UAC does indeed establish new requests in
>> 			parallel to the two temporary contacts for B1 and B2. The proxy
>> 			would need to treat the two requests as related and if both are
>> 			unanswered after T seconds, reject one with a 4xx and the other with
>> 			a 302 (this time with a third temporary contact URI mapped to B3).
>> 			So policy CAN be met by the proxy, although with dependence on the
>> 			UAC doing the right thing.
>> 		=09
>>
>> 		So here, we either need your home proxy to be a home B2BUA (After all,
>> 		what happens when B1 and B2 both send back early media, or both get
>> 		answered by a bot?), OR we need a richer 302 response that conveys
>> 		"Try these two destinations in parallel, and if neither replies in X
>> 		seconds, try this third destination).
>> 	=09
>> 	But, the UAC still doesn't know whether B1 and B2 would send early medi=
a.
>> =09
>> By offering different ports to B1 and B2 (especially with the media-to-s=
ignaling coupling extensions that have been proposed) UAC at least knows wh=
ich one of B1 and B2 each stream is coming from.
>=20
> Yes. But, there are other mechanisms to solve that, and it still doesn't =
solve the fundamental problem: the UAC can receive multiple media streams a=
ssociated with multiple early dialogs.
>=20
> Also, in bandwidth sensitive access networks there is only limited bandwi=
dth, normally calculated based on the need for media associated with one ea=
rly dialog. Even if the UAC is able to select and play only one stream to t=
he user, the voice quality will suffer due to too little bandwidth.
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From pkyzivat@cisco.com  Fri Mar  5 15:30:33 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Christer Holmberg wrote:
> Hi,
> 
>> If you want parallel forking, even if done by the UAC, there is no good solution to this problem. The parallel early media is going to be generated and something must happen to it. The only question is 
>> which of the bad solutions you prefer.
>>
>> For instance, you may have one stream asking you to enter an access code, another playing an out-of-service tone, and a third sending no media which will eventually pick up with a real person. What would 
>> you *like* for the caller to hear?
> 
> For this reasons you have B2BUAs, application servers, gating nodes etc, that at least TRY to deal with the problem, to ensure that whatever the user eventually hears, at least the quality will be good.

For what that's worth! You get a good quality rendering of that 
out-of-service tone, and so hang up even though the call was ringing 
elsewhere, but without any ringback.

I guess we just have to "let a 1000 flowers bloom" on this one.

	Paul

> Regards,
> 
> Christer
> 
> 
> 
> Christer Holmberg wrote:
>> Hi,
>> 		
>>
>>> 			I would like to expand on this issue a little. Suppose I have at my			
>>> 			home proxy a policy whereby calls go to B1 and B2 in parallel, and
>>> 			after T seconds of no answer calls instead go to B3 (voicemail,
>>> 			perhaps). There is no way that the proxy could convey that policy in
>>> 			a 3xx response to the UAC. To make this policy (or similar policies)
>>> 			work, the proxy would need to manufacturer temporary contact URIs
>>> 			that map to B1 and B2 and send these back in a 302. It would then
>>> 			need to hope that the UAC does indeed establish new requests in
>>> 			parallel to the two temporary contacts for B1 and B2. The proxy
>>> 			would need to treat the two requests as related and if both are
>>> 			unanswered after T seconds, reject one with a 4xx and the other with
>>> 			a 302 (this time with a third temporary contact URI mapped to B3).
>>> 			So policy CAN be met by the proxy, although with dependence on the
>>> 			UAC doing the right thing.
>>> 			
>>>
>>> 		So here, we either need your home proxy to be a home B2BUA (After all,
>>> 		what happens when B1 and B2 both send back early media, or both get
>>> 		answered by a bot?), OR we need a richer 302 response that conveys
>>> 		"Try these two destinations in parallel, and if neither replies in X
>>> 		seconds, try this third destination).
>>> 		
>>> 	But, the UAC still doesn't know whether B1 and B2 would send early media.
>>> 	
>>> By offering different ports to B1 and B2 (especially with the media-to-signaling coupling extensions that have been proposed) UAC at least knows which one of B1 and B2 each stream is coming from.
>> Yes. But, there are other mechanisms to solve that, and it still doesn't solve the fundamental problem: the UAC can receive multiple media streams associated with multiple early dialogs.
>>
>> Also, in bandwidth sensitive access networks there is only limited bandwidth, normally calculated based on the need for media associated with one early dialog. Even if the UAC is able to select and play only one stream to the user, the voice quality will suffer due to too little bandwidth.
>>
>> Regards,
>>
>> Christer
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> 

From christer.holmberg@ericsson.com  Sat Mar  6 01:10:14 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Sat, 6 Mar 2010 10:10:09 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>>>If you want parallel forking, even if done by the UAC, there is no good =
solution to this problem. The=20
>>>parallel early media is going to be generated and something must happen =
to it. The only question is
>>>which of the bad solutions you prefer.
>>>
>>>For instance, you may have one stream asking you to enter an access code=
, another playing an out-of-
>>>service tone, and a third sending no media which will eventually pick up=
 with a real person. What would
>>>you *like* for the caller to hear?
>>
>>For this reasons you have B2BUAs, application servers, gating nodes etc, =
that at least TRY to deal with the=20
>>problem, to ensure that whatever the user eventually hears, at least the =
quality will be good.
>
>For what that's worth! You get a good quality rendering of that
>out-of-service tone, and so hang up even though the call was ringing
>elsewhere, but without any ringback.
>
>I guess we just have to "let a 1000 flowers bloom" on this one.

As I've said before, I don't think there is a very big risk that the "wrong=
" early media would be presented to you.

Leaving PSTN interworking aside, most early media (at least the one of real=
 importance) will be generated by application servers/B2BUAs in the network=
, and they will normally be able to control the media pretty well. A SIP UA=
 might generate something, but unless it's a Henry Sinnreich device (where =
all the services are implemented in the end device :) I don't think UAs wil=
l normally send service related announcement etc.

Of course, if you have application servers sending media both in the origin=
ating and terminating network, there need to be some policy procedures on w=
hich media has higher priority etc.

Regards,

Christer






> Christer Holmberg wrote:
>> Hi,
>>
>>
>>>                     I would like to expand on this issue a little. Supp=
ose I have at my
>>>                     home proxy a policy whereby calls go to B1 and B2 i=
n parallel, and
>>>                     after T seconds of no answer calls instead go to B3=
 (voicemail,
>>>                     perhaps). There is no way that the proxy could conv=
ey that policy in
>>>                     a 3xx response to the UAC. To make this policy (or =
similar policies)
>>>                     work, the proxy would need to manufacturer temporar=
y contact URIs
>>>                     that map to B1 and B2 and send these back in a 302.=
 It would then
>>>                     need to hope that the UAC does indeed establish new=
 requests in
>>>                     parallel to the two temporary contacts for B1 and B=
2. The proxy
>>>                     would need to treat the two requests as related and=
 if both are
>>>                     unanswered after T seconds, reject one with a 4xx a=
nd the other with
>>>                     a 302 (this time with a third temporary contact URI=
 mapped to B3).
>>>                     So policy CAN be met by the proxy, although with de=
pendence on the
>>>                     UAC doing the right thing.
>>>
>>>
>>>             So here, we either need your home proxy to be a home B2BUA =
(After all,
>>>             what happens when B1 and B2 both send back early media, or =
both get
>>>             answered by a bot?), OR we need a richer 302 response that =
conveys
>>>             "Try these two destinations in parallel, and if neither rep=
lies in X
>>>             seconds, try this third destination).
>>>
>>>     But, the UAC still doesn't know whether B1 and B2 would send early =
media.
>>>
>>> By offering different ports to B1 and B2 (especially with the media-to-=
signaling coupling extensions that have been proposed) UAC at least knows w=
hich one of B1 and B2 each stream is coming from.
>> Yes. But, there are other mechanisms to solve that, and it still doesn't=
 solve the fundamental problem: the UAC can receive multiple media streams =
associated with multiple early dialogs.
>>
>> Also, in bandwidth sensitive access networks there is only limited bandw=
idth, normally calculated based on the need for media associated with one e=
arly dialog. Even if the UAC is able to select and play only one stream to =
the user, the voice quality will suffer due to too little bandwidth.
>>
>> Regards,
>>
>> Christer
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>=

From dean.willis@softarmor.com  Sat Mar  6 10:13:30 2010
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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On Mar 5, 2010, at 5:04 PM, Paul Kyzivat wrote:

> If you want parallel forking, even if done by the UAC, there is no  
> good solution to this problem. The parallel early media is going to  
> be generated and something must happen to it. The only question is  
> which of the bad solutions you prefer.
>
> For instance, you may have one stream asking you to enter an access  
> code, another playing an out-of-service tone, and a third sending no  
> media which will eventually pick up with a real person. What would  
> you *like* for the caller to hear?
>

I at least want the caller to be able to correlate a media flow with a  
signaling leg so they know which one is which.

--
Dean

From pkyzivat@cisco.com  Sun Mar  7 08:47:37 2010
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Date: Sun, 07 Mar 2010 11:47:34 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Christer Holmberg wrote:
> Hi,
> 
>>>> If you want parallel forking, even if done by the UAC, there is no good solution to this problem. The 
>>>> parallel early media is going to be generated and something must happen to it. The only question is
>>>> which of the bad solutions you prefer.
>>>>
>>>> For instance, you may have one stream asking you to enter an access code, another playing an out-of-
>>>> service tone, and a third sending no media which will eventually pick up with a real person. What would
>>>> you *like* for the caller to hear?
>>> For this reasons you have B2BUAs, application servers, gating nodes etc, that at least TRY to deal with the 
>>> problem, to ensure that whatever the user eventually hears, at least the quality will be good.
>> For what that's worth! You get a good quality rendering of that
>> out-of-service tone, and so hang up even though the call was ringing
>> elsewhere, but without any ringback.
>>
>> I guess we just have to "let a 1000 flowers bloom" on this one.
> 
> As I've said before, I don't think there is a very big risk that the "wrong" early media would be presented to you.


> Leaving PSTN interworking aside, most early media (at least the one of real importance) will be generated by application servers/B2BUAs in the network, and they will normally be able to control the media pretty well. A SIP UA might generate something, but unless it's a Henry Sinnreich device (where all the services are implemented in the end device :) I don't think UAs will normally send service related announcement etc.

I gather you are arguing that there just aren't many calls with early 
media. But "leaving PSTN interworking aside" isn't really fair, since 
there is so much of that. For instance there could be a lot of the 
google service that simply forks calls to multiple PSTN lines. And don't 
most pstn calls have in-band ringback? That would get multiple early 
media, though if its just ringback it doesn't matter if you play it or 
not. But if one of those becomes defective and starts playing out of 
service messages then it does matter which one you play.

> Of course, if you have application servers sending media both in the originating and terminating network, there need to be some policy procedures on which media has higher priority etc.

Policy doesn't help a lot. It content, not policy, that determines which 
is important to hear.

Perhaps the best we can do is discourage the use of early media for 
anything important.

	Thanks,
	Paul

> Regards,
> 
> Christer
> 
> 
> 
> 
> 
> 
>> Christer Holmberg wrote:
>>> Hi,
>>>
>>>
>>>>                     I would like to expand on this issue a little. Suppose I have at my
>>>>                     home proxy a policy whereby calls go to B1 and B2 in parallel, and
>>>>                     after T seconds of no answer calls instead go to B3 (voicemail,
>>>>                     perhaps). There is no way that the proxy could convey that policy in
>>>>                     a 3xx response to the UAC. To make this policy (or similar policies)
>>>>                     work, the proxy would need to manufacturer temporary contact URIs
>>>>                     that map to B1 and B2 and send these back in a 302. It would then
>>>>                     need to hope that the UAC does indeed establish new requests in
>>>>                     parallel to the two temporary contacts for B1 and B2. The proxy
>>>>                     would need to treat the two requests as related and if both are
>>>>                     unanswered after T seconds, reject one with a 4xx and the other with
>>>>                     a 302 (this time with a third temporary contact URI mapped to B3).
>>>>                     So policy CAN be met by the proxy, although with dependence on the
>>>>                     UAC doing the right thing.
>>>>
>>>>
>>>>             So here, we either need your home proxy to be a home B2BUA (After all,
>>>>             what happens when B1 and B2 both send back early media, or both get
>>>>             answered by a bot?), OR we need a richer 302 response that conveys
>>>>             "Try these two destinations in parallel, and if neither replies in X
>>>>             seconds, try this third destination).
>>>>
>>>>     But, the UAC still doesn't know whether B1 and B2 would send early media.
>>>>
>>>> By offering different ports to B1 and B2 (especially with the media-to-signaling coupling extensions that have been proposed) UAC at least knows which one of B1 and B2 each stream is coming from.
>>> Yes. But, there are other mechanisms to solve that, and it still doesn't solve the fundamental problem: the UAC can receive multiple media streams associated with multiple early dialogs.
>>>
>>> Also, in bandwidth sensitive access networks there is only limited bandwidth, normally calculated based on the need for media associated with one early dialog. Even if the UAC is able to select and play only one stream to the user, the voice quality will suffer due to too little bandwidth.
>>>
>>> Regards,
>>>
>>> Christer
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>

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To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
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Cc: 'SIPCORE' <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Paul Kyzivat wrote:
> 
> Perhaps the best we can do is discourage the use of early media for
> anything important.
> 

I'm not sure this scenario is that common, but if it is something of
interest, perhaps it's possible to add a header/parameter conveying the
nature of the media stream (ringback, service message, voice, etc.). This is
similar to the content-disposition handling parameter, but more informative
than "optional" and "required". This way, a B2BUA would be able to choose
the stream to forward and even without a B2BUA, an endpoint receiving
several streams would know which stream to play back to the user.



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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?] //copy from the sipcore Discussion Archive
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Hi,

Sorry for copy the mail from archive(I can not recv sipcore mails these 
days).

inlines.

Thanks,

Gao

From: Paul Kyzivat <pkyzivat at cisco.com> 
To: Christer Holmberg <christer.holmberg at ericsson.com> 
Cc: SIPCORE <sipcore at ietf.org> 
Date: Sun, 07 Mar 2010 11:47:34 -0500 
In-reply-to: <FF84A09F50A6DC48ACB6714F4666CC745C10EE4166 at 
ESESSCMS0354.eemea.ericsson.se> 
References: <FF84A09F50A6DC48ACB6714F4666CC745C10EE415F at 
ESESSCMS0354.eemea.ericsson.se> <AEA0E9BF-8B6A-4AAA-81B6-A7C68F6B7309 at 
softarmor.com> <FF84A09F50A6DC48ACB6714F4666CC745C1102CCCD at 
ESESSCMS0354.eemea.ericsson.se> <4B918DED.6090108 at cisco.com> 
<FF84A09F50A6DC48ACB6714F4666CC745C1102CCD1 at 
ESESSCMS0354.eemea.ericsson.se>, <4B919413.60506 at cisco.com> 
<FF84A09F50A6DC48ACB6714F4666CC745C10EE4166 at 
ESESSCMS0354.eemea.ericsson.se> 

Christer Holmberg wrote:

Hi,


If you want parallel forking, even if done by the UAC, there is no good 
solution to this problem. The parallel early media is going to be 
generated and something must happen to it. The only question is 
which of the bad solutions you prefer.

For instance, you may have one stream asking you to enter an access code, 
another playing an out-of-
service tone, and a third sending no media which will eventually pick up 
with a real person. What would
you *like* for the caller to hear?

For this reasons you have B2BUAs, application servers, gating nodes etc, 
that at least TRY to deal with the problem, to ensure that whatever the 
user eventually hears, at least the quality will be good. 
For what that's worth! You get a good quality rendering of that
out-of-service tone, and so hang up even though the call was ringing
elsewhere, but without any ringback.

I guess we just have to "let a 1000 flowers bloom" on this one.


As I've said before, I don't think there is a very big risk that the 
"wrong" early media would be presented to you.




Leaving PSTN interworking aside, most early media (at least the one of 
real importance) will be generated by application servers/B2BUAs in the 
network, and they will normally be able to control the media pretty well. 
A SIP UA might generate something, but unless it's a Henry Sinnreich 
device (where all the services are implemented in the end device :) I 
don't think UAs will normally send service related announcement etc.



I gather you are arguing that there just aren't many calls with early 
media. But "leaving PSTN interworking aside" isn't really fair, since 
there is so much of that. For instance there could be a lot of the google 
service that simply forks calls to multiple PSTN lines. And don't most 
pstn calls have in-band ringback? That would get multiple early media, 
though if its just ringback it doesn't matter if you play it or not. But 
if one of those becomes defective and starts playing out of service 
messages then it does matter which one you play. 

[Gao] I agree. The forking proxy/AS have no way to know the contents of 
early media currently. So, it is not proper to judge which one is more 
important and do filter things. 

Of course, if you have application servers sending media both in the 
originating and terminating network, there need to be some policy 
procedures on which media has higher priority etc.



Policy doesn't help a lot. It content, not policy, that determines which 
is important to hear. 

[Gao] Yes.

Perhaps the best we can do is discourage the use of early media for 
anything important. 

[Gao] As many operating equipment/network has used early media for 
someting important as DTMF number collecting or code collecting, 
discourage the usage may cause interworking and compatible problem.


===================================
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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">Sorry for copy the mail from archive(I
can not recv sipcore mails these days).</font>
<br>
<br><font size=2 face="sans-serif">inlines.</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao</font>
<br>
<br><font size=2 face="sans-serif">From: Paul Kyzivat &lt;pkyzivat at cisco.com&gt;
</font>
<br><font size=2 face="sans-serif">To: Christer Holmberg &lt;christer.holmberg
at ericsson.com&gt; </font>
<br><font size=2 face="sans-serif">Cc: SIPCORE &lt;sipcore at ietf.org&gt;
</font>
<br><font size=2 face="sans-serif">Date: Sun, 07 Mar 2010 11:47:34 -0500
</font>
<br><font size=2 face="sans-serif">In-reply-to: &lt;FF84A09F50A6DC48ACB6714F4666CC745C10EE4166
at ESESSCMS0354.eemea.ericsson.se&gt; </font>
<br><font size=2 face="sans-serif">References: &lt;FF84A09F50A6DC48ACB6714F4666CC745C10EE415F
at ESESSCMS0354.eemea.ericsson.se&gt; &lt;AEA0E9BF-8B6A-4AAA-81B6-A7C68F6B7309
at softarmor.com&gt; &lt;FF84A09F50A6DC48ACB6714F4666CC745C1102CCCD at
ESESSCMS0354.eemea.ericsson.se&gt; &lt;4B918DED.6090108 at cisco.com&gt;
&lt;FF84A09F50A6DC48ACB6714F4666CC745C1102CCD1 at ESESSCMS0354.eemea.ericsson.se&gt;,
&lt;4B919413.60506 at cisco.com&gt; &lt;FF84A09F50A6DC48ACB6714F4666CC745C10EE4166
at ESESSCMS0354.eemea.ericsson.se&gt; </font>
<br>
<br><font size=2 color=#008080 face="sans-serif">Christer Holmberg wrote:</font>
<br>
<br><font size=2 color=#008080 face="sans-serif">Hi,</font>
<br>
<br>
<br><font size=2 color=#008080 face="sans-serif">If you want parallel forking,
even if done by the UAC, there is no good solution to this problem. The
parallel early media is going to be generated and something must happen
to it. The only question is </font>
<br><font size=2 color=#008080 face="sans-serif">which of the bad solutions
you prefer.</font>
<br>
<br><font size=2 color=#008080 face="sans-serif">For instance, you may
have one stream asking you to enter an access code, another playing an
out-of-</font>
<br><font size=2 color=#008080 face="sans-serif">service tone, and a third
sending no media which will eventually pick up with a real person. What
would</font>
<br><font size=2 color=#008080 face="sans-serif">you *like* for the caller
to hear?</font>
<br>
<br><font size=2 color=#008080 face="sans-serif">For this reasons you have
B2BUAs, application servers, gating nodes etc, that at least TRY to deal
with the problem, to ensure that whatever the user eventually hears, at
least the quality will be good. </font>
<br><font size=2 color=#008080 face="sans-serif">For what that's worth!
You get a good quality rendering of that</font>
<br><font size=2 color=#008080 face="sans-serif">out-of-service tone, and
so hang up even though the call was ringing</font>
<br><font size=2 color=#008080 face="sans-serif">elsewhere, but without
any ringback.</font>
<br>
<br><font size=2 color=#008080 face="sans-serif">I guess we just have to
&quot;let a 1000 flowers bloom&quot; on this one.</font>
<br>
<br>
<br><font size=2 color=#008080 face="sans-serif">As I've said before, I
don't think there is a very big risk that the &quot;wrong&quot; early media
would be presented to you.</font>
<br>
<br>
<br>
<br>
<br><font size=2 color=#008080 face="sans-serif">Leaving PSTN interworking
aside, most early media (at least the one of real importance) will be generated
by application servers/B2BUAs in the network, and they will normally be
able to control the media pretty well. A SIP UA might generate something,
but unless it's a Henry Sinnreich device (where all the services are implemented
in the end device :) I don't think UAs will normally send service related
announcement etc.</font>
<br>
<br>
<br>
<br><font size=2 color=blue face="sans-serif">I gather you are arguing
that there just aren't many calls with early media. But &quot;leaving PSTN
interworking aside&quot; isn't really fair, since there is so much of that.
For instance there could be a lot of the google service that simply forks
calls to multiple PSTN lines. And don't most pstn calls have in-band ringback?
That would get multiple early media, though if its just ringback it doesn't
matter if you play it or not. But if one of those becomes defective and
starts playing out of service messages then it does matter which one you
play. </font>
<br>
<br><font size=2 face="sans-serif">[Gao] I agree. The forking proxy/AS
have no way to know the contents of early media currently. So, it is not
proper to judge which one is more important and do filter things. </font>
<br>
<br><font size=2 color=#008080 face="sans-serif">Of course, if you have
application servers sending media both in the originating and terminating
network, there need to be some policy procedures on which media has higher
priority etc.</font>
<br>
<br>
<br>
<br><font size=2 color=blue face="sans-serif">Policy doesn't help a lot.
It content, not policy, that determines which is important to hear. </font>
<br>
<br><font size=2 face="sans-serif">[Gao] Yes.</font>
<br>
<br><font size=2 color=blue face="sans-serif">Perhaps the best we can do
is discourage the use of early media for anything important. </font>
<br>
<br><font size=2 face="sans-serif">[Gao] As many operating equipment/network
has used early media for someting important as DTMF number collecting or
code collecting, discourage the usage may cause interworking and compatible
problem.</font>
<br>
<br>
<br><font size=2 face="sans-serif">===================================<br>
 Zip &nbsp; &nbsp;: 210012<br>
 Tel &nbsp; &nbsp;: 87211<br>
 Tel2 &nbsp; :(+86)-025-52877211<br>
 e_mail : gao.yang2@zte.com.cn<br>
===================================</font><br><pre>
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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I'd like to talk about how our equipment handle this issue(I still think 
the early media issue of forking is exist).

Logically, we define two logical point(can be in one physical equipment):
1. Caller's Gateway, such as MGCF(of IMS) for PSTN.
2. Forking Equipment.

As Forking Equipment has no way to know the contents of early media. So it 
is not proper for the Forking Equipment to do any filter of early medias. 
So, we make the Forking Equipment transfer all early medias in different 
dialogs to the caller direction.

The Caller's Gateway is the equipment familiar with caller's capability 
and its preference. So, if caller can accept multiple early medias, the 
Caller's Gateway just render the all; if caller can only afford one early 
media, the Caller's Gateway just send one early media(Usually, our 
Caller's Gateway send its own ringing backtone, insteand of choosing one).

In fact, if some early medias are used as collecting DTMF for call 
re-dispatch or code, the Caller's Gateway may still discard them and cause 
problem. While caller's capability may cause more basic problem of 
forking, such as caller can only afford three forking dialog, but it is 
the fourth one has real call answer. Such problems caused by UAC's 
capability can only be handled as best effort, IMO.

Thanks,

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

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<br><font size=2 face="sans-serif">I'd like to talk about how our equipment
handle this issue(I still think the early media issue of forking is exist).</font>
<br>
<br><font size=2 face="sans-serif">Logically, we define two logical point(can
be in one physical equipment):</font>
<br><font size=2 face="sans-serif">1. Caller's Gateway, such as MGCF(of
IMS) for PSTN.</font>
<br><font size=2 face="sans-serif">2. Forking Equipment.</font>
<br>
<br><font size=2 face="sans-serif">As Forking Equipment has no way to know
the contents of early media. So it is not proper for the Forking Equipment
to do any filter of early medias. So, we make the Forking Equipment transfer
all early medias in different dialogs to the caller direction.</font>
<br>
<br><font size=2 face="sans-serif">The Caller's Gateway is the equipment
familiar with caller's capability and its preference. So, if caller can
accept multiple early medias, the Caller's Gateway just render the all;
if caller can only afford one early media, the Caller's Gateway just send
one early media(Usually, our Caller's Gateway send its own ringing backtone,
insteand of choosing one).</font>
<br>
<br><font size=2 face="sans-serif">In fact, if some early medias are used
as collecting DTMF for call re-dispatch or code, the Caller's Gateway may
still discard them and cause problem. While caller's capability may cause
more basic problem of forking, such as caller can only afford three forking
dialog, but it is the fourth one has real call answer. Such problems caused
by UAC's capability can only be handled as best effort, IMO.</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao</font>
<br><font size=2 face="sans-serif">&nbsp;</font>
<br><font size=2 face="sans-serif">===================================<br>
 Zip &nbsp; &nbsp;: 210012<br>
 Tel &nbsp; &nbsp;: 87211<br>
 Tel2 &nbsp; :(+86)-025-52877211<br>
 e_mail : gao.yang2@zte.com.cn<br>
===================================</font><br><pre>
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Subject: [sipcore] I can send but can not recv. Maybe this information is useful for diagnose. //Hi Oscar, I can not get mails from sipcore maillist(other lists are OK) from 2 March. Could you help me to fix it? Thanks.
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Subject: [sipcore] I-D Action:draft-ietf-sipcore-invfix-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.


	Title           : Correct transaction handling for 2xx responses to Session Initiation Protocol (SIP) INVITE requests
	Author(s)       : R. Sparks, T. Zourzouvillys
	Filename        : draft-ietf-sipcore-invfix-01.txt
	Pages           : 20
	Date            : 2010-03-07

This document normatively updates RFC 3261, the Session Initiation
Protocol (SIP), to address an error in the specified handling of
success (2xx class) responses to INVITE requests.  Elements following
RFC 3261 exactly will misidentify retransmissions of the request as a
new, unassociated, request.  The correction involves modifying the
INVITE transaction state machines.  The correction also changes the
way responses that cannot be matched to an existing transaction are
handled to address a security risk.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-invfix-01.txt

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ftp://ftp.ietf.org/internet-drafts/

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From pkyzivat@cisco.com  Mon Mar  8 06:37:50 2010
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Date: Mon, 08 Mar 2010 09:37:52 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: 'SIPCORE' <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Gilad Shaham wrote:
> Paul Kyzivat wrote:
>> Perhaps the best we can do is discourage the use of early media for
>> anything important.
>>
> 
> I'm not sure this scenario is that common, but if it is something of
> interest, perhaps it's possible to add a header/parameter conveying the
> nature of the media stream (ringback, service message, voice, etc.). This is
> similar to the content-disposition handling parameter, but more informative
> than "optional" and "required". This way, a B2BUA would be able to choose
> the stream to forward and even without a B2BUA, an endpoint receiving
> several streams would know which stream to play back to the user.

I guess this would be something like the Reason header, or maybe it 
would really be the Reason header, though it would take some changes to 
allow it for that. Another possibility for this is Alert-Info.

The response code, possibly augmented with the Reason header, would work 
for things like out of service that are going to cause the call to fail. 
However the callee has to have confidence that the caller will be 
properly informed of the case.

The Alert-Info containing a URN can potentially work for common things 
that aren't going to cause the call to fail. But that isn't sufficient 
if there is variable information to convey to the caller, unless it is 
done with with a URI that can be dereferenced and rendered. The latter 
would be an alternative if we could convince UAs to render such things. 
My impression is that there is reluctance to do so, though it really 
isn't much different from rendering an early media string.

But none of this will help with many pstn interop cases where the reason 
isn't known except through the content of the media.

	Thanks,
	Paul

From dean.willis@softarmor.com  Mon Mar  8 07:44:53 2010
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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On Mar 8, 2010, at 8:37 AM, Paul Kyzivat wrote:

>
>
> Gilad Shaham wrote:
>> Paul Kyzivat wrote:
>>> Perhaps the best we can do is discourage the use of early media for
>>> anything important.
>>>
>> I'm not sure this scenario is that common, but if it is something of
>> interest, perhaps it's possible to add a header/parameter conveying  
>> the
>> nature of the media stream (ringback, service message, voice,  
>> etc.). This is
>> similar to the content-disposition handling parameter, but more  
>> informative
>> than "optional" and "required". This way, a B2BUA would be able to  
>> choose
>> the stream to forward and even without a B2BUA, an endpoint receiving
>> several streams would know which stream to play back to the user.
>
> I guess this would be something like the Reason header, or maybe it  
> would really be the Reason header, though it would take some changes  
> to allow it for that. Another possibility for this is Alert-Info.

Nut at the moment, we have UAs sending back media that have sent no  
SIP messages yet. So there's no place to add a  Reason header.

The only thing that's under the UAC's control is the offer, and AFAIK  
the only thing the UAC can do to differentiate early media streams is  
somehow assure that a different offer (with a different listening  
port) is used for each offer. The UAC cannot do this if the offer is  
forked by a proxy.

> ...

> But none of this will help with many pstn interop cases where the  
> reason isn't known except through the content of the media.


And even worse: assuming that the UAC has the ability to play-out all  
the potentially interesting media streams, there's no way for the UAC  
or the user thereof to CANCEL any leg and thereby "stop" early media  
with today's model. That's because one offer is used for all the early- 
media.

So for a UA that's really smart and can do some level of analysis  
within the media path (like say "Press 1 if you want to accept this  
call" and then wait for a tone, or perhaps do speech-recognition),  
there's still no ability for it to clean up the early media by  
signaling until the early media becomes a regular media session.

People claim to be worried about the media bandwidth impact of doing a  
302 back to the UA and having it do the forking, but the simple fact  
is that at least in this scenario the UA can control the result,  
whereas with proxy parallel forking, the UA potentially gets hammered  
with multiple parallel early media streams and has no way to do  
anything about it.

--
Dean




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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 8, 2010, =
at 8:37 AM, Paul Kyzivat wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div><br><br>Gilad Shaham wrote:<br><blockquote =
type=3D"cite">Paul Kyzivat wrote:<br></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">Perhaps the best we can do is =
discourage the use of early media =
for<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">anything =
important.<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote =
type=3D"cite"><br></blockquote></blockquote><blockquote type=3D"cite">I'm =
not sure this scenario is that common, but if it is something =
of<br></blockquote><blockquote type=3D"cite">interest, perhaps it's =
possible to add a header/parameter conveying =
the<br></blockquote><blockquote type=3D"cite">nature of the media stream =
(ringback, service message, voice, etc.). This =
is<br></blockquote><blockquote type=3D"cite">similar to the =
content-disposition handling parameter, but more =
informative<br></blockquote><blockquote type=3D"cite">than "optional" =
and "required". This way, a B2BUA would be able to =
choose<br></blockquote><blockquote type=3D"cite">the stream to forward =
and even without a B2BUA, an endpoint =
receiving<br></blockquote><blockquote type=3D"cite">several streams =
would know which stream to play back to the user.<br></blockquote><br>I =
guess this would be something like the Reason header, or maybe it would =
really be the Reason header, though it would take some changes to allow =
it for that. Another possibility for this is =
Alert-Info.<br></div></blockquote><div><br></div><div>Nut at the moment, =
we have UAs sending back media that have sent no SIP messages yet. So =
there's no place to add a &nbsp;Reason =
header.</div><div><br></div><div>The only thing that's under the UAC's =
control is the offer, and AFAIK the only thing the UAC can do to =
differentiate early media streams is somehow assure that a different =
offer (with a different listening port) is used for each offer. The UAC =
cannot do this if the offer is forked by a =
proxy.</div><div><br></div><blockquote type=3D"cite"><div><font =
class=3D"Apple-style-span" =
color=3D"#000000">...<br></font></div></blockquote><br><blockquote =
type=3D"cite"><div>But none of this will help with many pstn interop =
cases where the reason isn't known except through the content of the =
media.<br></div></blockquote></div><div><br></div><div>And even worse: =
assuming that the UAC has the ability to play-out all the potentially =
interesting media streams, there's no way for the UAC or the user =
thereof to CANCEL any leg and thereby "stop" early media with today's =
model. That's because one offer is used for all the =
early-media.</div><div><br></div><div>So for a UA that's really smart =
and can do some level of analysis within the media path (like say "Press =
1 if you want to accept this call" and then wait for a tone, or perhaps =
do speech-recognition), there's still no ability for it to clean up the =
early media by signaling until the early media becomes a regular media =
session.</div><div><br></div><div>People claim to be worried about the =
media bandwidth impact of doing a 302 back to the UA and having it do =
the forking, but the simple fact is that at least in this scenario the =
UA can control the result, whereas with proxy parallel forking, the UA =
potentially gets hammered with multiple parallel early media streams and =
has no way to do anything about =
it.</div><div><br></div><div>--</div><div>Dean</div><div><br></div><br><di=
v><br></div></body></html>=

--Apple-Mail-14--154407389--

From pkyzivat@cisco.com  Mon Mar  8 09:20:46 2010
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: 'SIPCORE' <sipcore@ietf.org>, Gilad Shaham <gilad@voxisoft.com>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Dean Willis wrote:
> 
> On Mar 8, 2010, at 8:37 AM, Paul Kyzivat wrote:
> 
>>
>>
>> Gilad Shaham wrote:
>>> Paul Kyzivat wrote:
>>>> Perhaps the best we can do is discourage the use of early media for
>>>> anything important.
>>>>
>>> I'm not sure this scenario is that common, but if it is something of
>>> interest, perhaps it's possible to add a header/parameter conveying the
>>> nature of the media stream (ringback, service message, voice, etc.). 
>>> This is
>>> similar to the content-disposition handling parameter, but more 
>>> informative
>>> than "optional" and "required". This way, a B2BUA would be able to choose
>>> the stream to forward and even without a B2BUA, an endpoint receiving
>>> several streams would know which stream to play back to the user.
>>
>> I guess this would be something like the Reason header, or maybe it 
>> would really be the Reason header, though it would take some changes 
>> to allow it for that. Another possibility for this is Alert-Info.
> 
> Nut at the moment, we have UAs sending back media that have sent no SIP 
> messages yet. So there's no place to add a  Reason header.

I wasn't thinking of discriminating the early media streams, but rather 
finding ways to eliminate them while retaining their intent.

> The only thing that's under the UAC's control is the offer, and AFAIK 
> the only thing the UAC can do to differentiate early media streams is 
> somehow assure that a different offer (with a different listening port) 
> is used for each offer. The UAC cannot do this if the offer is forked by 
> a proxy.
> 
>> ...
> 
>> But none of this will help with many pstn interop cases where the 
>> reason isn't known except through the content of the media.
> 
> And even worse: assuming that the UAC has the ability to play-out all 
> the potentially interesting media streams, there's no way for the UAC or 
> the user thereof to CANCEL any leg and thereby "stop" early media with 
> today's model. That's because one offer is used for all the early-media.

Several years ago Rohan proposed a model where the UAC did another O/A 
with each early dialog to move its media to a distinct port. That way it 
could then associate the signaling with a particular media stream. Seems 
cumbersome, but maybe it will take something like that to solve the problem.

> So for a UA that's really smart and can do some level of analysis within 
> the media path (like say "Press 1 if you want to accept this call" and 
> then wait for a tone, or perhaps do speech-recognition), there's still 
> no ability for it to clean up the early media by signaling until the 
> early media becomes a regular media session.
> 
> People claim to be worried about the media bandwidth impact of doing a 
> 302 back to the UA and having it do the forking, but the simple fact is 
> that at least in this scenario the UA can control the result, whereas 
> with proxy parallel forking, the UA potentially gets hammered with 
> multiple parallel early media streams and has no way to do anything 
> about it.

Yes. The cost and difficulty of the UAC handling the forking seems no 
worse that the extra O/As I mention above.

But even if you have each early dialog associated with a known distinct 
media session, the problem of deciding which of them to render remains 
and has no obvious solution.

Possibly it only works well with a UA that has a sophisticated UI that 
can show you the multiple sessions and let you flip between them.

	Thanks,
	Paul

From gilad@voxisoft.com  Mon Mar  8 09:47:36 2010
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From: Gilad Shaham <gilad@voxisoft.com>
Date: Mon, 8 Mar 2010 19:47:08 +0200
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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--001485f872ba9bf3d904814da813
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On Mon, Mar 8, 2010 at 7:20 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

>
>
> Dean Willis wrote:
>
>>
>> On Mar 8, 2010, at 8:37 AM, Paul Kyzivat wrote:
>>
>>
>>>
>>> Gilad Shaham wrote:
>>>
>>>> Paul Kyzivat wrote:
>>>>
>>>>> Perhaps the best we can do is discourage the use of early media for
>>>>> anything important.
>>>>>
>>>>>  I'm not sure this scenario is that common, but if it is something of
>>>> interest, perhaps it's possible to add a header/parameter conveying the
>>>> nature of the media stream (ringback, service message, voice, etc.).
>>>> This is
>>>> similar to the content-disposition handling parameter, but more
>>>> informative
>>>> than "optional" and "required". This way, a B2BUA would be able to
>>>> choose
>>>> the stream to forward and even without a B2BUA, an endpoint receiving
>>>> several streams would know which stream to play back to the user.
>>>>
>>>
>>> I guess this would be something like the Reason header, or maybe it would
>>> really be the Reason header, though it would take some changes to allow it
>>> for that. Another possibility for this is Alert-Info.
>>>
>>
>> Nut at the moment, we have UAs sending back media that have sent no SIP
>> messages yet. So there's no place to add a  Reason header.
>>
>
> I wasn't thinking of discriminating the early media streams, but rather
> finding ways to eliminate them while retaining their intent.
>

>> The only thing that's under the UAC's control is the offer, and AFAIK the
>> only thing the UAC can do to differentiate early media streams is somehow
>> assure that a different offer (with a different listening port) is used for
>> each offer. The UAC cannot do this if the offer is forked by a proxy.
>>
>>  ...
>>>
>>
>>  But none of this will help with many pstn interop cases where the reason
>>> isn't known except through the content of the media.
>>>
>>
>> And even worse: assuming that the UAC has the ability to play-out all the
>> potentially interesting media streams, there's no way for the UAC or the
>> user thereof to CANCEL any leg and thereby "stop" early media with today's
>> model. That's because one offer is used for all the early-media.
>>
>
> Several years ago Rohan proposed a model where the UAC did another O/A with
> each early dialog to move its media to a distinct port. That way it could
> then associate the signaling with a particular media stream. Seems
> cumbersome, but maybe it will take something like that to solve the problem.
>
>
Perhaps a less cumbersome way to utilize the offer-answer is to send an
offer without actually allocating the port and only choose a port on each
dialog that indicates early media. this way the initiating UA has full
control and is not overwhelmed with several incoming streams. This of course
has a disadvantage if the call is answered very quickly (perhaps
automatically) which may require an additional UPDATE to hear something.


>
>  So for a UA that's really smart and can do some level of analysis within
>> the media path (like say "Press 1 if you want to accept this call" and then
>> wait for a tone, or perhaps do speech-recognition), there's still no ability
>> for it to clean up the early media by signaling until the early media
>> becomes a regular media session.
>>
>> People claim to be worried about the media bandwidth impact of doing a 302
>> back to the UA and having it do the forking, but the simple fact is that at
>> least in this scenario the UA can control the result, whereas with proxy
>> parallel forking, the UA potentially gets hammered with multiple parallel
>> early media streams and has no way to do anything about it.
>>
>
> Yes. The cost and difficulty of the UAC handling the forking seems no worse
> that the extra O/As I mention above.
>
> But even if you have each early dialog associated with a known distinct
> media session, the problem of deciding which of them to render remains and
> has no obvious solution.
>
> Possibly it only works well with a UA that has a sophisticated UI that can
> show you the multiple sessions and let you flip between them.
>
> If we are relying on speech recognition or even tone detection, it sounds
very unlikely to become widely accepted. It's also very confusing for a
person to choose between those sessions and it doesn't make much sense for
the user to flip between them, unless the device could actually tell the
user there is something of great importance on that stream.

Once again, it means the other endpoints should indicate something about
their early media and as you said, PSTN is not covered by this option.
Still, maybe it's possible to leave PSTN as-is and concentrate on end-to-end
SIP.


Gilad

--001485f872ba9bf3d904814da813
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_quote">On Mon, Mar 8, 2010 at 7:20=
 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@cisco.co=
m">pkyzivat@cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0=
pt 0.8ex; padding-left: 1ex;">

<div class=3D"im"><br>
<br>
Dean Willis wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
On Mar 8, 2010, at 8:37 AM, Paul Kyzivat wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
<br>
Gilad Shaham wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Paul Kyzivat wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Perhaps the best we can do is discourage the use of early media for<br>
anything important.<br>
<br>
</blockquote>
I&#39;m not sure this scenario is that common, but if it is something of<br=
>
interest, perhaps it&#39;s possible to add a header/parameter conveying the=
<br>
nature of the media stream (ringback, service message, voice, etc.). This i=
s<br>
similar to the content-disposition handling parameter, but more informative=
<br>
than &quot;optional&quot; and &quot;required&quot;. This way, a B2BUA would=
 be able to choose<br>
the stream to forward and even without a B2BUA, an endpoint receiving<br>
several streams would know which stream to play back to the user.<br>
</blockquote>
<br>
I guess this would be something like the Reason header, or maybe it would r=
eally be the Reason header, though it would take some changes to allow it f=
or that. Another possibility for this is Alert-Info.<br>
</blockquote>
<br>
Nut at the moment, we have UAs sending back media that have sent no SIP mes=
sages yet. So there&#39;s no place to add a =A0Reason header.<br>
</blockquote>
<br></div>
I wasn&#39;t thinking of discriminating the early media streams, but rather=
 finding ways to eliminate them while retaining their intent.<br><div class=
=3D"im"></div></blockquote>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class=3D"im"=
><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204,=
 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">


<br>

The only thing that&#39;s under the UAC&#39;s control is the offer, and AFA=
IK the only thing the UAC can do to differentiate early media streams is so=
mehow assure that a different offer (with a different listening port) is us=
ed for each offer. The UAC cannot do this if the offer is forked by a proxy=
.<br>


<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
...<br>
</blockquote>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
But none of this will help with many pstn interop cases where the reason is=
n&#39;t known except through the content of the media.<br>
</blockquote>
<br>
And even worse: assuming that the UAC has the ability to play-out all the p=
otentially interesting media streams, there&#39;s no way for the UAC or the=
 user thereof to CANCEL any leg and thereby &quot;stop&quot; early media wi=
th today&#39;s model. That&#39;s because one offer is used for all the earl=
y-media.<br>


</blockquote>
<br></div>
Several years ago Rohan proposed a model where the UAC did another O/A with=
 each early dialog to move its media to a distinct port. That way it could =
then associate the signaling with a particular media stream. Seems cumberso=
me, but maybe it will take something like that to solve the problem.<div cl=
ass=3D"im">

<br></div></blockquote><div><div><br>
Perhaps a less cumbersome way to utilize the offer-answer is to send an
offer without actually allocating the port and only choose a port on
each dialog that indicates early media. this way the initiating UA has
full control and is not overwhelmed with several incoming streams. This
of course has a disadvantage if the call is answered very quickly
(perhaps automatically) which may require an additional UPDATE to hear
something.<br>
=A0=A0
<br></div></div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px=
 solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><=
div class=3D"im">
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
So for a UA that&#39;s really smart and can do some level of analysis withi=
n the media path (like say &quot;Press 1 if you want to accept this call&qu=
ot; and then wait for a tone, or perhaps do speech-recognition), there&#39;=
s still no ability for it to clean up the early media by signaling until th=
e early media becomes a regular media session.<br>


<br>
People claim to be worried about the media bandwidth impact of doing a 302 =
back to the UA and having it do the forking, but the simple fact is that at=
 least in this scenario the UA can control the result, whereas with proxy p=
arallel forking, the UA potentially gets hammered with multiple parallel ea=
rly media streams and has no way to do anything about it.<br>


</blockquote>
<br></div>
Yes. The cost and difficulty of the UAC handling the forking seems no worse=
 that the extra O/As I mention above.<br>
<br>
But even if you have each early dialog associated with a known distinct med=
ia session, the problem of deciding which of them to render remains and has=
 no obvious solution.<br>
<br>
Possibly it only works well with a UA that has a sophisticated UI that can =
show you the multiple sessions and let you flip between them.<br>
<br></blockquote><div>If we are relying on speech recognition or even tone =
detection, it
sounds very unlikely to become widely accepted. It&#39;s also very
confusing for a person to choose between those sessions and it doesn&#39;t =
make much sense for the user to flip between them, unless the
device could actually tell the user there is something of great
importance on that stream.<br><br>Once again, it means the other
endpoints should indicate something about their early media and as you said=
, PSTN is
not covered by this option. Still, maybe it&#39;s possible to leave PSTN as=
-is and concentrate on end-to-end SIP.<br><br>=A0</div>Gilad<br></div></div=
>

--001485f872ba9bf3d904814da813--

From dean.willis@softarmor.com  Mon Mar  8 13:31:14 2010
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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On Mar 8, 2010, at 11:20 AM, Paul Kyzivat wrote:

>
> I wasn't thinking of discriminating the early media streams, but  
> rather finding ways to eliminate them while retaining their intent.

That would be cool. Requires a business model change at PSTN gateways,  
which IMHO should go to 200 OK (and a fully-connected dialog) before/ 
while starting the IAM or equivalent on the PSTN side. But too many  
yerkaloids want to bill on 200OK.

But for non-gateway nodes, we could certainly do something along the  
lines of a 1xx header or specific cause-codes as different 1XX  
response types.

>
> Several years ago Rohan proposed a model where the UAC did another O/ 
> A with each early dialog to move its media to a distinct port. That  
> way it could then associate the signaling with a particular media  
> stream. Seems cumbersome, but maybe it will take something like that  
> to solve the problem.

Only works once the UAS sends back a 1xx of some sort. Some nodes just  
send media. For gateways, we therefore have to differentiate in the  
OFFER as seen by the UAS, which means sending a different offer to  
each UAS, which means NO  FORKING!

Or maybe we could do yet-another BCP on gateways to require early- 
dialog before media cut-through. This might be worth exploring. Of  
course, this requires updating all the gateways before it can be  
relied on, or at least probing for compliant gateways with a Require.

--
Dean



From dean.willis@softarmor.com  Mon Mar  8 13:37:28 2010
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On Mar 8, 2010, at 11:47 AM, Gilad Shaham wrote:
>
> Perhaps a less cumbersome way to utilize the offer-answer is to send  
> an offer without actually allocating the port and only choose a port  
> on each dialog that indicates early media. this way the initiating  
> UA has full control and is not overwhelmed with several incoming  
> streams. This of course has a disadvantage if the call is answered  
> very quickly (perhaps automatically) which may require an additional  
> UPDATE to hear something.

So you're saying send an offerless INVITE and use early-dialogs to  
negotiate O/A with the UAS(gateway)  acting as the offering node? That  
might work.

> ...

> If we are relying on speech recognition or even tone detection, it  
> sounds very unlikely to become widely accepted. It's also very  
> confusing for a person to choose between those sessions and it  
> doesn't make much sense for the user to flip between them, unless  
> the device could actually tell the user there is something of great  
> importance on that stream.

