
From brett@broadsoft.com  Thu Aug  2 07:22:32 2012
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From: Brett Tate <brett@broadsoft.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>, "draft-barnes-sipcore-rfc4244bis-callflows@tools.ietf.org" <draft-barnes-sipcore-rfc4244bis-callflows@tools.ietf.org>
Date: Thu, 2 Aug 2012 07:22:29 -0700
Thread-Topic: draft-barnes-sipcore-rfc4244bis-callflows-03: comments
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Subject: [sipcore] draft-barnes-sipcore-rfc4244bis-callflows-03: comments
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Howdy,

The following are some draft-barnes-sipcore-rfc4244bis-callflows-03 comment=
s.  They are similar to those provided in April for draft-ietf-sipcore-rfc4=
244bis-07.

Thanks,
Brett

------


Section 3 general comments:

- If depicting an RFC 3261 compliant UAS, the 302 examples must include To =
tags.

- If depicting an RFC 3261 compliant UAS, the Via headers using an FQDN mus=
t cause the UAS to add a received parameter to the Via.

Section 3.1:

- In case not intentional, the 302's ACK is not depicted within the call fl=
ow.

- The 2xx's ACK is depicted a ladder; however example.com did not add a Rec=
ord-Route.

- F4: The top unique via branch should be changed since it was already used=
 within F2.

Section 3.2:

- F2: If depicting an RFC 3261 compliant UAS, must add a To tag.

Section 3.3:

- In case not intentional, the 302's ACK is not depicted within the call fl=
ow.

- The 2xx's ACK is depicted a ladder; however example.com did not add a Rec=
ord-Route.

- F4: The top unique via branch should be changed since it was already used=
 within F2.

- F5: The Contact should be changed since it reflects the proxy instead of =
UAS location.

- F6: The top unique via branch should be changed since it was already used=
 within F2.

- F7: The Contact should be changed since it reflects the proxy instead of =
UAS location.

Section 3.4:

- In case not intentional, the 302's ACK is not depicted within the call fl=
ow.

- The 2xx's ACK is depicted a ladder; however example.com did not add a Rec=
ord-Route.

- F4: The top unique via branch should be changed since it was already used=
 within F2.

- F5: The Contact should be changed since it reflects the proxy instead of =
UAS location.

- F6: The top unique via branch should be changed since it was already used=
 within F2.

- F7: The Contact should be changed since it reflects the proxy instead of =
UAS location.



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From shida@ntt-at.com  Thu Aug  2 07:45:42 2012
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To: Brett Tate <brett@broadsoft.com>
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Cc: "draft-barnes-sipcore-rfc4244bis-callflows@tools.ietf.org" <draft-barnes-sipcore-rfc4244bis-callflows@tools.ietf.org>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-barnes-sipcore-rfc4244bis-callflows-03: comments
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Brett;

 Many thanks for the comments.=20

 I will reflect them ASAP.=20
=20
 Thanks again

 Shida

On Aug 2, 2012, at 7:22 AM, Brett Tate wrote:

> Howdy,
>=20
> The following are some draft-barnes-sipcore-rfc4244bis-callflows-03 =
comments.  They are similar to those provided in April for =
draft-ietf-sipcore-rfc4244bis-07.
>=20
> Thanks,
> Brett
>=20
> ------
>=20
>=20
> Section 3 general comments:
>=20
> - If depicting an RFC 3261 compliant UAS, the 302 examples must =
include To tags.
>=20
> - If depicting an RFC 3261 compliant UAS, the Via headers using an =
FQDN must cause the UAS to add a received parameter to the Via.
>=20
> Section 3.1:
>=20
> - In case not intentional, the 302's ACK is not depicted within the =
call flow.
>=20
> - The 2xx's ACK is depicted a ladder; however example.com did not add =
a Record-Route.
>=20
> - F4: The top unique via branch should be changed since it was already =
used within F2.
>=20
> Section 3.2:
>=20
> - F2: If depicting an RFC 3261 compliant UAS, must add a To tag.
>=20
> Section 3.3:
>=20
> - In case not intentional, the 302's ACK is not depicted within the =
call flow.
>=20
> - The 2xx's ACK is depicted a ladder; however example.com did not add =
a Record-Route.
>=20
> - F4: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F5: The Contact should be changed since it reflects the proxy =
instead of UAS location.
>=20
> - F6: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F7: The Contact should be changed since it reflects the proxy =
instead of UAS location.
>=20
> Section 3.4:
>=20
> - In case not intentional, the 302's ACK is not depicted within the =
call flow.
>=20
> - The 2xx's ACK is depicted a ladder; however example.com did not add =
a Record-Route.
>=20
> - F4: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F5: The Contact should be changed since it reflects the proxy =
instead of UAS location.
>=20
> - F6: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F7: The Contact should be changed since it reflects the proxy =
instead of UAS location.
>=20
>=20
>=20
> This email is intended solely for the person or entity to which it is =
addressed and may contain confidential and/or privileged information.  =
If you are not the intended recipient and have received this email in =
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> https://www.ietf.org/mailman/listinfo/sipcore


From fluffy@cisco.com  Thu Aug  2 17:02:04 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "sipcore@ietf.org WG" <sipcore@ietf.org>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "jmillan@aliax.net" <jmillan@aliax.net>, "vpascual@acmepacket.com" <vpascual@acmepacket.com>
Thread-Topic: Comments on draft-ietf-sipcore-sip-websocket-01
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Date: Fri, 3 Aug 2012 00:01:59 +0000
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Subject: [sipcore] Comments on draft-ietf-sipcore-sip-websocket-01
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I am sorry to miss sipcore but unfortunately I am chairing another meeting =
at this time - I would like to add a few comments.=20

I like this draft and think it is going the right direction.

In section 5.3, I think a better solution would be just require that any br=
owser needs to use an outbound proxy over ws. So effectively the web server=
 needs to provide an outbound proxy that can be used. Once we have that, yo=
u never need to do a SIP URL resolution inside the JS.=20

In section 7, I think it is key that credential used to get to the WS Proxy=
 can be different that the SIP credentials used to register with. This may =
be what you are greeting at but it seems a bit unclear. So in 8.1, I would =
have expected to see an challenge to the F3.=20









From jhildebr@cisco.com  Thu Aug  2 18:53:36 2012
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From: "Joe Hildebrand (jhildebr)" <jhildebr@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "sipcore@ietf.org WG" <sipcore@ietf.org>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "jmillan@aliax.net" <jmillan@aliax.net>, "vpascual@acmepacket.com" <vpascual@acmepacket.com>
Thread-Topic: [sipcore] Comments on draft-ietf-sipcore-sip-websocket-01
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On 8/2/12 5:01 PM, "Cullen Jennings (fluffy)" <fluffy@cisco.com> wrote:

>In section 5.3, I think a better solution would be just require that any
>browser needs to use an outbound proxy over ws. So effectively the web
>server needs to provide an outbound proxy that can be used. Once we have
>that, you never need to do a SIP URL resolution inside the JS.

As long as you're authenticating to the proxy, it could choose to do CORS,
which might allow for some more interesting deployments.  I'd suggest you
don't rule that out.  However, in those use cases, you still don't need to
do SIP URL resolution; the web context around the JS will know where to
send you to connect.

--=20
Joe Hildebrand




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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [sipcore] Comments on draft-ietf-sipcore-sip-websocket-01
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2012/8/3 Cullen Jennings (fluffy) <fluffy@cisco.com>:
> I like this draft and think it is going the right direction.

Hi Cullen, please read comments inline:


> In section 5.3, I think a better solution would be just require that any =
browser needs to use an outbound proxy over ws. So effectively the web serv=
er needs to provide an outbound proxy that can be used. Once we have that, =
you never need to do a SIP URL resolution inside the JS.

The draft does not assume web browser scenarios. In a web browser
scenario the web server providing the JavaScript SIP stack can decide
how the UA has to connect to the appropriate WebSocket and how to
provisione it with a WS URI or a SIP URI (so the JavaScript code
should map it to a WS URI prior to establish the WS connection). I
don't think we should mandate how such a WebSocket server should be
provided to the client.

But there are other scenarios, i.e. a SIP UA running in a mobile
device using a WebSocket stack provided by the mobile's operating
system, and using SIP on top of it. In this case the UA can perform
NAPTR/SRV queries as any other non-webapp application.

We wouldn't like to focus this draft just on web browser scenarios,
neither making rules based on web browser/JavaScript limitations. IMHO
this has been previously discussed on the list, and the consensus was
not to focus on web browser scenarios.




> In section 7, I think it is key that credential used to get to the WS Pro=
xy can be different that the SIP credentials used to register with. This ma=
y be what you are greeting at but it seems a bit unclear. So in 8.1, I woul=
d have expected to see an challenge to the F3.

"credential used to get to the WS Proxy" is a bit unclear. RFC 6455
(WebSocket) leaves the door open for a WebSocket server to reply
401/407 response for a WebSocket handshake request, but the fact is
that no one web browser reacts on a 401/407 (instead they just abort
the connection and raise a JS error).

Also, by participating in IETF WebSocket WG, I've realized that WWW
people want to know *nothing* about HTTP authentication. Instead they
trust on current WWW extended approach (Cookies, HTML form based
authentication, OAuth...).

So, IMHO, adding a credencials procedure in the WebSocket handshake
flow in example 8.1 would cause some confusion. Take into account
that, for sure, most of the SIP WebSocket implementors will build
their own authentication mechanism for validating a WebSocket
connection. Probably they will rely on the Cookie header in the
WebSocket GET request (a Cookie returned by the *Web* server after the
user logged in the web) or in some custom parameters in the WebSocket
GET request-line.


Thanks a lot.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Preliminary minutes have been posted for the face-to-face meeting in 
Vancouver. Please send any comments to me and Paul:

http://www.ietf.org/proceedings/84/minutes/minutes-84-sipcore

/a

From dwing@cisco.com  Mon Aug  6 16:25:37 2012
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From: "Dan Wing" <dwing@cisco.com>
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Cc: oscar.gallego@vodafone.com, draft-holmberg-sipcore-rkeep@tools.ietf.org
Subject: [sipcore] comments on draft-holmberg-sipcore-rkeep-00
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The document insufficient describes the problem.  It only says "One reason
can be that the scheduling policy", with insufficient detail.  After bumping
into the SIPCORE chairs and mentioning that I had some feedback on 'rkeep',
I have a deeper understanding of the problem space.  However, the document
should provide that understanding.  Specifically, the problem is that
applications cannot get scheduled to send their periodic keepalive messages,
but that a network-originated packet will allow the application to get
scheduled to send its reply; it is that reply message that is critical to
RKEEP working (because the NAT pinhole is often only refreshed with outbound
traffic, not inbound traffic).  This needs to be clearly stated in the
document.  When that is done, we can more carefully understand the reason
that operating systems are reluctant to schedule applications to send
occasional keepalive traffic.  I would guess the reasoning is to save the
device's battery.  If my guess is accurate, then RKEEP destroys (ruins) that
purpose of application scheduling.  SIP is not the sole protocol that would
benefit from RKEEP.  We could see other long-lived quiescent connections
benefiting from RKEEP, such as IMAP or XMPP, to keep their TCP connections
open.  If RKEEP is used by those other connections, the servers will not
synchronize when they send their keepalive traffic.  In contrast, if the
applications were allowed to schedule their own application keepalives and
their own network keepalives, the applications (or the client operating
system) could coordinate those application and network keepalives to use the
battery most efficiently.  With RKEEP's network-based keepalives, such
efficiency can never be realized.


This text in the document:
   This section describes how SIP UAs and proxies negotiate usage of
   keep-alives associated with a registration, or a dialog, which types
   of SIP requests can be used in order to negotiate the usage, and the
   lifetime of the negotiated keep-alives.
and several other places in the document, intermixes two things, (a) SIP
dialog state and (b) NAT pinhole state.  These really need to be described
separately, because they are different things -- (a) is application state
(SIP dialog state) the (b) is network middlebox state (NAT state).
Combining them, for the purposes of optimization, is useful, but needs to be
clearly stated.  This is because if the SIP communication is over TCP or is
signaled with PCP, the network state maintenance changes whereas the
application state may well remain the same (the application may want a
'liveness' check every ~10 minutes, but the network elements no longer need
such frequent traffic).


The document needs to acknowledge and describe the firewalls -- and not just
NATs -- create state that occasionally needs refreshing.  Currently, the
word 'firewall' does not appear in the document at all.  This is important
because while it might be true that "NAT goes away with IPv6", the need and
value of firewalls to protect the network (from useless noise traffic) and
to protect hosts does not necessarily go away with IPv6.  And of course
NAT64, if it is used with a SIP service (e.g., over the top service), may
also need these similar functions and thus IPv6 (on the handset) didn't
eliminate the need for NAT keepalives if the traffic traverses a stateful
NAT64.

The document should recommend PCP as the better answer than RKEEP.


Nit:  removing the rkeep parameter (Section 5.4) isn't the best way to
signal support of rkeep, but I believe that feedback was already provided in
the WG meeting.

-d



From pkyzivat@alum.mit.edu  Mon Aug  6 16:43:17 2012
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Thank you Dan for providing a different perspective.
This demonstrates a reason not to rush this work into sipcore - that we 
should investigate if the problem deserves to be considered in a broader 
focus.

While we think about where to deal with this I encourage more discussion.

	Thanks,
	Paul (as sipcore co-chair)

On 8/6/12 4:25 PM, Dan Wing wrote:
> The document insufficient describes the problem.  It only says "One reason
> can be that the scheduling policy", with insufficient detail.  After bumping
> into the SIPCORE chairs and mentioning that I had some feedback on 'rkeep',
> I have a deeper understanding of the problem space.  However, the document
> should provide that understanding.  Specifically, the problem is that
> applications cannot get scheduled to send their periodic keepalive messages,
> but that a network-originated packet will allow the application to get
> scheduled to send its reply; it is that reply message that is critical to
> RKEEP working (because the NAT pinhole is often only refreshed with outbound
> traffic, not inbound traffic).  This needs to be clearly stated in the
> document.  When that is done, we can more carefully understand the reason
> that operating systems are reluctant to schedule applications to send
> occasional keepalive traffic.  I would guess the reasoning is to save the
> device's battery.  If my guess is accurate, then RKEEP destroys (ruins) that
> purpose of application scheduling.  SIP is not the sole protocol that would
> benefit from RKEEP.  We could see other long-lived quiescent connections
> benefiting from RKEEP, such as IMAP or XMPP, to keep their TCP connections
> open.  If RKEEP is used by those other connections, the servers will not
> synchronize when they send their keepalive traffic.  In contrast, if the
> applications were allowed to schedule their own application keepalives and
> their own network keepalives, the applications (or the client operating
> system) could coordinate those application and network keepalives to use the
> battery most efficiently.  With RKEEP's network-based keepalives, such
> efficiency can never be realized.
>
>
> This text in the document:
>     This section describes how SIP UAs and proxies negotiate usage of
>     keep-alives associated with a registration, or a dialog, which types
>     of SIP requests can be used in order to negotiate the usage, and the
>     lifetime of the negotiated keep-alives.
> and several other places in the document, intermixes two things, (a) SIP
> dialog state and (b) NAT pinhole state.  These really need to be described
> separately, because they are different things -- (a) is application state
> (SIP dialog state) the (b) is network middlebox state (NAT state).
> Combining them, for the purposes of optimization, is useful, but needs to be
> clearly stated.  This is because if the SIP communication is over TCP or is
> signaled with PCP, the network state maintenance changes whereas the
> application state may well remain the same (the application may want a
> 'liveness' check every ~10 minutes, but the network elements no longer need
> such frequent traffic).
>
>
> The document needs to acknowledge and describe the firewalls -- and not just
> NATs -- create state that occasionally needs refreshing.  Currently, the
> word 'firewall' does not appear in the document at all.  This is important
> because while it might be true that "NAT goes away with IPv6", the need and
> value of firewalls to protect the network (from useless noise traffic) and
> to protect hosts does not necessarily go away with IPv6.  And of course
> NAT64, if it is used with a SIP service (e.g., over the top service), may
> also need these similar functions and thus IPv6 (on the handset) didn't
> eliminate the need for NAT keepalives if the traffic traverses a stateful
> NAT64.
>
> The document should recommend PCP as the better answer than RKEEP.
>
>
> Nit:  removing the rkeep parameter (Section 5.4) isn't the best way to
> signal support of rkeep, but I believe that feedback was already provided in
> the WG meeting.
>
> -d
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From hannes.tschofenig@gmx.net  Tue Aug  7 00:35:11 2012
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Hi guys,=20

I had gotten a question from the ATIS guys since they are thinking about =
using the Geolocation header from RFC 6442 but in an interesting way.=20

Section 4.1 of RFC 6442 indicates what URIs can go into the Geolocation =
header, which includes SIP and SIPS URIs.=20

ATIS has a different deployment model where the SIP messages go to a =
gateway that allows interactions with a legacy PSAP. The legacy PSAP =
then performs a location lookup using a regular phone number (using =
legacy location protocols).=20

In order to convey that phone number for a lookup they are thinking =
about using the Geolocation header in the following form:
Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=3Dphone)

Since a SIP URI is allowed in the Geolocation header it seems that this =
functionality is supported but the semantics is quite different to what =
SIP location conveyance envisions. The envisioned functionality was that =
the receipient of that location URI would either look at the body of the =
message (in case of a CID), would use HELD (in case of a HTTP/HTTPS URI) =
or SIP (in case of a SIP, SIPS, or PRES URI).=20

I had spoken to a few of you during the meeting and I had gotten =
conflicting responses about whether this is allowed behavior or not.=20

Any feedback from folks on this list?=20

Ciao
Hannes


From pkyzivat@alum.mit.edu  Tue Aug  7 07:31:12 2012
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On 8/7/12 12:35 AM, Hannes Tschofenig wrote:
> Hi guys,
>
> I had gotten a question from the ATIS guys since they are thinking about using the Geolocation header from RFC 6442 but in an interesting way.
>
> Section 4.1 of RFC 6442 indicates what URIs can go into the Geolocation header, which includes SIP and SIPS URIs.
>
> ATIS has a different deployment model where the SIP messages go to a gateway that allows interactions with a legacy PSAP. The legacy PSAP then performs a location lookup using a regular phone number (using legacy location protocols).
>
> In order to convey that phone number for a lookup they are thinking about using the Geolocation header in the following form:
> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>
> Since a SIP URI is allowed in the Geolocation header it seems that this functionality is supported but the semantics is quite different to what SIP location conveyance envisions. The envisioned functionality was that the receipient of that location URI would either look at the body of the message (in case of a CID), would use HELD (in case of a HTTP/HTTPS URI) or SIP (in case of a SIP, SIPS, or PRES URI).
>
> I had spoken to a few of you during the meeting and I had gotten conflicting responses about whether this is allowed behavior or not.
>
> Any feedback from folks on this list?

Well, the problem I see with this is that including a sip uri in 
geolocation header field implies that the deref will be done via a 
subscribe. Apparently that isn't then intent here.

That presents two distinct problems:

- what happens if the dereferencer actually tries to use subscribe?

- what happens if dereferencer uses this ATIS-specific mechanism
   but the server for the designated URI only supports subscribe
   and not this new mechanism?

I have no problem with using this approach *if* both of the above cases 
work. But if either of them will fail then I think this is an abuse of 6442.

IMO it would be ok for the dereferencer to attempt some proprietary 
optimization of the deref as long as:
- it is also capable of falling back to subscribe if the optimization
   fails;
- the server for the supplied sip uri (mgcf.carrier.net in the example)
   is capable of handling the subscribe if it gets one;
- the results of the subscribe and the proprietary deref always
   yield the same result.

	Thanks,
	Paul (as individual)

(I'm speaking as an individual for now. Later we can discuss if we need 
a chair ruling.)

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Hannes Tschofenig <hannes.tschofenig@gmx.net>, "SIPCORE (Session Initiation Protocol Core) WG" <sipcore@ietf.org>
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> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>=20
> In order to convey that phone number for a lookup they are thinking
> about using the Geolocation header in the following form:
> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=3Dphone)

Naively, I'd say that since the goal is to carry an E.164 number, a
tel: URI would be the ideal choice.

Dale

From hannes.tschofenig@gmx.net  Tue Aug  7 08:31:35 2012
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I agree but the tel URI is not one of the supported location URIs in RFC =
6442.=20
Hence, the guys just use a different way to convey the same information.=20=


On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:

>> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>=20
>> In order to convey that phone number for a lookup they are thinking
>> about using the Geolocation header in the following form:
>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=3Dphone)
>=20
> Naively, I'd say that since the goal is to carry an E.164 number, a
> tel: URI would be the ideal choice.
>=20
> Dale


From hannes.tschofenig@gmx.net  Tue Aug  7 08:36:06 2012
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Hi Paul,=20

these are good points. Thanks for the feedback.=20

I could imagine that the interaction with the location server fails when =
one tries to dereference the location URI using a subscribe.

I will forward your comments to the respective ATIS persons.=20

Ciao
Hannes

On Aug 7, 2012, at 5:31 PM, Paul Kyzivat wrote:

> On 8/7/12 12:35 AM, Hannes Tschofenig wrote:
>> Hi guys,
>>=20
>> I had gotten a question from the ATIS guys since they are thinking =
about using the Geolocation header from RFC 6442 but in an interesting =
way.
>>=20
>> Section 4.1 of RFC 6442 indicates what URIs can go into the =
Geolocation header, which includes SIP and SIPS URIs.
>>=20
>> ATIS has a different deployment model where the SIP messages go to a =
gateway that allows interactions with a legacy PSAP. The legacy PSAP =
then performs a location lookup using a regular phone number (using =
legacy location protocols).
>>=20
>> In order to convey that phone number for a lookup they are thinking =
about using the Geolocation header in the following form:
>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=3Dphone)
>>=20
>> Since a SIP URI is allowed in the Geolocation header it seems that =
this functionality is supported but the semantics is quite different to =
what SIP location conveyance envisions. The envisioned functionality was =
that the receipient of that location URI would either look at the body =
of the message (in case of a CID), would use HELD (in case of a =
HTTP/HTTPS URI) or SIP (in case of a SIP, SIPS, or PRES URI).
>>=20
>> I had spoken to a few of you during the meeting and I had gotten =
conflicting responses about whether this is allowed behavior or not.
>>=20
>> Any feedback from folks on this list?
>=20
> Well, the problem I see with this is that including a sip uri in =
geolocation header field implies that the deref will be done via a =
subscribe. Apparently that isn't then intent here.
>=20
> That presents two distinct problems:
>=20
> - what happens if the dereferencer actually tries to use subscribe?
>=20
> - what happens if dereferencer uses this ATIS-specific mechanism
>  but the server for the designated URI only supports subscribe
>  and not this new mechanism?
>=20
> I have no problem with using this approach *if* both of the above =
cases work. But if either of them will fail then I think this is an =
abuse of 6442.
>=20
> IMO it would be ok for the dereferencer to attempt some proprietary =
optimization of the deref as long as:
> - it is also capable of falling back to subscribe if the optimization
>  fails;
> - the server for the supplied sip uri (mgcf.carrier.net in the =
example)
>  is capable of handling the subscribe if it gets one;
> - the results of the subscribe and the proprietary deref always
>  yield the same result.
>=20
> 	Thanks,
> 	Paul (as individual)
>=20
> (I'm speaking as an individual for now. Later we can discuss if we =
need a chair ruling.)
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: [sipcore] I-D Action: draft-ietf-sipcore-proxy-feature-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : Mechanism to indicate support of features and capabiliti=
es in the Session Initiation Protocol (SIP)
	Author(s)       : Christer Holmberg
                          Ivo Sedlacek
                          Hadriel Kaplan
	Filename        : draft-ietf-sipcore-proxy-feature-05.txt
	Pages           : 20
	Date            : 2012-08-08

Abstract:
   This specification defines a new SIP header field, Feature-Caps, to
   convey feature capability indicators, which are used by SIP entities
   not represented by the URI of the Contact header field to indicate
   support of features and capabilities, where media feature tags cannot
   be used to indicate the support.

   This specification also defines feature capability indicators, and
   creates a new IANA registry, "Proxy-Feature Feature Capability
   Indicator Trees", for registering feature capability indicators.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-proxy-feature

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-proxy-feature-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-proxy-feature-05


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


From christer.holmberg@ericsson.com  Wed Aug  8 09:09:05 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 8 Aug 2012 18:09:02 +0200
Thread-Topic: Draft new version: proxy-feature-05
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Subject: [sipcore] Draft new version: proxy-feature-05
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Hi,

I have submitted a new version (-05) of the proxy-feature draft, based on K=
eith's WGLC comments.

May the force be with you,

Christer

From christer.holmberg@ericsson.com  Wed Aug  8 12:50:57 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dan Wing <dwing@cisco.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 8 Aug 2012 21:50:52 +0200
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Subject: Re: [sipcore] comments on draft-holmberg-sipcore-rkeep-00
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Hi Dan,

>The document insufficient describes the problem.  It only says "One reason
>can be that the scheduling policy", with insufficient detail.  After bumpi=
ng
>into the SIPCORE chairs and mentioning that I had some feedback on 'rkeep'=
,
>I have a deeper understanding of the problem space.  However, the document
>should provide that understanding.  Specifically, the problem is that
>applications cannot get scheduled to send their periodic keepalive message=
s,
>but that a network-originated packet will allow the application to get
>scheduled to send its reply; it is that reply message that is critical to
>RKEEP working (because the NAT pinhole is often only refreshed with outbou=
nd
>traffic, not inbound traffic).  This needs to be clearly stated in the
>document.

In my presentation, I tried to make it very clear that it is the STUN *resp=
onse*, sent
from the UA, which acts as the actual keep-alive message, due to the reason
you also mention,

But, I do agree with you that it should be made more clear in the draft.

>When that is done, we can more carefully understand the reason
>that operating systems are reluctant to schedule applications to send
>occasional keepalive traffic.  I would guess the reasoning is to save the
>device's battery. If my guess is accurate, then RKEEP destroys (ruins) tha=
t
>purpose of application scheduling.

I don't think it's only about sending keep-alives, but to be able to perfor=
m any type of scheduled actions.

>SIP is not the sole protocol that would benefit from RKEEP.  We could see =
other long-lived quiescent connections
>benefiting from RKEEP, such as IMAP or XMPP, to keep their TCP connections=
 open.

Keep in mind that the actual STUN keep-alive mechanism is already defined i=
n RFC 5626. This draft only defines a *SIP*
mechanism for negotiating the usage of it.

If you want to define how *other* protocols negotiate the usage of STUN for=
 keep-alives, I don't think it belongs in this draft.

>If RKEEP is used by those other connections, the servers will not synchron=
ize when they send their keepalive traffic.  In contrast, if the
>applications were allowed to schedule their own application keepalives and=
 their own network keepalives, the=20
>applications (or the client operating system) could coordinate those appli=
cation and network keepalives to use the
>battery most efficiently.  With RKEEP's network-based keepalives, such eff=
iciency can never be realized.
>
>This text in the document:
>   This section describes how SIP UAs and proxies negotiate usage of
>   keep-alives associated with a registration, or a dialog, which types
>   of SIP requests can be used in order to negotiate the usage, and the
>   lifetime of the negotiated keep-alives.
>and several other places in the document, intermixes two things, (a) SIP
>dialog state and (b) NAT pinhole state.  These really need to be described
>separately, because they are different things -- (a) is application state
>(SIP dialog state) the (b) is network middlebox state (NAT state).
>Combining them, for the purposes of optimization, is useful, but needs to =
be
>clearly stated.

SIP dialog only refers to the scope of the negotiated keep-alives, ie that =
you can negotiate them to be sent within the lifetime of a SIP dialog.