I'm rather thinking more that a UA can engage in an "Are you there"  
challenge-response with the far end to see if there's something worth  
talking to.

Or the UA could record the media on each leg and play it back to the  
user later for diagnostic purposes.

> Once again, it means the other endpoints should indicate something  
> about their early media and as you said, PSTN is not covered by this  
> option. Still, maybe it's possible to leave PSTN as-is and  
> concentrate on end-to-end SIP.

Maybe. If gateways at least did "early session" with an answer in  
1xx,  along with some sort of SDP-to-RTP binding token (along the  
lines of the crypto one's we've discussed) we could make it work.

--
dean
--Apple-Mail-16--133334879
Content-Type: text/html;
	charset=US-ASCII
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 8, 2010, =
at 11:47 AM, Gilad Shaham wrote:</div><blockquote type=3D"cite"><div =
dir=3D"ltr"><div class=3D"gmail_quote"><div><div><font =
class=3D"Apple-style-span" color=3D"#000000"><br></font> Perhaps a less =
cumbersome way to utilize the offer-answer is to send an offer without =
actually allocating the port and only choose a port on each dialog that =
indicates early media. this way the initiating UA has full control and =
is not overwhelmed with several incoming streams. This of course has a =
disadvantage if the call is answered very quickly (perhaps =
automatically) which may require an additional UPDATE to hear =
something.<br> =
</div></div></div></div></blockquote><div><br></div><div>So you're =
saying send an offerless INVITE and use early-dialogs to negotiate O/A =
with the UAS(gateway) &nbsp;acting as the offering node? That might =
work.</div><div><br></div><blockquote type=3D"cite"><div dir=3D"ltr"><div =
class=3D"gmail_quote"><blockquote class=3D"gmail_quote" =
style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt =
0.8ex; padding-left: 1ex;"><font class=3D"Apple-style-span" =
color=3D"#000000">...</font></blockquote></div></div></blockquote><br><blo=
ckquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_quote"><div>If =
we are relying on speech recognition or even tone detection, it sounds =
very unlikely to become widely accepted. It's also very confusing for a =
person to choose between those sessions and it doesn't make much sense =
for the user to flip between them, unless the device could actually tell =
the user there is something of great importance on that =
stream.<br></div></div></div></blockquote><div><br></div><div>I'm rather =
thinking more that a UA can engage in an "Are you there" =
challenge-response with the far end to see if there's something worth =
talking to.</div><div><br></div><div>Or the UA could record the media on =
each leg and play it back to the user later for diagnostic =
purposes.</div><div><br></div><blockquote type=3D"cite"><div =
dir=3D"ltr"><div class=3D"gmail_quote"><div>Once again, it means the =
other endpoints should indicate something about their early media and as =
you said, PSTN is not covered by this option. Still, maybe it's possible =
to leave PSTN as-is and concentrate on end-to-end =
SIP.<br></div></div></div></blockquote></div><br><div>Maybe. If gateways =
at least did "early session" with an answer in 1xx, &nbsp;along with =
some sort of SDP-to-RTP binding token (along the lines of the crypto =
one's we've discussed) we could make it =
work.</div><div><br></div><div>--</div><div>dean</div></body></html>=

--Apple-Mail-16--133334879--

From adam@nostrum.com  Mon Mar  8 13:49:00 2010
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[as chair]

For some reason, ZTE.com.cn is blocking mail sent from the SIPCORE 
mailing list to several email accounts. They have also blocked my 
attempts to contact the affected participants to try to resolve the 
issue. If anyone can contact any of the following individuals, please 
ask them to get in touch with me using an alternate email service (that 
is, an email service other than ZTE.com.cn):

    *

      gao.yang2@zte.com.cn

    *

      wang.libo@zte.com.cn

    *

      yan.chengan@zte.com.cn

    *

      yu.xutao@zte.com.cn

    *

      zhou.peng2@zte.com.cn


Thanks, and sorry for the noise.

/a

--------------090206000003090805050102
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>

<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body bgcolor="#ffffff" text="#000000">
[as chair]<br>
<br>
For some reason, ZTE.com.cn is blocking mail sent from the SIPCORE
mailing list to several email accounts. They have also blocked my
attempts to contact the affected participants to try to resolve the
issue. If anyone can contact any of the following individuals, please
ask them to get in touch with me using an alternate email service (that
is, an email service other than ZTE.com.cn):<br>
<br>
<ul>
  <li>
    <pre id="line1"><a class="moz-txt-link-abbreviated" href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a></pre>
  </li>
  <li>
    <pre id="line1"><a class="moz-txt-link-abbreviated" href="mailto:wang.libo@zte.com.cn">wang.libo@zte.com.cn</a></pre>
  </li>
  <li>
    <pre id="line1"><a class="moz-txt-link-abbreviated" href="mailto:yan.chengan@zte.com.cn">yan.chengan@zte.com.cn</a></pre>
  </li>
  <li>
    <pre id="line1"><a class="moz-txt-link-abbreviated" href="mailto:yu.xutao@zte.com.cn">yu.xutao@zte.com.cn</a></pre>
  </li>
  <li>
    <pre id="line1"><a class="moz-txt-link-abbreviated" href="mailto:zhou.peng2@zte.com.cn">zhou.peng2@zte.com.cn</a></pre>
  </li>
</ul>
<br>
Thanks, and sorry for the noise.<br>
<br>
/a<br>
</body>
</html>

--------------090206000003090805050102--

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The SIP Forum is going to hold a open meeting for the RAI community during
IETF 77. The purpose of this meeting is remind the SIP community of what the
SIP Forum is doing and discuss relevant technical issues that relate to IETF
WG's. The agenda for this luncheon will be to review current activities in
the SIP Forum and seek additional technical input from the RAI community on
mutual concerns.

 

As many of you know the SIP Forum is the principal sponsor of the SIPit
interoperability events and currently has 4 Task groups operating. 

 

BTW registration for SIPit 26 will be open soon.

 

Our principal technical initiatives are:

 

1.      SIP Connect 1.1 which is developing an enhanced  profile of IETF
Standards for PBX to SSP interoperability. As part of that work the IETF
MARTINI WG was formed to redefine the so called Implicit registration
procedure in SIP.

2.      FAX over SIP is investigating the interoperability of SIP in T.38
networks with a goal of possible recommendations to the relevant SDO's on
how fax can work better.

3.      Client User Agent Configuration which has a pending Recommendation
on how CUA devices could auto streamline the locating, retrieving and
maintaining of configuration-related information for SIP User Agents

4.      SIP in the SmartGrid. This is a Special Interest Group that is
evaluating the appropriateness of using SIP as a protocol for a number of
Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) deployments.

 

Information on the SIP Forum is available at 

 

http//www.sipforum.org

 

If you are interested in attending this event we have set up a Doodle Poll
to indicate your interest. Seating is somewhat limited ( 150 Maximum) lunch
will be provided.

 

Location .. Monday 11:30 13:00 the Laguna room and its located on the 4th
floor of the Anaheim Hilton

The link to the poll is: 

http://www.doodle.com/hde57nbpcdykzqfz 

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype: rshockey101
LinkedIn : http://www.linkedin.com/in/rshockey101



 


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<div class=3DSection1>

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

<p class=3DMsoNormal>The SIP Forum is going to hold a open meeting for =
the RAI
community during IETF 77. The purpose of this meeting is remind the SIP
community of what the SIP Forum is doing and discuss relevant technical =
issues
that relate to IETF WG&#8217;s. The agenda for this luncheon will be to =
review
current activities in the SIP Forum and seek additional technical input =
from
the RAI community on mutual concerns.<o:p></o:p></p>

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

<p class=3DMsoNormal>As many of you know the SIP Forum is the principal =
sponsor
of the SIPit interoperability events and currently has 4 Task groups =
operating.
<o:p></o:p></p>

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

<p class=3DMsoNormal>BTW registration for SIPit 26 will be open =
soon.<o:p></o:p></p>

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

<p class=3DMsoNormal>Our principal technical initiatives =
are:<o:p></o:p></p>

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

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP Connect 1.1 which is developing an enhanced
&nbsp;profile of IETF Standards for PBX to SSP interoperability. As part =
of
that work the IETF MARTINI WG was formed to redefine the so called =
Implicit
registration procedure in SIP.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>FAX over SIP is investigating the =
interoperability of
SIP in T.38 networks with a goal of possible recommendations to the =
relevant
SDO&#8217;s on how fax can work better.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Client User Agent Configuration which has a =
pending
Recommendation on how CUA devices could auto streamline the locating,
retrieving and maintaining of configuration-related information for SIP =
User
Agents<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP in the SmartGrid. This is a Special Interest =
Group
that is evaluating the appropriateness of using SIP as a protocol for a =
number
of Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) =
deployments.<o:p></o:p></p>

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

<p class=3DMsoNormal>Information on the SIP Forum is available at =
<o:p></o:p></p>

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

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Times =
New Roman","serif"'>http//www.sipforum.org</span><o:p></o:p></p>

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

<p class=3DMsoNormal>If you are interested in attending this event we =
have set up
a Doodle Poll to indicate your interest. Seating is somewhat limited ( =
150
Maximum) lunch will be provided.<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>Location .. </span><span
style=3D'font-size:10.0pt'>Monday 11:30 13:00 </span><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>the Laguna room and its =
located on
the 4th floor of the Anaheim Hilton</span><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>The link to the poll =
is:</span><span
style=3D'font-size:10.0pt'> <br>
<br>
<a href=3D"http://www.doodle.com/hde57nbpcdykzqfz" =
target=3D"_blank">http://www.doodle.com/hde57nbpcdykzqfz</a>
</span><o:p></o:p></p>

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

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>
Shockey Consulting<br>
Chairman of the Board of Directors SIP Forum<br>
PSTN Mobile: +1 703.593.2683<br>
&lt;<a =
href=3D"mailto:richard(at)shockey.us">mailto:richard(at)shockey.us</a>&gt=
;<br>
skype: rshockey101<br>
LinkedIn : <a =
href=3D"http://www.linkedin.com/in/rshockey101">http://www.linkedin.com/i=
n/rshockey101</a><br>
<br>
</span><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

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

</div>

</body>

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From rjsparks@nostrum.com  Fri Mar 12 07:57:04 2010
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Subject: [sipcore] RAI WG chair changes
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WIth the upcoming IESG and IAB changes, we need to make=20
a bunch of chair changes. After the end of the IETF 77 the chairs=20
for the following WGs will be:

SIPREC=20
Brian Rosen
John Elwell

SIPCORE
Adam Roach
Paul Kyzivat

CODEC
Michael Knappe
Jonathan Rosenberg
Cullen Jennings

DISPATCH
Mary Barnes
Cullen Jennings

ECRIT=20
Richard Barnes
Marc Linsner

XMPP
Joe Hildebrand
Ben Campbell

To help with continuity of the IETF77 meeting, we are going to add the =
new chairs now,=20
and not remove any existing chairs until after the meeting.=20

We would like to express many thinks for all the work all these chairs =
have done and continue to do.=20

Thanks,=20
Cullen, Gonzalo, & Robert








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Subject: [sipcore] Query on draft-ietf-sipcore-info-events-07
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Hi all

Are there any registered INFO packages and can the regular SIP event
packages based on RFC 3265 be used with INFO mechanism?

Thanks

Regards
Ranjit


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<P><FONT SIZE=3D2 FACE=3D"Arial">Hi all</FONT>
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<P><FONT SIZE=3D2 FACE=3D"Arial">Are there any registered INFO packages =
and can the regular SIP event packages based on RFC 3265 be used with =
INFO mechanism?</FONT></P>

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

<BR><FONT SIZE=3D2 FACE=3D"Arial">Ranjit</FONT>
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From adam@nostrum.com  Sat Mar 13 14:53:06 2010
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Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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Please don't cross-post to dispatch and sipcore.

On 3/12/10 23:26, Mar 12, Avasarala Ranjit-A20990 wrote:
>
> Are there any registered INFO packages
>

Not yet.

> and can the regular SIP event packages based on RFC 3265 be used with 
> INFO mechanism?
>

Absolutely not. No. Never. Under no circumstances. Nein, ???, ????, Non, 
Nej. They are _completely_ different things.

Really, if this is somehow unclear in the INFO draft, we need to pull it 
from publication request and work on it some more. It would be the worst 
possible outcome if implementors were led to believe that 3265 packages 
could be used with INFO. Where did you get the impression that this 
might be okay?

/a

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<span id="result_box" class="short_text"><span
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cross-post to dispatch and sipcore.</span></span><br>
<br>
On 3/12/10 23:26, Mar 12, Avasarala Ranjit-A20990 wrote:
<blockquote
 cite="mid:750BBC72E178114F9DC4872EBFF29A5B0A3BE70F@ZMY16EXM66.ds.mot.com"
 type="cite">
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  <p><font face="Arial" size="2">Are there any registered INFO packages</font></p>
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<br>
Not yet.<br>
<br>
<blockquote
 cite="mid:750BBC72E178114F9DC4872EBFF29A5B0A3BE70F@ZMY16EXM66.ds.mot.com"
 type="cite">
  <p><font face="Arial" size="2"> and can the regular SIP event
packages based on RFC 3265 be used with INFO mechanism?</font></p>
</blockquote>
<br>
Absolutely not. No. Never. Under no circumstances. Nein,
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
<span id="result_box" class="short_text"><span
 style="background-color: rgb(255, 255, 255);" title="No">&#1053;&#1077;&#1090;</span></span>,
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
<span id="result_box" class="short_text"><span
 style="background-color: rgb(255, 255, 255);" title="No">&#2344;&#2361;&#2368;&#2306;</span></span>,
Non,
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
<span id="result_box" class="short_text"><span
 style="background-color: rgb(255, 255, 255);" title="No">Nej. They are
_completely_ different things.<br>
<br>
Really, if this is somehow unclear in the INFO draft, we need to pull
it from publication request and work on it some more. It would be the
worst possible outcome if implementors were led to believe that 3265
packages could be used with INFO. Where did you get the impression that
this might be okay?<br>
</span></span><br>
/a<br>
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
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From eburger@standardstrack.com  Sun Mar 14 10:28:10 2010
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Cc: SIPCORE SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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The point of Section 7.4 is there is often a far superior method of =
doing user-user signaling besides INFO. For example, if you want to send =
DTMF digits, use KPML (he says modestly).

There is absolutely NO correspondence between an Info Package and an =
Event Package. Moreover, if someone tried to register such, my guess is =
they would not get the requisite Specification Required level needed to =
register a package.

On Mar 14, 2010, at 4:03 AM, Avasarala Ranjit-A20990 wrote:

> Hi Adam
> =20
> =46rom the latest version of the info =
draft-ietf-sipcore-info-events-07, the section 4 "Info Packages" does =
not talk about the kind of packages and their format.
> =20
> Though the section 7.4.1.2  talks that use of SUBSCRIBE / NOTIFY based =
on 3265 as an alternate mechanism, it is not clear what should be the =
format of a typical INFO package.
> =20
> Also further Section 7.4.1.3 mentions the usage of SIP MESSAGE method. =
So does this mean I can use MESSAGE method as an alternative to INFO?
> =20
> Thanks
> Regards=20
> Ranjit
> =20
>=20
> From: Adam Roach [mailto:adam@nostrum.com]=20
> Sent: Sunday, March 14, 2010 4:23 AM
> To: Avasarala Ranjit-A20990
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
>=20
> Please don't cross-post to dispatch and sipcore.
>=20
> On 3/12/10 23:26, Mar 12, Avasarala Ranjit-A20990 wrote:
>> Are there any registered INFO packages
>>=20
>=20
> Not yet.
>=20
>> and can the regular SIP event packages based on RFC 3265 be used with =
INFO mechanism?
>>=20
>=20
> Absolutely not. No. Never. Under no circumstances. Nein, =D0=9D=D0=B5=D1=
=82, =E0=A4=A8=E0=A4=B9=E0=A5=80=E0=A4=82, Non, Nej. They are =
_completely_ different things.
>=20
> Really, if this is somehow unclear in the INFO draft, we need to pull =
it from publication request and work on it some more. It would be the =
worst possible outcome if implementors were led to believe that 3265 =
packages could be used with INFO. Where did you get the impression that =
this might be okay?
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Hi Gonzalo/Chris/Gao,

Not sure if this is a nit - or if I am lost...

In Figure 3 the 200 OK for the second INVITE (which made an offer) does not
carry an answer. Is this deliberate? (I have the same issue in Figure 4
making me think perhaps I'm lost.)

If yes, then I think an explanatory comment would be helpful - since I would
expect a 200 with answer .

Regards,

Peter Musgrave

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Hi Gonzalo/Chris/Gao,=A0
<div><br></div><div>Not sure if this is a nit - or if I am lost...</div><di=
v><br></div><div>In Figure 3 the 200 OK for the second INVITE (which made a=
n offer) does not carry an answer. Is this deliberate? (I have the same iss=
ue in Figure 4 making me think perhaps I&#39;m lost.)</div>
<div><br></div><div>If yes, then I think an explanatory comment would be he=
lpful - since I would expect a 200 with answer .</div><div><br></div><div>R=
egards,=A0</div><div><br></div><div>Peter Musgrave</div>

--0016e68de3e46e35b90481c4acdd--

From brett@broadsoft.com  Sun Mar 14 11:16:47 2010
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From: Brett Tate <brett@broadsoft.com>
To: Peter Musgrave <peter.musgrave@magorcorp.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 14 Mar 2010 11:16:04 -0700
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It is deliberate; see RFC 3261 concerning when answer SDP needs to be withi=
n INVITE's 2xx response.  The 18x's were sent reliable; thus completing the=
 offer/answer exchange.  The following is mention above both figures: "PRAC=
K transactions are not shown for clarity".

-brett

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Peter Musgrave
Sent: Sunday, March 14, 2010 11:47 AM
To: Gonzalo.Camarillo@ericsson.com; Christer.Holmberg@ericsson.com; gao.yan=
g2@zte.com.cn
Cc: sipcore@ietf.org
Subject: [sipcore] draft-ietf-sipcore-reinvite-03

Hi Gonzalo/Chris/Gao,

Not sure if this is a nit - or if I am lost...

In Figure 3 the 200 OK for the second INVITE (which made an offer) does not=
 carry an answer. Is this deliberate? (I have the same issue in Figure 4 ma=
king me think perhaps I'm lost.)

If yes, then I think an explanatory comment would be helpful - since I woul=
d expect a 200 with answer .

Regards,

Peter Musgrave

--_000_747A6506A991724FB09B129B79D5FEB614806AF6E4EXMBXCLUS01ci_
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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>It is deliberate; see RFC 3261 concerning when answer SDP ne=
eds
to be within INVITE&#8217;s 2xx response.&nbsp; The 18x&#8217;s were sent
reliable; thus completing the offer/answer exchange.&nbsp; The following is
mention above both figures: &#8220;PRACK transactions are not shown for cla=
rity&#8221;.<o:p></o:p></span></p>

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

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

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

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

<div>

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

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] <b>On Behalf Of =
</b>Peter
Musgrave<br>
<b>Sent:</b> Sunday, March 14, 2010 11:47 AM<br>
<b>To:</b> Gonzalo.Camarillo@ericsson.com; Christer.Holmberg@ericsson.com;
gao.yang2@zte.com.cn<br>
<b>Cc:</b> sipcore@ietf.org<br>
<b>Subject:</b> [sipcore] draft-ietf-sipcore-reinvite-03<o:p></o:p></span><=
/p>

</div>

</div>

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

<p class=3DMsoNormal>Hi Gonzalo/Chris/Gao,&nbsp; <o:p></o:p></p>

<div>

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

</div>

<div>

<p class=3DMsoNormal>Not sure if this is a nit - or if I am lost...<o:p></o=
:p></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal>In Figure 3 the 200 OK for the second INVITE (which ma=
de an
offer) does not carry an answer. Is this deliberate? (I have the same issue=
 in
Figure 4 making me think perhaps I'm lost.)<o:p></o:p></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal>If yes, then I think an explanatory comment would be h=
elpful
- since I would expect a 200 with answer .<o:p></o:p></p>

</div>

<div>

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

</div>

<div>

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

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal>Peter Musgrave<o:p></o:p></p>

</div>

</div>

</div>

</body>

</html>

--_000_747A6506A991724FB09B129B79D5FEB614806AF6E4EXMBXCLUS01ci_--

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To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Cc: SIPCORE SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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SOME people think session management for SUBSCRIBE/NOTIFY is too hard, =
especially if the lifetime of the user-user exchange exactly matches the =
lifetime of the SIP dialog.

Simplicity and laziness?

Ah - MESSAGE (I was thinking SUBSCRIBE). Problem is you have no way of =
knowing whether the receiver wants to digest what you're sending.

When you say UAC for the INVITE, I presume you mean the UAC for the =
INFO, as well? If so, it is the UAS that must offer a Recv-Info header. =
What the UAC offers is irrelevant, unless the UAC wants to advertise =
what the UAC is willing to receive from the UAS.  All of that should be =
in the text.

On Mar 14, 2010, at 1:34 PM, Avasarala Ranjit-A20990 wrote:

>=20
> Hi Eric and all,
>=20
> Then what is the real purpose of an INFO package? Is there any example =
or use case where I would use an INFO package over SIP MESSAGE? Also if =
the INFO package is going to take the format of an XML document (I =
think), then how different would that be from the regular event =
packages?
>=20
> One more query I have is : say in case the UAC does not include the =
Recv-Info header in the initial INVITE request, then can it still use =
INFO with package with mid session INFO message?
>=20
> Thanks
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Eric Burger [mailto:eburger@standardstrack.com]=20
> Sent: Sunday, March 14, 2010 10:57 PM
> To: Avasarala Ranjit-A20990
> Cc: SIPCORE SIPCORE
> Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
>=20
> The point of Section 7.4 is there is often a far superior method of =
doing user-user signaling besides INFO. For example, if you want to send =
DTMF digits, use KPML (he says modestly).
>=20
> There is absolutely NO correspondence between an Info Package and an =
Event Package. Moreover, if someone tried to register such, my guess is =
they would not get the requisite Specification Required level needed to =
register a package.
>=20
> On Mar 14, 2010, at 4:03 AM, Avasarala Ranjit-A20990 wrote:
>=20
>> Hi Adam
>>=20
>> =46rom the latest version of the info =
draft-ietf-sipcore-info-events-07, the section 4 "Info Packages" does =
not talk about the kind of packages and their format.
>>=20
>> Though the section 7.4.1.2  talks that use of SUBSCRIBE / NOTIFY =
based on 3265 as an alternate mechanism, it is not clear what should be =
the format of a typical INFO package.
>>=20
>> Also further Section 7.4.1.3 mentions the usage of SIP MESSAGE =
method. So does this mean I can use MESSAGE method as an alternative to =
INFO?
>>=20
>> Thanks
>> Regards
>> Ranjit
>>=20
>>=20
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Sunday, March 14, 2010 4:23 AM
>> To: Avasarala Ranjit-A20990
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
>>=20
>> Please don't cross-post to dispatch and sipcore.
>>=20
>> On 3/12/10 23:26, Mar 12, Avasarala Ranjit-A20990 wrote:
>>> Are there any registered INFO packages
>>>=20
>>=20
>> Not yet.
>>=20
>>> and can the regular SIP event packages based on RFC 3265 be used =
with INFO mechanism?
>>>=20
>>=20
>> Absolutely not. No. Never. Under no circumstances. Nein, =D0=9D=D0=B5=D1=
=82, =E0=A4=A8=E0=A4=B9=E0=A5=80=E0=A4=82, Non, Nej. They are =
_completely_ different things.
>>=20
>> Really, if this is somehow unclear in the INFO draft, we need to pull =
it from publication request and work on it some more. It would be the =
worst possible outcome if implementors were led to believe that 3265 =
packages could be used with INFO. Where did you get the impression that =
this might be okay?
>>=20
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20


From nancy.greene@ericsson.com  Sun Mar 14 20:10:19 2010
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From: Nancy Greene <nancy.greene@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 14 Mar 2010 23:10:16 -0400
Thread-Topic: RFC 3263 - why require use port in  Request-URI?
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Subject: [sipcore] RFC 3263 - why require use port in  Request-URI?
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I have a question on section 4.2 of RFC 3263:=20
=20
Why is it that the port of the Request-URI is required to be used for an in=
termediate proxy? Is the issue that it is not known whether the next hop pr=
oxy is intermediate or is the actual destination in the Request-URI?=20

If so shouldn't there at least be a procedure described to use the port fro=
m DNS instead of the one in the Request-URI if sending to the one in the Re=
quest-URI fails?

Section 4.2 from RFC 3263 (locating SIP servers):

If the TARGET was not a numeric IP address, but a port is present       =20
in the URI, the client performs an A or AAAA record lookup of the
domain name. The result will be a list of IP addresses, each of
which can be contacted at the specific port from the URI and
transport protocol determined previously.

Nancy
=20


From ranjit@motorola.com  Sun Mar 14 23:30:31 2010
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References: <750BBC72E178114F9DC4872EBFF29A5B0A3BE70F@ZMY16EXM66.ds.mot.com><4B9C1752.1000905@nostrum.com><750BBC72E178114F9DC4872EBFF29A5B0A3BE74E@ZMY16EXM66.ds.mot.com><E01CA333-AF1E-4D26-B1E8-D99493838C13@standardstrack.com><750BBC72E178114F9DC4872EBFF29A5B0A3BE75A@ZMY16EXM66.ds.mot.com> <2AB9580C-6972-4E08-898E-E7E675E1DF81@standardstrack.com>
From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Eric Burger" <eburger@standardstrack.com>
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From pkyzivat@cisco.com  Mon Mar 15 06:10:23 2010
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Subject: Re: [sipcore] RFC 3263 - why require use port in  Request-URI?
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Nancy,

Questions such as this should be brought to 
<sip-implementors@lists.cs.columbia.edu>.

To answer your question...

The port from the URI is to be used *if* it is present in the URI.
(Would you prefer to *ignore* it?)

If the the port is *not* present in the URI, then the port is obtained 
from DNS via the SRV query. That is explained further on in the same 
section of the document.

	Thanks,
	Paul


Nancy Greene wrote:
> I have a question on section 4.2 of RFC 3263: 
>  
> Why is it that the port of the Request-URI is required to be used for an intermediate proxy? Is the issue that it is not known whether the next hop proxy is intermediate or is the actual destination in the Request-URI? 
> 
> If so shouldn't there at least be a procedure described to use the port from DNS instead of the one in the Request-URI if sending to the one in the Request-URI fails?
> 
> Section 4.2 from RFC 3263 (locating SIP servers):
> 
> If the TARGET was not a numeric IP address, but a port is present        
> in the URI, the client performs an A or AAAA record lookup of the
> domain name. The result will be a list of IP addresses, each of
> which can be contacted at the specific port from the URI and
> transport protocol determined previously.
> 
> Nancy
>  
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From nancy.greene@ericsson.com  Mon Mar 15 06:33:08 2010
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From: Nancy Greene <nancy.greene@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 15 Mar 2010 09:32:37 -0400
Thread-Topic: [sipcore] RFC 3263 - why require use port in  Request-URI?
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Subject: Re: [sipcore] RFC 3263 - why require use port in  Request-URI?
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Thanks for the answer, but I'd like to know why the RFC is written the way =
it is.=20

What if there is an intermediate SIP proxy between the target in the Reques=
t-URI and the proxy trying to reach it? That intermediate SIP proxy does no=
t allow traffic on the port mentioned in the Request-URI, so the request fa=
ils to reach the target.  This does not seem right. Shouldn't the RFC at th=
e very least require a retry at the port that the intermediate proxy expect=
s traffic on?

Nancy

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: March-15-10 9:08 AM
To: Nancy Greene
Cc: sipcore@ietf.org
Subject: Re: [sipcore] RFC 3263 - why require use port in Request-URI?

Nancy,

Questions such as this should be brought to <sip-implementors@lists.cs.colu=
mbia.edu>.

To answer your question...

The port from the URI is to be used *if* it is present in the URI.
(Would you prefer to *ignore* it?)

If the the port is *not* present in the URI, then the port is obtained from=
 DNS via the SRV query. That is explained further on in the same section of=
 the document.

	Thanks,
	Paul


Nancy Greene wrote:
> I have a question on section 4.2 of RFC 3263:=20
> =20
> Why is it that the port of the Request-URI is required to be used for an =
intermediate proxy? Is the issue that it is not known whether the next hop =
proxy is intermediate or is the actual destination in the Request-URI?=20
>=20
> If so shouldn't there at least be a procedure described to use the port f=
rom DNS instead of the one in the Request-URI if sending to the one in the =
Request-URI fails?
>=20
> Section 4.2 from RFC 3263 (locating SIP servers):
>=20
> If the TARGET was not a numeric IP address, but a port is present       =
=20
> in the URI, the client performs an A or AAAA record lookup of the=20
> domain name. The result will be a list of IP addresses, each of which=20
> can be contacted at the specific port from the URI and transport=20
> protocol determined previously.
>=20
> Nancy
> =20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From adam@nostrum.com  Mon Mar 15 07:17:03 2010
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Subject: Re: [sipcore] RFC 3263 - why require use port in  Request-URI?
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[as WG chair]

Nancy:

Paul is correct. This is the type of question that belongs on 
<sip-implementors@lists.cs.columbia.edu>, not the standards development 
mailing list. Please redirect your queries there.

In practice, I believe you are confused about what section 4.2 actually 
says. Someone on the sip-implementors list should be able to help 
clarify things for you.

Thanks.

/a

On 3/15/10 08:32, Mar 15, Nancy Greene wrote:
> Thanks for the answer, but I'd like to know why the RFC is written the way it is.
>
> What if there is an intermediate SIP proxy between the target in the Request-URI and the proxy trying to reach it? That intermediate SIP proxy does not allow traffic on the port mentioned in the Request-URI, so the request fails to reach the target.  This does not seem right. Shouldn't the RFC at the very least require a retry at the port that the intermediate proxy expects traffic on?
>
> Nancy
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: March-15-10 9:08 AM
> To: Nancy Greene
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] RFC 3263 - why require use port in Request-URI?
>
> Nancy,
>
> Questions such as this should be brought to<sip-implementors@lists.cs.columbia.edu>.
>
> To answer your question...
>
> The port from the URI is to be used *if* it is present in the URI.
> (Would you prefer to *ignore* it?)
>
> If the the port is *not* present in the URI, then the port is obtained from DNS via the SRV query. That is explained further on in the same section of the document.
>
> 	Thanks,
> 	Paul
>
>
> Nancy Greene wrote:
>    
>> I have a question on section 4.2 of RFC 3263:
>>
>> Why is it that the port of the Request-URI is required to be used for an intermediate proxy? Is the issue that it is not known whether the next hop proxy is intermediate or is the actual destination in the Request-URI?
>>
>> If so shouldn't there at least be a procedure described to use the port from DNS instead of the one in the Request-URI if sending to the one in the Request-URI fails?
>>
>> Section 4.2 from RFC 3263 (locating SIP servers):
>>
>> If the TARGET was not a numeric IP address, but a port is present
>> in the URI, the client performs an A or AAAA record lookup of the
>> domain name. The result will be a list of IP addresses, each of which
>> can be contacted at the specific port from the URI and transport
>> protocol determined previously.
>>
>> Nancy
>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>>      
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>    


From wwwrun@core3.amsl.com  Mon Mar 15 07:23:26 2010
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Subject: [sipcore] Last Call: draft-ietf-sipcore-info-events (Session Initiation Protocol (SIP) INFO Method and Package Framework) to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol Core 
WG (sipcore) to consider the following document:

- 'Session Initiation Protocol (SIP) INFO Method and Package Framework '
   <draft-ietf-sipcore-info-events-07.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2010-03-29. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-info-events-07.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=18772&rfc_flag=0


From adam@nostrum.com  Mon Mar 15 07:53:40 2010
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Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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On 3/14/10 19:01, Mar 14, Eric Burger wrote:
> Ah - MESSAGE (I was thinking SUBSCRIBE). Problem is you have no way of knowing whether the receiver wants to digest what you're sending.
>    

That's one problem.

The key problem is that MESSAGE means something. And that something is 
not "here's a pile of bits, figure out what to do with them." The 
meaning is "here is some kind of media that you need to present to your 
user."

For non-media payloads consumed by automata, you need something with 
different semantics.

/a

From adam@nostrum.com  Mon Mar 15 08:26:38 2010
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Cc: SIPCORE SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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[as participant]

On 3/15/10 01:28, Mar 15, Avasarala Ranjit-A20990 wrote:
> Ok. Say if I want to write a INFO package for a particular purpose, what should be the format of that package? Should it be an XML one like the event packages?

Not all event packages are XML -- see RFC 3842, for example. The format 
that is best suited for conveying information in a MIME body generally 
depends on what you're trying to convey. People seem to like XML for 
this kind of thing, but there is nothing preventing any appropriate data 
format.

> Then what would be the procedures for that package

If you want to publish an INFO package as part of the IETF process, you 
really need to step through section 7 -- and, in particular, section 7.4 
-- and explain why INFO is the right tool for the task you have in mind. 
You need to explain why the other mechanisms designed for inter-automata 
communication don't suit the purpose of the application you're 
designing. Describe why, for example, you want to use INFO instead of 
SUBSCRIBE and NOTIFY.

To be clear, you can define INFO packages outside the IETF process, as 
long as you do so in a way that can be referenced and which is 
effectively permanent. This would typically involve working with another 
SDO, although a company, organization, or individual could, in theory, 
publish a specification as well (as long as it met the criteria 
described in section 11.4 of the INFO draft).

In either case, you'll need to include enough information to allow 
interoperable implementation. Although it's not the only way to do 
things, the easy way to do this would be to include the following 12 
sections in your document, with contents as suggested by the 
corresponding parts of Section 10 of the INFO framework:

      1.  General
      2.  Overall Description
      3.  Applicability
      4.  Info Package Name
      5.  Info Package Parameters
      6.  SIP Option Tags
      7.  INFO Message Body Parts
      8.  Info Package Usage Restrictions
      9.  Rate of INFO Requests
      10. Info Package Security Considerations
      11. Implementation Details
      12. Examples


>   - like the UAC that intends to send a particular session or application level information generates the corresponding INFO package and sends it?
>
> E.g  UAC sends INVITE to UAS with Recv-Info: app-info
>       ...............................................
>       UAC sends INFO (app-info package)
>
>   will app-info package be something like this
>
>   <xml version="1.0" ?>
>    .................
>    <app-info>
>      ...
>    </app-info>
>   </xml>
>    

Except that it wouldn't be something as generic as "app-info." You'd 
need to define an info package that specified what kind of information 
is to be conveyed, and how the recipient processes it. Ideally, your 
event package name would reflect that application in some way.

You'll need a formal (and complete) definition of the body syntax. If 
you choose to use XML, this would be something like a DTD, Schema, or 
Relax NG definition. If you use some other data framework (e.g., ASN.1), 
you'll need to define your fields using the formal definition language 
provided by that framework. If you're rolling your own syntax, you 
probably want to use something like ABNF.

But I'm pretty much just repeating the guidance provided by the INFO 
draft itself. I would suggest a careful, detailed reading -- with 
special attention to sections 7 and 10 (and their respective subsections).

/a

From peter.musgrave@magorcorp.com  Mon Mar 15 09:34:44 2010
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Subject: [sipcore] INFO Package Candidate? RFC5168 Media Control
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--0016369208b43cf1540481d97556
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Hi,

I am finding the INFO event draft helpful (and have it in my code for my
'vendor specific' needs).

Is there interest in defining an INFO package for picture fast updates for a
video stream as per RFC5168?

I am currently not aware of a standard mechanism to see if an endpoint will
originate or accept such INFOs (although a number of implementations do).
Currently I am 'guessing'.

Would such a definition be done in sipcore or as part of mmusic (the
originators of RFC).

Regards,

Peter Musgrave

--0016369208b43cf1540481d97556
Content-Type: text/html; charset=ISO-8859-1
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Hi,=A0<div><br></div><div>I am finding the INFO event draft helpful (and ha=
ve it in my code for my &#39;vendor specific&#39; needs).<br><div><br></div=
><div>Is there interest in defining an INFO package for picture fast update=
s for a video stream as per RFC5168?</div>

<div><br></div><div>I am currently not aware of a standard mechanism to see=
 if an endpoint will originate or accept such INFOs (although a number of i=
mplementations do). Currently I am &#39;guessing&#39;.</div><div><br></div>

<div>Would such a definition be done in sipcore or as part of mmusic (the o=
riginators of RFC).</div><div><br></div><div>Regards,=A0</div><div><br></di=
v><div>Peter Musgrave</div></div>

--0016369208b43cf1540481d97556--

From pkyzivat@cisco.com  Mon Mar 15 09:41:57 2010
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Subject: Re: [sipcore] INFO Package Candidate? RFC5168 Media Control
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[I'm now quite sure what hat I'm wearing here. I don't think I have 
formally received by "chair" hat. I guess I would be wearing my chair 
hat if I had one.]

sipcore would *not* be the right place to do such an info package.
(Most likely it will never be the place to do *any* info package.)

mmusic does sound like a plausible place to do it. But of course they 
would have to agree.

	Thanks,
	Paul

Peter Musgrave wrote:
> Hi, 
> 
> I am finding the INFO event draft helpful (and have it in my code for my 
> 'vendor specific' needs).
> 
> Is there interest in defining an INFO package for picture fast updates 
> for a video stream as per RFC5168?
> 
> I am currently not aware of a standard mechanism to see if an endpoint 
> will originate or accept such INFOs (although a number of 
> implementations do). Currently I am 'guessing'.
> 
> Would such a definition be done in sipcore or as part of mmusic (the 
> originators of RFC).
> 
> Regards, 
> 
> Peter Musgrave
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From adam@nostrum.com  Mon Mar 15 09:45:21 2010
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Cc: sipcore@ietf.org, Peter Musgrave <peter.musgrave@magorcorp.com>
Subject: Re: [sipcore] INFO Package Candidate? RFC5168 Media Control
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[as chair]

And, when in doubt, I'd take things to DISPATCH. That's what they're 
there for.

/a

On 3/15/10 11:42, Mar 15, Paul Kyzivat wrote:
> [I'm now quite sure what hat I'm wearing here. I don't think I have 
> formally received by "chair" hat. I guess I would be wearing my chair 
> hat if I had one.]
>
> sipcore would *not* be the right place to do such an info package.
> (Most likely it will never be the place to do *any* info package.)
>
> mmusic does sound like a plausible place to do it. But of course they 
> would have to agree.
>
>     Thanks,
>     Paul
>
> Peter Musgrave wrote:
>> Hi,
>> I am finding the INFO event draft helpful (and have it in my code for 
>> my 'vendor specific' needs).
>>
>> Is there interest in defining an INFO package for picture fast 
>> updates for a video stream as per RFC5168?
>>
>> I am currently not aware of a standard mechanism to see if an 
>> endpoint will originate or accept such INFOs (although a number of 
>> implementations do). Currently I am 'guessing'.
>>
>> Would such a definition be done in sipcore or as part of mmusic (the 
>> originators of RFC).
>>
>> Regards,
>> Peter Musgrave
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Hi Peter,

those 200 (OK) responses do not carry answers because of one of the main
rules of the offer/answer model. That is, the UAS only provides one
answer to a given offer. In the cases you are pointing to, the answer
was provided in a reliable provisional response.

Cheers,

Gonzalo

On 14/03/2010 7:46 AM, Peter Musgrave wrote:
> Hi Gonzalo/Chris/Gao, 
> 
> Not sure if this is a nit - or if I am lost...
> 
> In Figure 3 the 200 OK for the second INVITE (which made an offer) does
> not carry an answer. Is this deliberate? (I have the same issue in
> Figure 4 making me think perhaps I'm lost.)
> 
> If yes, then I think an explanatory comment would be helpful - since I
> would expect a 200 with answer .
> 
> Regards, 
> 
> Peter Musgrave


From keith.drage@alcatel-lucent.com  Mon Mar 15 11:34:31 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Adam Roach <adam@nostrum.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 15 Mar 2010 19:34:28 +0100
Thread-Topic: [sipcore] INFO Package Candidate? RFC5168 Media Control
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Subject: Re: [sipcore] INFO Package Candidate? RFC5168 Media Control
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If you want to get it in front of a WG, dispatch is probably the best place=
 to start. (Do remember that dispatch is using advanced deadlines on the no=
rmal WG submission deadline).

Remember however that you do not need a WG charter item, or a proposed stan=
dard, to create an INFO package. Aside from getting some other group outsid=
e IETF to put it in one of their specifications, if it is simple, you may b=
e able to persuade an AD to sponsor it (as a more efficient task that sendi=
ng it to a WG), or it can just be submitted as informational and wait.

regards

Keith=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: Monday, March 15, 2010 4:45 PM
> To: Paul Kyzivat
> Cc: sipcore@ietf.org; Peter Musgrave
> Subject: Re: [sipcore] INFO Package Candidate? RFC5168 Media Control
>=20
> [as chair]
>=20
> And, when in doubt, I'd take things to DISPATCH. That's what=20
> they're there for.
>=20
> /a
>=20
> On 3/15/10 11:42, Mar 15, Paul Kyzivat wrote:
> > [I'm now quite sure what hat I'm wearing here. I don't think I have=20
> > formally received by "chair" hat. I guess I would be=20
> wearing my chair=20
> > hat if I had one.]
> >
> > sipcore would *not* be the right place to do such an info package.
> > (Most likely it will never be the place to do *any* info package.)
> >
> > mmusic does sound like a plausible place to do it. But of=20
> course they=20
> > would have to agree.
> >
> >     Thanks,
> >     Paul
> >
> > Peter Musgrave wrote:
> >> Hi,
> >> I am finding the INFO event draft helpful (and have it in=20
> my code for=20
> >> my 'vendor specific' needs).
> >>
> >> Is there interest in defining an INFO package for picture fast=20
> >> updates for a video stream as per RFC5168?
> >>
> >> I am currently not aware of a standard mechanism to see if an=20
> >> endpoint will originate or accept such INFOs (although a number of=20
> >> implementations do). Currently I am 'guessing'.
> >>
> >> Would such a definition be done in sipcore or as part of=20
> mmusic (the=20
> >> originators of RFC).
> >>
> >> Regards,
> >> Peter Musgrave
> >>
> >>
> >>=20
> ---------------------------------------------------------------------
> >> ---
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From dean.willis@softarmor.com  Mon Mar 15 16:29:37 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
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On Mar 15, 2010, at 12:35 PM, Gonzalo Camarillo wrote:

> Hi Peter,
>
> those 200 (OK) responses do not carry answers because of one of the  
> main
> rules of the offer/answer model. That is, the UAS only provides one
> answer to a given offer. In the cases you are pointing to, the answer
> was provided in a reliable provisional response.
>

There was, IIRC, some discussion on that, with my position being that  
the 200 OK SHOULD have a copy (and if present, it MUST be an identical  
copy!) of the answer provided in the reliable provisional response.

My goal here is that this makes certain debugging scenarios easier. It  
also makes these hideous call flows more palatable to my brain, which  
along with refusing to believe in the square root of negative one also  
seems to have trouble with reliable provisional media and the plethora  
of alternative O/A sattes that can result.

However, I do believe that the consensus was as Gonzalo reports.

--
Dean


From pkyzivat@cisco.com  Mon Mar 15 16:50:25 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
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[as individual]

I differ slightly from Dean on this:

- if the answer has been sent in a reliable provisional,
   it is not required, or expected, in the 2xx.

- for backward compatibility, it MAY, be *repeated* in
   the 2xx, but this is not advised.

The reason its not advised is that its presence there can only cause 
grief, not good. The only valid thing you can do if you receive it is 
ignore it. If you *use* it (i.e. switch to it) then you are going to be 
in a world of hurt if there has been more than one o/a exchange before 
the 2xx is sent. Specifically, in that case the only SDP that is 
permitted to appear in the 2xx is the *first* answer, which may well 
differ from the *last* answer that is to be used for the session.

	Thanks,
	Paul

Dean Willis wrote:
> 
> On Mar 15, 2010, at 12:35 PM, Gonzalo Camarillo wrote:
> 
>> Hi Peter,
>>
>> those 200 (OK) responses do not carry answers because of one of the main
>> rules of the offer/answer model. That is, the UAS only provides one
>> answer to a given offer. In the cases you are pointing to, the answer
>> was provided in a reliable provisional response.
>>
> 
> There was, IIRC, some discussion on that, with my position being that 
> the 200 OK SHOULD have a copy (and if present, it MUST be an identical 
> copy!) of the answer provided in the reliable provisional response.
> 
> My goal here is that this makes certain debugging scenarios easier. It 
> also makes these hideous call flows more palatable to my brain, which 
> along with refusing to believe in the square root of negative one also 
> seems to have trouble with reliable provisional media and the plethora 
> of alternative O/A sattes that can result.
> 
> However, I do believe that the consensus was as Gonzalo reports.
> 
> -- 
> Dean
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From alexander.milinski@nsn.com  Tue Mar 16 03:19:02 2010
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From: "Milinski, Alexander (NSN - DE/Munich)" <alexander.milinski@nsn.com>
To: "ext Paul Kyzivat" <pkyzivat@cisco.com>, "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
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Hi Paul, Dean,

now I am confused (though I got used to the square root of -1 being i).

Which RFC explains this repetition?

Moreover, how is the repeated answer distingusished from a new offer
(the previous o/a has been completed before),
which should be answered in the ACK?

Regards,
Alex


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of ext Paul Kyzivat
> Sent: Tuesday, March 16, 2010 12:50 AM
> To: Dean Willis
> Cc: gao.yang2@zte.com.cn; sipcore@ietf.org; Peter Musgrave
> Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
>=20
> [as individual]
>=20
> I differ slightly from Dean on this:
>=20
> - if the answer has been sent in a reliable provisional,
>    it is not required, or expected, in the 2xx.
>=20
> - for backward compatibility, it MAY, be *repeated* in
>    the 2xx, but this is not advised.
>=20
> The reason its not advised is that its presence there can only cause=20
> grief, not good. The only valid thing you can do if you receive it is=20
> ignore it. If you *use* it (i.e. switch to it) then you are=20
> going to be=20
> in a world of hurt if there has been more than one o/a=20
> exchange before=20
> the 2xx is sent. Specifically, in that case the only SDP that is=20
> permitted to appear in the 2xx is the *first* answer, which may well=20
> differ from the *last* answer that is to be used for the session.
>=20
> 	Thanks,
> 	Paul
>=20
> Dean Willis wrote:
> >=20
> > On Mar 15, 2010, at 12:35 PM, Gonzalo Camarillo wrote:
> >=20
> >> Hi Peter,
> >>
> >> those 200 (OK) responses do not carry answers because of=20
> one of the main
> >> rules of the offer/answer model. That is, the UAS only provides one
> >> answer to a given offer. In the cases you are pointing to,=20
> the answer
> >> was provided in a reliable provisional response.
> >>
> >=20
> > There was, IIRC, some discussion on that, with my position=20
> being that=20
> > the 200 OK SHOULD have a copy (and if present, it MUST be=20
> an identical=20
> > copy!) of the answer provided in the reliable provisional response.
> >=20
> > My goal here is that this makes certain debugging scenarios=20
> easier. It=20
> > also makes these hideous call flows more palatable to my=20
> brain, which=20
> > along with refusing to believe in the square root of=20
> negative one also=20
> > seems to have trouble with reliable provisional media and=20
> the plethora=20
> > of alternative O/A sattes that can result.
> >=20
> > However, I do believe that the consensus was as Gonzalo reports.
> >=20
> > --=20
> > Dean
> >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20

From peter.musgrave@magorcorp.com  Tue Mar 16 04:00:26 2010
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Date: Tue, 16 Mar 2010 07:00:26 -0400
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To: "Milinski, Alexander (NSN - DE/Munich)" <alexander.milinski@nsn.com>
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Hi all,

I would assume a new offer in a 200 OK would be distinguished by
increasing the sess-verssion in the o=3D line of the SDP.