>This is because if the SIP communication is over TCP or is
>signaled with PCP, the network state maintenance changes whereas the
>application state may well remain the same (the application may want a
>'liveness' check every ~10 minutes, but the network elements no longer nee=
d
>such frequent traffic).
>
>The document needs to acknowledge and describe the firewalls -- and not ju=
st
>NATs -- create state that occasionally needs refreshing.  Currently, the
>word 'firewall' does not appear in the document at all.  This is important
>because while it might be true that "NAT goes away with IPv6", the need an=
d
>value of firewalls to protect the network (from useless noise traffic) and
>to protect hosts does not necessarily go away with IPv6.  And of course
>NAT64, if it is used with a SIP service (e.g., over the top service), may
>also need these similar functions and thus IPv6 (on the handset) didn't
>eliminate the need for NAT keepalives if the traffic traverses a stateful
>NAT64.
>
> The document should recommend PCP as the better answer than RKEEP.

As said earlier, the STUN keep-alive mechanism is already defined in RFC 56=
26.=20

RFC 6233 defines a SIP mechanism which allows a UA behind a NAT to indicate=
 willingness to send such STUN requests, and this draft now defines a SIP m=
echanism (based on the 6233 mechanism) which allows a UA behind a NAT to in=
dicate willingess to receive such STUN requests.

>Nit:  removing the rkeep parameter (Section 5.4) isn't the best way to
>signal support of rkeep, but I believe that feedback was already provided =
in
>the WG meeting.

The details of the mechanism obviously needs to be discussed, if the work i=
s started.

Regards,

Christer=

From adam@nostrum.com  Wed Aug  8 13:07:02 2012
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Subject: Re: [sipcore] comments on draft-holmberg-sipcore-rkeep-00
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[as an individual]

On 8/8/12 2:50 PM, Christer Holmberg wrote:
> Keep in mind that the actual STUN keep-alive mechanism is already defined in RFC 5626. This draft only defines a*SIP*
> mechanism for negotiating the usage of it.
>
> If you want to define how*other*  protocols negotiate the usage of STUN for keep-alives, I don't think it belongs in this draft.

The concern, I believe -- and I've heard this from people other than Dan 
-- is that the issue of sending periodic NAT binding keepalives in the 
face of unpredictable OS scheduling is a *general* problem, and that it 
might be inadvisable to come up with application-by-application 
solutions to the problem.

/a

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    <div class="moz-cite-prefix">[as an individual]<br>
      <br>
      On 8/8/12 2:50 PM, Christer Holmberg wrote:<br>
    </div>
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A05853408683603@ESESSCMS0356.eemea.ericsson.se"
      type="cite">
      <pre wrap="">Keep in mind that the actual STUN keep-alive mechanism is already defined in RFC 5626. This draft only defines a <b class="moz-txt-star"><span class="moz-txt-tag">*</span>SIP<span class="moz-txt-tag">*</span></b>
mechanism for negotiating the usage of it.

If you want to define how <b class="moz-txt-star"><span class="moz-txt-tag">*</span>other<span class="moz-txt-tag">*</span></b> protocols negotiate the usage of STUN for keep-alives, I don't think it belongs in this draft.</pre>
    </blockquote>
    <br>
    The concern, I believe -- and I've heard this from people other than
    Dan -- is that the issue of sending periodic NAT binding keepalives
    in the face of unpredictable OS scheduling is a *general* problem,
    and that it might be inadvisable to come up with
    application-by-application solutions to the problem.<br>
    <br>
    /a<br>
  </body>
</html>

--------------010002030702000207040807--

From christer.holmberg@ericsson.com  Wed Aug  8 13:31:48 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 8 Aug 2012 22:28:39 +0200
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Hi,

>[as an individual]
>
>On 8/8/12 2:50 PM, Christer Holmberg wrote:
>
>>Keep in mind that the actual STUN keep-alive mechanism is already defined=
 in RFC 5626. This draft only defines a *SIP*
>>mechanism for negotiating the usage of it.
>>
>>If you want to define how *other* protocols negotiate the usage of STUN f=
or keep-alives, I don't think it belongs in this draft.
>
>The concern, I believe -- and I've heard this from people other than Dan -=
- is that the issue of sending periodic NAT binding keepalives=20
>in the face of unpredictable OS scheduling is a *general* problem, and tha=
t it might be inadvisable to come up with application-by-application=20
>solutions to the problem.

You might still need protocol-by-protocol solutions to negotiate the keep-a=
live, and the draft defines a negotiation solution for SIP. It can then be =
used by any SIP application.

Regards,

Christer=

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Cc: oscar.gallego@vodafone.com, draft-holmberg-sipcore-rkeep@tools.ietf.org
Subject: Re: [sipcore] comments on draft-holmberg-sipcore-rkeep-00
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Wednesday, August 08, 2012 12:51 PM
> To: Dan Wing; sipcore@ietf.org
> Cc: draft-holmberg-sipcore-rkeep@tools.ietf.org;
> oscar.gallego@vodafone.com
> Subject: RE: comments on draft-holmberg-sipcore-rkeep-00
> 
> Hi Dan,
> 
> >The document insufficient describes the problem.  It only says "One
> reason
> >can be that the scheduling policy", with insufficient detail.  After
> bumping
> >into the SIPCORE chairs and mentioning that I had some feedback on
> 'rkeep',
> >I have a deeper understanding of the problem space.  However, the
> document
> >should provide that understanding.  Specifically, the problem is that
> >applications cannot get scheduled to send their periodic keepalive
> messages,
> >but that a network-originated packet will allow the application to get
> >scheduled to send its reply; it is that reply message that is critical
> to
> >RKEEP working (because the NAT pinhole is often only refreshed with
> outbound
> >traffic, not inbound traffic).  This needs to be clearly stated in the
> >document.
> 
> In my presentation, I tried to make it very clear that it is the STUN
> *response*, sent
> from the UA, which acts as the actual keep-alive message, due to the
> reason you also mention,
> 
> But, I do agree with you that it should be made more clear in the
> draft.

That is the solution.

The underlying problem is caused by operating systems that are not
allowing client applications, running on handsets, to send their 
own keepalives.  It is that problem I would like to have better
described.  

> >When that is done, we can more carefully understand the reason
> >that operating systems are reluctant to schedule applications to send
> >occasional keepalive traffic.  I would guess the reasoning is to save
> the
> >device's battery. If my guess is accurate, then RKEEP destroys (ruins)
> that
> >purpose of application scheduling.
> 
> I don't think it's only about sending keep-alives, but to be able to
> perform any type of scheduled actions.

That is vague.  Can you be more specific by what you mean about 'scheduled
actions', if those are not keepalive messages.

> >SIP is not the sole protocol that would benefit from RKEEP.  We could
> see other long-lived quiescent connections
> >benefiting from RKEEP, such as IMAP or XMPP, to keep their TCP
> connections open.
> 
> Keep in mind that the actual STUN keep-alive mechanism is already
> defined in RFC 5626. This draft only defines a *SIP*
> mechanism for negotiating the usage of it.
> 
> If you want to define how *other* protocols negotiate the usage of STUN
> for keep-alives, I don't think it belongs in this draft.

I understand the scope of the draft.

My point is that the problem is larger than just the SIP application.

That is why I want the problem described better, so we can understand
the underlying cause of the problem -- and either get that fixed (because
it is flawed) or understand it better so we can quickly apply this
same RKEEP-style workaround to the other 213 applications that use
long-lived connections (the number 213 might be off by an order of
magnitude).


> >If RKEEP is used by those other connections, the servers will not
> synchronize when they send their keepalive traffic.  In contrast, if
> the
> >applications were allowed to schedule their own application keepalives
> and their own network keepalives, the
> >applications (or the client operating system) could coordinate those
> application and network keepalives to use the
> >battery most efficiently.  With RKEEP's network-based keepalives, such
> efficiency can never be realized.
> >
> >This text in the document:
> >   This section describes how SIP UAs and proxies negotiate usage of
> >   keep-alives associated with a registration, or a dialog, which
> types
> >   of SIP requests can be used in order to negotiate the usage, and
> the
> >   lifetime of the negotiated keep-alives.
> >and several other places in the document, intermixes two things, (a)
> SIP
> >dialog state and (b) NAT pinhole state.  These really need to be
> described
> >separately, because they are different things -- (a) is application
> state
> >(SIP dialog state) the (b) is network middlebox state (NAT state).
> >Combining them, for the purposes of optimization, is useful, but needs
> to be
> >clearly stated.
> 
> SIP dialog only refers to the scope of the negotiated keep-alives, ie
> that you can negotiate them to be sent within the lifetime of a SIP
> dialog.

I'll leave it to y'all, as SIP experts, how that relates to registration
refresh, if it relates at all.


> >This is because if the SIP communication is over TCP or is
> >signaled with PCP, the network state maintenance changes whereas the
> >application state may well remain the same (the application may want a
> >'liveness' check every ~10 minutes, but the network elements no longer
> need
> >such frequent traffic).
> >
> >The document needs to acknowledge and describe the firewalls -- and
> not just
> >NATs -- create state that occasionally needs refreshing.  Currently,
> the
> >word 'firewall' does not appear in the document at all.  This is
> important
> >because while it might be true that "NAT goes away with IPv6", the
> need and
> >value of firewalls to protect the network (from useless noise traffic)
> and
> >to protect hosts does not necessarily go away with IPv6.  And of
> course
> >NAT64, if it is used with a SIP service (e.g., over the top service),
> may
> >also need these similar functions and thus IPv6 (on the handset)
> didn't
> >eliminate the need for NAT keepalives if the traffic traverses a
> stateful
> >NAT64.
> >
> > The document should recommend PCP as the better answer than RKEEP.
> 
> As said earlier, the STUN keep-alive mechanism is already defined in
> RFC 5626.
> 
> RFC 6233 defines a SIP mechanism which allows a UA behind a NAT to
> indicate willingness to send such STUN requests, and this draft now
> defines a SIP mechanism (based on the 6233 mechanism) which allows a UA
> behind a NAT to indicate willingess to receive such STUN requests.

-d

> >Nit:  removing the rkeep parameter (Section 5.4) isn't the best way to
> >signal support of rkeep, but I believe that feedback was already
> provided in
> >the WG meeting.
> 
> The details of the mechanism obviously needs to be discussed, if the
> work is started.
> 
> Regards,
> 
> Christer=


From adam@nostrum.com  Fri Aug 10 11:38:42 2012
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Subject: [sipcore] WGLC: draft-ietf-sipcore-sip-websocket-02
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[as chair]

Based on feedback received from the authors and working group, we 
believe that the document "The WebSocket Protocol as a Transport for the 
Session Initiation Protocol (SIP)" is ready for working group last call.

Working group participants -- and, in particular, those participants who 
indicated a willingness to perform a dedicated review of the document -- 
are asked to read the current document and provide feedback. Even short 
statements along the lines of "I have carefully read the document and 
believe it is ready for publication" are useful.

The current version of the document is available here:

http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/

This WGLC runs until Friday, August 24th. Thank you.

/a

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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [sipcore] WGLC: draft-ietf-sipcore-sip-websocket-02
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My only concern with this draft is the "Appendix B. Topology Hiding" which
clearly states that SIP WebSocket Server would be non-compliant with
RFC3261 if it implements topology hiding which is actually recommended. I
would much rather see that this specification update RFC3261. It should
update section 18.2.1 of RFC 3261 stating that the server SHOULD not to add
"received" parameter to the via header if request was received
via WebSocket protocol and MUST not add the this parameter if via "sent-by"
header contains an address with ".invaid" domain.
_____________
Roman Shpount

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My only concern with this draft is the &quot;Appendix B. Topology Hiding&qu=
ot; which clearly states that SIP WebSocket Server would be non-compliant w=
ith RFC3261 if it implements topology hiding which is actually recommended.=
 I would much rather see that this specification update RFC3261. It should =
update section 18.2.1 of RFC 3261 stating that the server SHOULD not to add=
 &quot;received&quot; parameter to the via header if request was received v=
ia=A0WebSocket protocol and MUST not add the this parameter if via &quot;se=
nt-by&quot; header contains an address with &quot;.invaid&quot; domain.<div=
>
_____________<br>Roman Shpount<br><br></div>

--047d7b15a549f6446e04c6f011c9--

From pkyzivat@alum.mit.edu  Fri Aug 10 16:20:52 2012
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Subject: Re: [sipcore] WGLC: draft-ietf-sipcore-sip-websocket-02
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In addition to this issue, there is the one about the current 3261 
requirement to support TCP and UDP transports, which many 
implementations of this draft will have difficulty complying with.

I'd like to hear more people's thoughts on whether to have this draft 
update 3261 on either or both of these points.

	Thanks,
	Paul

On 8/10/12 2:31 PM, Roman Shpount wrote:
> My only concern with this draft is the "Appendix B. Topology Hiding"
> which clearly states that SIP WebSocket Server would be non-compliant
> with RFC3261 if it implements topology hiding which is actually
> recommended. I would much rather see that this specification update
> RFC3261. It should update section 18.2.1 of RFC 3261 stating that the
> server SHOULD not to add "received" parameter to the via header if
> request was received via WebSocket protocol and MUST not add the this
> parameter if via "sent-by" header contains an address with ".invaid" domain.
> _____________
> Roman Shpount
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Hannes Tschofenig <Hannes.Tschofenig@gmx.net>
Thread-Topic: [sipcore] Geolocation Header Question
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Date: Sat, 11 Aug 2012 17:57:21 +0000
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Cc: "Worley, Dale R \(Dale\)" <dworley@avaya.com>, "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Geolocation Header Question
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You could do this:
Geolocation: <sip:+358222222222@atis.invalid;user=3Dphone>
(although why ATIS would be defining how calls from Finland work is beyond =
me ;)

But I don't understand why they need it in a Geoloc header to begin with.  =
I assumed that header was for devices/networks that use its mechanisms (ie,=
 HELD, SUB, pidf).  They're not going to do that, so why use that header?

You said they're doing the location lookup on the PSTN side - presumably th=
en the location is being looked up based on the calling party info, which s=
hould be in another existing header.  Can't they use one of them as the que=
ry key for the lookup? (we have plenty to choose from for that already :)

-hadriel


On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:

> I agree but the tel URI is not one of the supported location URIs in RFC =
6442.=20
> Hence, the guys just use a different way to convey the same information.=
=20
>=20
> On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>=20
>>> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>=20
>>> In order to convey that phone number for a lookup they are thinking
>>> about using the Geolocation header in the following form:
>>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=3Dphone)
>>=20
>> Naively, I'd say that since the goal is to carry an E.164 number, a
>> tel: URI would be the ideal choice.
>>=20
>> Dale
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Cc: "Worley, Dale R \(Dale\)" <dworley@avaya.com>, "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Geolocation Header Question
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I don't think so.

A geolocation header with a sip uri has to lead to a presence server that
can respond to a presence subscription which would return a PIDF with
location.

Routing a tel uri (which user=phone is supposed to be interchangeable
with) to a presence service usually makes no sense.  I don't think you can
send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If you
can't, user=phone is not okay.

It's perfectly reasonable to make the uri
sip:+358222222222@presence.example.com
that is, a real domain, and no user=phone

Brian
>
> You could do this:
> Geolocation: <sip:+358222222222@atis.invalid;user=phone>
> (although why ATIS would be defining how calls from Finland work is beyond
> me ;)
>
> But I don't understand why they need it in a Geoloc header to begin with.
> I assumed that header was for devices/networks that use its mechanisms
> (ie, HELD, SUB, pidf).  They're not going to do that, so why use that
> header?
>
> You said they're doing the location lookup on the PSTN side - presumably
> then the location is being looked up based on the calling party info,
> which should be in another existing header.  Can't they use one of them as
> the query key for the lookup? (we have plenty to choose from for that
> already :)
>
> -hadriel
>
>
> On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>
>> I agree but the tel URI is not one of the supported location URIs in RFC
>> 6442.
>> Hence, the guys just use a different way to convey the same information.
>>
>> On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>
>>>> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>>
>>>> In order to convey that phone number for a lookup they are thinking
>>>> about using the Geolocation header in the following form:
>>>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>>>
>>> Naively, I'd say that since the goal is to carry an E.164 number, a
>>> tel: URI would be the ideal choice.
>>>
>>> Dale
>>
>> _______________________________________________
>> 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 pkyzivat@alum.mit.edu  Tue Aug 14 12:59:06 2012
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Subject: Re: [sipcore] Geolocation Header Question
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I already commented to the original question.
Here I'm just commenting on the comments.

On 8/14/12 1:55 PM, Brian Rosen wrote:
> I don't think so.
>
> A geolocation header with a sip uri has to lead to a presence server that
> can respond to a presence subscription which would return a PIDF with
> location.
>
> Routing a tel uri (which user=phone is supposed to be interchangeable
> with) to a presence service usually makes no sense.  I don't think you can
> send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If you
> can't, user=phone is not okay.

I disagree that a tel uri is *equivalent* to a sip URI with user=phone!

A sip URI with user=phone is supposed to be routed like any other sip 
URI until it reaches a server responsible for the domain of the URI. 
*Then* the user part is to be interpreted as representing a phone number.

Conversely, a tel URI has no domain. (And no user=phone.) The semantics 
of telephone numbers are public knowledge. So any server can decide 
where to route it next, based on the number and whatever other context 
it has.

IMO the distinction is important. If you want to enable anybody to 
decide how to reach the phone number (including going through a PSTN 
GW), then use a tel URI. If you want the call handled via SIP at least 
until it reaches a particular domain, then use a sip URI.

	Thanks,
	Paul

> It's perfectly reasonable to make the uri
> sip:+358222222222@presence.example.com
> that is, a real domain, and no user=phone
>
> Brian
>>
>> You could do this:
>> Geolocation:<sip:+358222222222@atis.invalid;user=phone>
>> (although why ATIS would be defining how calls from Finland work is beyond
>> me ;)
>>
>> But I don't understand why they need it in a Geoloc header to begin with.
>> I assumed that header was for devices/networks that use its mechanisms
>> (ie, HELD, SUB, pidf).  They're not going to do that, so why use that
>> header?
>>
>> You said they're doing the location lookup on the PSTN side - presumably
>> then the location is being looked up based on the calling party info,
>> which should be in another existing header.  Can't they use one of them as
>> the query key for the lookup? (we have plenty to choose from for that
>> already :)
>>
>> -hadriel
>>
>>
>> On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>>
>>> I agree but the tel URI is not one of the supported location URIs in RFC
>>> 6442.
>>> Hence, the guys just use a different way to convey the same information.
>>>
>>> On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>>
>>>>> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>>>
>>>>> In order to convey that phone number for a lookup they are thinking
>>>>> about using the Geolocation header in the following form:
>>>>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>>>>
>>>> Naively, I'd say that since the goal is to carry an E.164 number, a
>>>> tel: URI would be the ideal choice.
>>>>
>>>> Dale
>>>
>>> _______________________________________________
>>> 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
>


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From: "Brian Rosen" <br@brianrosen.net>
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Subject: Re: [sipcore] Geolocation Header Question
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Yes, agree, but probably not relevant to the discussion.

I went back and looked at the original email, and decided I either don't
understand something, or there is some larger problem.

Between what two elements is this Geolocation header running?

UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?

Regardless, it seems to be misusing the header, because it's not a sip or
http LbyR.
Better to get a real scheme somehow that does the right thing, use a
parameter in another header, or create a new header.  Depends on what this
actually represents.  It sounds like it really is LbyR, and you want to
communicate a key.  <scheme>:<tel>@domain is appropriate.  The issue is
what is the right scheme?  It's not sip - it's not sip presence.  It's not
HELD identity I take it.

Brian

> I already commented to the original question.
> Here I'm just commenting on the comments.
>
> On 8/14/12 1:55 PM, Brian Rosen wrote:
>> I don't think so.
>>
>> A geolocation header with a sip uri has to lead to a presence server
>> that
>> can respond to a presence subscription which would return a PIDF with
>> location.
>>
>> Routing a tel uri (which user=phone is supposed to be interchangeable
>> with) to a presence service usually makes no sense.  I don't think you
>> can
>> send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If
>> you
>> can't, user=phone is not okay.
>
> I disagree that a tel uri is *equivalent* to a sip URI with user=phone!
>
> A sip URI with user=phone is supposed to be routed like any other sip
> URI until it reaches a server responsible for the domain of the URI.
> *Then* the user part is to be interpreted as representing a phone number.
>
> Conversely, a tel URI has no domain. (And no user=phone.) The semantics
> of telephone numbers are public knowledge. So any server can decide
> where to route it next, based on the number and whatever other context
> it has.
>
> IMO the distinction is important. If you want to enable anybody to
> decide how to reach the phone number (including going through a PSTN
> GW), then use a tel URI. If you want the call handled via SIP at least
> until it reaches a particular domain, then use a sip URI.
>
> 	Thanks,
> 	Paul
>
>> It's perfectly reasonable to make the uri
>> sip:+358222222222@presence.example.com
>> that is, a real domain, and no user=phone
>>
>> Brian
>>>
>>> You could do this:
>>> Geolocation:<sip:+358222222222@atis.invalid;user=phone>
>>> (although why ATIS would be defining how calls from Finland work is
>>> beyond
>>> me ;)
>>>
>>> But I don't understand why they need it in a Geoloc header to begin
>>> with.
>>> I assumed that header was for devices/networks that use its mechanisms
>>> (ie, HELD, SUB, pidf).  They're not going to do that, so why use that
>>> header?
>>>
>>> You said they're doing the location lookup on the PSTN side -
>>> presumably
>>> then the location is being looked up based on the calling party info,
>>> which should be in another existing header.  Can't they use one of them
>>> as
>>> the query key for the lookup? (we have plenty to choose from for that
>>> already :)
>>>
>>> -hadriel
>>>
>>>
>>> On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>>>
>>>> I agree but the tel URI is not one of the supported location URIs in
>>>> RFC
>>>> 6442.
>>>> Hence, the guys just use a different way to convey the same
>>>> information.
>>>>
>>>> On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>>>
>>>>>> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>>>>
>>>>>> In order to convey that phone number for a lookup they are thinking
>>>>>> about using the Geolocation header in the following form:
>>>>>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>>>>>
>>>>> Naively, I'd say that since the goal is to carry an E.164 number, a
>>>>> tel: URI would be the ideal choice.
>>>>>
>>>>> Dale
>>>>
>>>> _______________________________________________
>>>> 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
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>



From pkyzivat@alum.mit.edu  Tue Aug 14 14:11:58 2012
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Subject: Re: [sipcore] Geolocation Header Question
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On 8/14/12 5:04 PM, Brian Rosen wrote:
> Yes, agree, but probably not relevant to the discussion.
>
> I went back and looked at the original email, and decided I either don't
> understand something, or there is some larger problem.
>
> Between what two elements is this Geolocation header running?
>
> UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?
>
> Regardless, it seems to be misusing the header, because it's not a sip or
> http LbyR.
> Better to get a real scheme somehow that does the right thing, use a
> parameter in another header, or create a new header.  Depends on what this
> actually represents.  It sounds like it really is LbyR, and you want to
> communicate a key.<scheme>:<tel>@domain is appropriate.  The issue is
> what is the right scheme?  It's not sip - it's not sip presence.  It's not
> HELD identity I take it.

As I already stated, I think sip: is ok if this special lookup can be 
viewed as an *optimization* over doing the presence lookup, and doing it 
the presence way still works.

Otherwise I agree with Brian.

	Thanks,
	Paul

> Brian
>
>> I already commented to the original question.
>> Here I'm just commenting on the comments.
>>
>> On 8/14/12 1:55 PM, Brian Rosen wrote:
>>> I don't think so.
>>>
>>> A geolocation header with a sip uri has to lead to a presence server
>>> that
>>> can respond to a presence subscription which would return a PIDF with
>>> location.
>>>
>>> Routing a tel uri (which user=phone is supposed to be interchangeable
>>> with) to a presence service usually makes no sense.  I don't think you
>>> can
>>> send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If
>>> you
>>> can't, user=phone is not okay.
>>
>> I disagree that a tel uri is *equivalent* to a sip URI with user=phone!
>>
>> A sip URI with user=phone is supposed to be routed like any other sip
>> URI until it reaches a server responsible for the domain of the URI.
>> *Then* the user part is to be interpreted as representing a phone number.
>>
>> Conversely, a tel URI has no domain. (And no user=phone.) The semantics
>> of telephone numbers are public knowledge. So any server can decide
>> where to route it next, based on the number and whatever other context
>> it has.
>>
>> IMO the distinction is important. If you want to enable anybody to
>> decide how to reach the phone number (including going through a PSTN
>> GW), then use a tel URI. If you want the call handled via SIP at least
>> until it reaches a particular domain, then use a sip URI.
>>
>> 	Thanks,
>> 	Paul
>>
>>> It's perfectly reasonable to make the uri
>>> sip:+358222222222@presence.example.com
>>> that is, a real domain, and no user=phone
>>>
>>> Brian
>>>>
>>>> You could do this:
>>>> Geolocation:<sip:+358222222222@atis.invalid;user=phone>
>>>> (although why ATIS would be defining how calls from Finland work is
>>>> beyond
>>>> me ;)
>>>>
>>>> But I don't understand why they need it in a Geoloc header to begin
>>>> with.
>>>> I assumed that header was for devices/networks that use its mechanisms
>>>> (ie, HELD, SUB, pidf).  They're not going to do that, so why use that
>>>> header?
>>>>
>>>> You said they're doing the location lookup on the PSTN side -
>>>> presumably
>>>> then the location is being looked up based on the calling party info,
>>>> which should be in another existing header.  Can't they use one of them
>>>> as
>>>> the query key for the lookup? (we have plenty to choose from for that
>>>> already :)
>>>>
>>>> -hadriel
>>>>
>>>>
>>>> On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>>>>
>>>>> I agree but the tel URI is not one of the supported location URIs in
>>>>> RFC
>>>>> 6442.
>>>>> Hence, the guys just use a different way to convey the same
>>>>> information.
>>>>>
>>>>> On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>>>>
>>>>>>> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>>>>>
>>>>>>> In order to convey that phone number for a lookup they are thinking
>>>>>>> about using the Geolocation header in the following form:
>>>>>>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>>>>>>
>>>>>> Naively, I'd say that since the goal is to carry an E.164 number, a
>>>>>> tel: URI would be the ideal choice.
>>>>>>
>>>>>> Dale
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>
>
>


From rjsparks@nostrum.com  Tue Aug 21 14:56:58 2012
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Subject: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05
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Summary: There are a few issues to address before progressing to IETF 
LC. I'm putting the document into revised ID needed to address at least 
the IANA points below.

The primary issue is with the instructions to IANA on creating the new 
trees and the registration template used to put things in those trees.

Sections 7.3.2 and 7.3.3 don't currently define the format of an entry 
in the trees in the new subregistry. This needs to be clearly defined, 
and the sections need to explicitly indicate that the trees are 
initially empty.

The template in section 8 was adapted from RFC2506, but it needs more 
tweaks and it needs to match whatever format is defined above.

   * We don't need a separate list for reviewing these requests. We 
certainly don't need a list at apps.ietf.org. The normal process for 
submitting a request for the global tree should be submitting the 
template by email to iana@iana.org, and for the sip tree, the normal 
process would be submitting an Internet-Draft to the IESG.

   * IANA is currently using a different for media feature tag 
registrations. It's available at 
<http://www.iana.org/protocols/media-feature-tag-template.txt>. If we 
really need to follow that registration format, please start from this 
template. But do we need all this information in the registry?