I do find the issue of repeating an SDP hard to pin down - after
poking into 3261 and 3262...am I looking in the right places?

Peter

On Tue, Mar 16, 2010 at 6:18 AM, Milinski, Alexander (NSN - DE/Munich)
<alexander.milinski@nsn.com> wrote:
> Hi Paul, Dean,
>
> now I am confused (though I got used to the square root of -1 being i).
>
> Which RFC explains this repetition?
>
> Moreover, how is the repeated answer distingusished from a new offer
> (the previous o/a has been completed before),
> which should be answered in the ACK?
>
> Regards,
> Alex
>
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of ext Paul Kyzivat
>> Sent: Tuesday, March 16, 2010 12:50 AM
>> To: Dean Willis
>> Cc: gao.yang2@zte.com.cn; sipcore@ietf.org; Peter Musgrave
>> Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
>>
>> [as individual]
>>
>> I differ slightly from Dean on this:
>>
>> - if the answer has been sent in a reliable provisional,
>> =A0 =A0it is not required, or expected, in the 2xx.
>>
>> - for backward compatibility, it MAY, be *repeated* in
>> =A0 =A0the 2xx, but this is not advised.
>>
>> The reason its not advised is that its presence there can only cause
>> grief, not good. The only valid thing you can do if you receive it is
>> ignore it. If you *use* it (i.e. switch to it) then you are
>> going to be
>> in a world of hurt if there has been more than one o/a
>> exchange before
>> the 2xx is sent. Specifically, in that case the only SDP that is
>> permitted to appear in the 2xx is the *first* answer, which may well
>> differ from the *last* answer that is to be used for the session.
>>
>> =A0 =A0 =A0 Thanks,
>> =A0 =A0 =A0 Paul
>>
>> Dean Willis wrote:
>> >
>> > On Mar 15, 2010, at 12:35 PM, Gonzalo Camarillo wrote:
>> >
>> >> Hi Peter,
>> >>
>> >> those 200 (OK) responses do not carry answers because of
>> one of the main
>> >> rules of the offer/answer model. That is, the UAS only provides one
>> >> answer to a given offer. In the cases you are pointing to,
>> the answer
>> >> was provided in a reliable provisional response.
>> >>
>> >
>> > There was, IIRC, some discussion on that, with my position
>> being that
>> > the 200 OK SHOULD have a copy (and if present, it MUST be
>> an identical
>> > copy!) of the answer provided in the reliable provisional response.
>> >
>> > My goal here is that this makes certain debugging scenarios
>> easier. It
>> > also makes these hideous call flows more palatable to my
>> brain, which
>> > along with refusing to believe in the square root of
>> negative one also
>> > seems to have trouble with reliable provisional media and
>> the plethora
>> > of alternative O/A sattes that can result.
>> >
>> > However, I do believe that the consensus was as Gonzalo reports.
>> >
>> > --
>> > Dean
>> >
>> > _______________________________________________
>> > sipcore mailing list
>> > sipcore@ietf.org
>> > https://www.ietf.org/mailman/listinfo/sipcore
>> >
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>

From thomas.belling@nsn.com  Tue Mar 16 05:08:35 2010
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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: "ext Peter Musgrave" <peter.musgrave@magorcorp.com>, "Milinski, Alexander (NSN - DE/Munich)" <alexander.milinski@nsn.com>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
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Hi Peter, Alexander
=20
> I do find the issue of repeating an SDP hard to pin down - after
poking into 3261 and 3262...am I looking in the right places?

Please read RFC 3261, Clause 13.2.1:
=20
In this specification, offers and answers
   can only appear in INVITE requests and responses, and ACK.  The usage
   of offers and answers is further restricted.  For the initial INVITE
   transaction, the rules are:
      o  The initial offer MUST be in either an INVITE or, if not there,
         in the first reliable non-failure message from the UAS back to
         the UAC.  In this specification, that is the final 2xx
         response.
      o  If the initial offer is in an INVITE, the answer MUST be in a
         reliable non-failure message from UAS back to UAC which is
         correlated to that INVITE.  For this specification, that is
         only the final 2xx response to that INVITE.  That same exact
         answer MAY also be placed in any provisional responses sent
         prior to the answer.  The UAC MUST treat the first session
         description it receives as the answer, and MUST ignore any
         session descriptions in subsequent responses to the initial
         INVITE.
      o  If the initial offer is in the first reliable non-failure
         message from the UAS back to UAC, the answer MUST be in the
         acknowledgement for that message (in this specification, ACK
         for a 2xx response).
      o  After having sent or received an answer to the first offer, the
         UAC MAY generate subsequent offers in requests based on rules
         specified for that method, but only if it has received answers
         to any previous offers, and has not sent any offers to which it
         hasn't gotten an answer.
      o  Once the UAS has sent or received an answer to the initial
         offer, it MUST NOT generate subsequent offers in any responses
         to the initial INVITE.  This means that a UAS based on this
         specification alone can never generate subsequent offers until
         completion of the initial transaction.
=20
Regards, Thomas

=20
________________________________

Von: sipcore-bounces@ietf.org im Auftrag von ext Peter Musgrave
Gesendet: Di 16.03.2010 12:00
An: Milinski, Alexander (NSN - DE/Munich)
Cc: gao.yang2@zte.com.cn; sipcore@ietf.org
Betreff: Re: [sipcore] draft-ietf-sipcore-reinvite-03



Hi all,

I would assume a new offer in a 200 OK would be distinguished by
increasing the sess-verssion in the o=3D line of the SDP.

I do find the issue of repeating an SDP hard to pin down - after
poking into 3261 and 3262...am I looking in the right places?

Peter

On Tue, Mar 16, 2010 at 6:18 AM, Milinski, Alexander (NSN - DE/Munich)
<alexander.milinski@nsn.com> wrote:
> Hi Paul, Dean,
>
> now I am confused (though I got used to the square root of -1 being =
i).
>
> Which RFC explains this repetition?
>
> Moreover, how is the repeated answer distingusished from a new offer
> (the previous o/a has been completed before),
> which should be answered in the ACK?
>
> Regards,
> Alex
>
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of ext Paul Kyzivat
>> Sent: Tuesday, March 16, 2010 12:50 AM
>> To: Dean Willis
>> Cc: gao.yang2@zte.com.cn; sipcore@ietf.org; Peter Musgrave
>> Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
>>
>> [as individual]
>>
>> I differ slightly from Dean on this:
>>
>> - if the answer has been sent in a reliable provisional,
>>    it is not required, or expected, in the 2xx.
>>
>> - for backward compatibility, it MAY, be *repeated* in
>>    the 2xx, but this is not advised.
>>
>> The reason its not advised is that its presence there can only cause
>> grief, not good. The only valid thing you can do if you receive it is
>> ignore it. If you *use* it (i.e. switch to it) then you are
>> going to be
>> in a world of hurt if there has been more than one o/a
>> exchange before
>> the 2xx is sent. Specifically, in that case the only SDP that is
>> permitted to appear in the 2xx is the *first* answer, which may well
>> differ from the *last* answer that is to be used for the session.
>>
>>       Thanks,
>>       Paul
>>
>> Dean Willis wrote:
>> >
>> > On Mar 15, 2010, at 12:35 PM, Gonzalo Camarillo wrote:
>> >
>> >> Hi Peter,
>> >>
>> >> those 200 (OK) responses do not carry answers because of
>> one of the main
>> >> rules of the offer/answer model. That is, the UAS only provides =
one
>> >> answer to a given offer. In the cases you are pointing to,
>> the answer
>> >> was provided in a reliable provisional response.
>> >>
>> >
>> > There was, IIRC, some discussion on that, with my position
>> being that
>> > the 200 OK SHOULD have a copy (and if present, it MUST be
>> an identical
>> > copy!) of the answer provided in the reliable provisional response.
>> >
>> > My goal here is that this makes certain debugging scenarios
>> easier. It
>> > also makes these hideous call flows more palatable to my
>> brain, which
>> > along with refusing to believe in the square root of
>> negative one also
>> > seems to have trouble with reliable provisional media and
>> the plethora
>> > of alternative O/A sattes that can result.
>> >
>> > However, I do believe that the consensus was as Gonzalo reports.
>> >
>> > --
>> > Dean
>> >
>> > _______________________________________________
>> > sipcore mailing list
>> > sipcore@ietf.org
>> > https://www.ietf.org/mailman/listinfo/sipcore
>> >
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore



From peter.musgrave@magorcorp.com  Tue Mar 16 05:27:06 2010
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Date: Tue, 16 Mar 2010 08:27:06 -0400
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Hi Thomas,

Thanks for the reference. I was reading that part of the spec. but
could not make my mind up about whether it forbids sending a copy of
the SDP in the 200 (if a reliable response had been provided). The
closest I came was the statement:

The UAC MUST treat the first session
        description it receives as the answer, and MUST ignore any
        session descriptions in subsequent responses to the initial
        INVITE.

Which makes me think if this does happen it MUST be ignored. So that
convinces me that it's not a good idea. I just didn't see any language
saying it was forbidden. Perhaps it should have been. I think what
makes this something I have to think/read carefully about is the lack
of examples in PRACK/3262.

Apologies to all if I am belabouring the point here.

Peter

On Tue, Mar 16, 2010 at 8:04 AM, Belling, Thomas (NSN - DE/Munich)
<thomas.belling@nsn.com> wrote:
> Hi Peter, Alexander
>
>> I do find the issue of repeating an SDP hard to pin down - after
> poking into 3261 and 3262...am I looking in the right places?
>
> Please read RFC 3261, Clause 13.2.1:
>
> In this specification, offers and answers
> =A0 can only appear in INVITE requests and responses, and ACK. =A0The usa=
ge
> =A0 of offers and answers is further restricted. =A0For the initial INVIT=
E
> =A0 transaction, the rules are:
> =A0 =A0 =A0o =A0The initial offer MUST be in either an INVITE or, if not =
there,
> =A0 =A0 =A0 =A0 in the first reliable non-failure message from the UAS ba=
ck to
> =A0 =A0 =A0 =A0 the UAC. =A0In this specification, that is the final 2xx
> =A0 =A0 =A0 =A0 response.
> =A0 =A0 =A0o =A0If the initial offer is in an INVITE, the answer MUST be =
in a
> =A0 =A0 =A0 =A0 reliable non-failure message from UAS back to UAC which i=
s
> =A0 =A0 =A0 =A0 correlated to that INVITE. =A0For this specification, tha=
t is
> =A0 =A0 =A0 =A0 only the final 2xx response to that INVITE. =A0That same =
exact
> =A0 =A0 =A0 =A0 answer MAY also be placed in any provisional responses se=
nt
> =A0 =A0 =A0 =A0 prior to the answer. =A0The UAC MUST treat the first sess=
ion
> =A0 =A0 =A0 =A0 description it receives as the answer, and MUST ignore an=
y
> =A0 =A0 =A0 =A0 session descriptions in subsequent responses to the initi=
al
> =A0 =A0 =A0 =A0 INVITE.
> =A0 =A0 =A0o =A0If the initial offer is in the first reliable non-failure
> =A0 =A0 =A0 =A0 message from the UAS back to UAC, the answer MUST be in t=
he
> =A0 =A0 =A0 =A0 acknowledgement for that message (in this specification, =
ACK
> =A0 =A0 =A0 =A0 for a 2xx response).
> =A0 =A0 =A0o =A0After having sent or received an answer to the first offe=
r, the
> =A0 =A0 =A0 =A0 UAC MAY generate subsequent offers in requests based on r=
ules
> =A0 =A0 =A0 =A0 specified for that method, but only if it has received an=
swers
> =A0 =A0 =A0 =A0 to any previous offers, and has not sent any offers to wh=
ich it
> =A0 =A0 =A0 =A0 hasn't gotten an answer.
> =A0 =A0 =A0o =A0Once the UAS has sent or received an answer to the initia=
l
> =A0 =A0 =A0 =A0 offer, it MUST NOT generate subsequent offers in any resp=
onses
> =A0 =A0 =A0 =A0 to the initial INVITE. =A0This means that a UAS based on =
this
> =A0 =A0 =A0 =A0 specification alone can never generate subsequent offers =
until
> =A0 =A0 =A0 =A0 completion of the initial transaction.
>
> Regards, Thomas

From gilad@voxisoft.com  Tue Mar 16 05:33:14 2010
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From: Gilad Shaham <gilad@voxisoft.com>
Date: Tue, 16 Mar 2010 14:32:56 +0200
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To: Peter Musgrave <peter.musgrave@magorcorp.com>
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Cc: gao.yang2@zte.com.cn, sipcore@ietf.org, "Belling, Thomas \(NSN - DE/Munich\)" <thomas.belling@nsn.com>
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
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> Which makes me think if this does happen it MUST be ignored. So that
> convinces me that it's not a good idea. I just didn't see any language
> saying it was forbidden. Perhaps it should have been. I think what
> makes this something I have to think/read carefully about is the lack
> of examples in PRACK/3262.
>
>
If you are looking for some offer answer usages in SIP you can also refer to
http://tools.ietf.org/html/draft-ietf-sipping-sip-offeranswer

In general, offer-answer has in practice many interop issues, so it's best
to keep your implementation as permissive as possible in terms of what you
expect to receive.

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<div dir="ltr"><br>
<div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Which makes me think if this does happen it MUST be ignored. So that<br>


convinces me that it&#39;s not a good idea. I just didn&#39;t see any language<br>
saying it was forbidden. Perhaps it should have been. I think what<br>
makes this something I have to think/read carefully about is the lack<br>
of examples in PRACK/3262.<br>
<br></blockquote></div><br>If you are looking for some offer answer usages in SIP you can also refer to <a href="http://tools.ietf.org/html/draft-ietf-sipping-sip-offeranswer">http://tools.ietf.org/html/draft-ietf-sipping-sip-offeranswer</a><br>

<br>In general, offer-answer has in practice many interop issues, so it&#39;s best to keep your implementation as permissive as possible in terms of what you expect to receive.<br><br></div>

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Cc: gao.yang2@zte.com.cn, sipcore@ietf.org, "Belling, Thomas \(NSN - DE/Munich\)" <thomas.belling@nsn.com>
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
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Peter Musgrave wrote:
> Hi Thomas,
> 
> Thanks for the reference. I was reading that part of the spec. but
> could not make my mind up about whether it forbids sending a copy of
> the SDP in the 200 (if a reliable response had been provided). The
> closest I came was the statement:
> 
> The UAC MUST treat the first session
>         description it receives as the answer, and MUST ignore any
>         session descriptions in subsequent responses to the initial
>         INVITE.

> Which makes me think if this does happen it MUST be ignored. So that
> convinces me that it's not a good idea. I just didn't see any language
> saying it was forbidden. Perhaps it should have been. I think what
> makes this something I have to think/read carefully about is the lack
> of examples in PRACK/3262.

The language is not very clear, and can only be discerned in total by 
integrating across several RFCs and making various inferences. This led 
to a very long email discussion/debate on the subject several years ago. 
That in turn led to the writing of the offeranswer draft, which attempts 
to capture all that. We have concluded that it is all just 
clarification, rather than normative changes.

> Apologies to all if I am belabouring the point here.

Don't feel too bad - it comes up every few months. If you haven't, 
please read the offeranswer draft:

http://www.ietf.org/id/draft-ietf-sipping-sip-offeranswer-12.txt

	Thanks,
	Paul

> Peter
> 
> On Tue, Mar 16, 2010 at 8:04 AM, Belling, Thomas (NSN - DE/Munich)
> <thomas.belling@nsn.com> wrote:
>> Hi Peter, Alexander
>>
>>> I do find the issue of repeating an SDP hard to pin down - after
>> poking into 3261 and 3262...am I looking in the right places?
>>
>> Please read RFC 3261, Clause 13.2.1:
>>
>> In this specification, offers and answers
>>   can only appear in INVITE requests and responses, and ACK.  The usage
>>   of offers and answers is further restricted.  For the initial INVITE
>>   transaction, the rules are:
>>      o  The initial offer MUST be in either an INVITE or, if not there,
>>         in the first reliable non-failure message from the UAS back to
>>         the UAC.  In this specification, that is the final 2xx
>>         response.
>>      o  If the initial offer is in an INVITE, the answer MUST be in a
>>         reliable non-failure message from UAS back to UAC which is
>>         correlated to that INVITE.  For this specification, that is
>>         only the final 2xx response to that INVITE.  That same exact
>>         answer MAY also be placed in any provisional responses sent
>>         prior to the answer.  The UAC MUST treat the first session
>>         description it receives as the answer, and MUST ignore any
>>         session descriptions in subsequent responses to the initial
>>         INVITE.
>>      o  If the initial offer is in the first reliable non-failure
>>         message from the UAS back to UAC, the answer MUST be in the
>>         acknowledgement for that message (in this specification, ACK
>>         for a 2xx response).
>>      o  After having sent or received an answer to the first offer, the
>>         UAC MAY generate subsequent offers in requests based on rules
>>         specified for that method, but only if it has received answers
>>         to any previous offers, and has not sent any offers to which it
>>         hasn't gotten an answer.
>>      o  Once the UAS has sent or received an answer to the initial
>>         offer, it MUST NOT generate subsequent offers in any responses
>>         to the initial INVITE.  This means that a UAS based on this
>>         specification alone can never generate subsequent offers until
>>         completion of the initial transaction.
>>
>> Regards, Thomas
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Cc: gao.yang2@zte.com.cn, sipcore@ietf.org, "Belling, Thomas \(NSN - DE/Munich\)" <thomas.belling@nsn.com>
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03
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Paul Kyzivat wrote:
> 
> ... We have concluded that it is all just 
> clarification, rather than normative changes.

I should have said more...

We decided what ambiguities required normative action should be dealt 
with separately from the o/a draft. draft-ietf-sipcore-reinvite is a 
normative draft doing just that.

	Thanks,
	Paul

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Begin forwarded message:

> From: Robert Sparks <rjsparks@nostrum.com>
> Date: March 16, 2010 11:15:36 AM CDT
> To: rai@ietf.org
> Subject: SIPit26 registration is open
>=20
> SIPit 26 will be held in Kista - the telecom valley outside Stockholm, =
Sweden on May 17-21, 2010. The event is hosted by Edvina and TANDBERG =
and sponsored by Ingate, Intertex and .se.
>=20
> See <http://www.sipit.net> and <http://sipit.edvina.se> for details =
and to register.
>=20
> RjS


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><br><div>Begin forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>From: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;">Robert Sparks =
&lt;<a =
href=3D"mailto:rjsparks@nostrum.com">rjsparks@nostrum.com</a>&gt;<br></spa=
n></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>Date: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;">March 16, 2010 =
11:15:36 AM CDT<br></span></div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>To: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><a =
href=3D"mailto:rai@ietf.org">rai@ietf.org</a><br></span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>Subject: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><b>SIPit26 =
registration is open</b><br></span></div><br><div>SIPit 26 will be held =
in Kista - the telecom valley outside Stockholm, Sweden on May 17-21, =
2010. The event is hosted by Edvina and TANDBERG and sponsored by =
Ingate, Intertex and .se.<br><br>See &lt;<a =
href=3D"http://www.sipit.net">http://www.sipit.net</a>&gt; and &lt;<a =
href=3D"http://sipit.edvina.se">http://sipit.edvina.se</a>&gt; for =
details and to =
register.<br><br>RjS</div></blockquote></div><br></body></html>=

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On Mar 16, 2010, at 5:18 AM, Milinski, Alexander (NSN - DE/Munich)  
wrote:

> Hi Paul, Dean,
>
> now I am confused (though I got used to the square root of -1 being  
> i).
>
> Which RFC explains this repetition?

None, really. We didn't do it like I wanted it.


> Moreover, how is the repeated answer distingusished from a new offer
> (the previous o/a has been completed before),
> which should be answered in the ACK?

The principle of repetition would say that you answer it again in the  
ACK, with an identical answer to that already sent. In other words,  
one is simply confirming in the "final" response that which was  
previously sent "provisionally". Provisional means "temporary until  
confirmed in a separate exchange", which means to me that there ought  
to be a final confirmation of the O/A executed in the provisional  
messages.

Note that I also argued that either we should have no O/A in reliable  
provisional, or that reliable provisional O/A should become mandatory  
and exclusive;  that is, that the "regular" offer/answer be discarded  
in favor of doing it in a reliable provisional exchange. This has some  
benefits when trying to grapple with the early media / early session  
problem in conjunction with PSTN gateways, and it's also useful for  
dealing with the folks who think that billing begins on 200 OK, who  
pretty much just annoy me.

--
Dean



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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 16, 2010, =
at 5:18 AM, Milinski, Alexander (NSN - DE/Munich) wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div>Hi =
Paul, Dean,<br><br>now I am confused (though I got used to the square =
root of -1 being i).<br><br>Which RFC explains this repetition?<font =
class=3D"Apple-style-span" color=3D"#000000"><font =
class=3D"Apple-style-span" =
color=3D"#144FAE"><br></font></font></div></blockquote><div><br></div>None=
, really. We didn't do it like I wanted =
it.</div><div><br><br><blockquote type=3D"cite"><div>Moreover, how is =
the repeated answer distingusished from a new offer<br>(the previous o/a =
has been completed before),<br>which should be answered in the =
ACK?<br></div></blockquote><br></div><div>The principle of repetition =
would say that you answer it again in the ACK, with an identical answer =
to that already sent. In other words, one is simply confirming in the =
"final" response that which was previously sent "provisionally". =
Provisional means "temporary until confirmed in a separate exchange", =
which means to me that there ought to be a final confirmation of the O/A =
executed in the provisional messages.</div><div><br></div><div>Note that =
I also argued that either we should have no O/A in reliable provisional, =
or that reliable provisional O/A should become mandatory and exclusive; =
&nbsp;that is, that the "regular" offer/answer be discarded in favor of =
doing it in a reliable provisional exchange. This has some benefits when =
trying to grapple with the early media / early session problem in =
conjunction with PSTN gateways, and it's also useful for dealing with =
the folks who think that billing begins on 200 OK, who pretty much just =
annoy =
me.</div><div><br></div><div>--</div><div>Dean</div><div><br></div><br></b=
ody></html>=

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To: Brett Tate <brett@broadsoft.com>, Peter Musgrave <peter.musgrave@magorcorp.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Bret=
t Tate [brett@broadsoft.com]
Sent: Sunday, March 14, 2010 2:16 PM
To: Peter Musgrave; sipcore@ietf.org
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-03

It is deliberate; see RFC 3261 concerning when answer SDP needs to be withi=
n INVITE=92s 2xx response.  The 18x=92s were sent reliable; thus completing=
 the offer/answer exchange.  The following is mention above both figures: =
=93PRACK transactions are not shown for clarity=94.
_________________________________________

It might avoid confusion if that note was extended with "Note that in some =
cases, the SDP answer is carried in a PRACK which is not shown."

Dale

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From dean.willis@softarmor.com  Tue Mar 16 22:43:22 2010
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On Mar 16, 2010, at 11:41 PM, gao.yang2@zte.com.cn wrote:

>
> [Gao] Hi Dean,
> I think as the UAS send the Answer in one reliable response, there's  
> no need to copy the same Answer as the confirmation in the final  
> respone.

That was also the consensus of the working group.

> There are UASs do not support 100rel, so they send the SDP(no as  
> Answer) in 1xx for early media usage. This is the case UAS should  
> copy the same SDP as Answer in the final respone.

So, a reliably sent answer is a "final Answer", and an unreliably-sent  
answer is just being hopeful.

>
> And send SDP in another reliable response after the initial O/A is  
> violation of current RFC3261's definition.

That's actually debatable; as quoted earlier in this thread, RFC 3261  
is not completely clear on whether a copy of the same answer is a  
different answer. Certainly, CHANGING the answer is forbidden, but  
repeating it might not be. That's pat of why we have further drafts to  
clarify this.

> I do not prefer there rules to be re-defined except for strong  
> reasons.

I concur. Of course, one of the best reasons for more definition is  
because the rule wasn't well defined to start with.

>
> >
> > Note that I also argued that either we should have no O/A in
> > reliable provisional, or that reliable provisional O/A should become
> > mandatory and exclusive;  that is, that the "regular" offer/answer
> > be discarded in favor of doing it in a reliable provisional
> > exchange. This has some benefits when trying to grapple with the
> > early media / early session problem in conjunction with PSTN
> > gateways, and it's also useful for dealing with the folks who think
> > that billing begins on 200 OK, who pretty much just annoy me.
>
> [Gao] I think the early media problem in conjunction with PSTN  
> gateways can be solved by using p-early-media and its policy.

Not without changing the gateway billing model, I think. But I look  
forward to somebody writing an I-D on how this might work.

--
Dean



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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 16, 2010, =
at 11:41 PM, <a =
href=3D"mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a> =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><br><tt><font size=3D"2">[Gao] Hi Dean,</font></tt> =
<br><tt><font size=3D"2">I think as the UAS send the Answer in one =
reliable response, there's no need to copy the same Answer as the =
confirmation in the final respone.</font></tt> =
<br></blockquote><div><br></div><div>That was also the consensus of the =
working group.</div><br><blockquote type=3D"cite"><tt><font =
size=3D"2">There are UASs do not support 100rel, so they send the SDP(no =
as Answer) in 1xx for early media usage. This is the case UAS should =
copy the same SDP as Answer in the final =
respone.</font></tt></blockquote><div><br></div>So, a reliably sent =
answer is a "final Answer", and an unreliably-sent answer is just being =
hopeful.</div><div><br><blockquote type=3D"cite"><tt><font size=3D"2"> =
</font></tt> <br><tt><font size=3D"2">And send SDP in another reliable =
response after the initial O/A is violation of current RFC3261's =
definition. </font></tt></blockquote><div><br></div><div>That's actually =
debatable; as quoted earlier in this thread, RFC 3261 is not completely =
clear on whether a copy of the same answer is a different answer. =
Certainly, CHANGING the answer is forbidden, but repeating it might not =
be. That's pat of why we have further drafts to clarify =
this.</div><br><blockquote type=3D"cite"><tt><font size=3D"2">I do not =
prefer there rules to be re-defined except for strong =
reasons.</font></tt> <br></blockquote><div><br></div><div>I concur. Of =
course, one of the best reasons for more definition is because the rule =
wasn't well defined to start with.</div><div><br></div><blockquote =
type=3D"cite"> <br><tt><font size=3D"2">&gt; <br> &gt; Note that I also =
argued that either we should have no O/A in <br> &gt; reliable =
provisional, or that reliable provisional O/A should become<br> &gt; =
mandatory and exclusive; &nbsp;that is, that the "regular" offer/answer =
<br> &gt; be discarded in favor of doing it in a reliable provisional =
<br> &gt; exchange. This has some benefits when trying to grapple with =
the <br> &gt; early media / early session problem in conjunction with =
PSTN <br> &gt; gateways, and it's also useful for dealing with the folks =
who think <br> &gt; that billing begins on 200 OK, who pretty much just =
annoy me.</font></tt> <br> <br><tt><font size=3D"2">[Gao] I think the =
early media problem in conjunction with PSTN gateways can be solved by =
using p-early-media and its policy.</font></tt> =
<br></blockquote></div><br><div>Not without changing the gateway billing =
model, I think. But I look forward to somebody writing an I-D on how =
this might =
work.</div><div><br></div><div>--</div><div>Dean</div><div><br></div><div>=
<br></div></body></html>=

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Cc: rai@ietf.org
Subject: [sipcore] Reminder Last Message: SIP Forum Luncheon at IETF 77 Monday 11:30 13:00
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The SIP Forum is going to hold a open meeting for the RAI community during
IETF 77. The purpose of this meeting is remind the SIP community of what the
SIP Forum is doing and discuss relevant technical issues that relate to IETF
WG's. The agenda for this luncheon will be to review current activities in
the SIP Forum and seek additional technical input from the RAI community on
mutual concerns.

 

As many of you know the SIP Forum is the principal sponsor of the SIPit
interoperability events and currently has 4 Task groups operating. 

 

BTW registration for SIPit 26 will be open soon.

 

Our principal technical initiatives are:

 

1.      SIP Connect 1.1 which is developing an enhanced  profile of IETF
Standards for PBX to SSP interoperability. As part of that work the IETF
MARTINI WG was formed to redefine the so called Implicit registration
procedure in SIP.

2.      FAX over SIP is investigating the interoperability of SIP in T.38
networks with a goal of possible recommendations to the relevant SDO's on
how fax can work better.

3.      Client User Agent Configuration which has a pending Recommendation
on how CUA devices could auto streamline the locating, retrieving and
maintaining of configuration-related information for SIP User Agents

4.      SIP in the SmartGrid. This is a Special Interest Group that is
evaluating the appropriateness of using SIP as a protocol for a number of
Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) deployments.

 

Information on the SIP Forum is available at 

 

http//www.sipforum.org

 

If you are interested in attending this event we have set up a Doodle Poll
to indicate your interest. Seating is somewhat limited ( 150 Maximum) lunch
will be provided.

 

Location .. Monday 11:30 13:00 the Laguna room and its located on the 4th
floor of the Anaheim Hilton

The link to the poll is: 

http://www.doodle.com/hde57nbpcdykzqfz 

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype: rshockey101
LinkedIn : http://www.linkedin.com/in/rshockey101

 


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<div class=3DSection1>

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

<p class=3DMsoNormal>The SIP Forum is going to hold a open meeting for =
the RAI
community during IETF 77. The purpose of this meeting is remind the SIP
community of what the SIP Forum is doing and discuss relevant technical =
issues
that relate to IETF WG&#8217;s. The agenda for this luncheon will be to =
review
current activities in the SIP Forum and seek additional technical input =
from
the RAI community on mutual concerns.<o:p></o:p></p>

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

<p class=3DMsoNormal>As many of you know the SIP Forum is the principal =
sponsor
of the SIPit interoperability events and currently has 4 Task groups =
operating.
<o:p></o:p></p>

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

<p class=3DMsoNormal>BTW registration for SIPit 26 will be open =
soon.<o:p></o:p></p>

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

<p class=3DMsoNormal>Our principal technical initiatives =
are:<o:p></o:p></p>

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

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP Connect 1.1 which is developing an enhanced
&nbsp;profile of IETF Standards for PBX to SSP interoperability. As part =
of that
work the IETF MARTINI WG was formed to redefine the so called Implicit
registration procedure in SIP.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>FAX over SIP is investigating the =
interoperability of
SIP in T.38 networks with a goal of possible recommendations to the =
relevant
SDO&#8217;s on how fax can work better.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Client User Agent Configuration which has a =
pending
Recommendation on how CUA devices could auto streamline the locating,
retrieving and maintaining of configuration-related information for SIP =
User
Agents<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP in the SmartGrid. This is a Special Interest =
Group
that is evaluating the appropriateness of using SIP as a protocol for a =
number
of Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) =
deployments.<o:p></o:p></p>

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

<p class=3DMsoNormal>Information on the SIP Forum is available at =
<o:p></o:p></p>

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

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Times =
New Roman","serif"'>http//www.sipforum.org</span><o:p></o:p></p>

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

<p class=3DMsoNormal>If you are interested in attending this event we =
have set up
a Doodle Poll to indicate your interest. Seating is somewhat limited ( =
150
Maximum) lunch will be provided.<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>Location .. </span><span
style=3D'font-size:10.0pt'>Monday 11:30 13:00 </span><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>the Laguna room and its =
located on
the 4th floor of the Anaheim Hilton</span><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>The link to the poll =
is:</span><span
style=3D'font-size:10.0pt'> <br>
<br>
<a href=3D"http://www.doodle.com/hde57nbpcdykzqfz" =
target=3D"_blank">http://www.doodle.com/hde57nbpcdykzqfz</a>
</span><o:p></o:p></p>

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

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>
Shockey Consulting<br>
Chairman of the Board of Directors SIP Forum<br>
PSTN Mobile: +1 703.593.2683<br>
&lt;<a =
href=3D"mailto:richard(at)shockey.us">mailto:richard(at)shockey.us</a>&gt=
;<br>
skype: rshockey101<br>
LinkedIn : <a =
href=3D"http://www.linkedin.com/in/rshockey101">http://www.linkedin.com/i=
n/rshockey101</a></span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

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

</div>

</body>

</html>

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Cc: sipcore@ietf.org, Avasarala Ranjit-A20990 <ranjit@motorola.com>
Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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Its worth looking to see if the document text is as explicit as people =
think it should be on this point.

Also, I remember some discussion about whether it's ok to reuse event =
package names? (Is it ok for someone to create a "dialog" event package =
for instance?)
The draft is silent on this at the moment - is that on purpose?

RjS

On Mar 13, 2010, at 4:53 PM, Adam Roach wrote:

> Please don't cross-post to dispatch and sipcore.
>=20
> On 3/12/10 23:26, Mar 12, Avasarala Ranjit-A20990 wrote:
>>=20
>> Are there any registered INFO packages
>>=20
>=20
> Not yet.
>=20
>> and can the regular SIP event packages based on RFC 3265 be used with =
INFO mechanism?
>>=20
>=20
> Absolutely not. No. Never. Under no circumstances. Nein, =D0=9D=D0=B5=D1=
=82, =E0=A4=A8=E0=A4=B9=E0=A5=80=E0=A4=82, Non, Nej. They are =
_completely_ different things.
>=20
> Really, if this is somehow unclear in the INFO draft, we need to pull =
it from publication request and work on it some more. It would be the =
worst possible outcome if implementors were led to believe that 3265 =
packages could be used with INFO. Where did you get the impression that =
this might be okay?
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Its =
worth looking to see if the document text is as explicit as people think =
it should be on this point.<div><br></div><div>Also, I remember some =
discussion about whether it's ok to reuse event package names? (Is it ok =
for someone to create a "dialog" event package for =
instance?)</div><div>The draft is silent on this at the moment - is that =
on purpose?</div><div><br></div><div>RjS<br><div><br><div><div>On Mar =
13, 2010, at 4:53 PM, Adam Roach wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite">
<div bgcolor=3D"#ffffff" text=3D"#000000">
<span id=3D"result_box" class=3D"short_text"><span =
style=3D"background-color: rgb(255, 255, 255);" title=3D"No">Please =
don't
cross-post to dispatch and sipcore.</span></span><br>
<br>
On 3/12/10 23:26, Mar 12, Avasarala Ranjit-A20990 wrote:
<blockquote =
cite=3D"mid:750BBC72E178114F9DC4872EBFF29A5B0A3BE70F@ZMY16EXM66.ds.mot.com=
" type=3D"cite">
 =20
 =20
 =20
<!-- Converted from text/rtf format --><p><font face=3D"Arial" =
size=3D"2">Are there any registered INFO packages</font></p>
</blockquote>
<br>
Not yet.<br>
<br>
<blockquote =
cite=3D"mid:750BBC72E178114F9DC4872EBFF29A5B0A3BE70F@ZMY16EXM66.ds.mot.com=
" type=3D"cite"><p><font face=3D"Arial" size=3D"2"> and can the regular =
SIP event
packages based on RFC 3265 be used with INFO mechanism?</font></p>
</blockquote>
<br>
Absolutely not. No. Never. Under no circumstances. Nein,

<span id=3D"result_box" class=3D"short_text"><span =
style=3D"background-color: rgb(255, 255, 255);" =
title=3D"No">=D0=9D=D0=B5=D1=82</span></span>,

<span id=3D"result_box" class=3D"short_text"><span =
style=3D"background-color: rgb(255, 255, 255);" =
title=3D"No">=E0=A4=A8=E0=A4=B9=E0=A5=80=E0=A4=82</span></span>,
Non,

<span id=3D"result_box" class=3D"short_text"><span =
style=3D"background-color: rgb(255, 255, 255);" title=3D"No">Nej. They =
are
_completely_ different things.<br>
<br>
Really, if this is somehow unclear in the INFO draft, we need to pull
it from publication request and work on it some more. It would be the
worst possible outcome if implementors were led to believe that 3265
packages could be used with INFO. Where did you get the impression that
this might be okay?<br>
</span></span><br>
/a<br>
</div>

_______________________________________________<br>sipcore mailing =
list<br><a =
href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>https://www.ietf.=
org/mailman/listinfo/sipcore<br></blockquote></div><br></div></div></body>=
</html>=

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From rjsparks@nostrum.com  Fri Mar 19 11:52:18 2010
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Subject: Re: [sipcore] 3265 Open Issue: 202 Response Code
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The notes were pointing to why we ended up with 202 in the first place. =
We introduced it when we figured out QAUTH wouldn't work because of the =
fixed non-INVITE transaction length. Shortly after, because of HERFP, we =
realized that we had to require the immediate NOTIFY from each branch. =
When we added the Subscription-State header to the NOTIFY (including =
it's potential value of "pending"), all utility for 202 vs 200 went =
away.

The problem we are trying to solve is that we have interoperability =
issues with implementations trying to treat 202 and 200 as meaning =
something
different to the application that receives them. Because of the issues =
pointed to above, they _can't_ and whenever I've found someone trying to
do something differently on receipt of a 202, they really wanted to be =
using Subscription-State from the NOTIFY anyhow.

This has also been a honeypot for inducing bad implementation. I've seen =
early implementations that would terminate a dialog if they
got a 202 to the SUBSCRIBE and a NOTIFY with a state of active. I've =
seen people try to build "conformance tests" that would complain
if that happened. Both are not helpful behaviors.

This is a rough spot (even if it is a very small one). We have a chance =
to clean it up.

RjS

On Mar 2, 2010, at 8:01 PM, DRAGE, Keith (Keith) wrote:

> Rather than change for change sake, what problem are we actually =
trying to fix here. The meeting notes refer to HERFP but that is hardly =
enlightening.
>=20
> regards
>=20
> Keith=20
>=20
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org=20
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
>> Sent: Tuesday, March 02, 2010 11:23 PM
>> To: SIPCORE
>> Subject: [sipcore] 3265 Open Issue: 202 Response Code
>>=20
>> [as a participant]
>>=20
>> In Stockholm, we had a discussion of the open issues in the=20
>> 3265bis One of the topics that we discussed was the utility=20
>> of the 202 response code in RFC 3265. A summary of the=20
>> conversation can be found in the minutes:
>>=20
>> http://www.ietf.org/proceedings/75/minutes/sipcore.html#Open%2
>> 0Issue:%20The%20Fate%20of%20202
>>=20
>> There was no clear agreement in that conversation, and=20
>> several participants seemed to hold very nuanced opinions on=20
>> the topic.
>>=20
>> In order to drive conversation on the topic, I will propose=20
>> the following text as a starting point.
>>=20
>> The key change is in Section 8.3.1, which I propose to be updated as
>> follows:
>>=20
>>    For historical purposes, the 202 (Accepted) response code is
>>    added to the "Success" header field definition.
>>=20
>>    This document does not specify the use of the 202 response code
>>    in conjunction with the SUBSCRIBE or NOTIFY methods. Previous
>>    versions of the SIP Events Framework assigned specific semantics
>>    to the 202 response code. Implementations conformant with this
>>    specification MUST treat an incoming 202 response as identical
>>    to a 200 response, and MUST NOT generate 202 response codes to
>>    SUBSCRIBE or NOTIFY messages.
>>=20
>> That is the change in policy that I am proposing. Several=20
>> other changes in the document follow from this change; I list=20
>> them below.
>>=20
>> In section 4.1.2.1, the third paragraph will be truncated=20
>> after the word "immedately." In its revised form, it will simply =
read:
>>=20
>>    This SUBSCRIBE request will be confirmed with a final response.
>>    200- class responses indicate that the subscription has been
>>    accepted, and that a NOTIFY will be sent immediately.
>>=20
>> In section 4.2.1.1, the 6th and 7th paragraphs (both starting=20
>> wht "If the notifier...") will be combined into a single=20
>> paragraph reading:
>>=20
>>    Once the notifier determines that it has enough information to
>>    create the subscription (i.e., it understands the event package,
>>    the subscription pertains to a known resource, and there are no
>>    other barriers to creating the subscription), it creates the
>>    subscription and a dialog, and returns a 200 (OK) response.
>>=20
>> In section 4.2.1.3, the fifth paragraph will replace "202 Accept"
>> with "200 OK", with the final result being:
>>=20
>>    If the notifier owner is interactively queried to determine
>>    whether a subscription is allowed, a 200 (OK) response is returned
>>    immediately.  Note that a NOTIFY message is still formed and
>>    sent under these circumstances, as described in the previous
>>    section.
>>=20
>> In section 5.4.6, the final sentance of the final paragraph=20
>> will be removed, resulting in the following text:
>>=20
>>    Information in this section includes details of how to=20
>> authenticate
>>    subscribers and authorization issues for the package.
>>=20
>> The first paragraph of section 6.2 will be updated to read:
>>=20
>>    The mere act of returning certain 4xx and 6xx responses to
>>    SUBSCRIBE requests may, under certain circumstances, create
>>    privacy concerns by revealing sensitive policy information.  In
>>    these cases, the notifier SHOULD always return a 200 (OK)=20
>> response.
>>    While the subsequent NOTIFY message may not convey true state,
>>    it MUST appear to contain a potentially correct piece of data
>>    from the point of view of the subscriber, indistinguishable from
>>    a valid response.  Information about whether a user is authorized
>>    to subscribe to the requested state is never conveyed back to
>>    the original user under these circumstances.
>>=20
>>=20
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Dale Worley" <dworley@avaya.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] 3265 Open Issue: 202 Response Code
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On Tue, 2010-03-02 at 17:22 -0600, Adam Roach wrote:
> The key change is in Section 8.3.1, which I propose to be updated as 
> follows:
> 
>     For historical purposes, the 202 (Accepted) response code is
>     added to the "Success" header field definition.
> 
>     This document does not specify the use of the 202 response code
>     in conjunction with the SUBSCRIBE or NOTIFY methods. Previous
>     versions of the SIP Events Framework assigned specific semantics
>     to the 202 response code. Implementations conformant with this
>     specification MUST treat an incoming 202 response as identical
>     to a 200 response, and MUST NOT generate 202 response codes to
>     SUBSCRIBE or NOTIFY messages.
> 
> That is the change in policy that I am proposing. Several other changes 
> in the document follow from this change; I list them below.

My understanding is that the SIP convention is that all 2xx responses
are considered successful and must be accepted as such by all SIP
elements.  If there is further semantics attached to a particular 2xx
response, an element that knows those semantics may act on them.  The
above text suggests that a SIP element might legitimately not treat
certain 2xx responses as successful in certain circumstances.  I would
prefer something like this:

        Previous versions of the SIP Events Framework assigned specific
        semantics to the 202 response code as a response to a SUBSCRIBE
        request, viz., that the SUBSCRIBE request was understood but the
        subscription had not yet been approved by the notifier's policy.
        This version merges this meaning into the semantics of the 200
        response, and hence conforming implementations SHOULD respond to
        a SUBSCRIBE request with a 200 response rather than a 202
        response in this case (like all other success cases).  As is
        true in all circumstances where a UA receives a 2xx response
        that does not have specifically defined semantics, a conforming
        implementation that receives a 202 response MUST treat it as
        identical to a 200 response.

You can change that "SHOULD" to "MUST", but I like it as SHOULD, as
there might be special circumstances that dictate some other 2xx
response.  In any case, the subscriber will accept any 2xx as
successful.

Dale



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Date: Mon, 22 Mar 2010 17:28:38 +0100
Thread-Topic: [sipcore] Query on draft-ietf-sipcore-info-events-07
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From adam@nostrum.com  Mon Mar 22 10:54:37 2010
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Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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On 3/22/10 9:28 AM, Christer Holmberg wrote:
>
> Hi,
>
>    
>> Also, I remember some discussion about whether it's ok to reuse event package names? (Is it
>> ok for someone to create a "dialog" event package for instance?)
>> The draft is silent on this at the moment - is that on purpose?
>>      
> I see no reason to explicitly forbid it, since we're dealing with different namespaces.
>    

I agree.

In fact, I believe that doing anything that implies that they're drawing 
from the same namespace is actually harmful: doing so can convey the 
incorrect impression that there might be some way to use an Event 
Package in INFO, or an INFO package in SUBSCRIBE.

/a

From christer.holmberg@ericsson.com  Mon Mar 22 23:36:02 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Tue, 23 Mar 2010 07:36:13 +0100
Thread-Topic: SUBSCRIBE 3xx - can it solve issues raised in 3265bis?  [was INVITE 3xx and early media]
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Subject: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
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Hi,

Would it be possible to use the 3xx mechanism in order to solve the forking=
 related problems with SUB/NOT (as described in 3265bis)?

The forking proxy would return a 3xx response to the SUBSCRIBE request, and=
 the UAC can then send individual SUBSCRIBE requests to each UAS?

Regards,

Christer=

From adam@nostrum.com  Tue Mar 23 00:13:50 2010
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References: <FF84A09F50A6DC48ACB6714F4666CC745C10EE415F@ESESSCMS0354.eemea.ericsson.se>	<4B918DED.6090108@cisco.com>	<FF84A09F50A6DC48ACB6714F4666CC745C1102CCD1@ESESSCMS0354.eemea.ericsson.se>	<4B919413.60506@cisco.com>	<FF84A09F50A6DC48ACB6714F4666CC745C10EE4166@ESESSCMS0354.eemea.ericsson.se>	<4B93D8A6.1040105@cisco.com> <001a01cabe3e$e7033f80$b509be80$@com>	<4B950BC0.5090601@cisco.com>	<0BD9B443-C89F-4D16-819D-C35D74BAFD1B@softarmor.com>	<4B9531DA.7050002@cisco.com>	<f97bcb1a1003080947mc41b645sbd8c3816e61ada88@mail.gmail.com>	<A648C787-10F4-4895-8F13-61FE2CFD0EC2@softarmor.com> <FF84A09F50A6DC48ACB6714F4666CC745E21B5A8C4@ESESSCMS0354.eemea.ericsson.se>
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Subject: Re: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
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The only real way you could deploy this would be with a 'proxy-require' 
option tag that would have associated semantics that forbid proxies from 
forking. I don't think that's very tractable at this point.

/a

On 3/22/10 11:36 PM, Christer Holmberg wrote:
> Hi,
>
> Would it be possible to use the 3xx mechanism in order to solve the forking related problems with SUB/NOT (as described in 3265bis)?
>
> The forking proxy would return a 3xx response to the SUBSCRIBE request, and the UAC can then send individual SUBSCRIBE requests to each UAS?
>
> Regards,
>
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>    


From christer.holmberg@ericsson.com  Tue Mar 23 00:29:04 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Adam Roach' <adam@nostrum.com>
Date: Tue, 23 Mar 2010 08:28:56 +0100
Thread-Topic: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
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References: <FF84A09F50A6DC48ACB6714F4666CC745C10EE415F@ESESSCMS0354.eemea.ericsson.se> <4B918DED.6090108@cisco.com> <FF84A09F50A6DC48ACB6714F4666CC745C1102CCD1@ESESSCMS0354.eemea.ericsson.se> <4B919413.60506@cisco.com> <FF84A09F50A6DC48ACB6714F4666CC745C10EE4166@ESESSCMS0354.eemea.ericsson.se> <4B93D8A6.1040105@cisco.com> <001a01cabe3e$e7033f80$b509be80$@com> <4B950BC0.5090601@cisco.com> <0BD9B443-C89F-4D16-819D-C35D74BAFD1B@softarmor.com> <4B9531DA.7050002@cisco.com> <f97bcb1a1003080947mc41b645sbd8c3816e61ada88@mail.gmail.com> <A648C787-10F4-4895-8F13-61FE2CFD0EC2@softarmor.com> <FF84A09F50A6DC48ACB6714F4666CC745E21B5A8C4@ESESSCMS0354.eemea.ericsson.se> <4BA86A35.90707@nostrum.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
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Hi,

I don't think we can "forbid" the proxy from forking, and if it does we hav=
e to live with the consequences.