Other major issues:

1) Should the expert reviewer be explicitly required to consider 
security issues as part of that review? (see section 4.6 of 
<https://datatracker.ietf.org/doc/draft-ietf-appsawg-media-type-regs/> 
for example).

2) The security considerations section claims that the aspects of the 
considerations in rfc3840 that apply to end user devices don't apply 
here since these are not typically used to convey information about end 
user devices. "Not typically" is weak. Can this sentence be made more 
specific about what concerns do and don't apply? The concerns in 3840 
about influencing application behavior, and in 2506 about providing 
information that can be used to exploit implementation weakeness at the 
elements that do provide these tags should be reflected here.

3) Consider clarifying the behavior of SUBSCRIBE NOTIFY dialogs in 
section 4.3.2. Only one of the 200 OKs to a forked SUBSCRIBE will get 
back to the UA that sent the subscribe. The information that creates the 
other dialogs for it must appear in the immediate NOTIFYs. Is there a 
requirement for the end accepting a subscription to use the same feature 
capabilities in the 200 OK to the subscription and in the immediate NOTIFY?

4) Similarly, consider clarifying whether UPDATES nested within an 
INVITE constrained to keep the same feature capability indicators as in 
the INVITE?

Minor Issues:

Section 4.3.4: A SIP Stand-Alone Transaction is by definition a 
non-INVITE transaction. RFC4320 made 18x to non-INVITES a MUST NOT. I 
suggest s/18x or 2xx/2xx/ in this section.

The two places where you talk about overriding registration policy in 
section 5.2 isn't quite right. We should be specific that we're only 
talking about SIP header field parameters. In the first case, we're 
using IETF Consensus (this document) to delegate parameters starting 
with a 'g.' for _this_ header field to a policy of Expert Review. In the 
second case, there's no overriding at all - registrations use IETF 
Consensus there.  Here's a stab at something more accurate. Maybe others 
can come up with something better. I suggest replacing the first 
instance with "Note that the general policy for registering SIP header 
field parameters is 'IETF Consensus' as specified in [RFC3968]. This 
document delegates the registration of parameters to the Feature-Caps 
header field starting with 'g.' to a policy of 'Expert Review'", and 
deleting the second instance.

The NOTE in section 6.2.1 is not useful unless you know the history of 
that led to the syntax in 3840 and 3841. I suggest replacing it with 
'NOTE: The "*" value is present in order to follow the guidelines for 
syntac in [RFC4485] and to maintain a consistent format with [RFC3840] 
and [RFC3841].'

Nits:

There is a mix of "xxx" "XXX" and "XXXX" to indicate to the RFC Editor 
that the number assigned to this RFC should appear here. Please choose 
one format and make the instructions to the RFC Editor for making the 
substitution clear.

Section 4.1: I suggest s/in a non-priority order, and any/as an 
unordered set. Any/

Why is the second paragraph of 4.1 in the introduction to this section? 
It's getting pretty deep into specification detail. It would fit better 
at the end of 4.2.1.

Section 4.2.2: The beginning of paragraphs 2 an 4 start "When a UA", but 
don't mean any UA. They rely on the context of the previous paragraphs 
to scope it to a UA that is part of a B2BUA. To help avoid confusion I 
suggest changing both of these this way: s/When a UA/When such a UA/

The NOTE in section 4.3.1 is confusing. What's the distinction between a 
SIP method and a request type that's important here? What methods does 
this specification leave the behavior undefined for. I don't think there 
are any except perhaps CANCEL. Is the note important to keep?

Section 4.3.2: s/is no long supported/is no longer supported/

Section 4.3.3: The NOTE says support applies to a Contact. It doesn't 
really - it applies to messages sent with that Contact as the target 
perhaps.

Section 5.1 has some edititis. Parts of it look like a "structure of 
this document" description, which should be in the Introduction to the 
document. It's especially odd that it points to earlier in the document. 
I suggest either deleting the last 3 1-sentence paragraphs, or moving 
them to section 1, and making the document roadmap more complete.

Section 5.2.2: I suggest s/ensure that the feature capability indicator 
meets/ensure that the definition of the feature capability indicator meets/

Section 5.3.3: STRONGLY RECOMMENDED isn't 2119. I suspect some 2119 
purists will object that this sentence doesn't need 2119 at all. I 
suggest at least changing STRONGLY to lower-case, and consider changing 
both words.

Section 7.3.2: s/through the Expert Review policies/through the Expert 
Review policy/



From pkyzivat@alum.mit.edu  Tue Aug 21 16:00:59 2012
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05
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Thanks Robert!

I'm going to defer to Christer on most of this, but want to comment on a 
couple of things:

On 8/21/12 5:56 PM, Robert Sparks wrote:
> Summary: There are a few issues to address before progressing to IETF
> LC. I'm putting the document into revised ID needed to address at least
> the IANA points below.
>
> The primary issue is with the instructions to IANA on creating the new
> trees and the registration template used to put things in those trees.
>
> Sections 7.3.2 and 7.3.3 don't currently define the format of an entry
> in the trees in the new subregistry. This needs to be clearly defined,
> and the sections need to explicitly indicate that the trees are
> initially empty.
>
> The template in section 8 was adapted from RFC2506, but it needs more
> tweaks and it needs to match whatever format is defined above.
>
>    * We don't need a separate list for reviewing these requests. We
> certainly don't need a list at apps.ietf.org. The normal process for
> submitting a request for the global tree should be submitting the
> template by email to iana@iana.org, and for the sip tree, the normal
> process would be submitting an Internet-Draft to the IESG.
>
>    * IANA is currently using a different for media feature tag
> registrations. It's available at
> <http://www.iana.org/protocols/media-feature-tag-template.txt>. If we
> really need to follow that registration format, please start from this
> template. But do we need all this information in the registry?

I find this one, and the old one, a bit strange.
The template asks for a bunch of information that IANA doesn't capture 
in its subregistry. Presumably that info could be used by the expert 
reviewer, but it wouldn't be available to users. In practice it appears 
that all the things registered have a reference to a document that might 
contain the same info. (Or it could contain conflicting info.)

So I'm thinking we need to do better than has been done for media 
feature tags.

> Other major issues:
>
> 1) Should the expert reviewer be explicitly required to consider
> security issues as part of that review? (see section 4.6 of
> <https://datatracker.ietf.org/doc/draft-ietf-appsawg-media-type-regs/>
> for example).
>
> 2) The security considerations section claims that the aspects of the
> considerations in rfc3840 that apply to end user devices don't apply
> here since these are not typically used to convey information about end
> user devices. "Not typically" is weak. Can this sentence be made more
> specific about what concerns do and don't apply? The concerns in 3840
> about influencing application behavior, and in 2506 about providing
> information that can be used to exploit implementation weakeness at the
> elements that do provide these tags should be reflected here.
>
> 3) Consider clarifying the behavior of SUBSCRIBE NOTIFY dialogs in
> section 4.3.2. Only one of the 200 OKs to a forked SUBSCRIBE will get
> back to the UA that sent the subscribe. The information that creates the
> other dialogs for it must appear in the immediate NOTIFYs. Is there a
> requirement for the end accepting a subscription to use the same feature
> capabilities in the 200 OK to the subscription and in the immediate NOTIFY?
>
> 4) Similarly, consider clarifying whether UPDATES nested within an
> INVITE constrained to keep the same feature capability indicators as in
> the INVITE?

I presume you mean an UPDATE to a before the final response to the 
dialog establishing INVITE is received.

It seems that UPDATES after the dialog is established aren't permitted 
to carry the header.

> Minor Issues:
>
> Section 4.3.4: A SIP Stand-Alone Transaction is by definition a
> non-INVITE transaction. RFC4320 made 18x to non-INVITES a MUST NOT. I
> suggest s/18x or 2xx/2xx/ in this section.
>
> The two places where you talk about overriding registration policy in
> section 5.2 isn't quite right. We should be specific that we're only
> talking about SIP header field parameters. In the first case, we're
> using IETF Consensus (this document) to delegate parameters starting
> with a 'g.' for _this_ header field to a policy of Expert Review. In the
> second case, there's no overriding at all - registrations use IETF
> Consensus there.  Here's a stab at something more accurate. Maybe others
> can come up with something better. I suggest replacing the first
> instance with "Note that the general policy for registering SIP header
> field parameters is 'IETF Consensus' as specified in [RFC3968]. This
> document delegates the registration of parameters to the Feature-Caps
> header field starting with 'g.' to a policy of 'Expert Review'", and
> deleting the second instance.

You and I already talked about this, but I realize now that we have to 
work on the wording more. The issue is that from a sip header field 
perspective the parameter includes the "+", and its parameters that 
begin with "+" that we are reserving in the header field parameter registry.

But the "+" isn't part of the name of the feature capability indicator.

So we are delegating parameters that start with "+g." to the expert 
review procedure.

And while things in the sip tree (things beginning with "+sip.", while 
still requiring IETF consensus, will still go in the sip tree, not as 
individuals in the sip header fields parameter registry. But maybe we 
don't need to "note" this part.

> The NOTE in section 6.2.1 is not useful unless you know the history of
> that led to the syntax in 3840 and 3841. I suggest replacing it with
> 'NOTE: The "*" value is present in order to follow the guidelines for
> syntac in [RFC4485] and to maintain a consistent format with [RFC3840]
> and [RFC3841].'
>
> Nits:
>
> There is a mix of "xxx" "XXX" and "XXXX" to indicate to the RFC Editor
> that the number assigned to this RFC should appear here. Please choose
> one format and make the instructions to the RFC Editor for making the
> substitution clear.

I recently discovered that xml2rfc has '&rfc.number;' to get this value, 
though I haven't tried it yet. Apparently it expands to XXXX for drafts.

	Thanks,
	Paul

From jmpolk@cisco.com  Tue Aug 21 17:05:10 2012
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Brian Rosen <br@brianrosen.net>
From: James Polk <jmpolk@cisco.com>
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Subject: Re: [sipcore] Geolocation Header Question
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At 04:11 PM 8/14/2012, Paul Kyzivat wrote:
>On 8/14/12 5:04 PM, Brian Rosen wrote:
>>Yes, agree, but probably not relevant to the discussion.
>>
>>I went back and looked at the original email, and decided I either don't
>>understand something, or there is some larger problem.
>>
>>Between what two elements is this Geolocation header running?
>>
>>UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?
>>
>>Regardless, it seems to be misusing the header, because it's not a sip or
>>http LbyR.
>>Better to get a real scheme somehow that does the right thing, use a
>>parameter in another header, or create a new header.  Depends on what this
>>actually represents.  It sounds like it really is LbyR, and you want to
>>communicate a key.<scheme>:<tel>@domain is appropriate.  The issue is
>>what is the right scheme?  It's not sip - it's not sip presence.  It's not
>>HELD identity I take it.
>
>As I already stated, I think sip: is ok if this special lookup can 
>be viewed as an *optimization* over doing the presence lookup, and 
>doing it the presence way still works.
>
>Otherwise I agree with Brian.

<tel:> isn't an allowed URI in RFC 6442, neither is the use=phone 
tag, so IMO that makes this offering a no go as far as 'being in line 
with an IETF standards track RFC' (to be formal about it).

James


>         Thanks,
>         Paul
>
>>Brian
>>
>>>I already commented to the original question.
>>>Here I'm just commenting on the comments.
>>>
>>>On 8/14/12 1:55 PM, Brian Rosen wrote:
>>>>I don't think so.
>>>>
>>>>A geolocation header with a sip uri has to lead to a presence server
>>>>that
>>>>can respond to a presence subscription which would return a PIDF with
>>>>location.
>>>>
>>>>Routing a tel uri (which user=phone is supposed to be interchangeable
>>>>with) to a presence service usually makes no sense.  I don't think you
>>>>can
>>>>send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If
>>>>you
>>>>can't, user=phone is not okay.
>>>
>>>I disagree that a tel uri is *equivalent* to a sip URI with user=phone!
>>>
>>>A sip URI with user=phone is supposed to be routed like any other sip
>>>URI until it reaches a server responsible for the domain of the URI.
>>>*Then* the user part is to be interpreted as representing a phone number.
>>>
>>>Conversely, a tel URI has no domain. (And no user=phone.) The semantics
>>>of telephone numbers are public knowledge. So any server can decide
>>>where to route it next, based on the number and whatever other context
>>>it has.
>>>
>>>IMO the distinction is important. If you want to enable anybody to
>>>decide how to reach the phone number (including going through a PSTN
>>>GW), then use a tel URI. If you want the call handled via SIP at least
>>>until it reaches a particular domain, then use a sip URI.
>>>
>>>         Thanks,
>>>         Paul
>>>
>>>>It's perfectly reasonable to make the uri
>>>>sip:+358222222222@presence.example.com
>>>>that is, a real domain, and no user=phone
>>>>
>>>>Brian
>>>>>
>>>>>You could do this:
>>>>>Geolocation:<sip:+358222222222@atis.invalid;user=phone>
>>>>>(although why ATIS would be defining how calls from Finland work is
>>>>>beyond
>>>>>me ;)
>>>>>
>>>>>But I don't understand why they need it in a Geoloc header to begin
>>>>>with.
>>>>>I assumed that header was for devices/networks that use its mechanisms
>>>>>(ie, HELD, SUB, pidf).  They're not going to do that, so why use that
>>>>>header?
>>>>>
>>>>>You said they're doing the location lookup on the PSTN side -
>>>>>presumably
>>>>>then the location is being looked up based on the calling party info,
>>>>>which should be in another existing header.  Can't they use one of them
>>>>>as
>>>>>the query key for the lookup? (we have plenty to choose from for that
>>>>>already :)
>>>>>
>>>>>-hadriel
>>>>>
>>>>>
>>>>>On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>>>>>
>>>>>>I agree but the tel URI is not one of the supported location URIs in
>>>>>>RFC
>>>>>>6442.
>>>>>>Hence, the guys just use a different way to convey the same
>>>>>>information.
>>>>>>
>>>>>>On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>>>>>
>>>>>>>>From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>>>>>>
>>>>>>>>In order to convey that phone number for a lookup they are thinking
>>>>>>>>about using the Geolocation header in the following form:
>>>>>>>>Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>>>>>>>
>>>>>>>Naively, I'd say that since the goal is to carry an E.164 number, a
>>>>>>>tel: URI would be the ideal choice.
>>>>>>>
>>>>>>>Dale
>>>>>>
>>>>>>_______________________________________________
>>>>>>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
>>>
>>>_______________________________________________
>>>sipcore mailing list
>>>sipcore@ietf.org
>>>https://www.ietf.org/mailman/listinfo/sipcore
>>
>>
>
>_______________________________________________
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>sipcore@ietf.org
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Subject: Re: [sipcore] Geolocation Header Question
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On 8/21/12 8:05 PM, James Polk wrote:
> At 04:11 PM 8/14/2012, Paul Kyzivat wrote:
>> On 8/14/12 5:04 PM, Brian Rosen wrote:
>>> Yes, agree, but probably not relevant to the discussion.
>>>
>>> I went back and looked at the original email, and decided I either don't
>>> understand something, or there is some larger problem.
>>>
>>> Between what two elements is this Geolocation header running?
>>>
>>> UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?
>>>
>>> Regardless, it seems to be misusing the header, because it's not a
>>> sip or
>>> http LbyR.
>>> Better to get a real scheme somehow that does the right thing, use a
>>> parameter in another header, or create a new header.  Depends on what
>>> this
>>> actually represents.  It sounds like it really is LbyR, and you want to
>>> communicate a key.<scheme>:<tel>@domain is appropriate.  The issue is
>>> what is the right scheme?  It's not sip - it's not sip presence.
>>> It's not
>>> HELD identity I take it.
>>
>> As I already stated, I think sip: is ok if this special lookup can be
>> viewed as an *optimization* over doing the presence lookup, and doing
>> it the presence way still works.
>>
>> Otherwise I agree with Brian.
>
> <tel:> isn't an allowed URI in RFC 6442, neither is the use=phone tag,
> so IMO that makes this offering a no go as far as 'being in line with an
> IETF standards track RFC' (to be formal about it).

how does it forbid user=phone? (It's a sip uri parameter, not a header 
parameter.)

> James
>
>
>>         Thanks,
>>         Paul
>>
>>> Brian
>>>
>>>> I already commented to the original question.
>>>> Here I'm just commenting on the comments.
>>>>
>>>> On 8/14/12 1:55 PM, Brian Rosen wrote:
>>>>> I don't think so.
>>>>>
>>>>> A geolocation header with a sip uri has to lead to a presence server
>>>>> that
>>>>> can respond to a presence subscription which would return a PIDF with
>>>>> location.
>>>>>
>>>>> Routing a tel uri (which user=phone is supposed to be interchangeable
>>>>> with) to a presence service usually makes no sense.  I don't think you
>>>>> can
>>>>> send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If
>>>>> you
>>>>> can't, user=phone is not okay.
>>>>
>>>> I disagree that a tel uri is *equivalent* to a sip URI with user=phone!
>>>>
>>>> A sip URI with user=phone is supposed to be routed like any other sip
>>>> URI until it reaches a server responsible for the domain of the URI.
>>>> *Then* the user part is to be interpreted as representing a phone
>>>> number.
>>>>
>>>> Conversely, a tel URI has no domain. (And no user=phone.) The semantics
>>>> of telephone numbers are public knowledge. So any server can decide
>>>> where to route it next, based on the number and whatever other context
>>>> it has.
>>>>
>>>> IMO the distinction is important. If you want to enable anybody to
>>>> decide how to reach the phone number (including going through a PSTN
>>>> GW), then use a tel URI. If you want the call handled via SIP at least
>>>> until it reaches a particular domain, then use a sip URI.
>>>>
>>>>         Thanks,
>>>>         Paul
>>>>
>>>>> It's perfectly reasonable to make the uri
>>>>> sip:+358222222222@presence.example.com
>>>>> that is, a real domain, and no user=phone
>>>>>
>>>>> Brian
>>>>>>
>>>>>> You could do this:
>>>>>> Geolocation:<sip:+358222222222@atis.invalid;user=phone>
>>>>>> (although why ATIS would be defining how calls from Finland work is
>>>>>> beyond
>>>>>> me ;)
>>>>>>
>>>>>> But I don't understand why they need it in a Geoloc header to begin
>>>>>> with.
>>>>>> I assumed that header was for devices/networks that use its
>>>>>> mechanisms
>>>>>> (ie, HELD, SUB, pidf).  They're not going to do that, so why use that
>>>>>> header?
>>>>>>
>>>>>> You said they're doing the location lookup on the PSTN side -
>>>>>> presumably
>>>>>> then the location is being looked up based on the calling party info,
>>>>>> which should be in another existing header.  Can't they use one of
>>>>>> them
>>>>>> as
>>>>>> the query key for the lookup? (we have plenty to choose from for that
>>>>>> already :)
>>>>>>
>>>>>> -hadriel
>>>>>>
>>>>>>
>>>>>> On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>>>>>>
>>>>>>> I agree but the tel URI is not one of the supported location URIs in
>>>>>>> RFC
>>>>>>> 6442.
>>>>>>> Hence, the guys just use a different way to convey the same
>>>>>>> information.
>>>>>>>
>>>>>>> On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>>>>>>
>>>>>>>>> From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>>>>>>>
>>>>>>>>> In order to convey that phone number for a lookup they are
>>>>>>>>> thinking
>>>>>>>>> about using the Geolocation header in the following form:
>>>>>>>>> Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>>>>>>>>
>>>>>>>> Naively, I'd say that since the goal is to carry an E.164 number, a
>>>>>>>> tel: URI would be the ideal choice.
>>>>>>>>
>>>>>>>> Dale
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>
>>>> _______________________________________________
>>>> 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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On 8/22/12 7:31 AM, "ext Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:

>> <tel:> isn't an allowed URI in RFC 6442, neither is the use=phone tag,
>> so IMO that makes this offering a no go as far as 'being in line with an
>> IETF standards track RFC' (to be formal about it).
> 
> how does it forbid user=phone? (It's a sip uri parameter, not a header
> parameter.)

Reading through the SIP location conveyance document I couldn't find a place
that said sip:+1222222222@foo-bar.com;user=phone isn't allowed.

So, I agree with Paul. 


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: James Polk <jmpolk@cisco.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Brian Rosen <br@brianrosen.net>
Date: Wed, 22 Aug 2012 03:15:50 -0400
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Um... why isn't sip:<tel>@domain valid? I see sip-uri as valid in the BNF, =
and that implies tel in the user portion.

Jon Peterson
NeuStar, Inc.
________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Jame=
s Polk [jmpolk@cisco.com]
Sent: Tuesday, August 21, 2012 5:05 PM
To: Paul Kyzivat; Brian Rosen
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Geolocation Header Question

At 04:11 PM 8/14/2012, Paul Kyzivat wrote:
>On 8/14/12 5:04 PM, Brian Rosen wrote:
>>Yes, agree, but probably not relevant to the discussion.
>>
>>I went back and looked at the original email, and decided I either don't
>>understand something, or there is some larger problem.
>>
>>Between what two elements is this Geolocation header running?
>>
>>UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?
>>
>>Regardless, it seems to be misusing the header, because it's not a sip or
>>http LbyR.
>>Better to get a real scheme somehow that does the right thing, use a
>>parameter in another header, or create a new header.  Depends on what thi=
s
>>actually represents.  It sounds like it really is LbyR, and you want to
>>communicate a key.<scheme>:<tel>@domain is appropriate.  The issue is
>>what is the right scheme?  It's not sip - it's not sip presence.  It's no=
t
>>HELD identity I take it.
>
>As I already stated, I think sip: is ok if this special lookup can
>be viewed as an *optimization* over doing the presence lookup, and
>doing it the presence way still works.
>
>Otherwise I agree with Brian.

<tel:> isn't an allowed URI in RFC 6442, neither is the use=3Dphone
tag, so IMO that makes this offering a no go as far as 'being in line
with an IETF standards track RFC' (to be formal about it).

James


>         Thanks,
>         Paul
>
>>Brian
>>
>>>I already commented to the original question.
>>>Here I'm just commenting on the comments.
>>>
>>>On 8/14/12 1:55 PM, Brian Rosen wrote:
>>>>I don't think so.
>>>>
>>>>A geolocation header with a sip uri has to lead to a presence server
>>>>that
>>>>can respond to a presence subscription which would return a PIDF with
>>>>location.
>>>>
>>>>Routing a tel uri (which user=3Dphone is supposed to be interchangeable
>>>>with) to a presence service usually makes no sense.  I don't think you
>>>>can
>>>>send a SUBSCRIBE to a tel uri.  If you could, user=3Dphone is okay.  If
>>>>you
>>>>can't, user=3Dphone is not okay.
>>>
>>>I disagree that a tel uri is *equivalent* to a sip URI with user=3Dphone=
!
>>>
>>>A sip URI with user=3Dphone is supposed to be routed like any other sip
>>>URI until it reaches a server responsible for the domain of the URI.
>>>*Then* the user part is to be interpreted as representing a phone number=
.
>>>
>>>Conversely, a tel URI has no domain. (And no user=3Dphone.) The semantic=
s
>>>of telephone numbers are public knowledge. So any server can decide
>>>where to route it next, based on the number and whatever other context
>>>it has.
>>>
>>>IMO the distinction is important. If you want to enable anybody to
>>>decide how to reach the phone number (including going through a PSTN
>>>GW), then use a tel URI. If you want the call handled via SIP at least
>>>until it reaches a particular domain, then use a sip URI.
>>>
>>>         Thanks,
>>>         Paul
>>>
>>>>It's perfectly reasonable to make the uri
>>>>sip:+358222222222@presence.example.com
>>>>that is, a real domain, and no user=3Dphone
>>>>
>>>>Brian
>>>>>
>>>>>You could do this:
>>>>>Geolocation:<sip:+358222222222@atis.invalid;user=3Dphone>
>>>>>(although why ATIS would be defining how calls from Finland work is
>>>>>beyond
>>>>>me ;)
>>>>>
>>>>>But I don't understand why they need it in a Geoloc header to begin
>>>>>with.
>>>>>I assumed that header was for devices/networks that use its mechanisms
>>>>>(ie, HELD, SUB, pidf).  They're not going to do that, so why use that
>>>>>header?
>>>>>
>>>>>You said they're doing the location lookup on the PSTN side -
>>>>>presumably
>>>>>then the location is being looked up based on the calling party info,
>>>>>which should be in another existing header.  Can't they use one of the=
m
>>>>>as
>>>>>the query key for the lookup? (we have plenty to choose from for that
>>>>>already :)
>>>>>
>>>>>-hadriel
>>>>>
>>>>>
>>>>>On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>>>>>
>>>>>>I agree but the tel URI is not one of the supported location URIs in
>>>>>>RFC
>>>>>>6442.
>>>>>>Hence, the guys just use a different way to convey the same
>>>>>>information.
>>>>>>
>>>>>>On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>>>>>
>>>>>>>>From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>>>>>>>
>>>>>>>>In order to convey that phone number for a lookup they are thinking
>>>>>>>>about using the Geolocation header in the following form:
>>>>>>>>Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=3Dphone)
>>>>>>>
>>>>>>>Naively, I'd say that since the goal is to carry an E.164 number, a
>>>>>>>tel: URI would be the ideal choice.
>>>>>>>
>>>>>>>Dale
>>>>>>
>>>>>>_______________________________________________
>>>>>>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
>>>
>>>_______________________________________________
>>>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 christer.holmberg@ericsson.com  Wed Aug 22 02:19:01 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-proxy-feature@tools.ietf.org" <draft-ietf-sipcore-proxy-feature@tools.ietf.org>
Date: Wed, 22 Aug 2012 11:18:54 +0200
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - minor issues and nits
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - minor issues and nits
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Hi Robert,

(I will address the major issue in a separate reply)

Minor Issues:
------------------------------------------------

> Section 4.3.4: A SIP Stand-Alone Transaction is by definition a non-INVIT=
E transaction. RFC4320 made 18x to non-INVITES a MUST NOT. I suggest s/18x =
or 2xx/2xx/ in this section.

Ok. I will change as suggested.

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

> The two places where you talk about overriding registration policy in sec=
tion 5.2 isn't quite right. We should be specific that=20
> we're only talking about SIP header field parameters. In the first case, =
we're using IETF Consensus (this document) to delegate=20
> parameters starting with a 'g.' for _this_ header field to a policy of Ex=
pert Review. In the second case, there's no overriding at=20
> all - registrations use IETF Consensus there.  Here's a stab at something=
 more accurate. Maybe others can come up with=20
> something better. I suggest replacing the first instance with "Note that =
the general policy for registering SIP header field=20
> parameters is 'IETF Consensus' as specified in [RFC3968]. This document d=
elegates the registration of parameters to the
> Feature-Caps header field starting with 'g.' to a policy of 'Expert Revie=
w'", and deleting the second instance.

> The NOTE in section 6.2.1 is not useful unless you know the history of th=
at led to the syntax in 3840 and 3841. I suggest replacing it with
> 'NOTE: The "*" value is present in order to follow the guidelines for syn=
tac in [RFC4485] and to maintain a consistent format with [RFC3840] and [RF=
C3841].'

Ok. I will change as suggested.


Nits:
------------------------------------------------

> There is a mix of "xxx" "XXX" and "XXXX" to indicate to the RFC Editor th=
at the number assigned to this RFC should appear here. Please choose one fo=
rmat and make the instructions to the RFC Editor for making the substitutio=
n clear.

Ok. I'll fix that.

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

> Section 4.1: I suggest s/in a non-priority order, and any/as an unordered=
 set. Any/

Ok. I will change as suggested.

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

> Why is the second paragraph of 4.1 in the introduction to this section?=20
> It's getting pretty deep into specification detail. It would fit better a=
t the end of 4.2.1.

Ok. I will move it there.