The proposal for OPTIONS 3xx (and, I guess it would be the same for INVITE =
3xx?) is to use the Request-Disposition to header to indicate a wish to red=
irect. But, then if of course depends on whether support for R-D is impleme=
nted in proxies or not. If not implemented things will work as today - with=
 all the consequences.

Note, however, that I am not suggesting to replace 3265bis with SUB 3xx - I=
 was only asking whether it could be used to solve the problems described.

Regards,

Christer=20

-----Original Message-----
From: Adam Roach [mailto:adam@nostrum.com]=20
Sent: 23. maaliskuuta 2010 9:14
To: Christer Holmberg
Cc: SIPCORE
Subject: Re: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bi=
s? [was INVITE 3xx and early media]

The only real way you could deploy this would be with a 'proxy-require'=20
option tag that would have associated semantics that forbid proxies from fo=
rking. I don't think that's very tractable at this point.

/a

On 3/22/10 11:36 PM, Christer Holmberg wrote:
> Hi,
>
> Would it be possible to use the 3xx mechanism in order to solve the forki=
ng related problems with SUB/NOT (as described in 3265bis)?
>
> The forking proxy would return a 3xx response to the SUBSCRIBE request, a=
nd the UAC can then send individual SUBSCRIBE requests to each UAS?
>
> Regards,
>
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>   =20


From pkyzivat@cisco.com  Tue Mar 23 07:46:53 2010
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
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Yes, of course its possible to use R-D to request redirect, as long as 
you are satisfied with the possibility that it won't be honored.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> 
> I don't think we can "forbid" the proxy from forking, and if it does we have to live with the consequences.
> 
> The proposal for OPTIONS 3xx (and, I guess it would be the same for INVITE 3xx?) is to use the Request-Disposition to header to indicate a wish to redirect. But, then if of course depends on whether support for R-D is implemented in proxies or not. If not implemented things will work as today - with all the consequences.
> 
> Note, however, that I am not suggesting to replace 3265bis with SUB 3xx - I was only asking whether it could be used to solve the problems described.
> 
> Regards,
> 
> Christer 
> 
> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com] 
> Sent: 23. maaliskuuta 2010 9:14
> To: Christer Holmberg
> Cc: SIPCORE
> Subject: Re: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
> 
> The only real way you could deploy this would be with a 'proxy-require' 
> option tag that would have associated semantics that forbid proxies from forking. I don't think that's very tractable at this point.
> 
> /a
> 
> On 3/22/10 11:36 PM, Christer Holmberg wrote:
>> Hi,
>>
>> Would it be possible to use the 3xx mechanism in order to solve the forking related problems with SUB/NOT (as described in 3265bis)?
>>
>> The forking proxy would return a 3xx response to the SUBSCRIBE request, and the UAC can then send individual SUBSCRIBE requests to each UAS?
>>
>> Regards,
>>
>> Christer
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>    
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Adam Roach" <adam@nostrum.com>, "Christer Holmberg" <christer.holmberg@ericsson.com>
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Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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Hi

Now since the I-D does not specify any particular format or template for
an INFO package, there is a very likely chance that someone could
propose an INFO package as a XML package which is quite similar to an
3265 based event package. Now how can someone prevent use of this
package with SUBSCRIBE / NOTIFY?

Regards
Ranjit

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of Adam Roach
Sent: Monday, March 22, 2010 11:25 PM
To: Christer Holmberg
Cc: sipcore@ietf.org; Avasarala Ranjit-A20990
Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07

On 3/22/10 9:28 AM, Christer Holmberg wrote:
>
> Hi,
>
>   =20
>> Also, I remember some discussion about whether it's ok to reuse event

>> package names? (Is it ok for someone to create a "dialog" event=20
>> package for instance?) The draft is silent on this at the moment - is
that on purpose?
>>     =20
> I see no reason to explicitly forbid it, since we're dealing with
different namespaces.
>   =20

I agree.

In fact, I believe that doing anything that implies that they're drawing
from the same namespace is actually harmful: doing so can convey the
incorrect impression that there might be some way to use an Event
Package in INFO, or an INFO package in SUBSCRIBE.

/a
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

From christer.holmberg@ericsson.com  Tue Mar 23 09:51:15 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Avasarala Ranjit-A20990' <ranjit@motorola.com>, Adam Roach <adam@nostrum.com>
Date: Tue, 23 Mar 2010 17:51:13 +0100
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Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
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Hi,=20

>Now since the I-D does not specify any particular format or template for a=
n INFO package,=20
>there is a very likely chance that someone could propose an INFO package a=
s a XML package=20
>which is quite similar to an
>3265 based event package. Now how can someone prevent use of this package =
with SUBSCRIBE /=20
>NOTIFY?

Technically you can use the same MIME message bodies both for INFO packages=
 and event packages, but if you want to use the message body as part of an =
event package you need to register that event package.

Regards,

Christer


-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: Monday, March 22, 2010 11:25 PM
To: Christer Holmberg
Cc: sipcore@ietf.org; Avasarala Ranjit-A20990
Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07

On 3/22/10 9:28 AM, Christer Holmberg wrote:
>
> Hi,
>
>   =20
>> Also, I remember some discussion about whether it's ok to reuse event

>> package names? (Is it ok for someone to create a "dialog" event=20
>> package for instance?) The draft is silent on this at the moment - is
that on purpose?
>>     =20
> I see no reason to explicitly forbid it, since we're dealing with
different namespaces.
>   =20

I agree.

In fact, I believe that doing anything that implies that they're drawing fr=
om the same namespace is actually harmful: doing so can convey the incorrec=
t impression that there might be some way to use an Event Package in INFO, =
or an INFO package in SUBSCRIBE.

/a
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

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Subject: [sipcore] Clarifying the voicemail use-cases in RFC4244bis
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 As a service provider we always use the last target as a 
target of voicemail.

 But in a corporate world, call can be forwarded to different 
user and if no one picks up, policy is usually to retarget 
the call to the voicemail of the original target. 

 As different deployment has different practice and 
requirements, it is difficult to say "A" is the only one 
you need to look at to solve the voicemail problem. 

 We can add a normative text saying if used in PBX 
environment to use the first target within its administrative 
domain and say, in non-PBX environment use the 
last target. But that's not much different from what Mary 
had on the slide.. 

  If that's not sufficient then I guess  we can add 
an identifier which identifies a target with the notion of 
 "use this target for voicemail" (May be use RFC4458?). 

 Proxy that handles the call and diverts the call to the 
voicemail after executing whatever policy can add 
this identifier.

 Regards
  Shida

From mary.ietf.barnes@gmail.com  Tue Mar 23 10:30:29 2010
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Shida Schubert <shida@ntt-at.com>
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Subject: Re: [sipcore] Clarifying the voicemail use-cases in RFC4244bis
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I'm wondering if it wouldn't be useful to show these two different
voicemail scenarios and thus we might be able to have normative text
or at least recommended behavior. I believe the HI headers would be
populated slightly differently in these scenarios, so this should
work.  In looking at all the other examples, the HI information that
is relevant is deterministic.   Thus, we might be able to make the
text stronger although I'm not convinced we can do more than a BCP
level of specification.

Mary.

On Tue, Mar 23, 2010 at 12:02 PM, Shida Schubert <shida@ntt-at.com> wrote:
>
> =A0As a service provider we always use the last target as a
> target of voicemail.
>
> =A0But in a corporate world, call can be forwarded to different
> user and if no one picks up, policy is usually to retarget
> the call to the voicemail of the original target.
>
> =A0As different deployment has different practice and
> requirements, it is difficult to say "A" is the only one
> you need to look at to solve the voicemail problem.
>
> =A0We can add a normative text saying if used in PBX
> environment to use the first target within its administrative
> domain and say, in non-PBX environment use the
> last target. But that's not much different from what Mary
> had on the slide..
>
> =A0If that's not sufficient then I guess =A0we can add
> an identifier which identifies a target with the notion of
> =A0"use this target for voicemail" (May be use RFC4458?).
>
> =A0Proxy that handles the call and diverts the call to the
> voicemail after executing whatever policy can add
> this identifier.
>
> =A0Regards
> =A0Shida
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, SIPCORE <sipcore@ietf.org>
Date: Tue, 23 Mar 2010 14:01:20 -0400
Thread-Topic: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
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Subject: Re: [sipcore] SUBSCRIBE 3xx - can it solve issues raised in 3265bis? [was INVITE 3xx and early media]
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________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Chri=
ster Holmberg [christer.holmberg@ericsson.com]

Would it be possible to use the 3xx mechanism in order to solve the forking=
 related problems with SUB/NOT (as described in 3265bis)?

The forking proxy would return a 3xx response to the SUBSCRIBE request, and=
 the UAC can then send individual SUBSCRIBE requests to each UAS?
_______________________________________________

A UA can send SUBSCRIBE with "Request-Disposition: no-recurse", but there's=
 no guarantee that every proxy on the way will obey it.

Dale

From keith.drage@alcatel-lucent.com  Tue Mar 23 12:25:28 2010
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To: Avasarala Ranjit-A20990 <ranjit@motorola.com>, Adam Roach <adam@nostrum.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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You cannot prevent it, because IETF are not the protocol police.

However if you wish to make a claim of conformance to both the event packag=
e specifications and the info package specifications, you will have to impl=
ement each using an appropriately registered descriptor of the package. Pre=
sumably expert review on either path will raise some issues with you trying=
 to do this, unless you demonstrate that what you are doing actually makes =
sense, rather than just providing two ways of using the protocol to do the =
same thing.

regards

Keith=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Avasarala Ranjit-A20990
> Sent: Tuesday, March 23, 2010 4:34 PM
> To: Adam Roach; Christer Holmberg
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
>=20
> Hi
>=20
> Now since the I-D does not specify any particular format or=20
> template for an INFO package, there is a very likely chance=20
> that someone could propose an INFO package as a XML package=20
> which is quite similar to an
> 3265 based event package. Now how can someone prevent use of=20
> this package with SUBSCRIBE / NOTIFY?
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: Monday, March 22, 2010 11:25 PM
> To: Christer Holmberg
> Cc: sipcore@ietf.org; Avasarala Ranjit-A20990
> Subject: Re: [sipcore] Query on draft-ietf-sipcore-info-events-07
>=20
> On 3/22/10 9:28 AM, Christer Holmberg wrote:
> >
> > Hi,
> >
> >   =20
> >> Also, I remember some discussion about whether it's ok to=20
> reuse event
>=20
> >> package names? (Is it ok for someone to create a "dialog" event=20
> >> package for instance?) The draft is silent on this at the=20
> moment - is
> that on purpose?
> >>     =20
> > I see no reason to explicitly forbid it, since we're dealing with
> different namespaces.
> >   =20
>=20
> I agree.
>=20
> In fact, I believe that doing anything that implies that=20
> they're drawing from the same namespace is actually harmful:=20
> doing so can convey the incorrect impression that there might=20
> be some way to use an Event Package in INFO, or an INFO=20
> package in SUBSCRIBE.
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Clarifying the voicemail use-cases in RFC4244bis
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IMO both mentioned options are broken. The proposed scenario is when 
Alice calls Bob, Bob forwards to Carol, and Carol forwards to VM.

If Carol didn't have VM, the call would have failed back to Bob, and he 
might have forwarded to his VM, or he might have then forwarded the call 
to Dave. When Carol forwards to VM she preempts Bob's ability to make 
the next policy decision.

The given scenarios for handling this mis the point - they don't give 
the control back to Bob. Instead they propose to have Carol's VM server 
use Bob's VM mailbox. This at best works if Bob and Carol use the same 
VM server, or servers that have a special relationship with one another. 
  And they still presume that Carol's VM server knows the policy that 
Bob wants to use.

Having all the info in HI does not solve this problem. The real problem 
is not one of giving Carol's VM server a list of mailboxes to choose 
among. The problem is to convey part of Bob's policy to Carol - namely 
whether to delegate the handling of call failure to Carol, or if instead 
Carol should simply fail the call attempt if there is no answer from 
Carol, and let Bob deal with that.

That is all possible with Callerprefs. If implemented it would give as 
good or better a result in all cases, and doesn't require HI.

	Thanks,
	Paul

Mary Barnes wrote:
> I'm wondering if it wouldn't be useful to show these two different
> voicemail scenarios and thus we might be able to have normative text
> or at least recommended behavior. I believe the HI headers would be
> populated slightly differently in these scenarios, so this should
> work.  In looking at all the other examples, the HI information that
> is relevant is deterministic.   Thus, we might be able to make the
> text stronger although I'm not convinced we can do more than a BCP
> level of specification.
> 
> Mary.
> 
> On Tue, Mar 23, 2010 at 12:02 PM, Shida Schubert <shida@ntt-at.com> wrote:
>>  As a service provider we always use the last target as a
>> target of voicemail.
>>
>>  But in a corporate world, call can be forwarded to different
>> user and if no one picks up, policy is usually to retarget
>> the call to the voicemail of the original target.
>>
>>  As different deployment has different practice and
>> requirements, it is difficult to say "A" is the only one
>> you need to look at to solve the voicemail problem.
>>
>>  We can add a normative text saying if used in PBX
>> environment to use the first target within its administrative
>> domain and say, in non-PBX environment use the
>> last target. But that's not much different from what Mary
>> had on the slide..
>>
>>  If that's not sufficient then I guess  we can add
>> an identifier which identifies a target with the notion of
>>  "use this target for voicemail" (May be use RFC4458?).
>>
>>  Proxy that handles the call and diverts the call to the
>> voicemail after executing whatever policy can add
>> this identifier.
>>
>>  Regards
>>  Shida
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From mary.ietf.barnes@gmail.com  Tue Mar 23 14:19:07 2010
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Date: Tue, 23 Mar 2010 16:19:07 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Clarifying the voicemail use-cases in RFC4244bis
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Paul,

That's one reason why we note that the call flows are examples and not
prescriptive for implementing the services ;) I don't entirely agree
that callerprefs is all that you need, but I will agree that trying to
document some of these services as only needing History-Info is not
necessarily a good idea, although many of them do just need
History-Info.

We had an offline discussion and were considering that the best
approach would be to pull the call flows (except perhaps the first
one) out into a separate document to progress as a BCP - that way we
can include the use of other mechanisms and optimize the
recommendations. Along with doing that, we propose to update the UAS
processing and the Application consideration sections of 4244bis to
have more discussion of using the new tags for services (without
prescribing services per se) would hopefully resolve the concern
raised by Cullen (and others) about the core document not describing
how to use the header (which was the agreed approach for RFC 4244, but
it seems folks realize that wasn't particularly constructive).

Separating the detailed flows in this manner is also more consistent
with how we've done protocol work in most of the other RAI WGs.  We
would of course have to determine the best home for a call flow
document.

Thanks,
Mary.

On Tue, Mar 23, 2010 at 3:37 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:
> IMO both mentioned options are broken. The proposed scenario is when Alic=
e
> calls Bob, Bob forwards to Carol, and Carol forwards to VM.
>
> If Carol didn't have VM, the call would have failed back to Bob, and he
> might have forwarded to his VM, or he might have then forwarded the call =
to
> Dave. When Carol forwards to VM she preempts Bob's ability to make the ne=
xt
> policy decision.
>
> The given scenarios for handling this mis the point - they don't give the
> control back to Bob. Instead they propose to have Carol's VM server use
> Bob's VM mailbox. This at best works if Bob and Carol use the same VM
> server, or servers that have a special relationship with one another. =A0=
And
> they still presume that Carol's VM server knows the policy that Bob wants=
 to
> use.
>
> Having all the info in HI does not solve this problem. The real problem i=
s
> not one of giving Carol's VM server a list of mailboxes to choose among. =
The
> problem is to convey part of Bob's policy to Carol - namely whether to
> delegate the handling of call failure to Carol, or if instead Carol shoul=
d
> simply fail the call attempt if there is no answer from Carol, and let Bo=
b
> deal with that.
>
> That is all possible with Callerprefs. If implemented it would give as go=
od
> or better a result in all cases, and doesn't require HI.
>
> =A0 =A0 =A0 =A0Thanks,
> =A0 =A0 =A0 =A0Paul
>
> Mary Barnes wrote:
>>
>> I'm wondering if it wouldn't be useful to show these two different
>> voicemail scenarios and thus we might be able to have normative text
>> or at least recommended behavior. I believe the HI headers would be
>> populated slightly differently in these scenarios, so this should
>> work. =A0In looking at all the other examples, the HI information that
>> is relevant is deterministic. =A0 Thus, we might be able to make the
>> text stronger although I'm not convinced we can do more than a BCP
>> level of specification.
>>
>> Mary.
>>
>> On Tue, Mar 23, 2010 at 12:02 PM, Shida Schubert <shida@ntt-at.com> wrot=
e:
>>>
>>> =A0As a service provider we always use the last target as a
>>> target of voicemail.
>>>
>>> =A0But in a corporate world, call can be forwarded to different
>>> user and if no one picks up, policy is usually to retarget
>>> the call to the voicemail of the original target.
>>>
>>> =A0As different deployment has different practice and
>>> requirements, it is difficult to say "A" is the only one
>>> you need to look at to solve the voicemail problem.
>>>
>>> =A0We can add a normative text saying if used in PBX
>>> environment to use the first target within its administrative
>>> domain and say, in non-PBX environment use the
>>> last target. But that's not much different from what Mary
>>> had on the slide..
>>>
>>> =A0If that's not sufficient then I guess =A0we can add
>>> an identifier which identifies a target with the notion of
>>> =A0"use this target for voicemail" (May be use RFC4458?).
>>>
>>> =A0Proxy that handles the call and diverts the call to the
>>> voicemail after executing whatever policy can add
>>> this identifier.
>>>
>>> =A0Regards
>>> =A0Shida
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>

From christer.holmberg@ericsson.com  Tue Mar 23 16:05:25 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Wed, 24 Mar 2010 00:05:39 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,
=20
>>>>>If you want parallel forking, even if done by the UAC, there is no=20
>>>>>good solution to this problem. The parallel early media is going to=20
>>>>>be generated and something must happen to it. The only question is whi=
ch of the bad=20
>>>>>solutions you prefer.
>>>>>
>>>>>For instance, you may have one stream asking you to enter an access=20
>>>>>code, another playing an out-of- service tone, and a third sending=20
>>>>>no media which will eventually pick up with a real person. What would =
you *like* for the=20
>>>>>caller to hear?
>>>>For this reasons you have B2BUAs, application servers, gating nodes=20
>>>>etc, that at least TRY to deal with the problem, to ensure that whateve=
r the user=20
>>>>eventually hears, at least the quality will be good.
>>>For what that's worth! You get a good quality rendering of that=20
>>>out-of-service tone, and so hang up even though the call was ringing=20
>>>elsewhere, but without any ringback.
>>>
>>>I guess we just have to "let a 1000 flowers bloom" on this one.
>>=20
>>As I've said before, I don't think there is a very big risk that the "wro=
ng" early media=20
>>would be presented to you.
>>
>>
>>Leaving PSTN interworking aside, most early media (at least the one of re=
al importance) will=20
>>be generated by application servers/B2BUAs in the network, and they will =
normally be able to=20
>>control the media pretty well. A SIP UA might generate something, but unl=
ess it's a Henry=20
>>Sinnreich device (where all the services are implemented in the end devic=
e :) I don't think=20
>>UAs will normally send service related announcement etc.
>
>I gather you are arguing that there just aren't many calls with early medi=
a. But "leaving=20
>PSTN interworking aside" isn't really fair, since there is so much of that=
.

I agree. My intention was only to say that the only "endpoints" that are li=
kely to send early media in most cases are PSTN networks.

>For instance there could be a lot of the google service that simply forks =
calls to multiple=20
>PSTN lines. And don't most pstn calls have in-band ringback? That would ge=
t multiple early=20
>media, though if its just ringback it doesn't matter if you play it or not=
. But if one of=20
>those becomes defective and starts playing out of service messages then it=
 does matter which=20
>one you play.

Yes.

>>Of course, if you have application servers sending media both in the orig=
inating and=20
>>terminating network, there need to be some policy procedures on which med=
ia has higher=20
>>priority etc.
>
>Policy doesn't help a lot. It content, not policy, that determines which i=
s important to=20
>hear.

Sure, but there can be policy that says "My early media is more important t=
hat anything else".

Having said that, having information abouot the content would probably be v=
ery useful. However, I don't think it would help in the PSTN interworking c=
ase, becase the MGC most likely does not know what early media (in-band rin=
ging or service anouncement) comes from the PSTN network.

>Perhaps the best we can do is discourage the use of early media for anythi=
ng important.

I guess it depends on what you mean by "important", but there are a number =
of services that rely on early media.

Regards,

Christer






>> Christer Holmberg wrote:
>>> Hi,
>>>
>>>
>>>>                     I would like to expand on this issue a little. Sup=
pose I have at my
>>>>                     home proxy a policy whereby calls go to B1 and B2 =
in parallel, and
>>>>                     after T seconds of no answer calls instead go to B=
3 (voicemail,
>>>>                     perhaps). There is no way that the proxy could con=
vey that policy in
>>>>                     a 3xx response to the UAC. To make this policy (or=
 similar policies)
>>>>                     work, the proxy would need to manufacturer tempora=
ry contact URIs
>>>>                     that map to B1 and B2 and send these back in a 302=
. It would then
>>>>                     need to hope that the UAC does indeed establish ne=
w requests in
>>>>                     parallel to the two temporary contacts for B1 and =
B2. The proxy
>>>>                     would need to treat the two requests as related an=
d if both are
>>>>                     unanswered after T seconds, reject one with a 4xx =
and the other with
>>>>                     a 302 (this time with a third temporary contact UR=
I mapped to B3).
>>>>                     So policy CAN be met by the proxy, although with d=
ependence on the
>>>>                     UAC doing the right thing.
>>>>
>>>>
>>>>             So here, we either need your home proxy to be a home B2BUA=
 (After all,
>>>>             what happens when B1 and B2 both send back early media, or=
 both get
>>>>             answered by a bot?), OR we need a richer 302 response that=
 conveys
>>>>             "Try these two destinations in parallel, and if neither re=
plies in X
>>>>             seconds, try this third destination).
>>>>
>>>>     But, the UAC still doesn't know whether B1 and B2 would send early=
 media.
>>>>
>>>> By offering different ports to B1 and B2 (especially with the media-to=
-signaling coupling extensions that have been proposed) UAC at least knows =
which one of B1 and B2 each stream is coming from.
>>> Yes. But, there are other mechanisms to solve that, and it still doesn'=
t solve the fundamental problem: the UAC can receive multiple media streams=
 associated with multiple early dialogs.
>>>
>>> Also, in bandwidth sensitive access networks there is only limited band=
width, normally calculated based on the need for media associated with one =
early dialog. Even if the UAC is able to select and play only one stream to=
 the user, the voice quality will suffer due to too little bandwidth.
>>>
>>> Regards,
>>>
>>> Christer
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>

From pkyzivat@cisco.com  Tue Mar 23 16:11:02 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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My conclusion is that none of the suggestions being thrown around 
fundamentally improve the handling of early media in any way. Its just a 
mess, and there is no feasible solution.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
>  
>>>>>> If you want parallel forking, even if done by the UAC, there is no 
>>>>>> good solution to this problem. The parallel early media is going to 
>>>>>> be generated and something must happen to it. The only question is which of the bad 
>>>>>> solutions you prefer.
>>>>>>
>>>>>> For instance, you may have one stream asking you to enter an access 
>>>>>> code, another playing an out-of- service tone, and a third sending 
>>>>>> no media which will eventually pick up with a real person. What would you *like* for the 
>>>>>> caller to hear?
>>>>> For this reasons you have B2BUAs, application servers, gating nodes 
>>>>> etc, that at least TRY to deal with the problem, to ensure that whatever the user 
>>>>> eventually hears, at least the quality will be good.
>>>> For what that's worth! You get a good quality rendering of that 
>>>> out-of-service tone, and so hang up even though the call was ringing 
>>>> elsewhere, but without any ringback.
>>>>
>>>> I guess we just have to "let a 1000 flowers bloom" on this one.
>>> As I've said before, I don't think there is a very big risk that the "wrong" early media 
>>> would be presented to you.
>>>
>>>
>>> Leaving PSTN interworking aside, most early media (at least the one of real importance) will 
>>> be generated by application servers/B2BUAs in the network, and they will normally be able to 
>>> control the media pretty well. A SIP UA might generate something, but unless it's a Henry 
>>> Sinnreich device (where all the services are implemented in the end device :) I don't think 
>>> UAs will normally send service related announcement etc.
>> I gather you are arguing that there just aren't many calls with early media. But "leaving 
>> PSTN interworking aside" isn't really fair, since there is so much of that.
> 
> I agree. My intention was only to say that the only "endpoints" that are likely to send early media in most cases are PSTN networks.
> 
>> For instance there could be a lot of the google service that simply forks calls to multiple 
>> PSTN lines. And don't most pstn calls have in-band ringback? That would get multiple early 
>> media, though if its just ringback it doesn't matter if you play it or not. But if one of 
>> those becomes defective and starts playing out of service messages then it does matter which 
>> one you play.
> 
> Yes.
> 
>>> Of course, if you have application servers sending media both in the originating and 
>>> terminating network, there need to be some policy procedures on which media has higher 
>>> priority etc.
>> Policy doesn't help a lot. It content, not policy, that determines which is important to 
>> hear.
> 
> Sure, but there can be policy that says "My early media is more important that anything else".
> 
> Having said that, having information abouot the content would probably be very useful. However, I don't think it would help in the PSTN interworking case, becase the MGC most likely does not know what early media (in-band ringing or service anouncement) comes from the PSTN network.
> 
>> Perhaps the best we can do is discourage the use of early media for anything important.
> 
> I guess it depends on what you mean by "important", but there are a number of services that rely on early media.
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> 
> 
>>> Christer Holmberg wrote:
>>>> Hi,
>>>>
>>>>
>>>>>                     I would like to expand on this issue a little. Suppose I have at my
>>>>>                     home proxy a policy whereby calls go to B1 and B2 in parallel, and
>>>>>                     after T seconds of no answer calls instead go to B3 (voicemail,
>>>>>                     perhaps). There is no way that the proxy could convey that policy in
>>>>>                     a 3xx response to the UAC. To make this policy (or similar policies)
>>>>>                     work, the proxy would need to manufacturer temporary contact URIs
>>>>>                     that map to B1 and B2 and send these back in a 302. It would then
>>>>>                     need to hope that the UAC does indeed establish new requests in
>>>>>                     parallel to the two temporary contacts for B1 and B2. The proxy
>>>>>                     would need to treat the two requests as related and if both are
>>>>>                     unanswered after T seconds, reject one with a 4xx and the other with
>>>>>                     a 302 (this time with a third temporary contact URI mapped to B3).
>>>>>                     So policy CAN be met by the proxy, although with dependence on the
>>>>>                     UAC doing the right thing.
>>>>>
>>>>>
>>>>>             So here, we either need your home proxy to be a home B2BUA (After all,
>>>>>             what happens when B1 and B2 both send back early media, or both get
>>>>>             answered by a bot?), OR we need a richer 302 response that conveys
>>>>>             "Try these two destinations in parallel, and if neither replies in X
>>>>>             seconds, try this third destination).
>>>>>
>>>>>     But, the UAC still doesn't know whether B1 and B2 would send early media.
>>>>>
>>>>> By offering different ports to B1 and B2 (especially with the media-to-signaling coupling extensions that have been proposed) UAC at least knows which one of B1 and B2 each stream is coming from.
>>>> Yes. But, there are other mechanisms to solve that, and it still doesn't solve the fundamental problem: the UAC can receive multiple media streams associated with multiple early dialogs.
>>>>
>>>> Also, in bandwidth sensitive access networks there is only limited bandwidth, normally calculated based on the need for media associated with one early dialog. Even if the UAC is able to select and play only one stream to the user, the voice quality will suffer due to too little bandwidth.
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
> 

From christer.holmberg@ericsson.com  Tue Mar 23 16:16:05 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>, Gilad Shaham <gilad@voxisoft.com>
Date: Wed, 24 Mar 2010 00:16:21 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: 'SIPCORE' <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

Maybe RFC 4574 could be used.

Regards,

Christer
=20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: 8. maaliskuuta 2010 16:38
To: Gilad Shaham
Cc: Christer Holmberg; 'SIPCORE'
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork=
 or not to fork?]



Gilad Shaham wrote:
> Paul Kyzivat wrote:
>> Perhaps the best we can do is discourage the use of early media for=20
>> anything important.
>>
>=20
> I'm not sure this scenario is that common, but if it is something of=20
> interest, perhaps it's possible to add a header/parameter conveying=20
> the nature of the media stream (ringback, service message, voice,=20
> etc.). This is similar to the content-disposition handling parameter,=20
> but more informative than "optional" and "required". This way, a B2BUA=20
> would be able to choose the stream to forward and even without a=20
> B2BUA, an endpoint receiving several streams would know which stream to p=
lay back to the user.

I guess this would be something like the Reason header, or maybe it would r=
eally be the Reason header, though it would take some changes to allow it f=
or that. Another possibility for this is Alert-Info.

The response code, possibly augmented with the Reason header, would work fo=
r things like out of service that are going to cause the call to fail.=20
However the callee has to have confidence that the caller will be properly =
informed of the case.

The Alert-Info containing a URN can potentially work for common things that=
 aren't going to cause the call to fail. But that isn't sufficient if there=
 is variable information to convey to the caller, unless it is done with wi=
th a URI that can be dereferenced and rendered. The latter would be an alte=
rnative if we could convince UAs to render such things.=20
My impression is that there is reluctance to do so, though it really isn't =
much different from rendering an early media string.

But none of this will help with many pstn interop cases where the reason is=
n't known except through the content of the media.

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Tue Mar 23 16:28:44 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Dean Willis' <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 24 Mar 2010 00:28:59 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: 'SIPCORE' <sipcore@ietf.org>, Gilad Shaham <gilad@voxisoft.com>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork	or not to fork?]
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Hi,=20

>>I wasn't thinking of discriminating the early media streams, but=20
>>rather finding ways to eliminate them while retaining their intent.
>
>That would be cool. Requires a business model change at PSTN gateways, whi=
ch IMHO should go=20
>to 200 OK (and a fully-connected dialog) before/ while starting the IAM or=
 equivalent on the=20
>PSTN side. But too many yerkaloids want to bill on 200OK.

It is not only about business models.

Using 200 OK would ensure that all but one dialog is terminated, but there =
can still be multiple sources of early media.

>But for non-gateway nodes, we could certainly do something along the lines=
 of a 1xx header or=20
>specific cause-codes as different 1XX response types.
>
>
>Several years ago Rohan proposed a model where the UAC did another O/=20
>A with each early dialog to move its media to a distinct port. That=20
>way it could then associate the signaling with a particular media=20
>stream. Seems cumbersome, but maybe it will take something like that=20
>to solve the problem.
>
>Only works once the UAS sends back a 1xx of some sort. Some nodes just sen=
d media. For=20
>gateways, we therefore have to differentiate in the OFFER as seen by the U=
AS, which means=20
>sending a different offer to each UAS, which means NO  FORKING!

I think that UASs should send back a 1xx of some sort.

>Or maybe we could do yet-another BCP on gateways to require early- dialog =
before media cut-
>through. This might be worth exploring. Of course, this requires updating =
all the gateways=20
>before it can be relied on, or at least probing for compliant gateways wit=
h a Require.

Or, we may want to only SDP only to establish a control channel for some ot=
her media negotiation protocol, used between the UAC and each UAS ;)

Regards,

Christer







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From christer.holmberg@ericsson.com  Tue Mar 23 16:34:04 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Wed, 24 Mar 2010 00:34:19 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

Since there is no protocol mechanism to handle this, I guess policy (whatev=
er it is based upon) is the only option. It may work well or bad, but if it=
's the only option...

Regards,

Christer=20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: 24. maaliskuuta 2010 1:11
To: Christer Holmberg
Cc: 'Dean Willis'; SIPCORE
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork=
 or not to fork?]

My conclusion is that none of the suggestions being thrown around fundament=
ally improve the handling of early media in any way. Its just a mess, and t=
here is no feasible solution.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
> =20
>>>>>> If you want parallel forking, even if done by the UAC, there is=20
>>>>>> no good solution to this problem. The parallel early media is=20
>>>>>> going to be generated and something must happen to it. The only=20
>>>>>> question is which of the bad solutions you prefer.
>>>>>>
>>>>>> For instance, you may have one stream asking you to enter an=20
>>>>>> access code, another playing an out-of- service tone, and a third=20
>>>>>> sending no media which will eventually pick up with a real=20
>>>>>> person. What would you *like* for the caller to hear?
>>>>> For this reasons you have B2BUAs, application servers, gating=20
>>>>> nodes etc, that at least TRY to deal with the problem, to ensure=20
>>>>> that whatever the user eventually hears, at least the quality will be=
 good.
>>>> For what that's worth! You get a good quality rendering of that=20
>>>> out-of-service tone, and so hang up even though the call was=20
>>>> ringing elsewhere, but without any ringback.
>>>>
>>>> I guess we just have to "let a 1000 flowers bloom" on this one.
>>> As I've said before, I don't think there is a very big risk that the=20
>>> "wrong" early media would be presented to you.
>>>
>>>
>>> Leaving PSTN interworking aside, most early media (at least the one=20
>>> of real importance) will be generated by application servers/B2BUAs=20
>>> in the network, and they will normally be able to control the media=20
>>> pretty well. A SIP UA might generate something, but unless it's a=20
>>> Henry Sinnreich device (where all the services are implemented in the e=
nd device :) I don't think UAs will normally send service related announcem=
ent etc.
>> I gather you are arguing that there just aren't many calls with early=20
>> media. But "leaving PSTN interworking aside" isn't really fair, since th=
ere is so much of that.
>=20
> I agree. My intention was only to say that the only "endpoints" that are =
likely to send early media in most cases are PSTN networks.
>=20
>> For instance there could be a lot of the google service that simply=20
>> forks calls to multiple PSTN lines. And don't most pstn calls have=20
>> in-band ringback? That would get multiple early media, though if its=20
>> just ringback it doesn't matter if you play it or not. But if one of=20
>> those becomes defective and starts playing out of service messages then =
it does matter which one you play.
>=20
> Yes.
>=20
>>> Of course, if you have application servers sending media both in the=20
>>> originating and terminating network, there need to be some policy=20
>>> procedures on which media has higher priority etc.
>> Policy doesn't help a lot. It content, not policy, that determines=20
>> which is important to hear.
>=20
> Sure, but there can be policy that says "My early media is more important=
 that anything else".
>=20
> Having said that, having information abouot the content would probably be=
 very useful. However, I don't think it would help in the PSTN interworking=
 case, becase the MGC most likely does not know what early media (in-band r=
inging or service anouncement) comes from the PSTN network.
>=20
>> Perhaps the best we can do is discourage the use of early media for anyt=
hing important.
>=20
> I guess it depends on what you mean by "important", but there are a numbe=
r of services that rely on early media.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>>> Christer Holmberg wrote:
>>>> Hi,
>>>>
>>>>
>>>>>                     I would like to expand on this issue a little. Su=
ppose I have at my
>>>>>                     home proxy a policy whereby calls go to B1 and B2=
 in parallel, and
>>>>>                     after T seconds of no answer calls instead go to =
B3 (voicemail,
>>>>>                     perhaps). There is no way that the proxy could co=
nvey that policy in
>>>>>                     a 3xx response to the UAC. To make this policy (o=
r similar policies)
>>>>>                     work, the proxy would need to manufacturer tempor=
ary contact URIs
>>>>>                     that map to B1 and B2 and send these back in a 30=
2. It would then
>>>>>                     need to hope that the UAC does indeed establish n=
ew requests in
>>>>>                     parallel to the two temporary contacts for B1 and=
 B2. The proxy
>>>>>                     would need to treat the two requests as related a=
nd if both are
>>>>>                     unanswered after T seconds, reject one with a 4xx=
 and the other with
>>>>>                     a 302 (this time with a third temporary contact U=
RI mapped to B3).
>>>>>                     So policy CAN be met by the proxy, although with =
dependence on the
>>>>>                     UAC doing the right thing.
>>>>>
>>>>>
>>>>>             So here, we either need your home proxy to be a home B2BU=
A (After all,
>>>>>             what happens when B1 and B2 both send back early media, o=
r both get
>>>>>             answered by a bot?), OR we need a richer 302 response tha=
t conveys
>>>>>             "Try these two destinations in parallel, and if neither r=
eplies in X
>>>>>             seconds, try this third destination).
>>>>>
>>>>>     But, the UAC still doesn't know whether B1 and B2 would send earl=
y media.
>>>>>
>>>>> By offering different ports to B1 and B2 (especially with the media-t=
o-signaling coupling extensions that have been proposed) UAC at least knows=
 which one of B1 and B2 each stream is coming from.
>>>> Yes. But, there are other mechanisms to solve that, and it still doesn=
't solve the fundamental problem: the UAC can receive multiple media stream=
s associated with multiple early dialogs.
>>>>
>>>> Also, in bandwidth sensitive access networks there is only limited ban=
dwidth, normally calculated based on the need for media associated with one=
 early dialog. Even if the UAC is able to select and play only one stream t=
o the user, the voice quality will suffer due to too little bandwidth.
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>=20

From dean.willis@softarmor.com  Tue Mar 23 19:06:07 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Paul Kyzivat wrote:
> My conclusion is that none of the suggestions being thrown around
> fundamentally improve the handling of early media in any way. Its just a
> mess, and there is no feasible solution.

Arguably, not forking in a proxy helps. It least lets you know how to
kill an early media stream that is being spewed back at you.

Offerless INVITEs help more, since the media can't come back to you
until the O/A has completed.

What if we split the O/A stage into three phases:

1) pre-offer: Alice INVITES Bob, includes a description of media she can
send and receive but not an IP address or port number

2) offer: Bob sends Alice a description of the media he expects to send
and to receive, along with the IP addresses and port numbers he is going
to use.

3) answer: Alice sends Bob the IP addresses and port number to whichs he
should direct media and from which Alice will be sending media.

Note that this eliminates early media. Bob can't send Alice any media
until he receives the answer.


--
Dean

From dean.willis@softarmor.com  Tue Mar 23 19:11:25 2010
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Cc: 'SIPCORE' <sipcore@ietf.org>, Gilad Shaham <gilad@voxisoft.com>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Christer Holmberg wrote:

>>> I wasn't thinking of discriminating the early media streams, but
>>>  rather finding ways to eliminate them while retaining their
>>> intent.
>> That would be cool. Requires a business model change at PSTN
>> gateways, which IMHO should go to 200 OK (and a fully-connected
>> dialog) before/ while starting the IAM or equivalent on the PSTN
>> side. But too many yerkaloids want to bill on 200OK.
> 
> It is not only about business models.
> 
> Using 200 OK would ensure that all but one dialog is terminated, but
> there can still be multiple sources of early media.
> 


No, I'm suggesting that all of the forks respond with a 200 OK and move
to a connected state, not just the first fork to respond. It's not a
race, and there is NO early media -- just media being sent before the
PSTN side of the gateway moves to a fully-connected state. Sorting out
multiple media streams is a separate problem, but there are ways to
automate many cases in a B2BUA or really smart UA.

But this breaks the business model, which assumes that everything
everywhere is fully-connected and in a "billing starts" state when the
first 200OK flies.

--
dean

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Dean Willis wrote:
> Paul Kyzivat wrote:
>> My conclusion is that none of the suggestions being thrown around
>> fundamentally improve the handling of early media in any way. Its just a
>> mess, and there is no feasible solution.
> 
> Arguably, not forking in a proxy helps. It least lets you know how to
> kill an early media stream that is being spewed back at you.

Well, that depends.

Its really only parallel forking thats a problem, so we can probably 
ignore serial forking.

For parallel forking, moving it from the proxy to the UAC doesn't really 
solve the problem - you can still end up with multiple media streams and 
no clue what to do with them.

You can say "don't do that", but there are desirable features that 
depend on it - namely "simultaneous ring". Potentially that can be 
replaced with presence-sensitive ringing to choose the *right* on to 
ring first, but that requires data that often isn't available.

> Offerless INVITEs help more, since the media can't come back to you
> until the O/A has completed.

That is just sticking your head in the sand. The media is still being 
generated, even if you don't know about it.

It does have the advantage that you can associate the media stream to 
the signaling source, but it still doesn't tell you which stream is 
interesting to listen to.

ISTM that if you want "simultaneous ring", and the things ringing are 
often PSTN devices, there is still no solution. Perhaps there is some 
possibility of automated media analysis - the pstn GW figuring out that 
the media being received is something uninteresting - just conventional 
ringback - and suppressing it. But thats a lot of trouble and is likely 
to have a high error rate.

	Thanks,
	Paul

> What if we split the O/A stage into three phases:
> 
> 1) pre-offer: Alice INVITES Bob, includes a description of media she can
> send and receive but not an IP address or port number
> 
> 2) offer: Bob sends Alice a description of the media he expects to send
> and to receive, along with the IP addresses and port numbers he is going
> to use.
> 
> 3) answer: Alice sends Bob the IP addresses and port number to whichs he
> should direct media and from which Alice will be sending media.
> 
> Note that this eliminates early media. Bob can't send Alice any media
> until he receives the answer.
> 
> 
> --
> Dean
> 

From christer.holmberg@ericsson.com  Wed Mar 24 09:56:02 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Wed, 24 Mar 2010 17:51:44 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: 'SIPCORE' <sipcore@ietf.org>, Gilad, Shaham <gilad@voxisoft.com>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>>>>I wasn't thinking of discriminating the early media streams, but
>>>>rather finding ways to eliminate them while retaining their
>>>>intent.
>>>That would be cool. Requires a business model change at PSTN
>>>gateways, which IMHO should go to 200 OK (and a fully-connected
>>>dialog) before/ while starting the IAM or equivalent on the PSTN
>>>side. But too many yerkaloids want to bill on 200OK.
>>
>>It is not only about business models.
>>
>>Using 200 OK would ensure that all but one dialog is terminated, but
>>there can still be multiple sources of early media.
>
>
>No, I'm suggesting that all of the forks respond with a 200 OK and move
>to a connected state, not just the first fork to respond. It's not a
>race, and there is NO early media -- just media being sent before the
>PSTN side of the gateway moves to a fully-connected state.
>
>Sorting out multiple media streams is a separate problem, but there are wa=
ys to
>automate many cases in a B2BUA or really smart UA.
>
>But this breaks the business model, which assumes that everything
>everywhere is fully-connected and in a "billing starts" state when the
>first 200OK flies.

But, what problem would that solve?=20

The only difference is in what state you are (established-dialog instead of=
 early-dialog), but you still have the same problem as duing the early dial=
og phase: you might receive early media from multiple locations.=20

Regards,

Christer=

From christer.holmberg@ericsson.com  Wed Mar 24 10:08:29 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Dean Willis <dean.willis@softarmor.com>
Date: Wed, 24 Mar 2010 18:08:28 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>>>My conclusion is that none of the suggestions being thrown around
>>>fundamentally improve the handling of early media in any way. Its just a
>>>mess, and there is no feasible solution.
>>
>>Arguably, not forking in a proxy helps. It least lets you know how to
>>kill an early media stream that is being spewed back at you.
>
>Well, that depends.
>
>Its really only parallel forking thats a problem, so we can probably
>ignore serial forking.
>
>For parallel forking, moving it from the proxy to the UAC doesn't really
>solve the problem - you can still end up with multiple media streams and
>no clue what to do with them.
>
>You can say "don't do that", but there are desirable features that
>depend on it - namely "simultaneous ring". Potentially that can be
>replaced with presence-sensitive ringing to choose the *right* on to
>ring first, but that requires data that often isn't available.
>
>>Offerless INVITEs help more, since the media can't come back to you until=
 the O/A has completed.
>
>That is just sticking your head in the sand. The media is still being gene=
rated, even if you don't know about it.
>
>It does have the advantage that you can associate the media stream to
>the signaling source, but it still doesn't tell you which stream is
>interesting to listen to.

Exactly. And, in my opinion THAT is the big problem.

Being able to associate a media stream with dialog is of course very useful=
, and there are ways to do that (including the one presented by Dean), but =
it doesn't solve the problem with multiple early media.

>ISTM that if you want "simultaneous ring", and the things ringing are
>often PSTN devices, there is still no solution. Perhaps there is some
>possibility of automated media analysis - the pstn GW figuring out that
>the media being received is something uninteresting - just conventional
>ringback - and suppressing it. But thats a lot of trouble and is likely
>to have a high error rate.

Correct. And, as long as the protocol allows forking, I think it is not pos=
sible to define a protocol mechanism how to solve the problem.

We can define information elements which contain information about the earl=
y media (eventhough it most likely won't work in the PSTN case), but at the=
 end of the day you will need policy rules in order to decide what media is=
 forwarded or played to the calling user.=20

Regards,

Christer=

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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Even if you would define policy, that doesn't mean it doesn't make sense to
provide information on the nature of the media from non-PSTN sources. So you
could say (in policy), for example, that PSTN is more important than just a
ringback tone, but not more important than an IVR.

My main question is whether we actually see in practice these scenarios to
warrant a header/parameter/attribute that indicates the nature of the early
media. In any case, I personally don't believe a B2BUA or phone would ever
be smart enough to know which stream to pass through. Even the smart ones
usually allow the first stream and block the rest. The reason is simple - if
it were easy to do that, solving the PSTN question would also be easy.