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

> Section 4.2.2: The beginning of paragraphs 2 an 4 start "When a UA", but =
don't mean any UA. They=20
> rely on the context of the previous paragraphs to scope it to a UA that i=
s part of a B2BUA. To help
> avoid confusion I suggest changing both of these this way: s/When a UA/Wh=
en such a UA/

Ok. I will change as suggested.

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

> The NOTE in section 4.3.1 is confusing. What's the distinction between a =
SIP method and a request type=20
> that's important here? What methods does this specification leave the beh=
avior undefined for. I don't think
> there are any except perhaps CANCEL. Is the note important to keep?

The content of the note is quite obvious, so I can remove it.

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

> Section 4.3.2: s/is no long supported/is no longer supported/

Ok. I will change as suggested.

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

>Section 4.3.3: The NOTE says support applies to a Contact. It doesn't real=
ly - it applies to messages sent with that Contact as the target perhaps.

I suggest the new modified note text:

  	"NOTE: While a REGISTER response can contain contacts that have been
   	registered as part of other registration transactions, support of any
   	indicated feature only applies to requests sent to the contact(s) that=
=20
	were explicitly conveyed in the associated REGISTER request."


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

> Section 5.1 has some edititis. Parts of it look like a "structure of this=
 document" description, which should be in the=20
> Introduction to the document. It's especially odd that it points to earli=
er in the document.=20
> I suggest either deleting the last 3 1-sentence paragraphs, or moving the=
m to section 1, and making the document=20
> roadmap more complete.

I suggest to remove them. Adding them to section 1 (where I do agree that t=
hey belong if we would keep them) makes little sense, as the table of conte=
nts comes just before section 1.

So, section 5.1 would look like:

"5.1.  Introduction

   Feature capability indicators are used by SIP entities not
   represented by the URI of the Contact header field to indicate
   support of features and capabilities, where media feature tags cannot
   be used to indicate the support.

   A value, or a list of values, that provides additional information
   about the supported feature or capability, can be associated with a
   feature capability indicator."

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

> Section 5.2.2: I suggest s/ensure that the feature capability indicator m=
eets/ensure that the definition of the feature capability indicator meets/

Ok. I will change as suggested.

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

> Section 5.3.3: STRONGLY RECOMMENDED isn't 2119. I suspect some 2119 puris=
ts will object that this sentence doesn't need 2119 at all. I suggest at le=
ast changing STRONGLY to lower-case, and consider changing both words.

Ok. I will change as suggested.

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

>Section 7.3.2: s/through the Expert Review policies/through the Expert Rev=
iew policy/

Ok. I will change as suggested.

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

Thanks!

Regards,

Christer


From christer.holmberg@ericsson.com  Wed Aug 22 05:09:47 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-proxy-feature@tools.ietf.org" <draft-ietf-sipcore-proxy-feature@tools.ietf.org>
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Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Hi Robert,

This reply addresses your comment regarding the IANA registration template/=
procedure, and the major issues.

> Summary: There are a few issues to address before progressing to IETF LC.=
 I'm putting the document into revised ID needed to address at least the IA=
NA points below.
>
> The primary issue is with the instructions to IANA on creating the new tr=
ees and the registration template used to put things in those trees.
>
> Sections 7.3.2 and 7.3.3 don't currently define the format of an entry in=
 the trees in the new subregistry. This needs to be clearly defined, and th=
e sections need to explicitly indicate that the trees are initially empty.

I suggest to add the following text to the end of sections 7.3.2 and 7.3.3:

	"The format of the [global/SIP] tree is as described below:

	Decimal      Name   Description   Reference
	---------------------------------------------------

	Decimal contains an integer value, incremented by IANA every time a new fe=
ature capability indicator is added to the table.=20

	Name contains the Feature Capability Indicator Name, provided in the regis=
tration feature capability indication registration template.

	Description contains the Abstract, provided in the registration feature ca=
pability indication registration template.

	Reference contains the Feature Capability Indicator Specification Referenc=
e, provided in the registration feature capability indication registration =
template."

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

> The template in section 8 was adapted from RFC2506, but it needs more twe=
aks and it needs to match whatever format is defined above.
>
>   * We don't need a separate list for reviewing these requests. We certai=
nly don't need a list at apps.ietf.org. The normal process for submitting a=
 request for the global tree should be submitting the template by email to =
iana@iana.org, and for the sip=20
> tree, the normal process would be submitting an Internet-Draft to the IES=
G.

I suggest the following change:

OLD:

   	"To: sip-feature-capability-indicators@apps.ietf.org
   	(feature capability indicators mailing list)
   	Subject: Registration of feature capability indicator XXXX"

NEW:

 	"Registration requests for the global tree are submitted by e-mail to ian=
a@iana.org.
	Registration requests for the sip tree requires submitting an Internet-Dra=
ft to the IESG."

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

>   * IANA is currently using a different for media feature tag registratio=
ns. It's available at <http://www.iana.org/protocols/media-feature-tag-temp=
late.txt>. If we really need to follow that registration format, please sta=
rt from this template. But do we=20
> need all this information in the registry?

I did start from the media feature tag template, and then removed stuff :)

However, I do agree that not all of the remaining stuff needs to be in the =
template. Unlike media feature tags, we do mandate a reference to a specifi=
cation for feature capability indicators, so whatever information that IANA=
 doesn't need can be put in that specification.

So, my suggestion would be that the registration template would look like t=
his:


   	"| Instructions are preceded by '|'.  All fields are mandatory.

   	Feature capability indicator name:

   	Description:

   	| The description should be no longer than 4 lines. More
   	| detailed information can be provided in the feature
   	| capability indicator specification.

   	Feature capability indicator name:

   	| The referenced specification MUST contain the information
   	| listed in Section 5.3 of XXXX (IANA: Replace XXXX with
   	| assigned RFC number of this specification.

   	Contact:

   	| Name(s) & email address(es) of person(s) to
   	| contact for further information."


The following information would be in the feature capability indication spe=
cification:


5.3.  Feature Capability Indicator Specification Requirements

5.3.1.  General

   A feature capability indicator specification MUST address the issues
   defined in the following subsections, or document why an issue is not
   applicable for the specific feature capability indicator.  A
   reference to the specification MUST be provided when the feature
   capability indicator is registered with IANA (see Section 8).

   It is bad practice for feature capability indicator specifications to
   repeat procedures (e.g. general procedures on the usage of the
   Feature-Caps header field and feature capability indicators) defined
   in this specification, unless needed for clarification or emphasis
   purpose.  A feature capability indicator specification MUST NOT
   modify the Feature-Caps header field rules and semantics defined in
   Section 4.

   A feature capability indicator specification MUST NOT weaken any
   behavior designated with "SHOULD" or "MUST" in this specification.
   However, a specification MAY strengthen "SHOULD", "MAY", or
   "RECOMMENDED" requirements to "MUST" strength if features and
   capabilities associated with the SIP feature capability indicator

5.3.2.  Overall Description

   The feature capability indicator specification MUST contain an
   overall description of the feature capability indicator: how it is
   used to indicate support of a feature, a description of the feature
   associated with the SIP feature cap, a description of any additional
   information (conveyed using one or more feature capability indicator
   values) that can be conveyed together with the feature capability
   indicator, and a description of how the associated feature may be
   exercised/invoked.

5.3.3.  Feature Capability Indicator Values

   A feature capability indicator can have an associated value, or a
   list of values.

   The feature capability indicator specification MUST define the syntax
   and semantics of any value defined for the feature capability
   indicator, including possible restrictions related to the usage of a
   specific value.  The feature capability indicator specification MUST=20
   define the value(s) in accordance with the ABNF defined in Section
   6.3.2. The feature capability indicator specification MUST define
   the default value, if applicable.

   If not values are defined for the feature capability indicator, it MUST
   be indicated in the feature capability indicator specification.

   A feature capability indicator value is only applicable for the
   feature capability indicator for which it has been defined.  For
   other feature capability indicators, the value has to be defined
   explicitly, even if the semantics are identical.

   It is strongly RECOMMENDED to not re-use a value that already has
   been defined for another feature capability indicator, unless the
   semantics of the values are the same.

5.3.4.  Usage Restrictions

   If there are restrictions on how SIP entities can insert a feature=20
   capability indicator, the feature capability indicator specification MUS=
T
   document such restrictions.

   There might be restrictions related to whether entities are allowed
   to insert a feature capability indicator in registration related
   messages, standalone transaction messages, or dialog related
   messages, whether entities are allowed to insert a feature capability
   indicator in requests or responses, whether entities also need to
   support other features and capabilities in order to insert a feature
   capability indicator, and whether entities are allowed to indicate
   support of a feature in conjunction with another feature.

5.3.5.  Interoperability Considerations

  If there are specific interoperability considerations that apply to the=20
  feature capability indicator, the feature capability indicator specificat=
ion
  MUST document such considerations.

5.3.6.  Security Considerations
=09
  If there are specific security considerations that apply to the feature
  capability indicator, the feature capability indicator specification
  MUST document such considerations.

5.3.7.  Examples

   It is RECOMMENDED that the feature capability indicator specification
   provide demonstrative message flow diagrams, paired with complete
   messages and message descriptions.

   Note that example message flows are by definition informative, and do
   not replace normative text.

5.3.8.  Other Information

   If there is additional information about the feature capability indicato=
r,
   it is RECOMMENDED to describe such information. It can include e.g.
   names of related feature capability indicators."
  =20


Other major issues:
-----------------------------------------

> 1) Should the expert reviewer be explicitly required to consider security=
 issues as part of that review? (see section 4.6 of <https://datatracker.ie=
tf.org/doc/draft-ietf-appsawg-media-type-regs/>
> for example).

Per my suggestion above, about moving 'Security considerations' from the te=
mplate into the feature capability indicator specification, I guess the exp=
ert reviewer would also look at the Security Considerations. I don't think =
we need to explicitly say anything.

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

> 2) The security considerations section claims that the aspects of the con=
siderations in rfc3840 that apply to end user devices don't apply here sinc=
e these are not typically used to convey information about end user devices=
. "Not typically" is weak. Can this > sentence be made more specific about =
what concerns do and don't apply? The concerns in 3840 about influencing ap=
plication behavior, and in 2506 about providing information that can be use=
d to exploit implementation weakeness at the elements that > do provide the=
se tags should be reflected here.

I suggest some additional security considerations text:

"9.  Security Considerations

   The security issues for feature capability indicators are similar to
   the ones defined in RFC 3840 for media feature tags.  Media
   feature tags can reveal information about end-users and end-user=20
   equipment, which can be used for industrial espionage. The knowledge
   about end-user equipment capabilities can also be used to influence
   application behavior. As feature capability indicators are not intended =
=20
   to convey  capability information of end-user devices, such end-user=20
   security aspects of   RFC 3840 do not apply to feature capability indica=
tors.

   In addition, the RFC 3840 security issue regarding an attacker using
   the SIP caller preferences extension [RFC3841] in order to affect
   routing decisions does not apply, as the mechanism is not defined to
   be used with feature capability indicators.

   Feature capability indicators can, however, provide capability and=20
   characteristics information about the SIP entity, some of which might
   be sensitive.  The Feature- Caps header field does not convey address=20
   information about SIP entities.  However, individual feature capability=
=20
   indicators might provide address information as feature capability indic=
ator
   values. Therefore, mechanisms for guaranteeing confidentiality and
   authenticity SHOULD be provided."

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

>3) Consider clarifying the behavior of SUBSCRIBE NOTIFY dialogs in section=
 4.3.2. Only one of the 200 OKs to a forked SUBSCRIBE=20
>will get back to the UA that sent the subscribe. The information that crea=
tes the other dialogs for it must appear in the immediate=20
>NOTIFYs. Is there a requirement for the end accepting a subscription to us=
e the same feature capabilities in the 200 OK to the subscription=20
>and in the immediate NOTIFY?

I suggest the following new note to section 4.3.2.

   	"NOTE: As it cannot be guaranteed that 2xx responses associated with
   	SIP SUBSCRIBE requests will reach the UAC, due to forking of the reques=
t,
	entities need to indicate supported features and capabilities in the SIP N=
OTIFY=20
	request that will be sent for each of the created subscription dialogs."

I don't think we need to mandate the capabilities in the 200 OK (if inserte=
d in the first place) need to be identical to the NOTIFY. As the NOTIFY is =
a target refresh request, it is allowed to indicate different capabilities.

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

> 4) Similarly, consider clarifying whether UPDATES nested within an INVITE=
 constrained to keep the same feature capability indicators as in the INVIT=
E?

Again, as UPDATEs are target refresh requests (even when nested within an I=
NVITE), they can indicate a new set of supported capabilities.

Note that the indication of supported features is not "offer/answer" - the =
support is indicated independently in each direction.

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

Thanks!

Regards,

Christer

From christer.holmberg@ericsson.com  Wed Aug 22 05:14:50 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Robert Sparks <rjsparks@nostrum.com>
Date: Wed, 22 Aug 2012 14:14:47 +0200
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05
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Hi Paul,

>> 4) Similarly, consider clarifying whether UPDATES nested within an=20
>> INVITE constrained to keep the same feature capability indicators as=20
>> in the INVITE?
>
> I presume you mean an UPDATE to a before the final response to the dialog=
 establishing INVITE is received.
>
> It seems that UPDATES after the dialog is established aren't permitted to=
 carry the header.

Where do you read that?

The text says that the header field can be carried in target refresh reques=
ts.