On Wed, Mar 24, 2010 at 7:08 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>>My conclusion is that none of the suggestions being thrown around
> >>>fundamentally improve the handling of early media in any way. Its just a
> >>>mess, and there is no feasible solution.
> >>
> >>Arguably, not forking in a proxy helps. It least lets you know how to
> >>kill an early media stream that is being spewed back at you.
> >
> >Well, that depends.
> >
> >Its really only parallel forking thats a problem, so we can probably
> >ignore serial forking.
> >
> >For parallel forking, moving it from the proxy to the UAC doesn't really
> >solve the problem - you can still end up with multiple media streams and
> >no clue what to do with them.
> >
> >You can say "don't do that", but there are desirable features that
> >depend on it - namely "simultaneous ring". Potentially that can be
> >replaced with presence-sensitive ringing to choose the *right* on to
> >ring first, but that requires data that often isn't available.
> >
> >>Offerless INVITEs help more, since the media can't come back to you until
> the O/A has completed.
> >
> >That is just sticking your head in the sand. The media is still being
> generated, even if you don't know about it.
> >
> >It does have the advantage that you can associate the media stream to
> >the signaling source, but it still doesn't tell you which stream is
> >interesting to listen to.
>
> Exactly. And, in my opinion THAT is the big problem.
>
> Being able to associate a media stream with dialog is of course very
> useful, and there are ways to do that (including the one presented by Dean),
> but it doesn't solve the problem with multiple early media.
>
> >ISTM that if you want "simultaneous ring", and the things ringing are
> >often PSTN devices, there is still no solution. Perhaps there is some
> >possibility of automated media analysis - the pstn GW figuring out that
> >the media being received is something uninteresting - just conventional
> >ringback - and suppressing it. But thats a lot of trouble and is likely
> >to have a high error rate.
>
> Correct. And, as long as the protocol allows forking, I think it is not
> possible to define a protocol mechanism how to solve the problem.
>
> We can define information elements which contain information about the
> early media (eventhough it most likely won't work in the PSTN case), but at
> the end of the day you will need policy rules in order to decide what media
> is forwarded or played to the calling user.
>
> Regards,
>
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

--001485f44d1650e5c904828f8aa5
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<div dir=3D"ltr">Even if you would define policy, that doesn&#39;t mean it =
doesn&#39;t make sense to provide information on the nature of the media fr=
om non-PSTN sources. So you could say (in policy), for example, that PSTN i=
s more important than just a ringback tone, but not more important than an =
IVR.<br>

<br>My main question is whether we actually see in practice these scenarios=
 to warrant a header/parameter/attribute that indicates the nature of the e=
arly media. In any case, I personally don&#39;t believe a B2BUA or phone wo=
uld ever be smart enough to know which stream to pass through. Even the sma=
rt ones usually allow the first stream and block the rest. The reason is si=
mple - if it were easy to do that, solving the PSTN question would also be =
easy.<br>

<br><div class=3D"gmail_quote">On Wed, Mar 24, 2010 at 7:08 PM, Christer Ho=
lmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); marg=
in: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

Hi,<br>
<div class=3D"im"><br>
&gt;&gt;&gt;My conclusion is that none of the suggestions being thrown arou=
nd<br>
&gt;&gt;&gt;fundamentally improve the handling of early media in any way. I=
ts just a<br>
&gt;&gt;&gt;mess, and there is no feasible solution.<br>
&gt;&gt;<br>
&gt;&gt;Arguably, not forking in a proxy helps. It least lets you know how =
to<br>
&gt;&gt;kill an early media stream that is being spewed back at you.<br>
&gt;<br>
&gt;Well, that depends.<br>
&gt;<br>
&gt;Its really only parallel forking thats a problem, so we can probably<br=
>
&gt;ignore serial forking.<br>
&gt;<br>
&gt;For parallel forking, moving it from the proxy to the UAC doesn&#39;t r=
eally<br>
&gt;solve the problem - you can still end up with multiple media streams an=
d<br>
&gt;no clue what to do with them.<br>
&gt;<br>
&gt;You can say &quot;don&#39;t do that&quot;, but there are desirable feat=
ures that<br>
&gt;depend on it - namely &quot;simultaneous ring&quot;. Potentially that c=
an be<br>
&gt;replaced with presence-sensitive ringing to choose the *right* on to<br=
>
&gt;ring first, but that requires data that often isn&#39;t available.<br>
&gt;<br>
&gt;&gt;Offerless INVITEs help more, since the media can&#39;t come back to=
 you until the O/A has completed.<br>
&gt;<br>
&gt;That is just sticking your head in the sand. The media is still being g=
enerated, even if you don&#39;t know about it.<br>
&gt;<br>
&gt;It does have the advantage that you can associate the media stream to<b=
r>
&gt;the signaling source, but it still doesn&#39;t tell you which stream is=
<br>
&gt;interesting to listen to.<br>
<br>
</div>Exactly. And, in my opinion THAT is the big problem.<br>
<br>
Being able to associate a media stream with dialog is of course very useful=
, and there are ways to do that (including the one presented by Dean), but =
it doesn&#39;t solve the problem with multiple early media.<br>
<div class=3D"im"><br>
&gt;ISTM that if you want &quot;simultaneous ring&quot;, and the things rin=
ging are<br>
&gt;often PSTN devices, there is still no solution. Perhaps there is some<b=
r>
&gt;possibility of automated media analysis - the pstn GW figuring out that=
<br>
&gt;the media being received is something uninteresting - just conventional=
<br>
&gt;ringback - and suppressing it. But thats a lot of trouble and is likely=
<br>
&gt;to have a high error rate.<br>
<br>
</div>Correct. And, as long as the protocol allows forking, I think it is n=
ot possible to define a protocol mechanism how to solve the problem.<br>
<br>
We can define information elements which contain information about the earl=
y media (eventhough it most likely won&#39;t work in the PSTN case), but at=
 the end of the day you will need policy rules in order to decide what medi=
a is forwarded or played to the calling user.<br>


<br>
Regards,<br>
<font color=3D"#888888"><br>
Christer<br>
</font><div><div></div><div class=3D"h5">__________________________________=
_____________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br></div>

--001485f44d1650e5c904828f8aa5--

From christer.holmberg@ericsson.com  Wed Mar 24 11:06:08 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Gilad Shaham <gilad@voxisoft.com>
Date: Wed, 24 Mar 2010 19:01:47 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or 	not to fork?]
Thread-Index: AcrLekTsXipJy3I3SJePj6pnTPx3lgAAc0wn
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>Even if you would define policy, that doesn't mean it doesn't make sense t=
o provide information on the nature of the media from
>non-PSTN sources.

I agree. I think it would be very useful, and you would be able to define b=
etter and more reliable policies.

>So you could say (in policy), for example, that PSTN is more important tha=
n just a ringback tone, but not more important than an
>IVR.

Yes.

>My main question is whether we actually see in practice these scenarios to=
 warrant a header/parameter/attribute that indicates the
>nature of the early media. In any case, I personally don't believe a B2BUA=
 or phone would ever be smart enough to know which
>stream to pass through. Even the smart ones usually allow the first stream=
 and block the rest. The reason is simple - if it were easy
>to do that, solving the PSTN question would also be easy.

I haven't seen any headers/parameters, but at least in 3GPP there is curren=
tly an ongoing discussion on whether some header/parameter would be needed =
for some specific services using early media.

Regards,

Christer





On Wed, Mar 24, 2010 at 7:08 PM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

>>>My conclusion is that none of the suggestions being thrown around
>>>fundamentally improve the handling of early media in any way. Its just a
>>>mess, and there is no feasible solution.
>>
>>Arguably, not forking in a proxy helps. It least lets you know how to
>>kill an early media stream that is being spewed back at you.
>
>Well, that depends.
>
>Its really only parallel forking thats a problem, so we can probably
>ignore serial forking.
>
>For parallel forking, moving it from the proxy to the UAC doesn't really
>solve the problem - you can still end up with multiple media streams and
>no clue what to do with them.
>
>You can say "don't do that", but there are desirable features that
>depend on it - namely "simultaneous ring". Potentially that can be
>replaced with presence-sensitive ringing to choose the *right* on to
>ring first, but that requires data that often isn't available.
>
>>Offerless INVITEs help more, since the media can't come back to you until=
 the O/A has completed.
>
>That is just sticking your head in the sand. The media is still being gene=
rated, even if you don't know about it.
>
>It does have the advantage that you can associate the media stream to
>the signaling source, but it still doesn't tell you which stream is
>interesting to listen to.

Exactly. And, in my opinion THAT is the big problem.

Being able to associate a media stream with dialog is of course very useful=
, and there are ways to do that (including the one presented by Dean), but =
it doesn't solve the problem with multiple early media.

>ISTM that if you want "simultaneous ring", and the things ringing are
>often PSTN devices, there is still no solution. Perhaps there is some
>possibility of automated media analysis - the pstn GW figuring out that
>the media being received is something uninteresting - just conventional
>ringback - and suppressing it. But thats a lot of trouble and is likely
>to have a high error rate.

Correct. And, as long as the protocol allows forking, I think it is not pos=
sible to define a protocol mechanism how to solve the problem.

We can define information elements which contain information about the earl=
y media (eventhough it most likely won't work in the PSTN case), but at the=
 end of the day you will need policy rules in order to decide what media is=
 forwarded or played to the calling user.

Regards,

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


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From: Brett Tate <brett@broadsoft.com>
To: Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 24 Mar 2010 12:01:49 -0700
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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If rfc3959 was actually supported, doesn't it solve many of the forking rel=
ated early media issues?  And the use of 199 addresses the ambiguity of kno=
wing if an early dialog/session actually still exists.

What still needs to be addressed for early media to work when forking is in=
volved?

________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Dean=
 Willis [dean.willis@softarmor.com]
Sent: Tuesday, March 23, 2010 10:06 PM
To: Paul Kyzivat
Cc: SIPCORE
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork=
 or not to fork?]

Paul Kyzivat wrote:
> My conclusion is that none of the suggestions being thrown around
> fundamentally improve the handling of early media in any way. Its just a
> mess, and there is no feasible solution.

Arguably, not forking in a proxy helps. It least lets you know how to
kill an early media stream that is being spewed back at you.

Offerless INVITEs help more, since the media can't come back to you
until the O/A has completed.

What if we split the O/A stage into three phases:

1) pre-offer: Alice INVITES Bob, includes a description of media she can
send and receive but not an IP address or port number

2) offer: Bob sends Alice a description of the media he expects to send
and to receive, along with the IP addresses and port numbers he is going
to use.

3) answer: Alice sends Bob the IP addresses and port number to whichs he
should direct media and from which Alice will be sending media.

Note that this eliminates early media. Bob can't send Alice any media
until he receives the answer.


--
Dean
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

From christer.holmberg@ericsson.com  Wed Mar 24 15:10:51 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Brett Tate <brett@broadsoft.com>, Dean Willis <dean.willis@softarmor.com>,  Paul Kyzivat <pkyzivat@cisco.com>, SIPCORE <sipcore@ietf.org>
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Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>If rfc3959 was actually supported, doesn't it solve many of the forking re=
lated early media issues?

3959 allows you to provide an early-media specific SDP (there are other way=
s of doing that also).

The early-session SDP also works as an indicator that there will be early m=
edia in the first place, but there are other ways of doing that also (e.g. =
using the P-Early-Media header).

But, the problem is still there: there might be multiple early media stream=
s.

>And the use of 199 addresses the ambiguity of knowing if an early dialog/s=
ession actually still exists.

True. It helps to indicate on which dialogs there will NOT be any additiona=
l early media.

But, it also doesn't solve the issue of having multiple active early media =
streams.

>What still needs to be addressed for early media to work when forking is i=
nvolved?

I think the main problem is having multiple early media streams. How do you=
 decide which to play or forward to the calling user?

Regards,

Christer



________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Dean=
 Willis [dean.willis@softarmor.com]
Sent: Tuesday, March 23, 2010 10:06 PM
To: Paul Kyzivat
Cc: SIPCORE
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork=
 or not to fork?]

Paul Kyzivat wrote:
> My conclusion is that none of the suggestions being thrown around
> fundamentally improve the handling of early media in any way. Its just a
> mess, and there is no feasible solution.

Arguably, not forking in a proxy helps. It least lets you know how to
kill an early media stream that is being spewed back at you.

Offerless INVITEs help more, since the media can't come back to you
until the O/A has completed.

What if we split the O/A stage into three phases:

1) pre-offer: Alice INVITES Bob, includes a description of media she can
send and receive but not an IP address or port number

2) offer: Bob sends Alice a description of the media he expects to send
and to receive, along with the IP addresses and port numbers he is going
to use.

3) answer: Alice sends Bob the IP addresses and port number to whichs he
should direct media and from which Alice will be sending media.

Note that this eliminates early media. Bob can't send Alice any media
until he receives the answer.


--
Dean
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

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Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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> > What still needs to be addressed for early media to work when forking i=
s involved?
>=20
> I think the main problem is having multiple early media streams. How do y=
ou decide=20
> which to play or forward to the calling user?

RFC 3959 basically allows the caller or caller's device to decide.  It can =
play, ignore, reject, or conference the early media and then switch to the =
answered media when appropriate.

Concerning "forward to the calling user", they all can be forwarded unless =
there is a middle box attempting to control things with or without input fr=
om calling party.  Without input from the calling party, I agree that it ca=
n potentially lead to non desirable experience for the user.

If the user's device and outbound middle boxes are not able to handle the s=
imultaneous media for RFC 3959, I assume something could be defined (if cal=
ler prefs is not adequate) to allow calling party and/or middle boxes to fo=
rce/request sequential forking.  However similar to solely relying upon 3xx=
's and attempting targets sequentially, this can lead to long call-setup ti=
mes before finally reaching voice mail or a party likely to answer.  In com=
parison to a 3xx solution, the sequential solution at least still allows th=
e calling party to maintain a little more control over the order/timing rel=
ated to trying the various targets.=

From gilad@voxisoft.com  Thu Mar 25 10:36:09 2010
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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In my experience, many implementations may actually do wrong things when
faced with multipart containing several SDPs. They should not, but in
practice this happens. Even if we do choose this over any other mechanism
proposed (such as 3xx), I would recommend we do not rely on RFC 3959 to
indicate the nature of the media.

On Thu, Mar 25, 2010 at 6:53 PM, Brett Tate <brett@broadsoft.com> wrote:

> > > What still needs to be addressed for early media to work when forking
> is involved?
> >
> > I think the main problem is having multiple early media streams. How do
> you decide
> > which to play or forward to the calling user?
>
> RFC 3959 basically allows the caller or caller's device to decide.  It can
> play, ignore, reject, or conference the early media and then switch to the
> answered media when appropriate.
>
> Concerning "forward to the calling user", they all can be forwarded unless
> there is a middle box attempting to control things with or without input
> from calling party.  Without input from the calling party, I agree that it
> can potentially lead to non desirable experience for the user.
>
> If the user's device and outbound middle boxes are not able to handle the
> simultaneous media for RFC 3959, I assume something could be defined (if
> caller prefs is not adequate) to allow calling party and/or middle boxes to
> force/request sequential forking.  However similar to solely relying upon
> 3xx's and attempting targets sequentially, this can lead to long call-setup
> times before finally reaching voice mail or a party likely to answer.  In
> comparison to a 3xx solution, the sequential solution at least still allows
> the calling party to maintain a little more control over the order/timing
> related to trying the various targets.
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr">In my experience, many implementations may actually do wro=
ng things when faced with multipart containing several SDPs. They should no=
t, but in practice this happens. Even if we do choose this over any other m=
echanism proposed (such as 3xx), I would recommend we do not rely on RFC 39=
59 to indicate the nature of the media.<br>

<br><div class=3D"gmail_quote">On Thu, Mar 25, 2010 at 6:53 PM, Brett Tate =
<span dir=3D"ltr">&lt;<a href=3D"mailto:brett@broadsoft.com">brett@broadsof=
t.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"b=
order-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; paddin=
g-left: 1ex;">

<div class=3D"im">&gt; &gt; What still needs to be addressed for early medi=
a to work when forking is involved?<br>
&gt;<br>
&gt; I think the main problem is having multiple early media streams. How d=
o you decide<br>
&gt; which to play or forward to the calling user?<br>
<br>
</div>RFC 3959 basically allows the caller or caller&#39;s device to decide=
. =A0It can play, ignore, reject, or conference the early media and then sw=
itch to the answered media when appropriate.<br>
<br>
Concerning &quot;forward to the calling user&quot;, they all can be forward=
ed unless there is a middle box attempting to control things with or withou=
t input from calling party. =A0Without input from the calling party, I agre=
e that it can potentially lead to non desirable experience for the user.<br=
>


<br>
If the user&#39;s device and outbound middle boxes are not able to handle t=
he simultaneous media for RFC 3959, I assume something could be defined (if=
 caller prefs is not adequate) to allow calling party and/or middle boxes t=
o force/request sequential forking. =A0However similar to solely relying up=
on 3xx&#39;s and attempting targets sequentially, this can lead to long cal=
l-setup times before finally reaching voice mail or a party likely to answe=
r. =A0In comparison to a 3xx solution, the sequential solution at least sti=
ll allows the calling party to maintain a little more control over the orde=
r/timing related to trying the various targets.<br>


<div><div></div><div class=3D"h5">_________________________________________=
______<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br></div>

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From dean.willis@softarmor.com  Thu Mar 25 11:12:06 2010
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Cc: 'SIPCORE' <sipcore@ietf.org>, Gilad Shaham <gilad@voxisoft.com>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Christer Holmberg wrote:
>> No, I'm suggesting that all of the forks respond with a 200 OK and
>> move to a connected state, not just the first fork to respond. It's
>> not a race, and there is NO early media -- just media being sent
>> before the PSTN side of the gateway moves to a fully-connected
>> state.
>> 
>> Sorting out multiple media streams is a separate problem, but there
>> are ways to automate many cases in a B2BUA or really smart UA.
>> 
>> But this breaks the business model, which assumes that everything 
>> everywhere is fully-connected and in a "billing starts" state when
>> the first 200OK flies.
> 
> But, what problem would that solve?
> 
> The only difference is in what state you are (established-dialog
> instead of early-dialog), but you still have the same problem as
> duing the early dialog phase: you might receive early media from
> multiple locations.
> 


It's not "early media" if it is in an established dialog. Once you have
a full dialog, you can do  things like send a "BYE", have a bot do a
verbal challenge-response on the media path, play one channel to the
left ear and one to the right, play all of the media flows
simultaneously with variable volume weighting using a hover-touch over a
per-session icon (click to pick "this call" and drop the other ones),
etc. But "early media" messes everything up.

--
dean

From dean.willis@softarmor.com  Thu Mar 25 11:22:05 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Paul Kyzivat wrote:

> Its really only parallel forking thats a problem, so we can probably
> ignore serial forking.

Not really. Early media is always a problem. How do you tell it to stop
without whacking all your forks?

> 
> For parallel forking, moving it from the proxy to the UAC doesn't really
> solve the problem - you can still end up with multiple media streams and
> no clue what to do with them.

Knowing which stream is which lets you do MORE things than you can with
just a bunch of garbled media coming in.


> 
> You can say "don't do that", but there are desirable features that
> depend on it - namely "simultaneous ring". Potentially that can be
> replaced with presence-sensitive ringing to choose the *right* on to
> ring first, but that requires data that often isn't available.

Google Voice does simultaneous ring quite nicely. If you call my GV
number, it will ring all my phones. If somebody answers a phone, GV will
say "Press 1 to accept a call from Paul Kyzivat", and if I press 1, GV
will terminate the other forks.

Webster did this years ago using the dynamicsoft stuff as a call controller.

But to do this kind of work, the call controller needs signaling control
on each fork, which you don't get with early media.

>> Offerless INVITEs help more, since the media can't come back to you
>> until the O/A has completed.
> 
> That is just sticking your head in the sand. The media is still being
> generated, even if you don't know about it.

No, really. They CANNOT send you media until you've sent your SDP.  If
you don't send your SDP (in the ACK) until they have sent theirs (in the
200OK), then there is NO EARLY MEDIA.

Yes, this means that PSTN gateways would have to complete ACK before
doing IAM on the PSTN side. I'm OK with that.

> It does have the advantage that you can associate the media stream to
> the signaling source, but it still doesn't tell you which stream is
> interesting to listen to.

But having associated signaling lets you do things to decide which one
is interesting.

> ISTM that if you want "simultaneous ring", and the things ringing are
> often PSTN devices, there is still no solution. Perhaps there is some
> possibility of automated media analysis - the pstn GW figuring out that
> the media being received is something uninteresting - just conventional
> ringback - and suppressing it. But thats a lot of trouble and is likely
> to have a high error rate.

Go play with Google Voice or something like it for awhile and then get
back to us.

--
Dean

From pkyzivat@cisco.com  Thu Mar 25 12:07:50 2010
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Date: Thu, 25 Mar 2010 15:08:03 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Brett Tate <brett@broadsoft.com>
References: <FF84A09F50A6DC48ACB6714F4666CC745C10EE415F@ESESSCMS0354.eemea.ericsson.se>	<AEA0E9BF-8B6A-4AAA-81B6-A7C68F6B7309@softarmor.com>	<FF84A09F50A6DC48ACB6714F4666CC745C1102CCCD@ESESSCMS0354.eemea.ericsson.se>	<4B918DED.6090108@cisco.com>	<FF84A09F50A6DC48ACB6714F4666CC745C1102CCD1@ESESSCMS0354.eemea.ericsson.se>, <4B919413.60506@cisco.com>	<FF84A09F50A6DC48ACB6714F4666CC745C10EE4166@ESESSCMS0354.eemea.ericsson.se>	<4B93D8A6.1040105@cisco.com>	<FF84A09F50A6DC48ACB6714F4666CC745E21B5A8D4@ESESSCMS0354.eemea.ericsson.se>	<4BA94A93.4010909@cisco.com>, <4BA9739B.9060903@softarmor.com>, <747A6506A991724FB09B129B79D5FEB614808FB60B@EXMBXCLUS01.citservers.local>, <FF84A09F50A6DC48ACB6714F4666CC745E21B30A3C@ESESSCMS0354.eemea.ericsson.se> <747A6506A991724FB09B129B79D5FEB614808FB60D@EXMBXCLUS01.citservers.local>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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[speaking as an individual]

Brett,

Its true that the caller MAY choose to render whatever it wants from the 
multiple media streams coming back. And thats fine if none of them 
contain anything of consequence.

But it doesn't alter reality. One, or more, of those early media streams 
may contain information that the caller needs to hear. When the caller's 
UA chooses, it may pick one that is unimportant over one that is 
important. And even picking one that is important isn't good enough if 
another was also important.

If the streams were identifiable (by automata) as to content, then at 
least an important one might be chosen over an unimportant one. I can 
only see two answers for the case where more than one has important content:

- indicate to the caller that there are multiple, and permit the
   caller to toggle between them

- indicate to the callee that its early media will not be rendered.
   Force it to wait to send until such time as it can be rendered.

Those two are actually complementary.

Unfortunately, the pstn media is probably by far the most common case. 
And with it:
- we have no indicatino of what the content is (do we?)
- indicating refusal to accept the media will not delay it,
   only cause it to be dropped.

	Thanks,
	Paul

Brett Tate wrote:
>>> What still needs to be addressed for early media to work when forking is involved?
>> I think the main problem is having multiple early media streams. How do you decide 
>> which to play or forward to the calling user?
> 
> RFC 3959 basically allows the caller or caller's device to decide.  It can play, ignore, reject, or conference the early media and then switch to the answered media when appropriate.
> 
> Concerning "forward to the calling user", they all can be forwarded unless there is a middle box attempting to control things with or without input from calling party.  Without input from the calling party, I agree that it can potentially lead to non desirable experience for the user.
> 
> If the user's device and outbound middle boxes are not able to handle the simultaneous media for RFC 3959, I assume something could be defined (if caller prefs is not adequate) to allow calling party and/or middle boxes to force/request sequential forking.  However similar to solely relying upon 3xx's and attempting targets sequentially, this can lead to long call-setup times before finally reaching voice mail or a party likely to answer.  In comparison to a 3xx solution, the sequential solution at least still allows the calling party to maintain a little more control over the order/timing related to trying the various targets.
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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From jon.peterson@neustar.biz  Thu Mar 25 14:27:48 2010
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Date: Thu, 25 Mar 2010 14:27:55 -0700
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Subject: [sipcore] rebooting location-conveyance
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James Polk and I sat down this week and took a hard look at the current s=
ipcore-location-conveyance draft, and we believe that there's an opportun=
ity to simplify (and shorten) the document.

The simplification derives mostly from a single change to the operation o=
f the mechanism. Today, the location-conveyance draft has a good deal of =
machinery to support the case where a proxy wants to replace existing loc=
ation information in a request with more accurate or preferred location i=
nformation. The current approach replaces the URI in the Geolocation head=
er (which probably points to a MIME body) with a URI desired by the proxy=
, presumably of an Internet resource containing the desired PIDF-LO body.=
 This solution isn't optimal for several reasons - most notably, since th=
e MIME body is not removed by said proxy, it will still be delivered to t=
he eventual location recipient, even if the URI in the Geolocation header=
 does not point to said body. In order to handle this case, the current d=
raft employs a disambiguation mechanism (relying on the "used-for-routing=
" and "inserted-by" parameters, as well as the "inserter") to inform the =
location recipient about the source of this location information. Moreove=
r, since this change is effected unilaterally, the originating user agent=
 has no idea that the location it supplied has been altered in transit, a=
nd may thus receive a surprising response to its request.

While this use case is an important one, we believe it can be addressed w=
ith a lighter mechanism. The basic idea is that a proxy server that wants=
 to replace the location information provided by a user agent would rejec=
t a request with the 424 Bad Location response code, optionally enclosing=
 in the body the location information that it would like the user agent t=
o incorporate.  The UA can insert the LI from the 424 in its subsequent r=
eissuance of the request; if no LI is supplied in the 424, then the user =
agent resends the request without LI. This brings the location-conveyance=
 mechanism back to one its original constraints: that only one PIDF-LO bo=
dy is associated with a particular SIP request. This allows us to remove =
the "used-for-routing" parameter as well as the "inserted-by"/"inserter" =
mechanism.

In a related clean-up, several of the error codes can probably be consoli=
dated or replaced with SIP response codes (for example, the Retry-After s=
emantics of the 200/300 error codes).

If no one objects, we'd therefore like to try to come back with a new ver=
sion that has fewer moving parts and fewer pages.

Thoughts?

Jon Peterson
NeuStar, Inc.



From keith.drage@alcatel-lucent.com  Thu Mar 25 14:54:27 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Jon Peterson <jon.peterson@neustar.biz>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 25 Mar 2010 22:54:44 +0100
Thread-Topic: [sipcore] rebooting location-conveyance
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Subject: Re: [sipcore] rebooting location-conveyance
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So, I believe the current solution allows a scenario where the originating =
UA inserts a location, and various proxies along the path could insert a lo=
cation, in addition to the existing one, providing they all represent their=
 view of the state of the user.

By making this proposal, are you eliminating this scenario, or allowing it =
to continue to exist.

regards

Keith

=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> Sent: Thursday, March 25, 2010 9:28 PM
> To: sipcore@ietf.org
> Subject: [sipcore] rebooting location-conveyance
>=20
> James Polk and I sat down this week and took a hard look at=20
> the current sipcore-location-conveyance draft, and we believe=20
> that there's an opportunity to simplify (and shorten) the document.
>=20
> The simplification derives mostly from a single change to the=20
> operation of the mechanism. Today, the location-conveyance=20
> draft has a good deal of machinery to support the case where=20
> a proxy wants to replace existing location information in a=20
> request with more accurate or preferred location information.=20
> The current approach replaces the URI in the Geolocation=20
> header (which probably points to a MIME body) with a URI=20
> desired by the proxy, presumably of an Internet resource=20
> containing the desired PIDF-LO body. This solution isn't=20
> optimal for several reasons - most notably, since the MIME=20
> body is not removed by said proxy, it will still be delivered=20
> to the eventual location recipient, even if the URI in the=20
> Geolocation header does not point to said body. In order to=20
> handle this case, the current draft employs a disambiguation=20
> mechanism (relying on the "used-for-routing" and=20
> "inserted-by" parameters, as well as the "inserter") to=20
> inform the location recipient about the source of th  is=20
> location information. Moreover, since this change is effected=20
> unilaterally, the originating user agent has no idea that the=20
> location it supplied has been altered in transit, and may=20
> thus receive a surprising response to its request.
>=20
> While this use case is an important one, we believe it can be=20
> addressed with a lighter mechanism. The basic idea is that a=20
> proxy server that wants to replace the location information=20
> provided by a user agent would reject a request with the 424=20
> Bad Location response code, optionally enclosing in the body=20
> the location information that it would like the user agent to=20
> incorporate.  The UA can insert the LI from the 424 in its=20
> subsequent reissuance of the request; if no LI is supplied in=20
> the 424, then the user agent resends the request without LI.=20
> This brings the location-conveyance mechanism back to one its=20
> original constraints: that only one PIDF-LO body is=20
> associated with a particular SIP request. This allows us to=20
> remove the "used-for-routing" parameter as well as the=20
> "inserted-by"/"inserter" mechanism.
>=20
> In a related clean-up, several of the error codes can=20
> probably be consolidated or replaced with SIP response codes=20
> (for example, the Retry-After semantics of the 200/300 error codes).
>=20
> If no one objects, we'd therefore like to try to come back=20
> with a new version that has fewer moving parts and fewer pages.
>=20
> Thoughts?
>=20
> Jon Peterson
> NeuStar, Inc.
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From Martin.Thomson@andrew.com  Thu Mar 25 17:05:35 2010
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From: "Thomson, Martin" <Martin.Thomson@andrew.com>
To: Jon Peterson <jon.peterson@neustar.biz>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 26 Mar 2010 07:42:59 +0800
Thread-Topic: [sipcore] rebooting location-conveyance
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From jon.peterson@neustar.biz  Thu Mar 25 17:08:56 2010
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From: Jon Peterson <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] rebooting location-conveyance
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Semantically, a single SIP request can contain multiple locations within 
the constraints of the presence data model - PIDF-LO is after all PIDF, 
and like any presence document it can make fairly complicated statements 
about the relationship of location information to users or devices, 
including enumerating multiple locations. However, the intention of the 
revision is to return this to an old design constraint where a SIP 
request may contain only a single Geolocation header field with one URI 
pointing to location information.

Jon Peterson
NeuStar, Inc.

DRAGE, Keith (Keith) wrote:
> So, I believe the current solution allows a scenario where the originating UA inserts a location, and various proxies along the path could insert a location, in addition to the existing one, providing they all represent their view of the state of the user.
>
> By making this proposal, are you eliminating this scenario, or allowing it to continue to exist.
>
> regards
>
> Keith
>
>  
>
>   
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org 
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
>> Sent: Thursday, March 25, 2010 9:28 PM
>> To: sipcore@ietf.org
>> Subject: [sipcore] rebooting location-conveyance
>>
>> James Polk and I sat down this week and took a hard look at 
>> the current sipcore-location-conveyance draft, and we believe 
>> that there's an opportunity to simplify (and shorten) the document.
>>
>> The simplification derives mostly from a single change to the 
>> operation of the mechanism. Today, the location-conveyance 
>> draft has a good deal of machinery to support the case where 
>> a proxy wants to replace existing location information in a 
>> request with more accurate or preferred location information. 
>> The current approach replaces the URI in the Geolocation 
>> header (which probably points to a MIME body) with a URI 
>> desired by the proxy, presumably of an Internet resource 
>> containing the desired PIDF-LO body. This solution isn't 
>> optimal for several reasons - most notably, since the MIME 
>> body is not removed by said proxy, it will still be delivered 
>> to the eventual location recipient, even if the URI in the 
>> Geolocation header does not point to said body. In order to 
>> handle this case, the current draft employs a disambiguation 
>> mechanism (relying on the "used-for-routing" and 
>> "inserted-by" parameters, as well as the "inserter") to 
>> inform the location recipient about the source of th  is 
>> location information. Moreover, since this change is effected 
>> unilaterally, the originating user agent has no idea that the 
>> location it supplied has been altered in transit, and may 
>> thus receive a surprising response to its request.
>>
>> While this use case is an important one, we believe it can be 
>> addressed with a lighter mechanism. The basic idea is that a 
>> proxy server that wants to replace the location information 
>> provided by a user agent would reject a request with the 424 
>> Bad Location response code, optionally enclosing in the body 
>> the location information that it would like the user agent to 
>> incorporate.  The UA can insert the LI from the 424 in its 
>> subsequent reissuance of the request; if no LI is supplied in 
>> the 424, then the user agent resends the request without LI. 
>> This brings the location-conveyance mechanism back to one its 
>> original constraints: that only one PIDF-LO body is 
>> associated with a particular SIP request. This allows us to 
>> remove the "used-for-routing" parameter as well as the 
>> "inserted-by"/"inserter" mechanism.
>>
>> In a related clean-up, several of the error codes can 
>> probably be consolidated or replaced with SIP response codes 
>> (for example, the Retry-After semantics of the 200/300 error codes).
>>
>> If no one objects, we'd therefore like to try to come back 
>> with a new version that has fewer moving parts and fewer pages.
>>
>> Thoughts?
>>
>> Jon Peterson
>> NeuStar, Inc.
>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>     


From jon.peterson@neustar.biz  Thu Mar 25 17:23:28 2010
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From: Jon Peterson <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] rebooting location-conveyance
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Broadly, any use case where you want to deliver two different locations 
to the same recipient is one that can be expressed semantically in 
PIDF-LO. I think that trying to implement this sort of semantics in the 
SIP layer has to be the wrong approach - it is far less brittle to 
extend PIDF than to add this to SIP. Ultimately, a by value PIDF-LO MIME 
body in a SIP request could, for example, contain a URI where someone 
could go to get location by reference in the future. That sounds like it 
could cover your use case at a high level, though of course we'd have to 
explore that in more detail.

Jon Peterson
NeuStar, Inc.

Thomson, Martin wrote:
> Hi Jon,
>
> I'd welcome any attempt to make this document simpler.  Over time, it's built up a fair amount of baggage as it's made it's way through four (?) working groups.  This sounds like a reasonable cut.
>
> The use case that is a victim of this is the one where a host has two locations: by value and by reference that they want to include.  This might be the "now" and "for later" use case, or it could be the location hiding use case where the value is poor quality and the reference might be used to get the really good stuff.  As long as we are able to address that use case, then I'm happy.
>
> I don't think that we need to re-build all that stuff that you are stripping out, but we currently don't have a good story for that.
>
> --Martin
>
>   
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf Of Jon Peterson
>> Sent: Thursday, 25 March 2010 2:28 PM
>> To: sipcore@ietf.org
>> Subject: [sipcore] rebooting location-conveyance
>>
>> James Polk and I sat down this week and took a hard look at the current
>> sipcore-location-conveyance draft, and we believe that there's an
>> opportunity to simplify (and shorten) the document.
>>
>> The simplification derives mostly from a single change to the operation
>> of the mechanism. Today, the location-conveyance draft has a good deal
>> of machinery to support the case where a proxy wants to replace
>> existing location information in a request with more accurate or
>> preferred location information. The current approach replaces the URI
>> in the Geolocation header (which probably points to a MIME body) with a
>> URI desired by the proxy, presumably of an Internet resource containing
>> the desired PIDF-LO body. This solution isn't optimal for several
>> reasons - most notably, since the MIME body is not removed by said
>> proxy, it will still be delivered to the eventual location recipient,
>> even if the URI in the Geolocation header does not point to said body.
>> In order to handle this case, the current draft employs a
>> disambiguation mechanism (relying on the "used-for-routing" and
>> "inserted-by" parameters, as well as the "inserter") to inform the
>> location recipient about the source of th
>>  is location information. Moreover, since this change is effected
>> unilaterally, the originating user agent has no idea that the location
>> it supplied has been altered in transit, and may thus receive a
>> surprising response to its request.
>>
>> While this use case is an important one, we believe it can be addressed
>> with a lighter mechanism. The basic idea is that a proxy server that
>> wants to replace the location information provided by a user agent
>> would reject a request with the 424 Bad Location response code,
>> optionally enclosing in the body the location information that it would
>> like the user agent to incorporate.  The UA can insert the LI from the
>> 424 in its subsequent reissuance of the request; if no LI is supplied
>> in the 424, then the user agent resends the request without LI. This
>> brings the location-conveyance mechanism back to one its original
>> constraints: that only one PIDF-LO body is associated with a particular
>> SIP request. This allows us to remove the "used-for-routing" parameter
>> as well as the "inserted-by"/"inserter" mechanism.
>>
>> In a related clean-up, several of the error codes can probably be
>> consolidated or replaced with SIP response codes (for example, the
>> Retry-After semantics of the 200/300 error codes).
>>
>> If no one objects, we'd therefore like to try to come back with a new
>> version that has fewer moving parts and fewer pages.
>>
>> Thoughts?
>>
>> Jon Peterson
>> NeuStar, Inc.
>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>     


From keith.drage@alcatel-lucent.com  Thu Mar 25 18:19:33 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Jon Peterson <jon.peterson@neustar.biz>
Date: Fri, 26 Mar 2010 02:19:33 +0100
Thread-Topic: [sipcore] rebooting location-conveyance
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] rebooting location-conveyance
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So on the basis that your answer appears to be NO to my question, I believe=
 I cannot support the change you are proposing.

regards

Keith=20

> -----Original Message-----
> From: Jon Peterson [mailto:jon.peterson@neustar.biz]=20
> Sent: Friday, March 26, 2010 12:09 AM
> To: DRAGE, Keith (Keith)
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] rebooting location-conveyance
>=20
>=20
> Semantically, a single SIP request can contain multiple=20
> locations within the constraints of the presence data model -=20
> PIDF-LO is after all PIDF, and like any presence document it=20
> can make fairly complicated statements about the relationship=20
> of location information to users or devices, including=20
> enumerating multiple locations. However, the intention of the=20
> revision is to return this to an old design constraint where=20
> a SIP request may contain only a single Geolocation header=20
> field with one URI pointing to location information.
>=20
> Jon Peterson
> NeuStar, Inc.
>=20
> DRAGE, Keith (Keith) wrote:
> > So, I believe the current solution allows a scenario where=20
> the originating UA inserts a location, and various proxies=20
> along the path could insert a location, in addition to the=20
> existing one, providing they all represent their view of the=20
> state of the user.
> >
> > By making this proposal, are you eliminating this scenario,=20
> or allowing it to continue to exist.
> >
> > regards
> >
> > Keith
> >
> > =20
> >
> >  =20
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org
> >> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> >> Sent: Thursday, March 25, 2010 9:28 PM
> >> To: sipcore@ietf.org
> >> Subject: [sipcore] rebooting location-conveyance
> >>
> >> James Polk and I sat down this week and took a hard look at the=20
> >> current sipcore-location-conveyance draft, and we believe that=20
> >> there's an opportunity to simplify (and shorten) the document.
> >>
> >> The simplification derives mostly from a single change to the=20
> >> operation of the mechanism. Today, the location-conveyance=20
> draft has=20
> >> a good deal of machinery to support the case where a proxy=20
> wants to=20
> >> replace existing location information in a request with=20
> more accurate=20
> >> or preferred location information.
> >> The current approach replaces the URI in the Geolocation header=20
> >> (which probably points to a MIME body) with a URI desired by the=20
> >> proxy, presumably of an Internet resource containing the desired=20
> >> PIDF-LO body. This solution isn't optimal for several=20
> reasons - most=20
> >> notably, since the MIME body is not removed by said proxy, it will=20
> >> still be delivered to the eventual location recipient, even if the=20
> >> URI in the Geolocation header does not point to said body.=20
> In order=20
> >> to handle this case, the current draft employs a disambiguation=20
> >> mechanism (relying on the "used-for-routing" and "inserted-by"=20
> >> parameters, as well as the "inserter") to inform the location=20
> >> recipient about the source of th  is location information.=20
> Moreover,=20
> >> since this change is effected unilaterally, the originating user=20
> >> agent has no idea that the location it supplied has been=20
> altered in=20
> >> transit, and may thus receive a surprising response to its request.
> >>
> >> While this use case is an important one, we believe it can be=20
> >> addressed with a lighter mechanism. The basic idea is that a proxy=20
> >> server that wants to replace the location information=20
> provided by a=20
> >> user agent would reject a request with the 424 Bad=20
> Location response=20
> >> code, optionally enclosing in the body the location=20
> information that=20
> >> it would like the user agent to incorporate.  The UA can=20
> insert the=20
> >> LI from the 424 in its subsequent reissuance of the=20
> request; if no LI=20
> >> is supplied in the 424, then the user agent resends the request=20
> >> without LI.
> >> This brings the location-conveyance mechanism back to one its=20
> >> original constraints: that only one PIDF-LO body is=20
> associated with a=20
> >> particular SIP request. This allows us to remove the=20
> >> "used-for-routing" parameter as well as the=20
> "inserted-by"/"inserter"=20
> >> mechanism.
> >>
> >> In a related clean-up, several of the error codes can probably be=20
> >> consolidated or replaced with SIP response codes (for example, the=20
> >> Retry-After semantics of the 200/300 error codes).
> >>
> >> If no one objects, we'd therefore like to try to come back=20
> with a new=20
> >> version that has fewer moving parts and fewer pages.
> >>
> >> Thoughts?
> >>
> >> Jon Peterson
> >> NeuStar, Inc.
> >>
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> >>    =20
>=20
> =

From fmenard@xittelecom.com  Thu Mar 25 19:09:19 2010
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Cc: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] rebooting location-conveyance
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I am concerned that this change may impact the ability for end-to-end =
location conveyance to work.

Am I wrong?

F.

On 2010-03-25, at 8:09 PM, Jon Peterson wrote:

>=20
> Semantically, a single SIP request can contain multiple locations =
within the constraints of the presence data model - PIDF-LO is after all =
PIDF, and like any presence document it can make fairly complicated =
statements about the relationship of location information to users or =
devices, including enumerating multiple locations. However, the =
intention of the revision is to return this to an old design constraint =
where a SIP request may contain only a single Geolocation header field =
with one URI pointing to location information.
>=20
> Jon Peterson
> NeuStar, Inc.
>=20
> DRAGE, Keith (Keith) wrote:
>> So, I believe the current solution allows a scenario where the =
originating UA inserts a location, and various proxies along the path =
could insert a location, in addition to the existing one, providing they =
all represent their view of the state of the user.
>>=20
>> By making this proposal, are you eliminating this scenario, or =
allowing it to continue to exist.
>>=20
>> regards
>>=20
>> Keith
>>=20
>>=20
>> =20
>>> -----Original Message-----
>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On =
Behalf Of Jon Peterson
>>> Sent: Thursday, March 25, 2010 9:28 PM
>>> To: sipcore@ietf.org
>>> Subject: [sipcore] rebooting location-conveyance
>>>=20
>>> James Polk and I sat down this week and took a hard look at the =
current sipcore-location-conveyance draft, and we believe that there's =
an opportunity to simplify (and shorten) the document.
>>>=20
>>> The simplification derives mostly from a single change to the =
operation of the mechanism. Today, the location-conveyance draft has a =
good deal of machinery to support the case where a proxy wants to =
replace existing location information in a request with more accurate or =
preferred location information. The current approach replaces the URI in =
the Geolocation header (which probably points to a MIME body) with a URI =
desired by the proxy, presumably of an Internet resource containing the =
desired PIDF-LO body. This solution isn't optimal for several reasons - =
most notably, since the MIME body is not removed by said proxy, it will =
still be delivered to the eventual location recipient, even if the URI =
in the Geolocation header does not point to said body. In order to =
handle this case, the current draft employs a disambiguation mechanism =
(relying on the "used-for-routing" and "inserted-by" parameters, as well =
as the "inserter") to inform the location recipient about the source of =
th  is location information. Moreover, since this change is effected =
unilaterally, the originating user agent has no idea that the location =
it supplied has been altered in transit, and may thus receive a =
surprising response to its request.
>>>=20
>>> While this use case is an important one, we believe it can be =
addressed with a lighter mechanism. The basic idea is that a proxy =
server that wants to replace the location information provided by a user =
agent would reject a request with the 424 Bad Location response code, =
optionally enclosing in the body the location information that it would =
like the user agent to incorporate.  The UA can insert the LI from the =
424 in its subsequent reissuance of the request; if no LI is supplied in =
the 424, then the user agent resends the request without LI. This brings =
the location-conveyance mechanism back to one its original constraints: =
that only one PIDF-LO body is associated with a particular SIP request. =
This allows us to remove the "used-for-routing" parameter as well as the =
"inserted-by"/"inserter" mechanism.
>>>=20
>>> In a related clean-up, several of the error codes can probably be =
consolidated or replaced with SIP response codes (for example, the =
Retry-After semantics of the 200/300 error codes).
>>>=20
>>> If no one objects, we'd therefore like to try to come back with a =
new version that has fewer moving parts and fewer pages.
>>>=20
>>> Thoughts?
>>>=20
>>> Jon Peterson
>>> NeuStar, Inc.
>>>=20
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>   =20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Thomson, Martin" <Martin.Thomson@andrew.com>
To: "Francois D. Menard" <fmenard@xittelecom.com>, Jon Peterson <jon.peterson@neustar.biz>
Date: Fri, 26 Mar 2010 17:52:20 +0800
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Cc: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] rebooting location-conveyance
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From john.elwell@siemens-enterprise.com  Fri Mar 26 07:47:54 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Jon Peterson <jon.peterson@neustar.biz>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 26 Mar 2010 15:45:54 +0100
Thread-Topic: [sipcore] rebooting location-conveyance
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Subject: Re: [sipcore] rebooting location-conveyance
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What do the emergency call people think about having to retry, with a possi=
ble impact on call establishment times?

John=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> Sent: 25 March 2010 14:28
> To: sipcore@ietf.org
> Subject: [sipcore] rebooting location-conveyance
>=20
> James Polk and I sat down this week and took a hard look at=20
> the current sipcore-location-conveyance draft, and we believe=20
> that there's an opportunity to simplify (and shorten) the document.
>=20
> The simplification derives mostly from a single change to the=20
> operation of the mechanism. Today, the location-conveyance=20
> draft has a good deal of machinery to support the case where=20
> a proxy wants to replace existing location information in a=20
> request with more accurate or preferred location information.=20
> The current approach replaces the URI in the Geolocation=20
> header (which probably points to a MIME body) with a URI=20
> desired by the proxy, presumably of an Internet resource=20
> containing the desired PIDF-LO body. This solution isn't=20
> optimal for several reasons - most notably, since the MIME=20
> body is not removed by said proxy, it will still be delivered=20
> to the eventual location recipient, even if the URI in the=20
> Geolocation header does not point to said body. In order to=20
> handle this case, the current draft employs a disambiguation=20
> mechanism (relying on the "used-for-routing" and=20
> "inserted-by" parameters, as well as the "inserter") to=20
> inform the location recipient about the source of th
>  is location information. Moreover, since this change is=20
> effected unilaterally, the originating user agent has no idea=20
> that the location it supplied has been altered in transit,=20
> and may thus receive a surprising response to its request.
>=20
> While this use case is an important one, we believe it can be=20
> addressed with a lighter mechanism. The basic idea is that a=20
> proxy server that wants to replace the location information=20
> provided by a user agent would reject a request with the 424=20
> Bad Location response code, optionally enclosing in the body=20
> the location information that it would like the user agent to=20
> incorporate.  The UA can insert the LI from the 424 in its=20
> subsequent reissuance of the request; if no LI is supplied in=20
> the 424, then the user agent resends the request without LI.=20
> This brings the location-conveyance mechanism back to one its=20
> original constraints: that only one PIDF-LO body is=20
> associated with a particular SIP request. This allows us to=20
> remove the "used-for-routing" parameter as well as the=20
> "inserted-by"/"inserter" mechanism.
>=20
> In a related clean-up, several of the error codes can=20
> probably be consolidated or replaced with SIP response codes=20
> (for example, the Retry-After semantics of the 200/300 error codes).
>=20
> If no one objects, we'd therefore like to try to come back=20
> with a new version that has fewer moving parts and fewer pages.
>=20
> Thoughts?
>=20
> Jon Peterson
> NeuStar, Inc.
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From jon.peterson@neustar.biz  Fri Mar 26 07:50:49 2010
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "Francois D. Menard" <fmenard@xittelecom.com>
Date: Fri, 26 Mar 2010 10:50:28 -0400
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Cc: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] rebooting location-conveyance
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--_000_C7D217C43BC30jonpetersonneustarbiz_
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I can't imagine it could - in fact, I think it makes end-to-end work better=
 - but if you can elaborate on your concern we could explore that further.

Jon Peterson
NeuStar, Inc.


On 3/25/10 7:09 PM, "Francois D. Menard" <fmenard@xittelecom.com> wrote:

I am concerned that this change may impact the ability for end-to-end locat=
ion conveyance to work.

Am I wrong?

F.

On 2010-03-25, at 8:09 PM, Jon Peterson wrote:

>
> Semantically, a single SIP request can contain multiple locations within =
the constraints of the presence data model - PIDF-LO is after all PIDF, and=
 like any presence document it can make fairly complicated statements about=
 the relationship of location information to users or devices, including en=
umerating multiple locations. However, the intention of the revision is to =
return this to an old design constraint where a SIP request may contain onl=
y a single Geolocation header field with one URI pointing to location infor=
mation.
>
> Jon Peterson
> NeuStar, Inc.
>
> DRAGE, Keith (Keith) wrote:
>> So, I believe the current solution allows a scenario where the originati=
ng UA inserts a location, and various proxies along the path could insert a=
 location, in addition to the existing one, providing they all represent th=
eir view of the state of the user.
>>
>> By making this proposal, are you eliminating this scenario, or allowing =
it to continue to exist.
>>
>> regards
>>
>> Keith
>>
>>
>>
>>> -----Original Message-----
>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Beh=
alf Of Jon Peterson
>>> Sent: Thursday, March 25, 2010 9:28 PM
>>> To: sipcore@ietf.org
>>> Subject: [sipcore] rebooting location-conveyance
>>>
>>> James Polk and I sat down this week and took a hard look at the current=
 sipcore-location-conveyance draft, and we believe that there's an opportun=
ity to simplify (and shorten) the document.
>>>
>>> The simplification derives mostly from a single change to the operation=
 of the mechanism. Today, the location-conveyance draft has a good deal of =
machinery to support the case where a proxy wants to replace existing locat=
ion information in a request with more accurate or preferred location infor=
mation. The current approach replaces the URI in the Geolocation header (wh=
ich probably points to a MIME body) with a URI desired by the proxy, presum=
ably of an Internet resource containing the desired PIDF-LO body. This solu=
tion isn't optimal for several reasons - most notably, since the MIME body =
is not removed by said proxy, it will still be delivered to the eventual lo=
cation recipient, even if the URI in the Geolocation header does not point =
to said body. In order to handle this case, the current draft employs a dis=
ambiguation mechanism (relying on the "used-for-routing" and "inserted-by" =
parameters, as well as the "inserter") to inform the location recipient abo=
ut the source of th  is location information. Moreover, since this change i=
s effected unilaterally, the originating user agent has no idea that the lo=
cation it supplied has been altered in transit, and may thus receive a surp=
rising response to its request.
>>>
>>> While this use case is an important one, we believe it can be addressed=
 with a lighter mechanism. The basic idea is that a proxy server that wants=
 to replace the location information provided by a user agent would reject =
a request with the 424 Bad Location response code, optionally enclosing in =
the body the location information that it would like the user agent to inco=
rporate.  The UA can insert the LI from the 424 in its subsequent reissuanc=
e of the request; if no LI is supplied in the 424, then the user agent rese=
nds the request without LI. This brings the location-conveyance mechanism b=
ack to one its original constraints: that only one PIDF-LO body is associat=
ed with a particular SIP request. This allows us to remove the "used-for-ro=
uting" parameter as well as the "inserted-by"/"inserter" mechanism.
>>>
>>> In a related clean-up, several of the error codes can probably be conso=
lidated or replaced with SIP response codes (for example, the Retry-After s=
emantics of the 200/300 error codes).
>>>
>>> If no one objects, we'd therefore like to try to come back with a new v=
ersion that has fewer moving parts and fewer pages.
>>>
>>> Thoughts?
>>>
>>> Jon Peterson
>>> NeuStar, Inc.
>>>
>>>
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore



--_000_C7D217C43BC30jonpetersonneustarbiz_
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<HTML>
<HEAD>
<TITLE>Re: [sipcore] rebooting location-conveyance</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
11pt'><BR>
I can&#8217;t imagine it could &#8211; in fact, I think it makes end-to-end=
 work better &#8211; but if you can elaborate on your concern we could expl=
ore that further.<BR>
<BR>
Jon Peterson<BR>
NeuStar, Inc.<BR>
<BR>
<BR>
On 3/25/10 7:09 PM, &quot;Francois D. Menard&quot; &lt;<a href=3D"fmenard@x=
ittelecom.com">fmenard@xittelecom.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>I am concerned that this change may impact =
the ability for end-to-end location conveyance to work.<BR>
<BR>
Am I wrong?<BR>
<BR>
F.<BR>
<BR>
On 2010-03-25, at 8:09 PM, Jon Peterson wrote:<BR>
<BR>
&gt;<BR>
&gt; Semantically, a single SIP request can contain multiple locations with=
in the constraints of the presence data model - PIDF-LO is after all PIDF, =
and like any presence document it can make fairly complicated statements ab=
out the relationship of location information to users or devices, including=
 enumerating multiple locations. However, the intention of the revision is =
to return this to an old design constraint where a SIP request may contain =
only a single Geolocation header field with one URI pointing to location in=
formation.<BR>
&gt;<BR>
&gt; Jon Peterson<BR>
&gt; NeuStar, Inc.<BR>
&gt;<BR>
&gt; DRAGE, Keith (Keith) wrote:<BR>
&gt;&gt; So, I believe the current solution allows a scenario where the ori=
ginating UA inserts a location, and various proxies along the path could in=
sert a location, in addition to the existing one, providing they all repres=
ent their view of the state of the user.<BR>
&gt;&gt;<BR>
&gt;&gt; By making this proposal, are you eliminating this scenario, or all=
owing it to continue to exist.<BR>
&gt;&gt;<BR>
&gt;&gt; regards<BR>
&gt;&gt;<BR>
&gt;&gt; Keith<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt; <BR>
&gt;&gt;&gt; -----Original Message-----<BR>
&gt;&gt;&gt; From: <a href=3D"sipcore-bounces@ietf.org">sipcore-bounces@iet=
f.org</a> [<a href=3D"mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounc=
es@ietf.org</a>] On Behalf Of Jon Peterson<BR>
&gt;&gt;&gt; Sent: Thursday, March 25, 2010 9:28 PM<BR>
&gt;&gt;&gt; To: <a href=3D"sipcore@ietf.org">sipcore@ietf.org</a><BR>
&gt;&gt;&gt; Subject: [sipcore] rebooting location-conveyance<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; James Polk and I sat down this week and took a hard look at th=
e current sipcore-location-conveyance draft, and we believe that there's an=
 opportunity to simplify (and shorten) the document.<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; The simplification derives mostly from a single change to the =
operation of the mechanism. Today, the location-conveyance draft has a good=
 deal of machinery to support the case where a proxy wants to replace exist=
ing location information in a request with more accurate or preferred locat=
ion information. The current approach replaces the URI in the Geolocation h=
eader (which probably points to a MIME body) with a URI desired by the prox=
y, presumably of an Internet resource containing the desired PIDF-LO body. =
This solution isn't optimal for several reasons - most notably, since the M=
IME body is not removed by said proxy, it will still be delivered to the ev=
entual location recipient, even if the URI in the Geolocation header does n=
ot point to said body. In order to handle this case, the current draft empl=
oys a disambiguation mechanism (relying on the &quot;used-for-routing&quot;=
 and &quot;inserted-by&quot; parameters, as well as the &quot;inserter&quot=
;) to inform the location recipient about the source of th &nbsp;is locatio=
n information. Moreover, since this change is effected unilaterally, the or=
iginating user agent has no idea that the location it supplied has been alt=
ered in transit, and may thus receive a surprising response to its request.=
<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; While this use case is an important one, we believe it can be =
addressed with a lighter mechanism. The basic idea is that a proxy server t=
hat wants to replace the location information provided by a user agent woul=
d reject a request with the 424 Bad Location response code, optionally encl=
osing in the body the location information that it would like the user agen=
t to incorporate. &nbsp;The UA can insert the LI from the 424 in its subseq=
uent reissuance of the request; if no LI is supplied in the 424, then the u=
ser agent resends the request without LI. This brings the location-conveyan=
ce mechanism back to one its original constraints: that only one PIDF-LO bo=
dy is associated with a particular SIP request. This allows us to remove th=
e &quot;used-for-routing&quot; parameter as well as the &quot;inserted-by&q=
uot;/&quot;inserter&quot; mechanism.<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; In a related clean-up, several of the error codes can probably=
 be consolidated or replaced with SIP response codes (for example, the Retr=
y-After semantics of the 200/300 error codes).<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; If no one objects, we'd therefore like to try to come back wit=
h a new version that has fewer moving parts and fewer pages.<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; Thoughts?<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; Jon Peterson<BR>
&gt;&gt;&gt; NeuStar, Inc.<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt;<BR>
&gt;&gt;&gt; _______________________________________________<BR>
&gt;&gt;&gt; sipcore mailing list<BR>
&gt;&gt;&gt; <a href=3D"sipcore@ietf.org">sipcore@ietf.org</a><BR>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">http=
s://www.ietf.org/mailman/listinfo/sipcore</a><BR>
&gt;&gt;&gt; &nbsp;&nbsp;<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; sipcore mailing list<BR>
&gt; <a href=3D"sipcore@ietf.org">sipcore@ietf.org</a><BR>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www.=
ietf.org/mailman/listinfo/sipcore</a><BR>
<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C7D217C43BC30jonpetersonneustarbiz_--

From jon.peterson@neustar.biz  Fri Mar 26 08:09:24 2010
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Subject: Re: [sipcore] rebooting location-conveyance
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In the emergency services case, I would not imagine intermediaries would 
confine themselves to the role of proxy servers - if they don't like the 
location attached to a message, I'm sure they'd just remove it, insert 
their own and forward the request. Why would we expect them to do otherwise?

The way the location-conveyance mechanism works now - where, in that 
case, the intermediary removes the cid pointer to the MIME body it 
doesn't like, but leaves the MIME body in the request and provides a new 
URL to the preferred location information - seems like it is an even 
worse way of accomplishing the aims of an emergency services 
intermediary, and one that surely would be supplanted in deployments by 
some kind of special-purpose intermediary that doesn't conform to the 
standard.

In the emergency services case, we need to do several things we wouldn't 
do otherwise, such as ignoring some classes of security violations 
because the consequences of not processing a call are so dire that we 
need to be generous in what we accept. The real question is if the 
steadily-growing set of parameters that have been defined in the draft 
to this point actually serve the practical requirements for emergency 
services or other use cases.

Jon Peterson
NeuStar, Inc.

Elwell, John wrote:
> What do the emergency call people think about having to retry, with a possible impact on call establishment times?
>
> John 
>
>   
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org 
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
>> Sent: 25 March 2010 14:28
>> To: sipcore@ietf.org
>> Subject: [sipcore] rebooting location-conveyance
>>
>> James Polk and I sat down this week and took a hard look at 
>> the current sipcore-location-conveyance draft, and we believe 
>> that there's an opportunity to simplify (and shorten) the document.
>>
>> The simplification derives mostly from a single change to the 
>> operation of the mechanism. Today, the location-conveyance 
>> draft has a good deal of machinery to support the case where 
>> a proxy wants to replace existing location information in a 
>> request with more accurate or preferred location information. 
>> The current approach replaces the URI in the Geolocation 
>> header (which probably points to a MIME body) with a URI 
>> desired by the proxy, presumably of an Internet resource 
>> containing the desired PIDF-LO body. This solution isn't 
>> optimal for several reasons - most notably, since the MIME 
>> body is not removed by said proxy, it will still be delivered 
>> to the eventual location recipient, even if the URI in the 
>> Geolocation header does not point to said body. In order to 
>> handle this case, the current draft employs a disambiguation 
>> mechanism (relying on the "used-for-routing" and 
>> "inserted-by" parameters, as well as the "inserter") to 
>> inform the location recipient about the source of th
>>  is location information. Moreover, since this change is 
>> effected unilaterally, the originating user agent has no idea 
>> that the location it supplied has been altered in transit, 
>> and may thus receive a surprising response to its request.
>>
>> While this use case is an important one, we believe it can be 
>> addressed with a lighter mechanism. The basic idea is that a 
>> proxy server that wants to replace the location information 
>> provided by a user agent would reject a request with the 424 
>> Bad Location response code, optionally enclosing in the body 
>> the location information that it would like the user agent to 
>> incorporate.  The UA can insert the LI from the 424 in its 
>> subsequent reissuance of the request; if no LI is supplied in 
>> the 424, then the user agent resends the request without LI. 
>> This brings the location-conveyance mechanism back to one its 
>> original constraints: that only one PIDF-LO body is 
>> associated with a particular SIP request. This allows us to 
>> remove the "used-for-routing" parameter as well as the 
>> "inserted-by"/"inserter" mechanism.
>>
>> In a related clean-up, several of the error codes can 
>> probably be consolidated or replaced with SIP response codes 
>> (for example, the Retry-After semantics of the 200/300 error codes).
>>
>> If no one objects, we'd therefore like to try to come back 
>> with a new version that has fewer moving parts and fewer pages.
>>
>> Thoughts?
>>
>> Jon Peterson
>> NeuStar, Inc.
>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>     


From ivo.sedlacek@ericsson.com  Fri Mar 26 08:51:36 2010
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From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 26 Mar 2010 16:51:50 +0100
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Subject: [sipcore] <identity> element in RFC 4235
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Hello,

Let's suppose that Alice called Bob and Bob forwarded the call to Cecil whi=
ch accepted the call.
When Alice's UE sent the INVITE, the To header was set to Bob's SIP URI (si=
p:bob@example.com).
Cecil's network provided P-Asserted-Identity header set to Cecil's SIP URI =
(sip:cecil@example.com) in 2xx response.

Now Duren subscribes to dialog event package at Alice's UE.
Alice's UE generates dialog event package notification including the create=
d dialog.

Is the <identity> element of the <remote> element of the created dialog sup=
posed to contain:
1) the original To header field URI (sip:bob@example.com);
2) the true identity of the participant (sip:cecil@example.com); or
3) none of them?

RFC4235 states:

---------------------------------------------------------------------------=
--------------------------
4.1.6.1.  Identity Element

   The "identity" element indicates a local or remote URI, as defined in
   [2] as appropriate.  It has an optional attribute, display, that
   contains the display name from the appropriate URI.

      Note that multiple identities (for example a sip: URI and a tel:
      URI) could be included if they all correspond to the participant.
      To avoid repeating identity information in each request, the
      subscriber can assume that the identity URIs are the same as in
      previous notifications if no identity elements are present in the
      corresponding local or remote element.  If any identity elements
      are present in the local or remote part of a notification, the new
      list of identity tags completely supersedes the old list in the
      corresponding part.

      <identity display=3D"Anonymous">
           sip:anonymous@anonymous.invalid</identity>
---------------------------------------------------------------------------=
--------------------------

remote URI as defined in [2] (RFC3261) is the original To header field URI =
(sip:bob@example.com).
However, the original To header field URI does not correspond to the true r=
emote participant (sip:cecil@example.com).
Does "as appropriate" have any particular meaning?

Thanks for clarification.

Kind regards

Ivo Sedlacek

Ericsson
Technology and Portfolio Management, Terminal Standardization
Sweden
Office: +46 10 711 9382
Fax: +46 10 713 5929
ivo.sedlacek@ericsson.com
www.ericsson.com

[http://www.ericsson.com/shared/images/Email.gif]


This communication is confidential. We only send and receive email on the b=
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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.6001.18385" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2>Hello,</FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>Let's sup=
pose that=20
</FONT></SPAN><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>=
Alice=20
called Bob and Bob forwarded the call to Cecil which accepted the=20
call.&nbsp;</FONT></SPAN></DIV>
<DIV><FONT face=3DArial><FONT size=3D2><SPAN class=3D573020815-26032010>Whe=
n Alice's=20
UE sent the INVITE, the To header was set to Bob's SIP URI=20
(sip:bob@example.com). </SPAN></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT size=3D2><SPAN=20
class=3D573020815-26032010></SPAN></FONT></FONT><FONT face=3DArial><FONT=20
size=3D2><SPAN class=3D573020815-26032010>Cecil's network provided=20
P-Asserted-Identity header set to Cecil's SIP URI (sip:cecil@example.com) i=
n 2xx=20
response.</SPAN></FONT></FONT></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>Now Duren=
 subscribes=20
to dialog event package at Alice's UE.</FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>Alice's U=
E generates=20
dialog event package notification including the created dialog.=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>Is the=20
&lt;identity&gt; element of the </FONT></SPAN><SPAN=20
class=3D573020815-26032010><FONT face=3DArial size=3D2>&lt;remote&gt; eleme=
nt of the=20
created dialog </FONT></SPAN><SPAN class=3D573020815-26032010><FONT face=3D=
Arial=20
size=3D2>supposed to contain:</FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>1)&nbsp;t=
he original=20
To header field URI (sip:bob@example.com); </FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><SPAN class=3D573020815-26032010><FON=
T=20
face=3DArial size=3D2>2)&nbsp;the true identity of the&nbsp;participant=20
(sip:cecil@example.com); or</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><SPAN class=3D573020815-26032010><FON=
T=20
face=3DArial size=3D2>3) none of them?</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><SPAN class=3D573020815-26032010><FON=
T=20
face=3DArial size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><SPAN class=3D573020815-26032010><FON=
T=20
face=3DArial size=3D2>RFC4235 states:</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2>------------------------------------------------------------------=
-----------------------------------</FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>4.1.6.1.&=
nbsp;=20
Identity Element</FONT></SPAN></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2>&nbsp;&nb=
sp; The=20
"identity" element indicates a local or remote URI, as defined=20
in<BR>&nbsp;&nbsp; [2] as appropriate.&nbsp; It has an optional attribute,=
=20
display, that<BR>&nbsp;&nbsp; contains the display name from the appropriat=
e=20
URI.</FONT></SPAN></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Note that multiple identities (for =
example=20
a sip: URI and a tel:<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; URI) could be inclu=
ded=20
if they all correspond to the participant.<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; To=20
avoid repeating identity information in each request,=20
the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subscriber can assume that the identi=
ty=20
URIs are the same as in<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; previous notifica=
tions=20
if no identity elements are present in the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;=20
corresponding local or remote element.&nbsp; If any identity=20
elements<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; are present in the local or remo=
te=20
part of a notification, the new<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; list of=20
identity tags completely supersedes the old list in=20
the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; corresponding part.</FONT></SPAN></DI=
V>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;identity=20
display=3D"Anonymous"&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
sip:anonymous@anonymous.invalid&lt;/identity&gt;<BR><SPAN=20
class=3D573020815-26032010><FONT face=3DArial=20
size=3D2>------------------------------------------------------------------=
-----------------------------------</FONT></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010></SPAN></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010>remote URI as defined in [2] (RFC3261) is the or=
iginal=20
To header field URI (sip:bob@example.com). </SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010></SPAN></FONT></SPAN><SPAN=20
class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010>However, the original To header field URI=20
</SPAN></FONT></SPAN><SPAN class=3D573020815-26032010><FONT face=3DArial=20
size=3D2><SPAN class=3D573020815-26032010>does not correspond to the true r=
emote=20
participant (sip:cecil@example.com). </SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010>Does "as appropriate" have any particular=20
meaning?</SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010></SPAN></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010>Thanks for clarification.</SPAN></FONT></SPAN></=
DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010></SPAN></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D573020815-26032010><FONT face=3DArial size=3D2><SPAN=20
class=3D573020815-26032010>Kind regards</SPAN></FONT></SPAN><SPAN=20
class=3D573020815-26032010><FONT face=3DArial size=3D2>&nbsp;</FONT></SPAN>=
<SPAN=20
class=3D573020815-26032010><FONT face=3DArial size=3D2></FONT></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft>
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt; COLOR: #404040"><B><SPAN=
=20
style=3D"FONT-SIZE: 10pt; COLOR: #404040; FONT-FAMILY: Arial">Ivo Sedlacek=
=20
</SPAN></B><B><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #404040; FONT-FAMILY: Arial"><BR></SPAN></=
B></P>
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt; COLOR: #404040"><BR><SPA=
N=20
style=3D"FONT-SIZE: 10pt; COLOR: #404040; FONT-FAMILY: Arial">Ericsson<BR>T=
echnology=20
and Portfolio Management, Terminal Standardization <BR>Sweden<BR>Office: +4=
6 10=20
711 9382<BR>Fax: +46 10 713=20
5929<BR>ivo.sedlacek@ericsson.com<BR>www.ericsson.com</SPAN></P>
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><BR><IMG alt=3D"Ericsson logo=
"=20
hspace=3D0 src=3D"http://www.ericsson.com/shared/images/Email.gif" align=3D=
bottom=20
border=3D0> </SPAN></P>
<P class=3DMsoNormal=20
style=3D"MARGIN: 0cm 0cm 0pt; mso-margin-top-alt: auto; mso-margin-bottom-a=
lt: auto"><SPAN=20
style=3D"FONT-SIZE: 8pt; COLOR: #404040; FONT-FAMILY: Arial"><BR><BR>This=20
communication is confidential. We only send and receive email on the basis =
of=20
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sclaimer</A>=20
</SPAN></P></DIV>
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From keith.drage@alcatel-lucent.com  Fri Mar 26 08:58:41 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Jon Peterson <jon.peterson@neustar.biz>, "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Fri, 26 Mar 2010 16:54:48 +0100
Thread-Topic: [sipcore] rebooting location-conveyance
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Subject: Re: [sipcore] rebooting location-conveyance
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1)	At least on the emergency calls I've been looking at, we've been doing o=
ur best to make sure intermediaries are proxies. Insertions could well be l=
ocation by reference, rather than location by value as a result. B2BUAs bre=
ak transparency - remember?

2)	From day 1) on emergency calls, the ethos has been - Do not make judgeme=
nts on the value of information to the PSAP. Therefore in the work I've bee=
n involved in, we never remove information from an emergency call. Everythi=
ng is passed to the PSAP in some form. This makes it impossible to fulfil t=
hat.

regards

Keith=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> Sent: Friday, March 26, 2010 3:10 PM
> To: Elwell, John
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] rebooting location-conveyance
>=20
>=20
> In the emergency services case, I would not imagine=20
> intermediaries would confine themselves to the role of proxy=20
> servers - if they don't like the location attached to a=20
> message, I'm sure they'd just remove it, insert their own and=20
> forward the request. Why would we expect them to do otherwise?
>=20
> The way the location-conveyance mechanism works now - where,=20
> in that case, the intermediary removes the cid pointer to the=20
> MIME body it doesn't like, but leaves the MIME body in the=20
> request and provides a new URL to the preferred location=20
> information - seems like it is an even worse way of=20
> accomplishing the aims of an emergency services intermediary,=20
> and one that surely would be supplanted in deployments by=20
> some kind of special-purpose intermediary that doesn't=20
> conform to the standard.
>=20
> In the emergency services case, we need to do several things=20
> we wouldn't do otherwise, such as ignoring some classes of=20
> security violations because the consequences of not=20
> processing a call are so dire that we need to be generous in=20
> what we accept. The real question is if the steadily-growing=20
> set of parameters that have been defined in the draft to this=20
> point actually serve the practical requirements for emergency=20
> services or other use cases.
>=20
> Jon Peterson
> NeuStar, Inc.
>=20
> Elwell, John wrote:
> > What do the emergency call people think about having to=20
> retry, with a possible impact on call establishment times?
> >
> > John
> >
> >  =20
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org
> >> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> >> Sent: 25 March 2010 14:28
> >> To: sipcore@ietf.org
> >> Subject: [sipcore] rebooting location-conveyance
> >>
> >> James Polk and I sat down this week and took a hard look at the=20
> >> current sipcore-location-conveyance draft, and we believe that=20
> >> there's an opportunity to simplify (and shorten) the document.
> >>
> >> The simplification derives mostly from a single change to the=20
> >> operation of the mechanism. Today, the location-conveyance=20
> draft has=20
> >> a good deal of machinery to support the case where a proxy=20
> wants to=20
> >> replace existing location information in a request with=20
> more accurate=20
> >> or preferred location information.
> >> The current approach replaces the URI in the Geolocation header=20
> >> (which probably points to a MIME body) with a URI desired by the=20
> >> proxy, presumably of an Internet resource containing the desired=20
> >> PIDF-LO body. This solution isn't optimal for several=20
> reasons - most=20
> >> notably, since the MIME body is not removed by said proxy, it will=20
> >> still be delivered to the eventual location recipient, even if the=20
> >> URI in the Geolocation header does not point to said body.=20
> In order=20
> >> to handle this case, the current draft employs a disambiguation=20
> >> mechanism (relying on the "used-for-routing" and "inserted-by"=20
> >> parameters, as well as the "inserter") to inform the location=20
> >> recipient about the source of th  is location information.=20
> Moreover,=20
> >> since this change is effected unilaterally, the originating user=20
> >> agent has no idea that the location it supplied has been=20
> altered in=20
> >> transit, and may thus receive a surprising response to its request.
> >>
> >> While this use case is an important one, we believe it can be=20
> >> addressed with a lighter mechanism. The basic idea is that a proxy=20
> >> server that wants to replace the location information=20
> provided by a=20
> >> user agent would reject a request with the 424 Bad=20
> Location response=20
> >> code, optionally enclosing in the body the location=20
> information that=20
> >> it would like the user agent to incorporate.  The UA can=20
> insert the=20
> >> LI from the 424 in its subsequent reissuance of the=20
> request; if no LI=20
> >> is supplied in the 424, then the user agent resends the request=20
> >> without LI.
> >> This brings the location-conveyance mechanism back to one its=20
> >> original constraints: that only one PIDF-LO body is=20
> associated with a=20
> >> particular SIP request. This allows us to remove the=20
> >> "used-for-routing" parameter as well as the=20
> "inserted-by"/"inserter"=20
> >> mechanism.
> >>
> >> In a related clean-up, several of the error codes can probably be=20
> >> consolidated or replaced with SIP response codes (for example, the=20
> >> Retry-After semantics of the 200/300 error codes).
> >>
> >> If no one objects, we'd therefore like to try to come back=20
> with a new=20
> >> version that has fewer moving parts and fewer pages.
> >>
> >> Thoughts?
> >>
> >> Jon Peterson
> >> NeuStar, Inc.
> >>
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> >>    =20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From pkyzivat@cisco.com  Fri Mar 26 09:36:42 2010
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [sipcore] <identity> element in RFC 4235
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[as individual]
comment at end...

Ivo Sedlacek wrote:
> Hello,
>  
> Let's suppose that Alice called Bob and Bob forwarded the call to Cecil 
> which accepted the call. 
> When Alice's UE sent the INVITE, the To header was set to Bob's SIP URI 
> (sip:bob@example.com).
> Cecil's network provided P-Asserted-Identity header set to Cecil's SIP 
> URI (sip:cecil@example.com) in 2xx response.
>  
> Now Duren subscribes to dialog event package at Alice's UE.
> Alice's UE generates dialog event package notification including the 
> created dialog.
>  
> Is the <identity> element of the <remote> element of the created dialog 
> supposed to contain:
> 1) the original To header field URI (sip:bob@example.com);
> 2) the true identity of the participant (sip:cecil@example.com); or
> 3) none of them?
>  
> RFC4235 states:
>  
> -----------------------------------------------------------------------------------------------------
> 4.1.6.1.  Identity Element
>  
>    The "identity" element indicates a local or remote URI, as defined in
>    [2] as appropriate.  It has an optional attribute, display, that
>    contains the display name from the appropriate URI.
>  
>       Note that multiple identities (for example a sip: URI and a tel:
>       URI) could be included if they all correspond to the participant.
>       To avoid repeating identity information in each request, the
>       subscriber can assume that the identity URIs are the same as in
>       previous notifications if no identity elements are present in the
>       corresponding local or remote element.  If any identity elements
>       are present in the local or remote part of a notification, the new
>       list of identity tags completely supersedes the old list in the
>       corresponding part.
>  
>       <identity display="Anonymous">
>            sip:anonymous@anonymous.invalid</identity>
> -----------------------------------------------------------------------------------------------------
>  
> remote URI as defined in [2] (RFC3261) is the original To header field 
> URI (sip:bob@example.com).
> However, the original To header field URI does not correspond to the 
> true remote participant (sip:cecil@example.com).
> Does "as appropriate" have any particular meaning?

You mention Cecil's PAI being returned in the 2xx. The semantics of that 
are not defined in ietf. (I believe this usage *is* defined in IMS.)

RFC 4916 (connected identity) defines a mechanism to deliver that back 
to the caller in a request. If 4916 is used, the actual remote identity 
that is part of dialog state according to 3261 will be changed, and 
presumably will then be reported in the notify.

The impact of PAI on returns from the dialog event package is, AFAIK, 
undefined. (Does IMS define it?)

	Thanks,
	Paul

From adam@nostrum.com  Fri Mar 26 09:40:52 2010
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Subject: Re: [sipcore] rebooting location-conveyance
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[as participant]

Actually, I read his response as a qualified "yes." If we go to base 
requirements and ask whether proxies on the path can take action to 
cause their own view of the user's location to be added to the request, 
then the requirement -- by my understanding -- is satisfied by the 
proposed mechanism.

/a

On 3/25/10 6:19 PM, DRAGE, Keith (Keith) wrote:
> So on the basis that your answer appears to be NO to my question, I believe I cannot support the change you are proposing.
>
> regards
>
> Keith
>
>    
>> -----Original Message-----
>> From: Jon Peterson [mailto:jon.peterson@neustar.biz]
>> Sent: Friday, March 26, 2010 12:09 AM
>> To: DRAGE, Keith (Keith)
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] rebooting location-conveyance
>>
>>
>> Semantically, a single SIP request can contain multiple
>> locations within the constraints of the presence data model -
>> PIDF-LO is after all PIDF, and like any presence document it
>> can make fairly complicated statements about the relationship
>> of location information to users or devices, including
>> enumerating multiple locations. However, the intention of the
>> revision is to return this to an old design constraint where
>> a SIP request may contain only a single Geolocation header
>> field with one URI pointing to location information.
>>
>> Jon Peterson
>> NeuStar, Inc.
>>
>> DRAGE, Keith (Keith) wrote:
>>      
>>> So, I believe the current solution allows a scenario where
>>>        
>> the originating UA inserts a location, and various proxies
>> along the path could insert a location, in addition to the
>> existing one, providing they all represent their view of the
>> state of the user.
>>      
>>> By making this proposal, are you eliminating this scenario,
>>>        
>> or allowing it to continue to exist.
>>      
>>> regards
>>>
>>> Keith
>>>
>>>
>>>
>>>
>>>        
>>>> -----Original Message-----
>>>> From: sipcore-bounces@ietf.org
>>>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
>>>> Sent: Thursday, March 25, 2010 9:28 PM
>>>> To: sipcore@ietf.org
>>>> Subject: [sipcore] rebooting location-conveyance
>>>>
>>>> James Polk and I sat down this week and took a hard look at the
>>>> current sipcore-location-conveyance draft, and we believe that
>>>> there's an opportunity to simplify (and shorten) the document.
>>>>
>>>> The simplification derives mostly from a single change to the
>>>> operation of the mechanism. Today, the location-conveyance
>>>>          
>> draft has
>>      
>>>> a good deal of machinery to support the case where a proxy
>>>>          
>> wants to
>>      
>>>> replace existing location information in a request with
>>>>          
>> more accurate
>>      
>>>> or preferred location information.
>>>> The current approach replaces the URI in the Geolocation header
>>>> (which probably points to a MIME body) with a URI desired by the
>>>> proxy, presumably of an Internet resource containing the desired
>>>> PIDF-LO body. This solution isn't optimal for several
>>>>          
>> reasons - most
>>      
>>>> notably, since the MIME body is not removed by said proxy, it will
>>>> still be delivered to the eventual location recipient, even if the
>>>> URI in the Geolocation header does not point to said body.
>>>>          
>> In order
>>      
>>>> to handle this case, the current draft employs a disambiguation
>>>> mechanism (relying on the "used-for-routing" and "inserted-by"
>>>> parameters, as well as the "inserter") to inform the location
>>>> recipient about the source of th  is location information.
>>>>          
>> Moreover,
>>      
>>>> since this change is effected unilaterally, the originating user
>>>> agent has no idea that the location it supplied has been
>>>>          
>> altered in
>>      
>>>> transit, and may thus receive a surprising response to its request.
>>>>
>>>> While this use case is an important one, we believe it can be
>>>> addressed with a lighter mechanism. The basic idea is that a proxy
>>>> server that wants to replace the location information
>>>>          
>> provided by a
>>      
>>>> user agent would reject a request with the 424 Bad
>>>>          
>> Location response
>>      
>>>> code, optionally enclosing in the body the location
>>>>          
>> information that
>>      
>>>> it would like the user agent to incorporate.  The UA can
>>>>          
>> insert the
>>      
>>>> LI from the 424 in its subsequent reissuance of the
>>>>          
>> request; if no LI
>>      
>>>> is supplied in the 424, then the user agent resends the request
>>>> without LI.
>>>> This brings the location-conveyance mechanism back to one its
>>>> original constraints: that only one PIDF-LO body is
>>>>          
>> associated with a
>>      
>>>> particular SIP request. This allows us to remove the
>>>> "used-for-routing" parameter as well as the
>>>>          
>> "inserted-by"/"inserter"
>>      
>>>> mechanism.
>>>>
>>>> In a related clean-up, several of the error codes can probably be
>>>> consolidated or replaced with SIP response codes (for example, the
>>>> Retry-After semantics of the 200/300 error codes).
>>>>
>>>> If no one objects, we'd therefore like to try to come back
>>>>          
>> with a new
>>      
>>>> version that has fewer moving parts and fewer pages.
>>>>
>>>> Thoughts?
>>>>
>>>> Jon Peterson
>>>> NeuStar, Inc.
>>>>
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>>          
>>
>>      
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>    


From jon.peterson@neustar.biz  Fri Mar 26 10:07:40 2010
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To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [sipcore] rebooting location-conveyance
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--_000_C7D237C63BC7Fjonpetersonneustarbiz_
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Taxonomy aside, the core architectural question is how much support is need=
ed for carrying multiple locations in a single  SIP request, and whether th=
at support should be staged at the SIP layer or the body (PIDF) layer. Para=
meters like "inserted-by" cause immediate suspicion precisely because they =
straddle that boundary - it is the job of PIDF to explain who generated thi=
s location information and what data model element is coupled to the locati=
on information. Merely pointing from the SIP layer to different PIDF bodies=
 describing different locations does not clarify the relationship of these =
elements to one another in data model terms.

While one can argue that you couldn't sort that out in a timeframe expedien=
t enough for the emergency services case, I continue to maintain that we sh=
ould expect exceptional behavior from intermediaries tailored to help emerg=
ency services calls along their path. The job of the network in this case i=
s to deliver quickly the best information it can to the right service - the=
 argument that you "never remove information from an emergency call" does n=
ot entail "encourage the accumulation of a bewildering variety of different=
 locations without describing their relationship to one another." At the en=
d of the day, of course, the result of placing an emergency services call w=
ill probably be the delivery of location information to someone who is stil=
l required to ask the caller personally for their location, and this only f=
urther illustrates why we treat this as a special case. For the common case=
, we want an architecturally clean way to associate a SIP request with loca=
tion information.

Jon Peterson
NeuStar, Inc.

On 3/26/10 8:54 AM, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>=
 wrote:

1)      At least on the emergency calls I've been looking at, we've been do=
ing our best to make sure intermediaries are proxies. Insertions could well=
 be location by reference, rather than location by value as a result. B2BUA=
s break transparency - remember?

2)      From day 1) on emergency calls, the ethos has been - Do not make ju=
dgements on the value of information to the PSAP. Therefore in the work I'v=
e been involved in, we never remove information from an emergency call. Eve=
rything is passed to the PSAP in some form. This makes it impossible to ful=
fil that.

regards

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> Sent: Friday, March 26, 2010 3:10 PM
> To: Elwell, John
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] rebooting location-conveyance
>
>
> In the emergency services case, I would not imagine
> intermediaries would confine themselves to the role of proxy
> servers - if they don't like the location attached to a
> message, I'm sure they'd just remove it, insert their own and
> forward the request. Why would we expect them to do otherwise?
>
> The way the location-conveyance mechanism works now - where,
> in that case, the intermediary removes the cid pointer to the
> MIME body it doesn't like, but leaves the MIME body in the
> request and provides a new URL to the preferred location
> information - seems like it is an even worse way of
> accomplishing the aims of an emergency services intermediary,
> and one that surely would be supplanted in deployments by
> some kind of special-purpose intermediary that doesn't
> conform to the standard.
>
> In the emergency services case, we need to do several things
> we wouldn't do otherwise, such as ignoring some classes of
> security violations because the consequences of not
> processing a call are so dire that we need to be generous in
> what we accept. The real question is if the steadily-growing
> set of parameters that have been defined in the draft to this
> point actually serve the practical requirements for emergency
> services or other use cases.
>
> Jon Peterson
> NeuStar, Inc.
>
> Elwell, John wrote:
> > What do the emergency call people think about having to
> retry, with a possible impact on call establishment times?
> >
> > John
> >
> >
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org
> >> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> >> Sent: 25 March 2010 14:28
> >> To: sipcore@ietf.org
> >> Subject: [sipcore] rebooting location-conveyance
> >>
> >> James Polk and I sat down this week and took a hard look at the
> >> current sipcore-location-conveyance draft, and we believe that
> >> there's an opportunity to simplify (and shorten) the document.
> >>
> >> The simplification derives mostly from a single change to the
> >> operation of the mechanism. Today, the location-conveyance
> draft has
> >> a good deal of machinery to support the case where a proxy
> wants to
> >> replace existing location information in a request with
> more accurate
> >> or preferred location information.
> >> The current approach replaces the URI in the Geolocation header
> >> (which probably points to a MIME body) with a URI desired by the
> >> proxy, presumably of an Internet resource containing the desired
> >> PIDF-LO body. This solution isn't optimal for several
> reasons - most
> >> notably, since the MIME body is not removed by said proxy, it will
> >> still be delivered to the eventual location recipient, even if the
> >> URI in the Geolocation header does not point to said body.
> In order
> >> to handle this case, the current draft employs a disambiguation
> >> mechanism (relying on the "used-for-routing" and "inserted-by"
> >> parameters, as well as the "inserter") to inform the location
> >> recipient about the source of th  is location information.
> Moreover,
> >> since this change is effected unilaterally, the originating user
> >> agent has no idea that the location it supplied has been
> altered in
> >> transit, and may thus receive a surprising response to its request.
> >>
> >> While this use case is an important one, we believe it can be
> >> addressed with a lighter mechanism. The basic idea is that a proxy
> >> server that wants to replace the location information
> provided by a
> >> user agent would reject a request with the 424 Bad
> Location response
> >> code, optionally enclosing in the body the location
> information that
> >> it would like the user agent to incorporate.  The UA can
> insert the
> >> LI from the 424 in its subsequent reissuance of the
> request; if no LI
> >> is supplied in the 424, then the user agent resends the request
> >> without LI.
> >> This brings the location-conveyance mechanism back to one its
> >> original constraints: that only one PIDF-LO body is
> associated with a
> >> particular SIP request. This allows us to remove the
> >> "used-for-routing" parameter as well as the
> "inserted-by"/"inserter"
> >> mechanism.
> >>
> >> In a related clean-up, several of the error codes can probably be
> >> consolidated or replaced with SIP response codes (for example, the
> >> Retry-After semantics of the 200/300 error codes).
> >>
> >> If no one objects, we'd therefore like to try to come back
> with a new
> >> version that has fewer moving parts and fewer pages.
> >>
> >> Thoughts?
> >>
> >> Jon Peterson
> >> NeuStar, Inc.
> >>
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> >>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


--_000_C7D237C63BC7Fjonpetersonneustarbiz_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [sipcore] rebooting location-conveyance</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
11pt'><BR>
Taxonomy aside, the core architectural question is how much support is need=
ed for carrying multiple locations in a single &nbsp;SIP request, and wheth=
er that support should be staged at the SIP layer or the body (PIDF) layer.=
 Parameters like &#8220;inserted-by&#8221; cause immediate suspicion precis=
ely because they straddle that boundary &#8211; it is the job of PIDF to ex=
plain who generated this location information and what data model element i=
s coupled to the location information. Merely pointing from the SIP layer t=
o different PIDF bodies describing different locations does not clarify the=
 relationship of these elements to one another in data model terms.<BR>
<BR>
While one can argue that you couldn&#8217;t sort that out in a timeframe ex=
pedient enough for the emergency services case, I continue to maintain that=
 we should expect exceptional behavior from intermediaries tailored to help=
 emergency services calls along their path. The job of the network in this =
case is to deliver quickly the best information it can to the right service=
 - the argument that you &#8220;never remove information from an emergency =
call&#8221; does not entail &#8220;encourage the accumulation of a bewilder=
ing variety of different locations without describing their relationship to=
 one another.&#8221; At the end of the day, of course, the result of placin=
g an emergency services call will probably be the delivery of location info=
rmation to someone who is still required to ask the caller personally for t=
heir location, and this only further illustrates why we treat this as a spe=
cial case. For the common case, we want an architecturally clean way to ass=
ociate a SIP request with location information.<BR>
<BR>
Jon Peterson<BR>
NeuStar, Inc.<BR>
<BR>
On 3/26/10 8:54 AM, &quot;DRAGE, Keith (Keith)&quot; &lt;<a href=3D"keith.d=
rage@alcatel-lucent.com">keith.drage@alcatel-lucent.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>1) &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;At least o=
n the emergency calls I've been looking at, we've been doing our best to ma=
ke sure intermediaries are proxies. Insertions could well be location by re=
ference, rather than location by value as a result. B2BUAs break transparen=
cy - remember?<BR>
<BR>
2) &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;From day 1) on emergency calls, the ethos =
has been - Do not make judgements on the value of information to the PSAP. =
Therefore in the work I've been involved in, we never remove information fr=
om an emergency call. Everything is passed to the PSAP in some form. This m=
akes it impossible to fulfil that.<BR>
<BR>
regards<BR>
<BR>
Keith<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: <a href=3D"sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a=
><BR>
&gt; [<a href=3D"mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounces@ie=
tf.org</a>] On Behalf Of Jon Peterson<BR>
&gt; Sent: Friday, March 26, 2010 3:10 PM<BR>
&gt; To: Elwell, John<BR>
&gt; Cc: <a href=3D"sipcore@ietf.org">sipcore@ietf.org</a><BR>
&gt; Subject: Re: [sipcore] rebooting location-conveyance<BR>
&gt;<BR>
&gt;<BR>
&gt; In the emergency services case, I would not imagine<BR>
&gt; intermediaries would confine themselves to the role of proxy<BR>
&gt; servers - if they don't like the location attached to a<BR>
&gt; message, I'm sure they'd just remove it, insert their own and<BR>
&gt; forward the request. Why would we expect them to do otherwise?<BR>
&gt;<BR>
&gt; The way the location-conveyance mechanism works now - where,<BR>
&gt; in that case, the intermediary removes the cid pointer to the<BR>
&gt; MIME body it doesn't like, but leaves the MIME body in the<BR>
&gt; request and provides a new URL to the preferred location<BR>
&gt; information - seems like it is an even worse way of<BR>
&gt; accomplishing the aims of an emergency services intermediary,<BR>
&gt; and one that surely would be supplanted in deployments by<BR>
&gt; some kind of special-purpose intermediary that doesn't<BR>
&gt; conform to the standard.<BR>
&gt;<BR>
&gt; In the emergency services case, we need to do several things<BR>
&gt; we wouldn't do otherwise, such as ignoring some classes of<BR>
&gt; security violations because the consequences of not<BR>
&gt; processing a call are so dire that we need to be generous in<BR>
&gt; what we accept. The real question is if the steadily-growing<BR>
&gt; set of parameters that have been defined in the draft to this<BR>
&gt; point actually serve the practical requirements for emergency<BR>
&gt; services or other use cases.<BR>
&gt;<BR>
&gt; Jon Peterson<BR>
&gt; NeuStar, Inc.<BR>
&gt;<BR>
&gt; Elwell, John wrote:<BR>
&gt; &gt; What do the emergency call people think about having to<BR>
&gt; retry, with a possible impact on call establishment times?<BR>
&gt; &gt;<BR>
&gt; &gt; John<BR>
&gt; &gt;<BR>
&gt; &gt; &nbsp;<BR>
&gt; &gt;&gt; -----Original Message-----<BR>
&gt; &gt;&gt; From: <a href=3D"sipcore-bounces@ietf.org">sipcore-bounces@ie=
tf.org</a><BR>
&gt; &gt;&gt; [<a href=3D"mailto:sipcore-bounces@ietf.org">mailto:sipcore-b=
ounces@ietf.org</a>] On Behalf Of Jon Peterson<BR>
&gt; &gt;&gt; Sent: 25 March 2010 14:28<BR>
&gt; &gt;&gt; To: <a href=3D"sipcore@ietf.org">sipcore@ietf.org</a><BR>
&gt; &gt;&gt; Subject: [sipcore] rebooting location-conveyance<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; James Polk and I sat down this week and took a hard look at t=
he<BR>
&gt; &gt;&gt; current sipcore-location-conveyance draft, and we believe tha=
t<BR>
&gt; &gt;&gt; there's an opportunity to simplify (and shorten) the document=
.<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; The simplification derives mostly from a single change to the=
<BR>
&gt; &gt;&gt; operation of the mechanism. Today, the location-conveyance<BR=
>
&gt; draft has<BR>
&gt; &gt;&gt; a good deal of machinery to support the case where a proxy<BR=
>
&gt; wants to<BR>
&gt; &gt;&gt; replace existing location information in a request with<BR>
&gt; more accurate<BR>
&gt; &gt;&gt; or preferred location information.<BR>
&gt; &gt;&gt; The current approach replaces the URI in the Geolocation head=
er<BR>
&gt; &gt;&gt; (which probably points to a MIME body) with a URI desired by =
the<BR>
&gt; &gt;&gt; proxy, presumably of an Internet resource containing the desi=
red<BR>
&gt; &gt;&gt; PIDF-LO body. This solution isn't optimal for several<BR>
&gt; reasons - most<BR>
&gt; &gt;&gt; notably, since the MIME body is not removed by said proxy, it=
 will<BR>
&gt; &gt;&gt; still be delivered to the eventual location recipient, even i=
f the<BR>
&gt; &gt;&gt; URI in the Geolocation header does not point to said body.<BR=
>
&gt; In order<BR>
&gt; &gt;&gt; to handle this case, the current draft employs a disambiguati=
on<BR>
&gt; &gt;&gt; mechanism (relying on the &quot;used-for-routing&quot; and &q=
uot;inserted-by&quot;<BR>
&gt; &gt;&gt; parameters, as well as the &quot;inserter&quot;) to inform th=
e location<BR>
&gt; &gt;&gt; recipient about the source of th &nbsp;is location informatio=
n.<BR>
&gt; Moreover,<BR>
&gt; &gt;&gt; since this change is effected unilaterally, the originating u=
ser<BR>
&gt; &gt;&gt; agent has no idea that the location it supplied has been<BR>
&gt; altered in<BR>
&gt; &gt;&gt; transit, and may thus receive a surprising response to its re=
quest.<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; While this use case is an important one, we believe it can be=
<BR>
&gt; &gt;&gt; addressed with a lighter mechanism. The basic idea is that a =
proxy<BR>
&gt; &gt;&gt; server that wants to replace the location information<BR>
&gt; provided by a<BR>
&gt; &gt;&gt; user agent would reject a request with the 424 Bad<BR>
&gt; Location response<BR>
&gt; &gt;&gt; code, optionally enclosing in the body the location<BR>
&gt; information that<BR>
&gt; &gt;&gt; it would like the user agent to incorporate. &nbsp;The UA can=
<BR>
&gt; insert the<BR>
&gt; &gt;&gt; LI from the 424 in its subsequent reissuance of the<BR>
&gt; request; if no LI<BR>
&gt; &gt;&gt; is supplied in the 424, then the user agent resends the reque=
st<BR>
&gt; &gt;&gt; without LI.<BR>
&gt; &gt;&gt; This brings the location-conveyance mechanism back to one its=
<BR>
&gt; &gt;&gt; original constraints: that only one PIDF-LO body is<BR>
&gt; associated with a<BR>
&gt; &gt;&gt; particular SIP request. This allows us to remove the<BR>
&gt; &gt;&gt; &quot;used-for-routing&quot; parameter as well as the<BR>
&gt; &quot;inserted-by&quot;/&quot;inserter&quot;<BR>
&gt; &gt;&gt; mechanism.<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; In a related clean-up, several of the error codes can probabl=
y be<BR>
&gt; &gt;&gt; consolidated or replaced with SIP response codes (for example=
, the<BR>
&gt; &gt;&gt; Retry-After semantics of the 200/300 error codes).<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; If no one objects, we'd therefore like to try to come back<BR=
>
&gt; with a new<BR>
&gt; &gt;&gt; version that has fewer moving parts and fewer pages.<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; Thoughts?<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; Jon Peterson<BR>
&gt; &gt;&gt; NeuStar, Inc.<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt;<BR>
&gt; &gt;&gt; _______________________________________________<BR>
&gt; &gt;&gt; sipcore mailing list<BR>
&gt; &gt;&gt; <a href=3D"sipcore@ietf.org">sipcore@ietf.org</a><BR>
&gt; &gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">htt=
ps://www.ietf.org/mailman/listinfo/sipcore</a><BR>
&gt; &gt;&gt; &nbsp;&nbsp;&nbsp;<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; sipcore mailing list<BR>
&gt; <a href=3D"sipcore@ietf.org">sipcore@ietf.org</a><BR>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www.=
ietf.org/mailman/listinfo/sipcore</a><BR>
&gt; <BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C7D237C63BC7Fjonpetersonneustarbiz_--

From keith.drage@alcatel-lucent.com  Fri Mar 26 10:41:08 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Adam Roach <adam@nostrum.com>
Date: Fri, 26 Mar 2010 18:41:22 +0100
Thread-Topic: [sipcore] rebooting location-conveyance
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] rebooting location-conveyance
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And I read it as no, as essentially only one location goes forward.

The original predicate from Jon was that we have multiple locations because=
 an intermediate devices wants to replace the existing location.

My view is that this never was a use case. The use case was that an interme=
diate entity had location information relating to the sender of the message=
, for which it did not know whether it was better or worse that that create=
d by other location inserters. Therefore the solution sends all the locatio=
ns to the location recipient, and the recipient then makes its own value di=
scussion.=20

There was also a use case where there was no location from the original sen=
der at all, but I read that as not being impacted by Jon's input.

For information the relevant requirement for this part of the work was:

"   Proxy-1  Proxy servers must be capable of adding a Location header=20
            field during processing of SIP requests.

   Motivation:  Provide network assertion of location
            when UACs are unable to do so, or when network assertion is
            more reliable than UAC assertion of location

   Note: Because UACs connected to SIP signaling networks may have=20
         widely varying access network arrangements, including VPN=20
         tunnels and roaming mechanisms, it may be difficult for a=20
         network to reliably know the location of the endpoint.  Proxy=20
         assertion of location is NOT RECOMMENDED unless the SIP=20
         signaling network has reliable knowledge of the actual=20
         location of the Targets.
"

However it probably does not help advance the argument from either side.

regards

Keith


> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]=20
> Sent: Friday, March 26, 2010 4:38 PM
> To: DRAGE, Keith (Keith)
> Cc: Jon Peterson; sipcore@ietf.org
> Subject: Re: [sipcore] rebooting location-conveyance
>=20
> [as participant]
>=20
> Actually, I read his response as a qualified "yes." If we go=20
> to base requirements and ask whether proxies on the path can=20
> take action to cause their own view of the user's location to=20
> be added to the request, then the requirement -- by my=20
> understanding -- is satisfied by the proposed mechanism.
>=20
> /a
>=20
> On 3/25/10 6:19 PM, DRAGE, Keith (Keith) wrote:
> > So on the basis that your answer appears to be NO to my=20
> question, I believe I cannot support the change you are proposing.
> >
> > regards
> >
> > Keith
> >
> >   =20
> >> -----Original Message-----
> >> From: Jon Peterson [mailto:jon.peterson@neustar.biz]
> >> Sent: Friday, March 26, 2010 12:09 AM
> >> To: DRAGE, Keith (Keith)
> >> Cc: sipcore@ietf.org
> >> Subject: Re: [sipcore] rebooting location-conveyance
> >>
> >>
> >> Semantically, a single SIP request can contain multiple locations=20
> >> within the constraints of the presence data model -=20
> PIDF-LO is after=20
> >> all PIDF, and like any presence document it can make fairly=20
> >> complicated statements about the relationship of location=20
> information=20
> >> to users or devices, including enumerating multiple locations.=20
> >> However, the intention of the revision is to return this to an old=20
> >> design constraint where a SIP request may contain only a single=20
> >> Geolocation header field with one URI pointing to location=20
> >> information.
> >>
> >> Jon Peterson
> >> NeuStar, Inc.
> >>
> >> DRAGE, Keith (Keith) wrote:
> >>     =20
> >>> So, I believe the current solution allows a scenario where
> >>>       =20
> >> the originating UA inserts a location, and various proxies=20
> along the=20
> >> path could insert a location, in addition to the existing one,=20
> >> providing they all represent their view of the state of the user.
> >>     =20
> >>> By making this proposal, are you eliminating this scenario,
> >>>       =20
> >> or allowing it to continue to exist.
> >>     =20
> >>> regards
> >>>
> >>> Keith
> >>>
> >>>
> >>>
> >>>
> >>>       =20
> >>>> -----Original Message-----
> >>>> From: sipcore-bounces@ietf.org
> >>>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Jon Peterson
> >>>> Sent: Thursday, March 25, 2010 9:28 PM
> >>>> To: sipcore@ietf.org
> >>>> Subject: [sipcore] rebooting location-conveyance
> >>>>
> >>>> James Polk and I sat down this week and took a hard look at the=20
> >>>> current sipcore-location-conveyance draft, and we believe that=20
> >>>> there's an opportunity to simplify (and shorten) the document.
> >>>>
> >>>> The simplification derives mostly from a single change to the=20
> >>>> operation of the mechanism. Today, the location-conveyance
> >>>>         =20
> >> draft has
> >>     =20
> >>>> a good deal of machinery to support the case where a proxy
> >>>>         =20
> >> wants to
> >>     =20
> >>>> replace existing location information in a request with
> >>>>         =20
> >> more accurate
> >>     =20
> >>>> or preferred location information.
> >>>> The current approach replaces the URI in the Geolocation header=20
> >>>> (which probably points to a MIME body) with a URI desired by the=20
> >>>> proxy, presumably of an Internet resource containing the desired=20
> >>>> PIDF-LO body. This solution isn't optimal for several
> >>>>         =20
> >> reasons - most
> >>     =20
> >>>> notably, since the MIME body is not removed by said=20
> proxy, it will=20
> >>>> still be delivered to the eventual location recipient,=20
> even if the=20
> >>>> URI in the Geolocation header does not point to said body.
> >>>>         =20
> >> In order
> >>     =20
> >>>> to handle this case, the current draft employs a disambiguation=20
> >>>> mechanism (relying on the "used-for-routing" and "inserted-by"
> >>>> parameters, as well as the "inserter") to inform the location=20
> >>>> recipient about the source of th  is location information.
> >>>>         =20
> >> Moreover,
> >>     =20
> >>>> since this change is effected unilaterally, the originating user=20
> >>>> agent has no idea that the location it supplied has been
> >>>>         =20
> >> altered in
> >>     =20
> >>>> transit, and may thus receive a surprising response to=20
> its request.
> >>>>
> >>>> While this use case is an important one, we believe it can be=20
> >>>> addressed with a lighter mechanism. The basic idea is=20
> that a proxy=20
> >>>> server that wants to replace the location information
> >>>>         =20
> >> provided by a
> >>     =20
> >>>> user agent would reject a request with the 424 Bad
> >>>>         =20
> >> Location response
> >>     =20
> >>>> code, optionally enclosing in the body the location
> >>>>         =20
> >> information that
> >>     =20
> >>>> it would like the user agent to incorporate.  The UA can
> >>>>         =20
> >> insert the
> >>     =20
> >>>> LI from the 424 in its subsequent reissuance of the
> >>>>         =20
> >> request; if no LI
> >>     =20
> >>>> is supplied in the 424, then the user agent resends the request=20
> >>>> without LI.
> >>>> This brings the location-conveyance mechanism back to one its=20
> >>>> original constraints: that only one PIDF-LO body is
> >>>>         =20
> >> associated with a
> >>     =20
> >>>> particular SIP request. This allows us to remove the=20
> >>>> "used-for-routing" parameter as well as the
> >>>>         =20
> >> "inserted-by"/"inserter"
> >>     =20
> >>>> mechanism.
> >>>>
> >>>> In a related clean-up, several of the error codes can=20
> probably be=20
> >>>> consolidated or replaced with SIP response codes (for=20
> example, the=20
> >>>> Retry-After semantics of the 200/300 error codes).
> >>>>
> >>>> If no one objects, we'd therefore like to try to come back
> >>>>         =20
> >> with a new
> >>     =20
> >>>> version that has fewer moving parts and fewer pages.
> >>>>
> >>>> Thoughts?
> >>>>
> >>>> Jon Peterson
> >>>> NeuStar, Inc.
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> sipcore mailing list
> >>>> sipcore@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/sipcore
> >>>>
> >>>>         =20
> >>
> >>     =20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >   =20
>=20
> =

From Martin.Thomson@andrew.com  Fri Mar 26 12:37:04 2010
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From: "Thomson, Martin" <Martin.Thomson@andrew.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: [sipcore] Mary's notes from SIPCORE session on Friday, March 26 13:00-14:00
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Below are my raw notes from the IETF-77 Friday SIPCORE session.

Mary.

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


Table 2 (Adam):
-----------------------
Call for consensus:
1. All documents defining new header fields and methods...define in
semantics and no longer table 2.
2. Produce an RFC that defines new procesdures for adding elements to
table 2 as a new header field and methods

Discussion of what would be for the form for item 2:
-  Adam: we're not defining this yet.
- Keith: what will be the form for item 2. webpage?  Adam: No.
- Adam: it could be an RFC.
- Jonathan Lennox: can we write a document that says that table 2 is
no longer normative. Adam: not deciding that yet.

Consensus call:
- Overwhelming hum for item 1. [Note: one in jabber room also support item 1]

Gonzalo: let's make sure that we don't just go update things for
completeness - address when there are real (e.g., interop problems)

Options (Christer)
--------------------------
Procedural Problems:
[Note: potential gap in notes 13:15-13:20]

1) Proc Problem:  response code.   3261 states that response code
should be the same in an INVITE
Q: does this mean the OPTIONS request should be rejected if the UAS is
currently in a call and would reject an INVITE?
Proposal: Clarify that OPTIONS is about returning general capabilities.

Discussion:
Adam: agree with John E. - we need to have a meta discussion before we
dive. But, other aspects of options are dependent on call state.
Christer: we need to provide guidance on this.

2) Proc Problem: to tag. If the OPTIONS request contains a TO tag, but
no associated dialog exists, procs seem to allow both 481 and 200
responses and even cause dialog to be recreated.
Q: Do we need to clarify behavior
Proposal: If dialog does not exist, return a 481.

Discussion:
Keith: if you use 481, then you're just dropping.  Suggests that you
can still allow folks to respond to OPTIONS, then use 200 (per
previous chart) - if you didn't want to respond to options then you
can send whatever you want.
Christer: asks for clarification.
Robert: Keith isn't arguing against this case. This isn't special about OPTIONS.
Brett: Lack of clarity (ie. not requiring 481)
Robert: suggests that Brett is implying to be careful with RFC 5057 implications

General comment: Adam - doesn't want to focus so much on details -
wants to figure out if group wants to work on problem.

3) Problem: Accept-contact. 3841 states that a request can be rejected
if UAS doesn't suipport feature tags in Accept-contact

Forking problems:
[refer to charts for details]

Questions:
1) How does proxy make a decision whether to return 3xx or fork?
Proposal: use Request-Disposition:redirect
Issue: if there is no RD header, does proxy still return 3xx or assume
default RD.

2) Does proxy return contact addr/gruu of the UAS(s) or something that
hides addresses
Proposal: Policy decision

3) If OPTIONS request contains Accept-contact with capability feature
tags, does the proxy only return contacts for UAS that matches
capabilities?
Proposal: Yes.

Discussion:
Adam: get back to general discussion of whether group wants to work on this.


Way Forward options:
1) Do nothing.  Adam: reworded do we want to work on this at all -
doesn't imply everything is clear)
2) BCP clarifying issues. Adam: this is too specific - do we want to
solve this problem at all?

Discussion:
John E: not certain yet. But do something simple with 2 - more inline
with warnings, etc.
Cullen: don't think we can get consensus on any of these problems.
Don't see a compelling problem. Prefers alternative 1)
James P: [missed this point]
Hadriel: hasn't seen these as problems - these are more features.
Prefers option 1.
Christer: even if we go with option 1 - is there someone that thinks
there is a problem?
Hadriel: has never seen a 3xx to an OPTIONS.
Christer: some folks need this, though.
Peanut Gallery: if you use proxies and not SBCs, you do see this.
Hadriel: to get the feature you want, you need a new capability - not
an interop problem
Brett: Don't think this can be useful to capture things in a way that
pre-conditions did.
Christer: agree - you need something else if you want to do beyond
what options does.
Brett: example of simring ... you can't guarantee subsequent invite
would be routed the same.

Adam: we're back to discussing fiddly bits - need to discuss meta issue.

Robert: tangental point - wants to raise level of awareness on the use
of the BCP acronym.  There will be mounting pushback on the use of
BCPs that are really INFO.

Keith: Will we have an Internet draft for the next discussion?
Adam: most folks are aware of this from ML discussion.

Conclusion: None -  take discussion to the list.

Draft-sipcore-location-conveyance (Jon P):
--------------------------------------------------------------

Proposing to simply - there's been a gradual accumulation of parms,
error codes, etc.
-  reduce to cover only necessary cases
-  Improve predictability from the perspective of the UA

Important use cases:
- Proxies asserting location

Propose to go back to UA just inserting PIDF-LO and proxy can change
or add to.  All the parms, etc. on who added what can be included in
PIDF-LO, thus architectural separation from SIP.

If we do this...
- no need for "used-for-routing", "inserted-by", etc. [Refer to charts]

Discussion:
James W: right now can support a Location URI and a PIDF LO.
John E: have we considered the possibility of looping?
Jon P: sure we did account for that. The possibility to reject is
necessary for a number of cases.
John E: if flow itself includes an indication.
Jon P: don't think we need new stuff, just new language.
Keith: this is not a good idea. There are use cases that we are
needing to support.
Jon P: you haven't elucidated use case.
Keith: in an emergency call, if UAC inserts a location. The network
adds to that. Wants the functionality of both.
Jon P: walks through - in example on chart 4, both original and
corrected location are included in PIDF LO
Keith: what if UAC doesn't include a header, would that also use a 424.
Jon P: if you think UAC supports geo location, then you could use 424.
Hadriel: nice idea in theory, but there's already stuff that is
deployed. In reality, UAs don't support "jack".
Jon P: don't think specialized intermediaries are going to listen to
anything that we say.  They just want location expediently. They'll
chop and dice how they want.
Hadriel: want to change mechanism but not use case?
Jon P: want to remove mechanisms for that use case that won't be used
Hadriel: people (proxies) are inserting PIDF LO and just forwarding it
along. The guy that is receiving this stuff should be the one doing
all the work to get the information.
Jon P: you don't really care about these headers being removed since
they're included in PIDF LO (i.e., who inserted).  Anything that SIP
doesn't need for routing should be in PIDF-LO
Christer: if UA doesn't support 424, UA won't add the new location in
the PIDF LO. Is proxy happy if it just has one location that it wants
(i.e., ignore others)
Jon P: that's always been a problem.  Only changing syntax and not semantics
James P: for Christer's concern, as long as a proxy's location is
included, then other doesn't matter
Martin Thomson: haven't been convinced by any arguments against
proposal. May be problems with stuff that is already deployed.
James W: only concern is around strong language about what should be
included...need to keep some of that.
James P: there can be multiple PIDF LOs in request. Maybe should do a
403...This might be okay if you agree to accept limitations.
Cullen: ..concern over end quality of what gets deployed. Thinks this
is far simpler - easier to get right. Understand impact on what's
existingly deployed. This should finish much sooner than what we have
now.

Conclusion: Good discussion ... need to drive this to conclusion.

From christer.holmberg@ericsson.com  Fri Mar 26 20:50:46 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Sat, 27 Mar 2010 04:51:08 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: 'SIPCORE' <sipcore@ietf.org>, Gilad, Shaham <gilad@voxisoft.com>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>>> No, I'm suggesting that all of the forks respond with a 200 OK and
>>> move to a connected state, not just the first fork to respond. It's
>>> not a race, and there is NO early media -- just media being sent
>>> before the PSTN side of the gateway moves to a fully-connected
>>> state.
>>>
>>> Sorting out multiple media streams is a separate problem, but there
>>> are ways to automate many cases in a B2BUA or really smart UA.
>>>
>>> But this breaks the business model, which assumes that everything
>>> everywhere is fully-connected and in a "billing starts" state when
>>> the first 200OK flies.
>>
>> But, what problem would that solve?
>>
>> The only difference is in what state you are (established-dialog
>> instead of early-dialog), but you still have the same problem as
>> duing the early dialog phase: you might receive early media from
>> multiple locnennations.
>>
>
>
>It's not "early media" if it is in an established dialog.

True. But I don't understand why that is relevant in this context. The prob=
lem is that you are receiving media from multiple locations, so I don't see=
 how it matters whether you are in the early dialog phase or not.

>Once you have a full dialog, you can do  things like send a "BYE"

I don't remember if you're allowed to send BYE in the forward direction on =
an early dialog (you cannot send it in the backward direction, which is one=
 reason we have this wonderful thing called 199 :), but if you can't you ca=
n always send a new offer with "inactive" (eventhough the early dialog itse=
lf will not be terminated).

However, I am not sure you always want to terminate dialogs. Just because y=
ou don't "like" the before-the-remote-user-has-picked-up-the-hook media it =
doesn't mean you necessarily want to terminate the whole dialog.

>have a bot do a verbal challenge-response on the media path,

Why can't you do that with early media?=20

> play one channel to the left ear and one to the right,

You can do that with early dialogs also.

And, if the user have to listen at both channels at the same time, it is st=
ill going to be messy even you use "stereo".

>play all of the media flows simultaneously with variable volume weighting =
using a hover-touch over a
>per-session icon (click to pick "this call" and drop the other ones),

Sure, it may work with geeky users using fancy equipment.

It doesn't work for PSTN (DTMF is the best you can get there), and I don't =
think it would work with the average user who just wants to make a call.

And, again, you may not even have enough bandwidth to forward media to the =
calling user in the first place. I don't think networks are designed that w=
ay.

>etc. But "early media" messes everything up.

Simultanous media from multiple remote locations messes things up.

Regards,

Christer=

From christer.holmberg@ericsson.com  Fri Mar 26 21:06:04 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Sat, 27 Mar 2010 05:06:24 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>>Its really only parallel forking thats a problem, so we can probably igno=
re serial forking.
>
>Not really. Early media is always a problem. How do you tell it to stop wi=
thout whacking all your forks?

"inactive".

Some anouncement based services also use DTMF control to control the media.

>>For parallel forking, moving it from the proxy to the UAC doesn't really =
solve the problem - you can still end up with=20
>>multiple media streams and no clue what to do with them.
>
>Knowing which stream is which lets you do MORE things than you can with
>just a bunch of garbled media coming in.

Knowing which stream is which - AND knowing something about the content of =
the media.

>>You can say "don't do that", but there are desirable features that
>>depend on it - namely "simultaneous ring". Potentially that can be
>>replaced with presence-sensitive ringing to choose the *right* on to
>>ring first, but that requires data that often isn't available.
>
>Google Voice does simultaneous ring quite nicely. If you call my GV
>number, it will ring all my phones. If somebody answers a phone, GV will
>say "Press 1 to accept a call from Paul Kyzivat", and if I press 1, GV
>will terminate the other forks.

But, at that poing the call has been answered by Paul. It is whatever happe=
ns before that (no matter whether the call, from a SIP perspective, is in t=
he early dialog phase or not) that is the problem.

>Webster did this years ago using the dynamicsoft stuff as a call controlle=
r.
>
>But to do this kind of work, the call controller needs signaling control
>on each fork, which you don't get with early media.

It does if you create a dialog, using 18x (or, as you suggest, 200) before =
playing the media.

(The fact that SIP allows media to be sent before any response has been sen=
t is a bug, in my opinion.)

>>> Offerless INVITEs help more, since the media can't come back to you
>>> until the O/A has completed.
>>
>> That is just sticking your head in the sand. The media is still being
>> generated, even if you don't know about it.
>
>No, really. They CANNOT send you media until you've sent your SDP.  If
>you don't send your SDP (in the ACK) until they have sent theirs (in the
>200OK), then there is NO EARLY MEDIA.
>
>Yes, this means that PSTN gateways would have to complete ACK before
>doing IAM on the PSTN side. I'm OK with that.

But, if the ACK is sent to multiple remote destinations at the same time yo=
u are still going to end up with the same media-from-multiple-locations pro=
blem - it will just occur a little later.

If you do NOT want to send ACK to multiple remote destinations at the same =
time, you can as well to serial forking to start with.

>>It does have the advantage that you can associate the media stream to
>>the signaling source, but it still doesn't tell you which stream is
>>interesting to listen to.
>
>But having associated signaling lets you do things to decide which one
>is interesting.

Having signalling which gives information about the content of the media, y=
es. I agree that would be useful.

But, using your offerless-INVITE-mechanish is not going to give you any mor=
e information about the content of the media. It only allows you to not rec=
eive media until you've sent the ACK.

>>ISTM that if you want "simultaneous ring", and the things ringing are
>>often PSTN devices, there is still no solution. Perhaps there is some
>>possibility of automated media analysis - the pstn GW figuring out that
>>the media being received is something uninteresting - just conventional
>>ringback - and suppressing it. But thats a lot of trouble and is likely
>>to have a high error rate.
>
>Go play with Google Voice or something like it for awhile and then get
>back to us.

Does GV have the possibility to receive media (other than "meaningless" rin=
gtones) from multiple locations?

Regards,

Christer=

From christer.holmberg@ericsson.com  Fri Mar 26 21:12:55 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>, Dean Willis <dean.willis@softarmor.com>
Date: Sat, 27 Mar 2010 05:10:20 +0100
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>IMO, if the caller's UE is as strong as Google Voice and it's software, th=
e network can just render all the early media to it.
>So, the key problem is about UE with less capability.
>
>I still think it is BCP thing. I once said how my equipment handle this in=
 the mail(http://www.ietf.org/mail->archive/web/sipcore/current/msg02114.ht=
ml).
>
>BTW, doing this can make PSTN user having the same experience as GV:
>PSTN gateway(such as MGCF in IMS) terminate the multiple forked early medi=
a and just send only one to the PSTN user. And
>using DTMF, the user can switch from one voice to another by press the num=
ber key.



That could work if the media isn't actually played until it is "chosen" (us=
ing DTMF). If the early media A and B is played at the same time, the user =
will miss B while listening to A, and when the user switches to B the messa=
ge may already be over...



Regards,



Christer





Paul Kyzivat wrote:

> Its really only parallel forking thats a problem, so we can probably
> ignore serial forking.

Not really. Early media is always a problem. How do you tell it to stop
without whacking all your forks?

>
> For parallel forking, moving it from the proxy to the UAC doesn't really
> solve the problem - you can still end up with multiple media streams and
> no clue what to do with them.

Knowing which stream is which lets you do MORE things than you can with
just a bunch of garbled media coming in.


>
> You can say "don't do that", but there are desirable features that
> depend on it - namely "simultaneous ring". Potentially that can be
> replaced with presence-sensitive ringing to choose the *right* on to
> ring first, but that requires data that often isn't available.

Google Voice does simultaneous ring quite nicely. If you call my GV
number, it will ring all my phones. If somebody answers a phone, GV will
say "Press 1 to accept a call from Paul Kyzivat", and if I press 1, GV
will terminate the other forks.

Webster did this years ago using the dynamicsoft stuff as a call controller=
.

But to do this kind of work, the call controller needs signaling control
on each fork, which you don't get with early media.

>> Offerless INVITEs help more, since the media can't come back to you
>> until the O/A has completed.
>
> That is just sticking your head in the sand. The media is still being
> generated, even if you don't know about it.

No, really. They CANNOT send you media until you've sent your SDP.  If
you don't send your SDP (in the ACK) until they have sent theirs (in the
200OK), then there is NO EARLY MEDIA.

Yes, this means that PSTN gateways would have to complete ACK before
doing IAM on the PSTN side. I'm OK with that.

> It does have the advantage that you can associate the media stream to
> the signaling source, but it still doesn't tell you which stream is
> interesting to listen to.

But having associated signaling lets you do things to decide which one
is interesting.

> ISTM that if you want "simultaneous ring", and the things ringing are
> often PSTN devices, there is still no solution. Perhaps there is some
> possibility of automated media analysis - the pstn GW figuring out that
> the media being received is something uninteresting - just conventional
> ringback - and suppressing it. But thats a lot of trouble and is likely
> to have a high error rate.

Go play with Google Voice or something like it for awhile and then get
back to us.

--
Dean
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From dean.willis@softarmor.com  Mon Mar 29 16:53:27 2010
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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On Sat, 2010-03-27 at 04:51 +0100, Christer Holmberg wrote:

> 
> True. But I don't understand why that is relevant in this context. The
>  problem is that you are receiving media from multiple locations, so I
>  don't see how it matters whether you are in the early dialog phase or
>  not.
> 

The problem is not just that you are receiving media from multiple
locations. It's that you're receiving data from multiple locations, you
don't know who those locations are or anything else about them, and you
have no way to stop the media nor manipulate it in any useful way
besides just discarding it.

--
Dean



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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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On Sat, 2010-03-27 at 05:06 +0100, Christer Holmberg wrote:
> Hi,
> 
> >>Its really only parallel forking thats a problem, so we can probably ignore serial forking.
> >
> >Not really. Early media is always a problem. How do you tell it to stop without whacking all your forks?
> 
> "inactive".

You can't send it an inactive because it hasn't sent you an answer yet,
so you can't target that fork with an inactive.

> 
> Some anouncement based services also use DTMF control to control the media.
> 
> >>For parallel forking, moving it from the proxy to the UAC doesn't really solve the problem - you can still end up with 
> >>multiple media streams and no clue what to do with them.
> >
> >Knowing which stream is which lets you do MORE things than you can with
> >just a bunch of garbled media coming in.
> 
> Knowing which stream is which - AND knowing something about the content of the media.
> 
> >>You can say "don't do that", but there are desirable features that
> >>depend on it - namely "simultaneous ring". Potentially that can be
> >>replaced with presence-sensitive ringing to choose the *right* on to
> >>ring first, but that requires data that often isn't available.
> >
> >Google Voice does simultaneous ring quite nicely. If you call my GV
> >number, it will ring all my phones. If somebody answers a phone, GV will
> >say "Press 1 to accept a call from Paul Kyzivat", and if I press 1, GV
> >will terminate the other forks.
> 
> But, at that poing the call has been answered by Paul. It is whatever happens before that (no matter whether the call, from a SIP perspective, is in the early dialog phase or not) that is the problem.
> 
> >Webster did this years ago using the dynamicsoft stuff as a call controller.
> >
> >But to do this kind of work, the call controller needs signaling control
> >on each fork, which you don't get with early media.
> 
> It does if you create a dialog, using 18x (or, as you suggest, 200) before playing the media.
> 
> (The fact that SIP allows media to be sent before any response has been sent is a bug, in my opinion.)
> 
> >>> Offerless INVITEs help more, since the media can't come back to you
> >>> until the O/A has completed.
> >>
> >> That is just sticking your head in the sand. The media is still being
> >> generated, even if you don't know about it.
> >
> >No, really. They CANNOT send you media until you've sent your SDP.  If
> >you don't send your SDP (in the ACK) until they have sent theirs (in the
> >200OK), then there is NO EARLY MEDIA.
> >
> >Yes, this means that PSTN gateways would have to complete ACK before
> >doing IAM on the PSTN side. I'm OK with that.
> 
> But, if the ACK is sent to multiple remote destinations at the same
>  time you are still going to end up with the same
>  media-from-multiple-locations problem - it will just occur a little
>  later.

There's a difference between having multiple active dialogs, each with
related media, and having a bunch of early-media streams just coming at
you outside of a dialog context.

In this case, it's a different set of ACKs -- the caller could, for
example, serialize the acks and resort the order based on
application-determined probability of an interesting result. Or pick one
to play to the user and three more to hand to a challenge-response bot
just in case a human answers.

> But, using your offerless-INVITE-mechanish is not going to give you any
>  more information about the content of the media. It only allows you to
>  not receive media until you've sent the ACK.

Which gives you the "identity" aof the fork-terminator and any further
related information that might be available before sending the ACK.


>  Does GV have the possibility to receive media (other than
>  "meaningless" ringtones) from multiple locations?
> 

Yes. It uses an IVR to challenge the recipient and decide which fork to
"complete".

--
Dean




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To: 'Dean Willis' <dean.willis@softarmor.com>
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Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Cc: 'SIPCORE' <sipcore@ietf.org>, Gilad, Shaham <gilad@voxisoft.com>
Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,=20

>>True. But I don't understand why that is relevant in this context. The =20
>>problem is that you are receiving media from multiple locations, so I =20
>>don't see how it matters whether you are in the early dialog phase or =20
>>not.
>=20
>The problem is not just that you are receiving media from multiple locatio=
ns. It's that you're receiving data from multiple locations, you don't know=
 who those locations are or anything else about them,=20
>and you have no way to stop the media nor manipulate it in any useful way =
besides just discarding it.

That is the case if you receive media before you have received ANY response=
 (18x or 200), yes, and personally I think it should not be allowed to send=
 media in that way. "Fortunately", in many environments, such early media w=
ill never reach the UAC (due to gating procedures etc), so hopefully it wil=
l encourage people not to send early media that way :)

But, IF the UAS sends a 18x, and establishes an early dialog, I don't see t=
he difference.

Regards,

Christer




From christer.holmberg@ericsson.com  Mon Mar 29 20:04:12 2010
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Date: Tue, 30 Mar 2010 05:04:37 +0200
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Hi,

>>>>Its really only parallel forking thats a problem, so we can probably ig=
nore serial forking.
>>>
>>>Not really. Early media is always a problem. How do you tell it to stop =
without whacking all your forks?
>>=20
>>"inactive".
>
>You can't send it an inactive because it hasn't sent you an answer yet, so=
 you can't target that fork with an inactive.

I think we talk about different "early media" scenarios.

You talk about the case when there is NO answer coming from the UAS before =
the early media, while I talk about the case when there IS an answer (recei=
ved in a 18x) before the early media.

I AGREE with the issues related to your case, and I have not been arguing a=
gainst those :)=20

As I've said earlier, sending early media before an answer is not good, in =
my opinion.

I think the issue is that you and I have different solutions to that proble=
m: you think that a 200 OK should be sent before any media is sent, while I=
 think it is enough to send a 18x.

>>>Yes, this means that PSTN gateways would have to complete ACK before=20
>>>doing IAM on the PSTN side. I'm OK with that.
>>>=20
>>But, if the ACK is sent to multiple remote destinations at the same =20
>>time you are still going to end up with the same =20
>>media-from-multiple-locations problem - it will just occur a little =20
>>later.
>
>There's a difference between having multiple active dialogs, each with rel=
ated media, and having a bunch of early-media streams just coming at you ou=
tside of a dialog context.

I totally agree. Each media stream should have an associated dialog. But, t=
hat dialog could be an early dialog.

>>Does GV have the possibility to receive media (other than "meaningless" r=
ingtones) from multiple locations?
>=20
>Yes. It uses an IVR to challenge the recipient and decide which fork to "c=
omplete".

But, does it play the IVR message at the same time as the "ringtones" (whic=
h could also be anouncements), or does it not even contact the remote termi=
nals before the work to complete has been chosen?

Regards,

Christer



From ivo.sedlacek@ericsson.com  Mon Mar 29 23:11:04 2010
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From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 30 Mar 2010 08:11:24 +0200
Thread-Topic: [sipcore] <identity> element in RFC 4235
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Thanks for clarification.

See comment at the end.

Kind regards

Ivo Sedlacek

Ericsson
Technology and Portfolio Management, Terminal Standardization Sweden
Office: +46 10 711 9382
Fax: +46 10 713 5929
ivo.sedlacek@ericsson.com
www.ericsson.com

This communication is confidential. We only send and receive email on the b=
asis of the term set out at www.ericsson.com/email_disclaimer


-----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 26. b=F8ezna 2010 17:32
> To: Ivo Sedlacek
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] <identity> element in RFC 4235
>
> [as individual]
> comment at end...
>
> Ivo Sedlacek wrote:
> > Hello,
> >
> > Let's suppose that Alice called Bob and Bob forwarded the call to
> > Cecil which accepted the call.
> > When Alice's UE sent the INVITE, the To header was set to Bob's SIP
> > URI (sip:bob@example.com).
> > Cecil's network provided P-Asserted-Identity header set to Cecil's SIP
> > URI (sip:cecil@example.com) in 2xx response.
> >
> > Now Duren subscribes to dialog event package at Alice's UE.
> > Alice's UE generates dialog event package notification including the
> > created dialog.
> >
> > Is the <identity> element of the <remote> element of the created
> > dialog supposed to contain:
> > 1) the original To header field URI (sip:bob@example.com);
> > 2) the true identity of the participant (sip:cecil@example.com); or
> > 3) none of them?
> >
> > RFC4235 states:
> >
> > ----------------------------------------------------------------------
> > -------------------------------
> > 4.1.6.1.  Identity Element
> >
> >    The "identity" element indicates a local or remote URI, as defined i=
n
> >    [2] as appropriate.  It has an optional attribute, display, that
> >    contains the display name from the appropriate URI.
> >
> >       Note that multiple identities (for example a sip: URI and a tel:
> >       URI) could be included if they all correspond to the participant.
> >       To avoid repeating identity information in each request, the
> >       subscriber can assume that the identity URIs are the same as in
> >       previous notifications if no identity elements are present in the
> >       corresponding local or remote element.  If any identity elements
> >       are present in the local or remote part of a notification, the ne=
w
> >       list of identity tags completely supersedes the old list in the
> >       corresponding part.
> >
> >       <identity display=3D"Anonymous">
> >            sip:anonymous@anonymous.invalid</identity>
> > ----------------------------------------------------------------------
> > -------------------------------
> >
> > remote URI as defined in [2] (RFC3261) is the original To header field
> > URI (sip:bob@example.com).
> > However, the original To header field URI does not correspond to the
> > true remote participant (sip:cecil@example.com).
> > Does "as appropriate" have any particular meaning?
>
> You mention Cecil's PAI being returned in the 2xx. The semantics of that =
are not
> defined in ietf. (I believe this usage *is* defined in IMS.)
>
> RFC 4916 (connected identity) defines a mechanism to deliver that back to=
 the caller
> in a request. If 4916 is used, the actual remote identity that is part of=
 dialog
> state according to 3261 will be changed, and presumably will then be repo=
rted in
> the notify.
>
> The impact of PAI on returns from the dialog event package is, AFAIK, und=
efined.
> (Does IMS define it?)
>
>       Thanks,
>       Paul

IMS (so far) does not define any further rules above RFC 4235 on what to pu=
t in the dialog event package notification body.

From adam@nostrum.com  Tue Mar 30 11:38:09 2010
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[as chair]

A draft version of the minutes from the recent IETF meeting in Anaheim 
has been posted to the proceedings site:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html

Please review the minutes and submit any corrections to the chairs no 
later than April 15th. Thank you.

/a

From adam@nostrum.com  Tue Mar 30 11:50:55 2010
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Subject: [sipcore] Consensus Call: SIP Table 2
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[as chair]

Last week, at the face-to-face meeting in Anaheim, the SIPCORE 
participants in attendance had a discussion regarding the future of 
Tables 2 and 3 in RFC 3261. A summary of the conversation, along with 
links to the presentations and an audio recording of the conversation, 
can be found in the minutes:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203

During that conversation, the chairs polled those present for support 
between two options:

   1. Produce an artifact stating that documents defining new header
      fields and methods define semantics in prose, and will no longer
      extend Table 2; or

   2. Produce an RFC that defines new procedures for adding elements to
      Table 2 as new header fields and methods are defined

Among those present, there was strong support for option 1. If you wish 
to raise an objection to this direction, please do so on the SIPCORE 
mailing list before April 15th.

Thank you.

/a

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</head>
<body bgcolor="#ffffff" text="#000000">
[as chair]<br>
<br>
Last week, at the face-to-face meeting in Anaheim, the SIPCORE
participants in attendance had a discussion regarding the future of
Tables 2 and 3 in RFC 3261. A summary of the conversation, along with
links to the presentations and an audio recording of the conversation,
can be found in the minutes:<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203">http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203</a><br>
<br>
During that conversation, the chairs polled those present for support
between two options:<br>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
<ol>
  <li>Produce an artifact stating that documents defining new header
fields and methods define semantics in prose, and will no longer extend
Table 2; or<br>
    <br>
  </li>
  <li>Produce an RFC that defines new procedures for adding elements to
Table 2 as new header fields and methods are defined</li>
</ol>
Among those present, there was strong support for option 1. If you wish
to raise an objection to this direction, please do so on the SIPCORE
mailing list before April 15th.<br>
<br>
Thank you.<br>
<br>
/a<br>
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--------------030001080907000408040909--

From adam@nostrum.com  Tue Mar 30 12:05:11 2010
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Subject: [sipcore] SIP OPTIONS: Shall we work on this?
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[as chair]

One of the topics of discussion at the recent face-to-face meeting in 
Anaheim was the SIP OPTIONS method. We have had significant discussion 
of some specific technical topics on the SIPCORE mailing list, such that 
all participants should have a reasonable overview of the scope that any 
related work may entail. Participants may also benefit from reading 
through the minutes of the corresponding conversation in Anaheim:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTIONS%20Method

Rather than continue discussion of the specific technical topics related 
to OPTIONS, the chairs would encourage the working group to engage in a 
discussion around whether we should, as a group, work on addressing some 
of the issues that have been identified.

In particular, we would direct the group to consider the nature of such 
work:

    * Are we proposing to introduce new features, or fix existing bugs?
    * If we are adding new features: is OPTIONS the best way to
      introduce these features, or would other mechanisms (presence,
      callee-caps) produce superior results?
    * If we are fixing bugs: are the bugs theoretical, or can we cite
      actual interoperability issues in the field?
    * If there are identified interoperability issues: is the level of
      effort involved in fixing them commensurate with the impact of the
      problems?

Finally, keep in mind that work does not happen unless participants are 
willing to invest time. If you speak out in favor of pursuing this work, 
please indicate the level of time commitment that you will personally 
invest in bringing such work to a successful conclusion (e.g., will 
author drafts; will perform dedicated reviews; etc).

Thanks.

/a

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>

<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body bgcolor="#ffffff" text="#000000">
[as chair]<br>
<br>
One of the topics of discussion at the recent face-to-face meeting in
Anaheim was the SIP OPTIONS method. We have had significant discussion
of some specific technical topics on the SIPCORE mailing list, such
that all participants should have a reasonable overview of the scope
that any related work may entail. Participants may also benefit from
reading through the minutes of the corresponding conversation in
Anaheim:<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTIONS%20Method">http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTIONS%20Method</a><br>
<br>
Rather than continue discussion of the specific technical topics
related to OPTIONS, the chairs would encourage the working group to
engage in a discussion around whether we should, as a group, work on
addressing some of the issues that have been identified.<br>
<br>
In particular, we would direct the group to consider the nature of such
work:<br>
<ul>
  <li>Are we proposing to introduce new features, or fix existing bugs?</li>
  <li>If we are adding new features: is OPTIONS the best way to
introduce these features, or would other mechanisms (presence,
callee-caps) produce superior results?<br>
  </li>
  <li>If we are fixing bugs: are the bugs theoretical, or can we cite
actual interoperability issues in the field?</li>
  <li>If there are identified interoperability issues: is the level of
effort involved in fixing them commensurate with the impact of the
problems?</li>
</ul>
Finally, keep in mind that work does not happen unless participants are
willing to invest time. If you speak out in favor of pursuing this
work, please indicate the level of time commitment that you will
personally invest in bringing such work to a successful conclusion
(e.g., will author drafts; will perform dedicated reviews; etc).<br>
<br>
Thanks.<br>
<br>
/a<br>
</body>
</html>

--------------070406010406090507020301--

From adam@nostrum.com  Tue Mar 30 13:04:47 2010
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Subject: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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[as participant]

During the Anaheim meeting, I made an assertion that the optional 
feature of re-creating dialogs upon receipt of an in-dialog INVITE 
(i.e., one with a "To" tag) that did not match an ongoing dialog had 
been removed between RFC 2543 and RFC 3261.

I was wrong.

What I was recalling was a related change in handling; between 2543 and 
3261, the creation of such zombie dialogs changed from mandatory to 
optional (instead of going from "optional" to "prohibited," as I 
mistakenly recalled).

In 2543, the arrival of an apparently-in-dialog INVITE that didn't match 
an ongoing "call-leg" (dialog) would always create a new "call leg"; the 
called party had no way of rejecting the call with an indication that 
the associated "call leg" had been previously terminated (RFC 2543, 
section 10.1.1) -- the 481 response code had semantics only for CANCEL 
and BYE.

In 3261, the UAS is given the _option_ of re-creating the dialog under 
such circumstances. From 3261, section 12.2.2:

>     If the request has a tag in the To header field, but the dialog
>     identifier does not match any existing dialogs, the UAS may have
>     crashed and restarted, or it may have received a request for a
>     different (possibly failed) UAS (the UASs can construct the To tags
>     so that a UAS can identify that the tag was for a UAS for which it is
>     providing recovery).  Another possibility is that the incoming
>     request has been simply misrouted.  Based on the To tag, the UAS MAY
>     either accept or reject the request.  Accepting the request for
>     acceptable To tags provides robustness, so that dialogs can persist
>     even through crashes.  UAs wishing to support this capability must
>     take into consideration some issues such as choosing monotonically
>     increasing CSeq sequence numbers even across reboots, reconstructing
>     the route set, and accepting out-of-range RTP timestamps and sequence
>     numbers.
>
>     If the UAS wishes to reject the request because it does not wish to
>     recreate the dialog, it MUST respond to the request with a 481
>     (Call/Transaction Does Not Exist) status code and pass that to the
>     server transaction.
>    

In the interest of not digging into the actual details of the 
OPTIONS-related issues, I am intentionally refraining from commenting on 
the relationship between this behavior and the associated OPTIONS 
discussion topic from the Anaheim meeting.

/a

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<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
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<body bgcolor="#ffffff" text="#000000">
[as participant]<br>
<br>
During the Anaheim meeting, I made an assertion that the optional
feature of re-creating dialogs upon receipt of an in-dialog INVITE
(i.e., one with a "To" tag) that did not match an ongoing dialog had
been removed between RFC 2543 and RFC 3261.<br>
<br>
I was wrong.<br>
<br>
What I was recalling was a related change in handling; between 2543 and
3261, the creation of such zombie dialogs changed from mandatory to
optional (instead of going from "optional" to "prohibited," as I
mistakenly recalled).<br>
<br>
In 2543, the arrival of an apparently-in-dialog INVITE that didn't
match an ongoing "call-leg" (dialog) would always create a new "call
leg"; the called party had no way of rejecting the call with an
indication that the associated "call leg" had been previously
terminated (RFC 2543, section 10.1.1) -- the 481 response code had
semantics only for CANCEL and BYE.<br>
<br>
In 3261, the UAS is given the _option_ of re-creating the dialog under
such circumstances. From 3261, section 12.2.2:<br>
<br>
<blockquote type="cite">
  <meta http-equiv="content-type"
 content="text/html; charset=ISO-8859-1">
  <pre>   If the request has a tag in the To header field, but the dialog
   identifier does not match any existing dialogs, the UAS may have
   crashed and restarted, or it may have received a request for a
   different (possibly failed) UAS (the UASs can construct the To tags
   so that a UAS can identify that the tag was for a UAS for which it is
   providing recovery).  Another possibility is that the incoming
   request has been simply misrouted.  Based on the To tag, the UAS MAY
   either accept or reject the request.  Accepting the request for
   acceptable To tags provides robustness, so that dialogs can persist
   even through crashes.  UAs wishing to support this capability must
   take into consideration some issues such as choosing monotonically
   increasing CSeq sequence numbers even across reboots, reconstructing
   the route set, and accepting out-of-range RTP timestamps and sequence
   numbers.

   If the UAS wishes to reject the request because it does not wish to
   recreate the dialog, it MUST respond to the request with a 481
   (Call/Transaction Does Not Exist) status code and pass that to the
   server transaction.
  </pre>
</blockquote>
<br>
In the interest of not digging into the actual details of the
OPTIONS-related issues, I am intentionally refraining from commenting
on the relationship between this behavior and the associated OPTIONS
discussion topic from the Anaheim meeting.<br>
<br>
/a<br>
</body>
</html>

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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Date: Wed, 31 Mar 2010 07:45:48 +0200
Thread-Topic: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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Subject: Re: [sipcore] INVITE 3xx and early media [was OPTIONS 3xx: To fork or not to fork?]
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From christer.holmberg@ericsson.com  Tue Mar 30 22:55:41 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Adam Roach' <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 31 Mar 2010 07:56:05 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,

Adam, thanks for sending out the the e-mail!

First, a clarification regarding "new features":

When we say "new features", I assume (based on Hadriel's comment in the mee=
ting) it means the possibility to, with a single OPTIONS request, retrieve =
the capabilities from ALL UAS(s) behind a forking proxy - instead of only o=
ne UAS (whoever sends the 200 OK to the forking proxy first).

The message I got from the meeting, and also previous e-mail discussions, w=
as that the "OPTIONS 3xx" solution already provideds this feature, and that=
 it can be used (without defining new protocol elements etc).

So, in my opinion the "adding" word is a little missleading. Instead I thin=
k the question is whether we want to DOCUMENT that existing feature.


Second, regarding bugs and interoperability issues in the field. Personally=
 I am not sure of any major ones at the moment, but that is probably becaus=
e OPTIONS haven't been very much deployed in the first place (it has been u=
sed for some keep-alive/ping purposes, but in those cases it really doesn't=
 matter what comes in the response etc). Now, however, at least I am starti=
ng to get questions regarding the usage of OPTIONS, so I guess we need to a=
sk ourselves whether we think the bugs WILL create interoperability issues.

And, the issue you raised regarding using the re-INVITE to re-create dialog=
s ("[sipcore] Mea Culpa: Dialog Necromancy in RFC 3261") also affects OPTIO=
NS, eventhough you want to keep them separate (which is ok).

Obviously I am in favor of pursuing the work, and will be willing to author=
 drafts :)

Regards,

Christer




________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: 30. maaliskuuta 2010 22:06
To: SIPCORE
Subject: [sipcore] SIP OPTIONS: Shall we work on this?

[as chair]

One of the topics of discussion at the recent face-to-face meeting in Anahe=
im was the SIP OPTIONS method. We have had significant discussion of some s=
pecific technical topics on the SIPCORE mailing list, such that all partici=
pants should have a reasonable overview of the scope that any related work =
may entail. Participants may also benefit from reading through the minutes =
of the corresponding conversation in Anaheim:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTI=
ONS%20Method

Rather than continue discussion of the specific technical topics related to=
 OPTIONS, the chairs would encourage the working group to engage in a discu=
ssion around whether we should, as a group, work on addressing some of the =
issues that have been identified.

In particular, we would direct the group to consider the nature of such wor=
k:

 *   Are we proposing to introduce new features, or fix existing bugs?
 *   If we are adding new features: is OPTIONS the best way to introduce th=
ese features, or would other mechanisms (presence, callee-caps) produce sup=
erior results?
 *   If we are fixing bugs: are the bugs theoretical, or can we cite actual=
 interoperability issues in the field?
 *   If there are identified interoperability issues: is the level of effor=
t involved in fixing them commensurate with the impact of the problems?

Finally, keep in mind that work does not happen unless participants are wil=
ling to invest time. If you speak out in favor of pursuing this work, pleas=
e indicate the level of time commitment that you will personally invest in =
bringing such work to a successful conclusion (e.g., will author drafts; wi=
ll perform dedicated reviews; etc).

Thanks.

/a

--_000_FF84A09F50A6DC48ACB6714F4666CC745E21BEF4EAESESSCMS0354e_
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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.6000.16982" name=3DGENERATOR></HEAD>
<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>Hi,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>Adam, thanks for sending out the the=20
e-mail!</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>First,&nbsp;a clarification regarding "new featu=
res":=20
</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>When&nbsp;we say&nbsp;"new features", I assume (=
based=20
on&nbsp;Hadriel's comment in the meeting)&nbsp;it means the possibility to,=
 with=20
a single OPTIONS request,&nbsp;retrieve the capabilities from ALL UAS(s) be=
hind=20
a forking proxy -&nbsp;instead of only one UAS&nbsp;(whoever sends the 200 =
OK to=20
the forking proxy first).</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>The message I got from the meeting, and also pre=
vious=20
e-mail discussions, was that the "OPTIONS 3xx" solution already provideds t=
his=20
feature, and that it can be used (without defining new protocol elements=20
etc).</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>So, in my opinion the "adding" word is a little=
=20
missleading. Instead I think the question is whether we want to DOCUMENT th=
at=20
existing feature.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>Second, regarding bugs and interoperability issu=
es in=20
the field. Personally I am not sure of any major ones at the moment, but th=
at is=20
probably because OPTIONS haven't been very much deployed&nbsp;in the first =
place=20
(it has been used for some keep-alive/ping purposes, but in those cases it=
=20
really doesn't matter what comes in the response etc). Now, however, at lea=
st I=20
am starting to get questions regarding the usage of OPTIONS, so I guess we =
need=20
to ask ourselves whether we think the bugs WILL create interoperability=20
issues.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>And, the issue you raised regarding&nbsp;using t=
he=20
re-INVITE to re-create dialogs ("[sipcore] Mea Culpa: Dialog Necromancy in =
RFC=20
3261") also affects OPTIONS, eventhough you want to keep them separate (whi=
ch is=20
ok).</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>Obviously I am in favor of pursuing the work, an=
d will=20
be willing to author drafts :)</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D634293302-31032010>Christer</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sipcore-bounces@ietf.org=20
[mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>Adam Roach<BR><B>Sent=
:</B>=20
30. maaliskuuta 2010 22:06<BR><B>To:</B> SIPCORE<BR><B>Subject:</B> [sipcor=
e]=20
SIP OPTIONS: Shall we work on this?<BR></FONT><BR></DIV>
<DIV></DIV>[as chair]<BR><BR>One of the topics of discussion at the recent=
=20
face-to-face meeting in Anaheim was the SIP OPTIONS method. We have had=20
significant discussion of some specific technical topics on the SIPCORE mai=
ling=20
list, such that all participants should have a reasonable overview of the s=
cope=20
that any related work may entail. Participants may also benefit from readin=
g=20
through the minutes of the corresponding conversation in Anaheim:<BR><BR><A=
=20
class=3Dmoz-txt-link-freetext=20
href=3D"http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SI=
P%20OPTIONS%20Method">http://www.ietf.org/proceedings/10mar/minutes/sipcore=
.html#The%20SIP%20OPTIONS%20Method</A><BR><BR>Rather=20
than continue discussion of the specific technical topics related to OPTION=
S,=20
the chairs would encourage the working group to engage in a discussion arou=
nd=20
whether we should, as a group, work on addressing some of the issues that h=
ave=20
been identified.<BR><BR>In particular, we would direct the group to conside=
r the=20
nature of such work:<BR>
<UL>
  <LI>Are we proposing to introduce new features, or fix existing bugs?=20
  <LI>If we are adding new features: is OPTIONS the best way to introduce t=
hese=20
  features, or would other mechanisms (presence, callee-caps) produce super=
ior=20
  results?<BR>
  <LI>If we are fixing bugs: are the bugs theoretical, or can we cite actua=
l=20
  interoperability issues in the field?=20
  <LI>If there are identified interoperability issues: is the level of effo=
rt=20
  involved in fixing them commensurate with the impact of the problems?=20
</LI></UL>Finally, keep in mind that work does not happen unless participan=
ts=20
are willing to invest time. If you speak out in favor of pursuing this work=
,=20
please indicate the level of time commitment that you will personally inves=
t in=20
bringing such work to a successful conclusion (e.g., will author drafts; wi=
ll=20
perform dedicated reviews; etc).<BR><BR>Thanks.<BR><BR>/a<BR></BODY></HTML>

--_000_FF84A09F50A6DC48ACB6714F4666CC745E21BEF4EAESESSCMS0354e_--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Wed, 31 Mar 2010 10:19:13 +0200
Thread-Topic: RFC3265/3265bis and draft-ietf-sipcore-subnot-etags issue regarding notification suppression
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Subject: [sipcore] RFC3265/3265bis and draft-ietf-sipcore-subnot-etags issue regarding notification suppression
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Hi,

A question regarding the notification suppression function in draft-ietf-si=
pcore-subnot-etags:

The draft allows NOTIFY requests to be suppressed for re/de-SUBSCRIBEs.

Section 5.8 of the draft says that the suppression might be an issue for B2=
BUAs that expect a NOTIFY. But, doesn't this issue also apply to stateful p=
roxies which keep track of subscription dialogs? Don't they also expect NOT=
IFY? Or, is a proxy which maintains subscription dialog state considered to=
 be a B2BUA?

So, when NOTIFY is suppressed for de-subscribe then stateful proxies (or wh=
atever they are called) which keep track of subscription dialogs and which =
do not support draft-ietf-sipcore-subnot-etags-04 will still believe the su=
bscription dialog exists.

Now, 3265bis defines timer N, so if it applies also to proxies (open issue =
in the draft) I assume that proxies will simply ignore the re/de-subscripti=
on in case of no associated NOTIFY within a certain time, and eventually th=
e subscription will expire and the proxy will release the state. However, I=
 think that is bad protocol design.

RFC3265 does not define timer N, though, so the behavior for RFC compliant =
proxies is unclear.

Now, since the NOTIFY request is so essential (at least in 3265bis the SUB =
200 OK becomes almost meaningless from a functional perspective) to the who=
le SUB/NOT framework, I wonder whether it is a very good idea to allow the =
NOTIFY request suppression. I DO like the idea of being able to suppress th=
e NOTIFY message body, and maybe suppressing the NOTIFY request for re-subs=
criptions isn't that dangerous. But, suppressing the NOTIFY for de-subscrip=
tions is bad, in my opinion.

Regards,

Christer








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>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<font face=3D"Arial, sans-serif" size=3D"3">
<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">A question regarding the notification suppression fun=
ction in draft-ietf-sipcore-subnot-etags:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">The draft allows NOTIFY requests to be suppressed for=
 re/de-SUBSCRIBEs.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Section 5.8 of the draft says that the suppression mi=
ght be an issue for B2BUAs that expect a NOTIFY. But, doesn't this issue al=
so apply to stateful proxies which keep track of subscription dialogs? Don'=
t they also expect NOTIFY? Or, is
a proxy which maintains subscription dialog state considered to be a B2BUA?=
</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">So, when NOTIFY is suppressed for de-subscribe then s=
tateful proxies (or whatever they are called) which keep track of subscript=
ion dialogs and which do not support draft-ietf-sipcore-subnot-etags-04 wil=
l still believe the subscription dialog
exists.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Now, 3265bis defines timer N, so if it applies also t=
o proxies (open issue in the draft) I assume that proxies will simply ignor=
e the re/de-subscription in case of no associated NOTIFY within a certain t=
ime, and eventually the subscription
will expire and the proxy will release the state. However, I think that is =
bad protocol design.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">RFC3265 does not define timer N, though, so the behav=
ior for RFC compliant proxies is unclear.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Now, since the NOTIFY request is so essential (at lea=
st in 3265bis the SUB 200 OK becomes almost meaningless from a functional p=
erspective) to the whole SUB/NOT framework, I wonder whether it is a very g=
ood idea to allow the NOTIFY request
suppression. I DO like the idea of being able to suppress the NOTIFY messag=
e body, and maybe suppressing the NOTIFY request for re-subscriptions isn't=
 that dangerous. But, suppressing the NOTIFY for de-subscriptions is bad, i=
n my opinion.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
</font>
</body>
</html>

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From christer.holmberg@ericsson.com  Wed Mar 31 01:30:38 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 31 Mar 2010 10:27:07 +0200
Thread-Topic: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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Subject: Re: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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Hi Adam,

Mabye you were thinking about RFC 4028 (session timer)? It's a while since =
we worked on that draft, but I THINK we in some early versions had some tex=
t about session re-creation, which was later removed. And, based on a quick=
 look, the RFC only talks about session re-fresh.

I think the usage of re-INVITE to re-create a dialog is bad, and will most =
likely cause all kind of issues. For example, there is no indication whethe=
r the UAC even wants the dialog to be re-created in the first place, in cas=
e it doesn't exist. And, in order to work any entity involved will have to =
be able to restore all resources (related to media, NAT traversal, security=
 etc etc etc) from when the dialog existed - which means they have to maint=
ain data for that dialog, and maybe even some procedures (keep-alives etc).

Now, I didn't see any question in your e-mail, e.g. wheter we should do som=
ething about this, but my opinion is still: do not allow it :)

Regards,

Christer




________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: 30. maaliskuuta 2010 23:05
To: SIPCORE
Subject: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261

[as participant]

During the Anaheim meeting, I made an assertion that the optional feature o=
f re-creating dialogs upon receipt of an in-dialog INVITE (i.e., one with a=
 "To" tag) that did not match an ongoing dialog had been removed between RF=
C 2543 and RFC 3261.

I was wrong.

What I was recalling was a related change in handling; between 2543 and 326=
1, the creation of such zombie dialogs changed from mandatory to optional (=
instead of going from "optional" to "prohibited," as I mistakenly recalled)=
.

In 2543, the arrival of an apparently-in-dialog INVITE that didn't match an=
 ongoing "call-leg" (dialog) would always create a new "call leg"; the call=
ed party had no way of rejecting the call with an indication that the assoc=
iated "call leg" had been previously terminated (RFC 2543, section 10.1.1) =
-- the 481 response code had semantics only for CANCEL and BYE.

In 3261, the UAS is given the _option_ of re-creating the dialog under such=
 circumstances. From 3261, section 12.2.2:


   If the request has a tag in the To header field, but the dialog
   identifier does not match any existing dialogs, the UAS may have
   crashed and restarted, or it may have received a request for a
   different (possibly failed) UAS (the UASs can construct the To tags
   so that a UAS can identify that the tag was for a UAS for which it is
   providing recovery).  Another possibility is that the incoming
   request has been simply misrouted.  Based on the To tag, the UAS MAY
   either accept or reject the request.  Accepting the request for
   acceptable To tags provides robustness, so that dialogs can persist
   even through crashes.  UAs wishing to support this capability must
   take into consideration some issues such as choosing monotonically
   increasing CSeq sequence numbers even across reboots, reconstructing
   the route set, and accepting out-of-range RTP timestamps and sequence
   numbers.

   If the UAS wishes to reject the request because it does not wish to
   recreate the dialog, it MUST respond to the request with a 481
   (Call/Transaction Does Not Exist) status code and pass that to the
   server transaction.


In the interest of not digging into the actual details of the OPTIONS-relat=
ed issues, I am intentionally refraining from commenting on the relationshi=
p between this behavior and the associated OPTIONS discussion topic from th=
e Anaheim meeting.

/a

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<META content=3D"MSHTML 6.00.6000.16982" name=3DGENERATOR></HEAD>
<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010>Hi Adam,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010>Mabye you were thinking about RFC 4028 (session =
timer)?=20
</SPAN></FONT><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D294491908-31032010>It's a while since we worked on that draft, but =
I THINK=20
we in some early versions had some text about session re-creation, which wa=
s=20
later removed. And,&nbsp;based on a quick look, the RFC only talks about se=
ssion=20
re-fresh.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010>I think the usage of re-INVITE to re-create a di=
alog is=20
bad, and will most likely cause all kind of issues. For example, there is n=
o=20
indication whether the UAC even wants the dialog to be re-created in the fi=
rst=20
place,&nbsp;in case it doesn't exist. And, in order to work any entity invo=
lved=20
will have to be able to restore all resources (related to media, NAT traver=
sal,=20
security etc etc etc) from when the dialog existed - which means they have =
to=20
maintain data for that dialog, and maybe even some procedures (keep-alives=
=20
etc).</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010>Now, I didn't see any question in your e-mail, e=
.g.=20
wheter we should do something about this, but my opinion is still: do not a=
llow=20
it :)</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D294491908-31032010>Christer</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipcore-bounces@ietf.org=20
  [mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>Adam=20
  Roach<BR><B>Sent:</B> 30. maaliskuuta 2010 23:05<BR><B>To:</B>=20
  SIPCORE<BR><B>Subject:</B> [sipcore] Mea Culpa: Dialog Necromancy in RFC=
=20
  3261<BR></FONT><BR></DIV>
  <DIV></DIV>[as participant]<BR><BR>During the Anaheim meeting, I made an=
=20
  assertion that the optional feature of re-creating dialogs upon receipt o=
f an=20
  in-dialog INVITE (i.e., one with a "To" tag) that did not match an ongoin=
g=20
  dialog had been removed between RFC 2543 and RFC 3261.<BR><BR>I was=20
  wrong.<BR><BR>What I was recalling was a related change in handling; betw=
een=20
  2543 and 3261, the creation of such zombie dialogs changed from mandatory=
 to=20
  optional (instead of going from "optional" to "prohibited," as I mistaken=
ly=20
  recalled).<BR><BR>In 2543, the arrival of an apparently-in-dialog INVITE =
that=20
  didn't match an ongoing "call-leg" (dialog) would always create a new "ca=
ll=20
  leg"; the called party had no way of rejecting the call with an indicatio=
n=20
  that the associated "call leg" had been previously terminated (RFC 2543,=
=20
  section 10.1.1) -- the 481 response code had semantics only for CANCEL an=
d=20
  BYE.<BR><BR>In 3261, the UAS is given the _option_ of re-creating the dia=
log=20
  under such circumstances. From 3261, section 12.2.2:<BR><BR>
  <BLOCKQUOTE type=3D"cite"><PRE>   If the request has a tag in the To head=
er field, but the dialog
   identifier does not match any existing dialogs, the UAS may have
   crashed and restarted, or it may have received a request for a
   different (possibly failed) UAS (the UASs can construct the To tags
   so that a UAS can identify that the tag was for a UAS for which it is
   providing recovery).  Another possibility is that the incoming
   request has been simply misrouted.  Based on the To tag, the UAS MAY
   either accept or reject the request.  Accepting the request for
   acceptable To tags provides robustness, so that dialogs can persist
   even through crashes.  UAs wishing to support this capability must
   take into consideration some issues such as choosing monotonically
   increasing CSeq sequence numbers even across reboots, reconstructing
   the route set, and accepting out-of-range RTP timestamps and sequence
   numbers.

   If the UAS wishes to reject the request because it does not wish to
   recreate the dialog, it MUST respond to the request with a 481
   (Call/Transaction Does Not Exist) status code and pass that to the
   server transaction.
  </PRE></BLOCKQUOTE><BR>In the interest of not digging into the actual=20
  details of the OPTIONS-related issues, I am intentionally refraining from=
=20
  commenting on the relationship between this behavior and the associated=20
  OPTIONS discussion topic from the Anaheim=20
meeting.<BR><BR>/a<BR></BLOCKQUOTE></BODY></HTML>

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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] SIP OPTIONS: Shall we work on this?
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Christer,

[as chair]

By accepting the 3xx solution to handling multiple UAs, you have removed 
the most obvious "new feature". There may or may not be other less 
obvious ones lurking in there. For instance, you (and I) have suggested 
using Request-Disposition:redirect to "ensure" the 3xx is returned. But 
honoring R-D is optional by the proxy. So at the moment there is no way 
to be certain that you are getting info for all the available UAs. 
Anything that somehow made honoring R-D mandatory for OPTIONS would be a 
new feature.

Also, there are other cases where the behavior is currently 
optional/ambiguous. "Fixing" those to be unambiguous/non-optional can 
also be construed as a *change*. (There is a fine line here about what 
is intended by the current texts.)

It will certainly be easier if no normative changes are required.

In any case, having you willing to do the work is a good thing. But 
there had better also be others that are interested in the work enough 
to review and comment on it, and to justify spending list and meeting 
time on.

[As individual]

I have "mild" interest in this from an academic perspective, but I don't 
see it as necessary work. I'm waiting to see how much other interest 
there is.

	Thanks,
	Paul

Christer Holmberg wrote:
> Hi,
>  
> Adam, thanks for sending out the the e-mail!
>  
> First, a clarification regarding "new features":
>  
> When we say "new features", I assume (based on Hadriel's comment in the 
> meeting) it means the possibility to, with a single OPTIONS 
> request, retrieve the capabilities from ALL UAS(s) behind a forking 
> proxy - instead of only one UAS (whoever sends the 200 OK to the forking 
> proxy first).
>  
> The message I got from the meeting, and also previous e-mail 
> discussions, was that the "OPTIONS 3xx" solution already provideds this 
> feature, and that it can be used (without defining new protocol elements 
> etc).
>  
> So, in my opinion the "adding" word is a little missleading. Instead I 
> think the question is whether we want to DOCUMENT that existing feature.
>  
>  
> Second, regarding bugs and interoperability issues in the field. 
> Personally I am not sure of any major ones at the moment, but that is 
> probably because OPTIONS haven't been very much deployed in the first 
> place (it has been used for some keep-alive/ping purposes, but in those 
> cases it really doesn't matter what comes in the response etc). Now, 
> however, at least I am starting to get questions regarding the usage of 
> OPTIONS, so I guess we need to ask ourselves whether we think the bugs 
> WILL create interoperability issues.
>  
> And, the issue you raised regarding using the re-INVITE to re-create 
> dialogs ("[sipcore] Mea Culpa: Dialog Necromancy in RFC 3261") also 
> affects OPTIONS, eventhough you want to keep them separate (which is ok).
>  
> Obviously I am in favor of pursuing the work, and will be willing to 
> author drafts :)
>  
> Regards,
>  
> Christer
>  
>  
>  
> 
> ------------------------------------------------------------------------
> *From:* sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] *On 
> Behalf Of *Adam Roach
> *Sent:* 30. maaliskuuta 2010 22:06
> *To:* SIPCORE
> *Subject:* [sipcore] SIP OPTIONS: Shall we work on this?
> 
> [as chair]
> 
> One of the topics of discussion at the recent face-to-face meeting in 
> Anaheim was the SIP OPTIONS method. We have had significant discussion 
> of some specific technical topics on the SIPCORE mailing list, such that 
> all participants should have a reasonable overview of the scope that any 
> related work may entail. Participants may also benefit from reading 
> through the minutes of the corresponding conversation in Anaheim:
> 
> http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%20OPTIONS%20Method
> 
> Rather than continue discussion of the specific technical topics related 
> to OPTIONS, the chairs would encourage the working group to engage in a 
> discussion around whether we should, as a group, work on addressing some 
> of the issues that have been identified.
> 
> In particular, we would direct the group to consider the nature of such 
> work:
> 
>     * Are we proposing to introduce new features, or fix existing bugs?
>     * If we are adding new features: is OPTIONS the best way to
>       introduce these features, or would other mechanisms (presence,
>       callee-caps) produce superior results?
>     * If we are fixing bugs: are the bugs theoretical, or can we cite
>       actual interoperability issues in the field?
>     * If there are identified interoperability issues: is the level of
>       effort involved in fixing them commensurate with the impact of the
>       problems? 
> 
> Finally, keep in mind that work does not happen unless participants are 
> willing to invest time. If you speak out in favor of pursuing this work, 
> please indicate the level of time commitment that you will personally 
> invest in bringing such work to a successful conclusion (e.g., will 
> author drafts; will perform dedicated reviews; etc).
> 
> Thanks.
> 
> /a
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From adam@nostrum.com  Wed Mar 31 07:22:39 2010
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Subject: Re: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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On 3/31/10 3:27 AM, Christer Holmberg wrote:
> Now, I didn't see any question in your e-mail, e.g. wheter we should 
> do something about this, but my opinion is still: do not allow it :)
>

Sorry if I was unclear in the original email. It wasn't a question; it 
was a correction. I didn't want people laboring under a misconception 
because of my confusion.

This behavior is hardly something we need to go back and change in RFC 3261.

/a

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 http-equiv="Content-Type">
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<body bgcolor="#ffffff" text="#000000">
On 3/31/10 3:27 AM, Christer Holmberg wrote:
<blockquote
 cite="mid:FF84A09F50A6DC48ACB6714F4666CC745E21BEF611@ESESSCMS0354.eemea.ericsson.se"
 type="cite">
  <div dir="ltr" align="left"><font color="#0000ff" face="Arial"
 size="2"><span class="294491908-31032010">Now, I didn't see any
question in your e-mail, e.g. wheter we should do something about this,
but my opinion is still: do not allow it :)</span></font></div>
  <br>
</blockquote>
<br>
Sorry if I was unclear in the original email. It wasn't a question; it
was a correction. I didn't want people laboring under a misconception
because of my confusion.<br>
<br>
This behavior is hardly something we need to go back and change in RFC
3261.<br>
<br>
/a<br>
</body>
</html>

--------------090404090605050405010007--

From christer.holmberg@ericsson.com  Wed Mar 31 12:48:20 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Adam Roach' <adam@nostrum.com>
Date: Wed, 31 Mar 2010 21:48:47 +0200
Thread-Topic: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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Subject: Re: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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So, you are saying that it is ok to re-crete dialogs using re-INVITEs?

If that's the way we want to go, fine. I just hope people actually won't st=
art re-creating dialogs, because I think that could cause interop issues an=
d state unsynchronization - without anyone to blame, since everyone can cla=
im that they are acting according to the RFC...

Also, as I said earlier, I thought be decided that the session timer functi=
onality could not be used to re-created dialogs. But, again, it's a while s=
ince we did that work, so...

Regards,

Christer

________________________________
From: Adam Roach [mailto:adam@nostrum.com]
Sent: 31. maaliskuuta 2010 17:23
To: Christer Holmberg
Cc: SIPCORE
Subject: Re: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261

On 3/31/10 3:27 AM, Christer Holmberg wrote:
Now, I didn't see any question in your e-mail, e.g. wheter we should do som=
ething about this, but my opinion is still: do not allow it :)


Sorry if I was unclear in the original email. It wasn't a question; it was =
a correction. I didn't want people laboring under a misconception because o=
f my confusion.

This behavior is hardly something we need to go back and change in RFC 3261=
.

/a

--_000_FF84A09F50A6DC48ACB6714F4666CC745E21B5A905ESESSCMS0354e_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.3790.4639" name=3DGENERATOR></HEAD>
<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>So, you are saying that it is ok to re-crete dialo=
gs using=20
re-INVITEs?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>If that's the way we want to go, fine. I just hope=
 people=20
actually won't start re-creating dialogs, because I think that could=20
cause&nbsp;interop issues and state unsynchronization - without anyone to b=
lame,=20
since everyone&nbsp;can claim that they are acting&nbsp;according to the=20
RFC...</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Also, as I said earlier, I thought be decided that=
 the=20
session timer functionality could not be used to re-created dialogs. But, a=
gain,=20
it's a while since we did that work, so...</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D027284319-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Adam Roach [mailto:adam@nostrum.c=
om]=20
<BR><B>Sent:</B> 31. maaliskuuta 2010 17:23<BR><B>To:</B> Christer=20
Holmberg<BR><B>Cc:</B> SIPCORE<BR><B>Subject:</B> Re: [sipcore] Mea Culpa:=
=20
Dialog Necromancy in RFC 3261<BR></FONT><BR></DIV>
<DIV></DIV>On 3/31/10 3:27 AM, Christer Holmberg wrote:=20
<BLOCKQUOTE=20
cite=3Dmid:FF84A09F50A6DC48ACB6714F4666CC745E21BEF611@ESESSCMS0354.eemea.er=
icsson.se=20
type=3D"cite">
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D294491908-31032010>Now, I didn't see any question in your e-mail,=
 e.g.=20
  wheter we should do something about this, but my opinion is still: do not=
=20
  allow it :)</SPAN></FONT></DIV><BR></BLOCKQUOTE><BR>Sorry if I was unclea=
r in=20
the original email. It wasn't a question; it was a correction. I didn't wan=
t=20
people laboring under a misconception because of my confusion.<BR><BR>This=
=20
behavior is hardly something we need to go back and change in RFC=20
3261.<BR><BR>/a<BR></BODY></HTML>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@cisco.com>
Date: Wed, 31 Mar 2010 22:16:39 +0200
Thread-Topic: [sipcore] SIP OPTIONS: Shall we work on this?
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Hi,=20

>[as chair]
>
>By accepting the 3xx solution to handling multiple UAs, you have removed t=
he most obvious "new feature". There may or may not be other less obvious o=
nes lurking in there. For instance, you (and I) have=20
>suggested using Request-Disposition:redirect to "ensure" the 3xx is return=
ed. But honoring R-D is optional by the proxy. So at the moment there is no=
 way to be certain that you are getting info for all=20
>the available UAs.=20
>Anything that somehow made honoring R-D mandatory for OPTIONS would be a n=
ew feature.

I agree.

To clarify, I have not suggested anything that would make honoring R-D mand=
atory. I am fully aware that proxies may not do that (due to policy rules, =
or the fact that they don't support R-D in the first place). That is true f=
or the 3841 mechanism in general. So, if the proxy does not honor R-D, thin=
gs will work as today.

I am not even sure how we could mandate the proxy to honor R-D. Proxy-Requi=
re is bad, since it affects all proxies, and we don't have a Proxy-Require-=
If-You-Intend-To-Fork header field :)


>Also, there are other cases where the behavior is currently optional/ambig=
uous. "Fixing" those to be unambiguous/non-optional can also be construed a=
s a *change*. (There is a fine line here about what is=20
>intended by the current texts.)
>
>It will certainly be easier if no normative changes are required.

I agree.

Personally, based on the studies and discussions so far, I think a BCP woul=
d be enough. Yes, people can claim that the RFC still allows this and that,=
 but I think that most people do take BCPs into considerations when they de=
sign their implementations.


>In any case, having you willing to do the work is a good thing. But there =
had better also be others that are interested in the work enough to review =
and comment on it, and to justify spending list and=20
>meeting time on.

Sure. For me the main issue was to find a solution for the forking issue, a=
nd the 3xx solution should probably be ok (even if it can't be mandated).=20

I DO think it would be good to document it, and also to document/clarify th=
e unambiguous stuff, but if people object I am not going to fight for it, e=
venthough I think that it generally is very bad to have that kind of unambi=
guous stuff in our protocol speicifcations :)

Regards,

Christer





Christer Holmberg wrote:
> Hi,
> =20
> Adam, thanks for sending out the the e-mail!
> =20
> First, a clarification regarding "new features":
> =20
> When we say "new features", I assume (based on Hadriel's comment in=20
> the
> meeting) it means the possibility to, with a single OPTIONS request,=20
> retrieve the capabilities from ALL UAS(s) behind a forking proxy -=20
> instead of only one UAS (whoever sends the 200 OK to the forking proxy=20
> first).
> =20
> The message I got from the meeting, and also previous e-mail=20
> discussions, was that the "OPTIONS 3xx" solution already provideds=20
> this feature, and that it can be used (without defining new protocol=20
> elements etc).
> =20
> So, in my opinion the "adding" word is a little missleading. Instead I=20
> think the question is whether we want to DOCUMENT that existing feature.
> =20
> =20
> Second, regarding bugs and interoperability issues in the field.=20
> Personally I am not sure of any major ones at the moment, but that is=20
> probably because OPTIONS haven't been very much deployed in the first=20
> place (it has been used for some keep-alive/ping purposes, but in=20
> those cases it really doesn't matter what comes in the response etc).=20
> Now, however, at least I am starting to get questions regarding the=20
> usage of OPTIONS, so I guess we need to ask ourselves whether we think=20
> the bugs WILL create interoperability issues.
> =20
> And, the issue you raised regarding using the re-INVITE to re-create=20
> dialogs ("[sipcore] Mea Culpa: Dialog Necromancy in RFC 3261") also=20
> affects OPTIONS, eventhough you want to keep them separate (which is ok).
> =20
> Obviously I am in favor of pursuing the work, and will be willing to=20
> author drafts :)
> =20
> Regards,
> =20
> Christer
> =20
> =20
> =20
>=20
> ----------------------------------------------------------------------
> --
> *From:* sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] *On=20
> Behalf Of *Adam Roach
> *Sent:* 30. maaliskuuta 2010 22:06
> *To:* SIPCORE
> *Subject:* [sipcore] SIP OPTIONS: Shall we work on this?
>=20
> [as chair]
>=20
> One of the topics of discussion at the recent face-to-face meeting in=20
> Anaheim was the SIP OPTIONS method. We have had significant discussion=20
> of some specific technical topics on the SIPCORE mailing list, such=20
> that all participants should have a reasonable overview of the scope=20
> that any related work may entail. Participants may also benefit from=20
> reading through the minutes of the corresponding conversation in Anaheim:
>=20
> http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#The%20SIP%2
> 0OPTIONS%20Method
>=20
> Rather than continue discussion of the specific technical topics=20
> related to OPTIONS, the chairs would encourage the working group to=20
> engage in a discussion around whether we should, as a group, work on=20
> addressing some of the issues that have been identified.
>=20
> In particular, we would direct the group to consider the nature of=20
> such
> work:
>=20
>     * Are we proposing to introduce new features, or fix existing bugs?
>     * If we are adding new features: is OPTIONS the best way to
>       introduce these features, or would other mechanisms (presence,
>       callee-caps) produce superior results?
>     * If we are fixing bugs: are the bugs theoretical, or can we cite
>       actual interoperability issues in the field?
>     * If there are identified interoperability issues: is the level of
>       effort involved in fixing them commensurate with the impact of the
>       problems?=20
>=20
> Finally, keep in mind that work does not happen unless participants=20
> are willing to invest time. If you speak out in favor of pursuing this=20
> work, please indicate the level of time commitment that you will=20
> personally invest in bringing such work to a successful conclusion=20
> (e.g., will author drafts; will perform dedicated reviews; etc).
>=20
> Thanks.
>=20
> /a
>=20
>=20
> ----------------------------------------------------------------------
> --
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From christer.holmberg@ericsson.com  Wed Mar 31 13:29:42 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Adam Roach' <adam@nostrum.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 31 Mar 2010 22:30:08 +0200
Thread-Topic: [sipcore] Consensus Call: SIP Table 2
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Hi,

Just to clarify: both alternatives will leave whatever is currently unclear=
 as it is, and only for new header fields and methods would be make things =
clear (whatever mechanism we would use)?

And, couldn't alternative 1 result in lots of "defined semantics" - especia=
lly if we define a new method and may have to take a large number of header=
 fields into consideration? Maybe you'll even end up with something that lo=
oks like Table 2...

Note that I am not objecting to option 1 - I just haven't been able to "pic=
ture" it yet :)

Regards,

Christer

________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach
Sent: 30. maaliskuuta 2010 21:51
To: SIPCORE
Subject: [sipcore] Consensus Call: SIP Table 2

[as chair]

Last week, at the face-to-face meeting in Anaheim, the SIPCORE participants=
 in attendance had a discussion regarding the future of Tables 2 and 3 in R=
FC 3261. A summary of the conversation, along with links to the presentatio=
ns and an audio recording of the conversation, can be found in the minutes:

http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%=
20/%203

During that conversation, the chairs polled those present for support betwe=
en two options:

 1.  Produce an artifact stating that documents defining new header fields =
and methods define semantics in prose, and will no longer extend Table 2; o=
r

 2.  Produce an RFC that defines new procedures for adding elements to Tabl=
e 2 as new header fields and methods are defined

Among those present, there was strong support for option 1. If you wish to =
raise an objection to this direction, please do so on the SIPCORE mailing l=
ist before April 15th.

Thank you.

/a

--_000_FF84A09F50A6DC48ACB6714F4666CC745E21B5A908ESESSCMS0354e_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.3790.4639" name=3DGENERATOR></HEAD>
<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Just to clarify: both alternatives will leave what=
ever is=20
currently unclear as it is, and only for new header fields and methods woul=
d be=20
make things clear (whatever mechanism we would use)?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>And, couldn't alternative 1 result in lots of "def=
ined=20
semantics" - especially if we define a new method and may have to take a la=
rge=20
number of header fields into consideration? Maybe you'll even end up with=20
something that looks like Table 2...</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Note that I am not objecting to option 1 - I just =
haven't=20
been able to "picture" it yet :)</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D700302120-31032010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sipcore-bounces@ietf.org=20
[mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>Adam Roach<BR><B>Sent=
:</B>=20
30. maaliskuuta 2010 21:51<BR><B>To:</B> SIPCORE<BR><B>Subject:</B> [sipcor=
e]=20
Consensus Call: SIP Table 2<BR></FONT><BR></DIV>
<DIV></DIV>[as chair]<BR><BR>Last week, at the face-to-face meeting in Anah=
eim,=20
the SIPCORE participants in attendance had a discussion regarding the futur=
e of=20
Tables 2 and 3 in RFC 3261. A summary of the conversation, along with links=
 to=20
the presentations and an audio recording of the conversation, can be found =
in=20
the minutes:<BR><BR><A class=3Dmoz-txt-link-freetext=20
href=3D"http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Ta=
ble%202%20/%203">http://www.ietf.org/proceedings/10mar/minutes/sipcore.html=
#SIP%20Table%202%20/%203</A><BR><BR>During=20
that conversation, the chairs polled those present for support between two=
=20
options:<BR>
<OL>
  <LI>Produce an artifact stating that documents defining new header fields=
 and=20
  methods define semantics in prose, and will no longer extend Table 2;=20
  or<BR><BR>
  <LI>Produce an RFC that defines new procedures for adding elements to Tab=
le 2=20
  as new header fields and methods are defined </LI></OL>Among those presen=
t,=20
there was strong support for option 1. If you wish to raise an objection to=
 this=20
direction, please do so on the SIPCORE mailing list before April=20
15th.<BR><BR>Thank you.<BR><BR>/a<BR></BODY></HTML>

--_000_FF84A09F50A6DC48ACB6714F4666CC745E21B5A908ESESSCMS0354e_--

From adam@nostrum.com  Wed Mar 31 14:38:32 2010
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Subject: Re: [sipcore] Consensus Call: SIP Table 2
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On 3/31/10 3:30 PM, Christer Holmberg wrote:
> Just to clarify: both alternatives will leave whatever is currently 
> unclear as it is....
[as individual]

In truth, I don't think anything is particularly unclear -- it's just 
not documented the same way. The documents that didn't expand table 2 
while adding new header fields all have prose that describes when the 
header field can appear. (Yes, I know this, because I looked at every 
one of them when I did the table 2 expansion that appears in the slides).

[as chair]

But, yes, the consensus call only deals with actions as they apply to 
future documents -- such as RFC3265bis -- and does not address actions 
that would affect RFCs or any other documents that have already cleared 
IETF review.

> And, couldn't alternative 1 result in lots of "defined semantics" - 
> especially if we define a new method and may have to take a large 
> number of header fields into consideration? Maybe you'll even end up 
> with something that looks like Table 2...

[as individual again]

I don't think so. As Keith has pointed out, statements like "both header 
fields defined in this document can appear only in methods that create 
new dialogs" fill out entire swaths of the table all at once. They also 
have the pleasantly future-proof quality that we don't need to say 
anything specifically about the header fields when someone creates a new 
method. A simple evaluation of the new method's properties against the 
header fields' normative text makes the answer brilliantly clear.

/a

> Note that I am not objecting to option 1 - I just haven't been able to 
> "picture" it yet :)
> Regards,
> Christer
>
> ------------------------------------------------------------------------
> *From:* sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] *On 
> Behalf Of *Adam Roach
> *Sent:* 30. maaliskuuta 2010 21:51
> *To:* SIPCORE
> *Subject:* [sipcore] Consensus Call: SIP Table 2
>
> [as chair]
>
> Last week, at the face-to-face meeting in Anaheim, the SIPCORE 
> participants in attendance had a discussion regarding the future of 
> Tables 2 and 3 in RFC 3261. A summary of the conversation, along with 
> links to the presentations and an audio recording of the conversation, 
> can be found in the minutes:
>
> http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203
>
> During that conversation, the chairs polled those present for support 
> between two options:
>
>    1. Produce an artifact stating that documents defining new header
>       fields and methods define semantics in prose, and will no longer
>       extend Table 2; or
>
>    2. Produce an RFC that defines new procedures for adding elements
>       to Table 2 as new header fields and methods are defined
>
> Among those present, there was strong support for option 1. If you 
> wish to raise an objection to this direction, please do so on the 
> SIPCORE mailing list before April 15th.
>
> Thank you.
>
> /a


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On 3/31/10 3:30 PM, Christer Holmberg wrote:
<blockquote
 cite="mid:FF84A09F50A6DC48ACB6714F4666CC745E21B5A908@ESESSCMS0354.eemea.ericsson.se"
 type="cite">
  <div dir="ltr" align="left"><span class="700302120-31032010"><font
 color="#0000ff" face="Arial" size="2">Just to clarify: both
alternatives will leave whatever is currently unclear as it is</font></span>....<br>
  </div>
</blockquote>
[as individual]<br>
<br>
In truth, I don't think anything is particularly unclear -- it's just
not documented the same way. The documents that didn't expand table 2
while adding new header fields all have prose that describes when the
header field can appear. (Yes, I know this, because I looked at every
one of them when I did the table 2 expansion that appears in the
slides).<br>
<br>
[as chair]<br>
<br>
But, yes, the consensus call only deals with actions as they apply to
future documents -- such as RFC3265bis -- and does not address actions
that would affect RFCs or any other documents that have already cleared
IETF review. <br>
<br>
<blockquote
 cite="mid:FF84A09F50A6DC48ACB6714F4666CC745E21B5A908@ESESSCMS0354.eemea.ericsson.se"
 type="cite">
  <div dir="ltr" align="left"><span class="700302120-31032010"><font
 color="#0000ff" face="Arial" size="2">And, couldn't alternative 1
result in lots of "defined semantics" - especially if we define a new
method and may have to take a large number of header fields into
consideration? Maybe you'll even end up with something that looks like
Table 2...</font></span></div>
</blockquote>
<br>
[as individual again]<br>
<br>
I don't think so. As Keith has pointed out, statements like "both
header fields defined in this document can appear only in methods that
create new dialogs" fill out entire swaths of the table all at once.
They also have the pleasantly future-proof quality that we don't need
to say anything specifically about the header fields when someone
creates a new method. A simple evaluation of the new method's
properties against the header fields' normative text makes the answer
brilliantly clear.<br>
<br>
/a<br>
<br>
<blockquote
 cite="mid:FF84A09F50A6DC48ACB6714F4666CC745E21B5A908@ESESSCMS0354.eemea.ericsson.se"
 type="cite">
  <div dir="ltr" align="left"><span class="700302120-31032010"></span> <span
 class="700302120-31032010"><font color="#0000ff" face="Arial" size="2">Note
that I am not objecting to option 1 - I just haven't been able to
"picture" it yet :)</font></span></div>
  <div dir="ltr" align="left"><span class="700302120-31032010"></span>&nbsp;</div>
  <div dir="ltr" align="left"><span class="700302120-31032010"><font
 color="#0000ff" face="Arial" size="2">Regards,</font></span></div>
  <div dir="ltr" align="left"><span class="700302120-31032010"></span>&nbsp;</div>
  <div dir="ltr" align="left"><span class="700302120-31032010"><font
 color="#0000ff" face="Arial" size="2">Christer</font></span></div>
  <br>
  <div class="OutlookMessageHeader" dir="ltr" align="left" lang="en-us">
  <hr tabindex="-1"><font face="Tahoma" size="2"><b>From:</b>
<a class="moz-txt-link-abbreviated" href="mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounces@ietf.org</a>] <b>On
Behalf Of </b>Adam Roach<br>
  <b>Sent:</b> 30. maaliskuuta 2010 21:51<br>
  <b>To:</b> SIPCORE<br>
  <b>Subject:</b> [sipcore] Consensus Call: SIP Table 2<br>
  </font><br>
  </div>
[as chair]<br>
  <br>
Last week, at the face-to-face meeting in Anaheim, the SIPCORE
participants in attendance had a discussion regarding the future of
Tables 2 and 3 in RFC 3261. A summary of the conversation, along with
links to the presentations and an audio recording of the conversation,
can be found in the minutes:<br>
  <br>
  <a moz-do-not-send="true" class="moz-txt-link-freetext"
 href="http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203">http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%202%20/%203</a><br>
  <br>
During that conversation, the chairs polled those present for support
between two options:<br>
  <ol>
    <li>Produce an artifact stating that documents defining new header
fields and methods define semantics in prose, and will no longer extend
Table 2; or<br>
      <br>
    </li>
    <li>Produce an RFC that defines new procedures for adding elements
to Table 2 as new header fields and methods are defined </li>
  </ol>
Among those present, there was strong support for option 1. If you wish
to raise an objection to this direction, please do so on the SIPCORE
mailing list before April 15th.<br>
  <br>
Thank you.<br>
  <br>
/a<br>
</blockquote>
<br>
</body>
</html>

--------------030809040708020206060204--

From adam@nostrum.com  Wed Mar 31 14:39:03 2010
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Subject: Re: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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[as individual]

On 3/31/10 2:48 PM, Christer Holmberg wrote:
> So, you are saying that it is ok to re-crete dialogs using re-INVITEs?

No, I'm pointing out that RFC 3261 says that. I have attached no moral 
value to that normative behavior.

Just because SIPCORE is chartered to work on core SIP protocol issues 
does not obligate us to challenge the validity of the normative 
statements in RFC 3261 whenever one is mentioned. The role of SIPCORE is 
to solve clear interoperability problems that people are actually 
experiencing, and extend the core protocol when doing so helps create 
new services. It does not serve us well to invent new problems where 
none exist, nor is it a worthy investment of time to create new tools 
that cannot be usefully made into actual features.

/a

--------------040605000107090903030006
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[as individual]<br>
<br>
On 3/31/10 2:48 PM, Christer Holmberg wrote:
<blockquote
 cite="mid:FF84A09F50A6DC48ACB6714F4666CC745E21B5A905@ESESSCMS0354.eemea.ericsson.se"
 type="cite">
  <meta http-equiv="Content-Type"
 content="text/html; charset=ISO-8859-1">
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  <div dir="ltr" align="left"><span class="027284319-31032010"><font
 color="#0000ff" face="Arial" size="2">So, you are saying that it is ok
to re-crete dialogs using re-INVITEs?</font></span></div>
</blockquote>
<br>
No, I'm pointing out that RFC 3261 says that. I have attached no moral
value to that normative behavior.<br>
<br>
Just because SIPCORE is chartered to work on core SIP protocol issues
does not obligate us to challenge the validity of the normative
statements in RFC 3261 whenever one is mentioned. The role of SIPCORE
is to solve clear interoperability problems that people are actually
experiencing, and extend the core protocol when doing so helps create
new services. It does not serve us well to invent new problems where
none exist, nor is it a worthy investment of time to create new tools
that cannot be usefully made into actual features.<br>
<br>
/a<br>
</body>
</html>

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From christer.holmberg@ericsson.com  Wed Mar 31 23:44:46 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Date: Thu, 1 Apr 2010 08:45:11 +0200
Thread-Topic: [sipcore] Consensus Call: SIP Table 2
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Hi,=20

>Just to clarify: both alternatives will leave whatever is currently unclea=
r as it is....
>	=09
>
>[as individual]
>=09
>In truth, I don't think anything is particularly unclear -- it's just not =
documented the same way. The documents that didn't expand table 2 while add=
ing new header fields all have prose that=20
>describes when the header field can appear. (Yes, I know this, because I l=
ooked at every one of them when I did the table 2 expansion that appears in=
 the slides).
>
>So, you are saying that by reading the specs we would be able to complete =
the table you had put toghether?
>=09
>	[as chair]
>=09
>But, yes, the consensus call only deals with actions as they apply to futu=
re documents -- such as RFC3265bis -- and does not address actions that wou=
ld affect RFCs or any other documents that=20
>have already cleared IETF review.=20
>=09
>And, couldn't alternative 1 result in lots of "defined semantics" - especi=
ally if we define a new method and may have to take a large number of heade=
r fields into consideration? Maybe you'll=20
>even end up with something that looks like Table 2...
>
>
>[as individual again]
>=09
>I don't think so. As Keith has pointed out, statements like "both header f=
ields defined in this document can appear only in methods that create new d=
ialogs" fill out entire swaths of the table=20
>all at once. They also have the pleasantly future-proof quality that we do=
n't need to say anything specifically about the header fields when someone =
creates a new method. A simple evaluation of=20
>the new method's properties against the header fields' normative text make=
s the answer brilliantly clear.

Sure. That is fine when you define new headers, and when you define new met=
hods and need to cover those new headers.

But, when you define a new method, you still also need to consider all "old=
" existing header fields - for which we unfortunately do not have statement=
s as those suggested by Keith.

Take 3265bis as an example, and assume you remove table 2. The text will of=
 course describe the usage of header fields directly associated with SUB/NO=
T, eg. Allow-Events. But, how will you deal with all the other header field=
s that we have defined over the years? I am sure there are header fields th=
at we do NOT need to cover, because they are associated with specific mecha=
nisms (e.g. RSeq and RAck). But, there is a huge number of more "generic" h=
eader fields that somehow need to be covered.

Anyway, there is only one way to figure out: by doing the work :)

Regards,

Christer







________________________________

		From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f Of Adam Roach
		Sent: 30. maaliskuuta 2010 21:51
		To: SIPCORE
		Subject: [sipcore] Consensus Call: SIP Table 2
	=09
	=09
		[as chair]
	=09
		Last week, at the face-to-face meeting in Anaheim, the SIPCORE participan=
ts in attendance had a discussion regarding the future of Tables 2 and 3 in=
 RFC 3261. A summary of the conversation, along with links to the presentat=
ions and an audio recording of the conversation, can be found in the minute=
s:
	=09
		http://www.ietf.org/proceedings/10mar/minutes/sipcore.html#SIP%20Table%20=
2%20/%203
	=09
		During that conversation, the chairs polled those present for support bet=
ween two options:
	=09

		1.	Produce an artifact stating that documents defining new header fields =
and methods define semantics in prose, and will no longer extend Table 2; o=
r
		=09
		=09
		2.	Produce an RFC that defines new procedures for adding elements to Tabl=
e 2 as new header fields and methods are defined=20

		Among those present, there was strong support for option 1. If you wish t=
o raise an objection to this direction, please do so on the SIPCORE mailing=
 list before April 15th.
	=09
		Thank you.
	=09
		/a
	=09



From christer.holmberg@ericsson.com  Wed Mar 31 23:57:27 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Date: Thu, 1 Apr 2010 08:57:56 +0200
Thread-Topic: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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Subject: Re: [sipcore] Mea Culpa: Dialog Necromancy in RFC 3261
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Hi,=20

>>So, you are saying that it is ok to re-crete dialogs using re-INVITEs?
>
>
>No, I'm pointing out that RFC 3261 says that. I have attached no moral val=
ue to that normative behavior.
>=09
>Just because SIPCORE is chartered to work on core SIP protocol issues does=
 not obligate us to challenge the validity of the normative statements in R=
FC 3261 whenever one is mentioned. The role=20
>of SIPCORE is to solve clear interoperability problems that people are act=
ually experiencing, and extend the core protocol when doing so helps create=
 new services. It does not serve us well to=20
>invent new problems where none exist, nor is it a worthy investment of tim=
e to create new tools that cannot be usefully made into actual features.

I agree.

But, before things are actually deployed, people are writing test specifica=
tions, designing protocol procedures for certain features and services, and=
 they may find that difficult/impossible if the spec is unclear on how enti=
ties are supposed to behave.

Regards,

Christer