>> Minor Issues:
>>
>> Section 4.3.4: A SIP Stand-Alone Transaction is by definition a=20
>> non-INVITE transaction. RFC4320 made 18x to non-INVITES a MUST NOT. I=20
>> suggest s/18x or 2xx/2xx/ in this section.
>>
>> The two places where you talk about overriding registration policy in=20
>> section 5.2 isn't quite right. We should be specific that we're only=20
>> talking about SIP header field parameters. In the first case, we're=20
>> using IETF Consensus (this document) to delegate parameters starting=20
>> with a 'g.' for _this_ header field to a policy of Expert Review. In=20
>> the second case, there's no overriding at all - registrations use IETF=20
>> Consensus there.  Here's a stab at something more accurate. Maybe=20
>> others can come up with something better. I suggest replacing the=20
>> first instance with "Note that the general policy for registering SIP=20
>> header field parameters is 'IETF Consensus' as specified in [RFC3968].=20
>> This document delegates the registration of parameters to the=20
>> Feature-Caps header field starting with 'g.' to a policy of 'Expert=20
>> Review'", and deleting the second instance.
>
> You and I already talked about this, but I realize now that we have to wo=
rk on the wording more. The issue is that from a sip=20
> header field perspective the parameter includes the "+", and its paramete=
rs that begin with "+" that we are reserving in the header field parameter =
registry.
>
> But the "+" isn't part of the name of the feature capability indicator.
>
> So we are delegating parameters that start with "+g." to the expert revie=
w procedure.
>
> And while things in the sip tree (things beginning with "+sip.", while st=
ill requiring IETF consensus, will still go in the sip tree, not as individ=
uals in the sip header fields parameter registry. But maybe we don't need t=
o "note" this part.

I am not sure I understand. The reviewer will check the feature capability =
indicator. The "+" is a SIP coding issue. The draft even says that the lead=
ing "+" is not part of the registration.


>> The NOTE in section 6.2.1 is not useful unless you know the history of=20
>> that led to the syntax in 3840 and 3841. I suggest replacing it with
>> 'NOTE: The "*" value is present in order to follow the guidelines for=20
>> syntac in [RFC4485] and to maintain a consistent format with [RFC3840]=20
>> and [RFC3841].'
>>
>> Nits:
>>
>> There is a mix of "xxx" "XXX" and "XXXX" to indicate to the RFC Editor=20
>> that the number assigned to this RFC should appear here. Please choose=20
>> one format and make the instructions to the RFC Editor for making the=20
>> substitution clear.
>
> I recently discovered that xml2rfc has '&rfc.number;' to get this value, =
though I haven't tried it yet. Apparently it expands to XXXX for drafts.

I'll take a look :)

Regards,

Christer

From brett@broadsoft.com  Wed Aug 22 05:19:34 2012
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From: Brett Tate <brett@broadsoft.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-proxy-feature@tools.ietf.org" <draft-ietf-sipcore-proxy-feature@tools.ietf.org>
Date: Wed, 22 Aug 2012 05:19:30 -0700
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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> >3) Consider clarifying the behavior of SUBSCRIBE NOTIFY dialogs in
> section 4.3.2. Only one of the 200 OKs to a forked SUBSCRIBE
> >will get back to the UA that sent the subscribe. The information that
> creates the other dialogs for it must appear in the immediate
> >NOTIFYs. Is there a requirement for the end accepting a subscription
> to use the same feature capabilities in the 200 OK to the subscription
> >and in the immediate NOTIFY?
>=20
> I suggest the following new note to section 4.3.2.
>=20
> "NOTE: As it cannot be guaranteed that 2xx responses associated with
>  SIP SUBSCRIBE requests will reach the UAC, due to forking of the request=
,
>  entities need to indicate supported features and capabilities in=20
>  the SIP NOTIFY request that will be sent for each of the created=20
>  subscription dialogs."
>=20
> I don't think we need to mandate the capabilities in the 200 OK=20
> (if inserted in the first place) need to be identical to the=20
> NOTIFY. As the NOTIFY is a target refresh request, it is allowed=20
> to indicate different capabilities.

Based upon the current wording of draft-ietf-sipcore-proxy-feature-05 secti=
on 4.3.2, the feature-caps within the initial SUBSCRIBE's 2xx (or deprecate=
d 18x) appears irrelevant from subscriber's perspective since the response =
does not appear to create a dialog.

RFC 6665 is a little ambiguous concerning if SUBSCRIBE's 2xx actually creat=
es a dialog.  From notifier perspective, it looks like sending 2xx does ind=
icate that the dialog usage was created.  However from a subscriber perspec=
tive, it looks like receiving the 2xx no longer creates a dialog.

RFC 6665 section 4.2.1.1

"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 usage, and returns a 200 (OK) response."

RFC 6665 section 4.4.1:

"Dialogs usages are created upon completion of a NOTIFY transaction
 for a new subscription, unless the NOTIFY request contains a
 "Subscription-State" of "terminated."

 Because the dialog usage is established by the NOTIFY request, the
 route set at the subscriber is taken from the NOTIFY request itself,
 as opposed to the route set present in the 200-class response to the
 SUBSCRIBE request."

From christer.holmberg@ericsson.com  Wed Aug 22 05:21:28 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-proxy-feature@tools.ietf.org" <draft-ietf-sipcore-proxy-feature@tools.ietf.org>
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Hi Brett,

Please see the note text I suggested in my reply to Robert.

Regards,

Christer

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Brett Tate
Sent: 22. elokuuta 2012 15:20
To: sipcore@ietf.org; draft-ietf-sipcore-proxy-feature@tools.ietf.org
Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Rob=
ert's comments - IANA issues and major issues

> >3) Consider clarifying the behavior of SUBSCRIBE NOTIFY dialogs in
> section 4.3.2. Only one of the 200 OKs to a forked SUBSCRIBE
> >will get back to the UA that sent the subscribe. The information that
> creates the other dialogs for it must appear in the immediate
> >NOTIFYs. Is there a requirement for the end accepting a subscription
> to use the same feature capabilities in the 200 OK to the subscription
> >and in the immediate NOTIFY?
>=20
> I suggest the following new note to section 4.3.2.
>=20
> "NOTE: As it cannot be guaranteed that 2xx responses associated with =20
> SIP SUBSCRIBE requests will reach the UAC, due to forking of the=20
> request,  entities need to indicate supported features and=20
> capabilities in  the SIP NOTIFY request that will be sent for each of=20
> the created  subscription dialogs."
>=20
> I don't think we need to mandate the capabilities in the 200 OK (if=20
> inserted in the first place) need to be identical to the NOTIFY. As=20
> the NOTIFY is a target refresh request, it is allowed to indicate=20
> different capabilities.

Based upon the current wording of draft-ietf-sipcore-proxy-feature-05 secti=
on 4.3.2, the feature-caps within the initial SUBSCRIBE's 2xx (or deprecate=
d 18x) appears irrelevant from subscriber's perspective since the response =
does not appear to create a dialog.

RFC 6665 is a little ambiguous concerning if SUBSCRIBE's 2xx actually creat=
es a dialog.  From notifier perspective, it looks like sending 2xx does ind=
icate that the dialog usage was created.  However from a subscriber perspec=
tive, it looks like receiving the 2xx no longer creates a dialog.

RFC 6665 section 4.2.1.1

"Once the notifier determines that it has enough information to create  the=
 subscription (i.e., it understands the event package, the  subscription pe=
rtains to a known resource, and there are no other  barriers to creating th=
e subscription), it creates the subscription  and a dialog usage, and retur=
ns a 200 (OK) response."

RFC 6665 section 4.4.1:

"Dialogs usages are created upon completion of a NOTIFY transaction  for a =
new subscription, unless the NOTIFY request contains a  "Subscription-State=
" of "terminated."

 Because the dialog usage is established by the NOTIFY request, the  route =
set at the subscriber is taken from the NOTIFY request itself,  as opposed =
to the route set present in the 200-class response to the  SUBSCRIBE reques=
t."
_______________________________________________
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From brett@broadsoft.com  Wed Aug 22 05:45:47 2012
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From: Brett Tate <brett@broadsoft.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-proxy-feature@tools.ietf.org" <draft-ietf-sipcore-proxy-feature@tools.ietf.org>
Date: Wed, 22 Aug 2012 05:45:44 -0700
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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> Please see the note text I suggested in my reply to Robert.
>=20
> > I suggest the following new note to section 4.3.2.
> >
> > "NOTE: As it cannot be guaranteed that 2xx responses=20
> > associated with SIP SUBSCRIBE requests will reach the=20
> > UAC, due to forking of the request,  entities need to=20
> > indicate supported features and capabilities in the=20
> > SIP NOTIFY request that will be sent for each of
> > the created subscription dialogs."

Good enough for me.  However, my understanding is that the SUBSCRIBE 2xx co=
mment is relevant to RFC 3265 compliant subscribers instead of RFC 6665 com=
pliant subscribers.


From pkyzivat@alum.mit.edu  Wed Aug 22 10:18:49 2012
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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A couple of comments now. Maybe more when I have more time to think 
about it:

On 8/22/12 8:09 AM, Christer Holmberg wrote:
> Hi Robert,
>
> This reply addresses your comment regarding the IANA registration template/procedure, and the major issues.
>
>> Summary: There are a few issues to address before progressing to IETF LC. I'm putting the document into revised ID needed to address at least the IANA points below.
>>
>> The primary issue is with the instructions to IANA on creating the new trees and the registration template used to put things in those trees.
>>
>> Sections 7.3.2 and 7.3.3 don't currently define the format of an entry in the trees in the new subregistry. This needs to be clearly defined, and the sections need to explicitly indicate that the trees are initially empty.
>
> I suggest to add the following text to the end of sections 7.3.2 and 7.3.3:
>
> 	"The format of the [global/SIP] tree is as described below:
>
> 	Decimal      Name   Description   Reference
> 	---------------------------------------------------
>
> 	Decimal contains an integer value, incremented by IANA every time a new feature capability indicator is added to the table.

Why? What value does this integer have?

>> 4) Similarly, consider clarifying whether UPDATES nested within an INVITE constrained to keep the same feature capability indicators as in the INVITE?
>
> Again, as UPDATEs are target refresh requests (even when nested within an INVITE), they can indicate a new set of supported capabilities.

Consider:

INVITE
1xx INVITE
UPDATE
200 UPDATE
200 INVITE

(Regardless of the direction of the UPDATE)

Do the P-F values in the 200 INVITE override those from the UPDATE or 
its response?

In theory the 200 UPDATE and 200 INVITE can be received out of order. 
And proxies that are including P-F may not be stateful about this, so 
that they can only indicate their state at the time the message goes by.

I don't see what could be done about it. In most cases it won't matter 
what the order is.

	Thanks,
	Paul




From pkyzivat@alum.mit.edu  Wed Aug 22 10:50:19 2012
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05
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On 8/22/12 8:14 AM, Christer Holmberg wrote:
> Hi Paul,
>
>>> 4) Similarly, consider clarifying whether UPDATES nested within an
>>> INVITE constrained to keep the same feature capability indicators as
>>> in the INVITE?
>>
>> I presume you mean an UPDATE to a before the final response to the dialog establishing INVITE is received.
>>
>> It seems that UPDATES after the dialog is established aren't permitted to carry the header.
>
> Where do you read that?

Oops. Sorry. At the time I wrote that I missed the part about being 
allowed in target refresh requests.

> The text says that the header field can be carried in target refresh requests.
>
>>> Minor Issues:
>>>
>>> Section 4.3.4: A SIP Stand-Alone Transaction is by definition a
>>> non-INVITE transaction. RFC4320 made 18x to non-INVITES a MUST NOT. I
>>> suggest s/18x or 2xx/2xx/ in this section.
>>>
>>> The two places where you talk about overriding registration policy in
>>> section 5.2 isn't quite right. We should be specific that we're only
>>> talking about SIP header field parameters. In the first case, we're
>>> using IETF Consensus (this document) to delegate parameters starting
>>> with a 'g.' for _this_ header field to a policy of Expert Review. In
>>> the second case, there's no overriding at all - registrations use IETF
>>> Consensus there.  Here's a stab at something more accurate. Maybe
>>> others can come up with something better. I suggest replacing the
>>> first instance with "Note that the general policy for registering SIP
>>> header field parameters is 'IETF Consensus' as specified in [RFC3968].
>>> This document delegates the registration of parameters to the
>>> Feature-Caps header field starting with 'g.' to a policy of 'Expert
>>> Review'", and deleting the second instance.
>>
>> You and I already talked about this, but I realize now that we have to work on the wording more. The issue is that from a sip
>> header field perspective the parameter includes the "+", and its parameters that begin with "+" that we are reserving in the header field parameter registry.
>>
>> But the "+" isn't part of the name of the feature capability indicator.
>>
>> So we are delegating parameters that start with "+g." to the expert review procedure.
>>
>> And while things in the sip tree (things beginning with "+sip.", while still requiring IETF consensus, will still go in the sip tree, not as individuals in the sip header fields parameter registry. But maybe we don't need to "note" this part.
>
> I am not sure I understand. The reviewer will check the feature capability indicator. The "+" is a SIP coding issue. The draft even says that the leading "+" is not part of the registration.

I brought it up because when you look at the header as a whole, the 
*parameter* does include the "+". That is pertinent to the discussion of 
which parameters are being delegated. What do you think of:

"Note that the general policy for registering SIP header field 
parameters is 'IETF Consensus' as specified in [RFC3968]. This document, 
itself subject to 'IETF Consensus' defines a new IANA procedure for 
registration of all parameters beginning with "+" of the "Proxy-Feature" 
sip header field. The registration of those parameters beginning with 
"+" followed by "g." are governed by 'Expert Review' policy, while 
others are 'IETF Consensus'.

	Thanks,
	Paul



From christer.holmberg@ericsson.com  Wed Aug 22 11:35:55 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 22 Aug 2012 20:35:51 +0200
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Hi,

>>> Summary: There are a few issues to address before progressing to IETF L=
C. I'm putting the document into revised ID needed to address at least the =
IANA points below.
>>>
>>> The primary issue is with the instructions to IANA on creating the new =
trees and the registration template used to put things in those trees.
>>>
>>> Sections 7.3.2 and 7.3.3 don't currently define the format of an entry =
in the trees in the new subregistry. This needs to be clearly defined, and =
the sections need to explicitly indicate that the trees are initially empty=
.
>>
>> I suggest to add the following text to the end of sections 7.3.2 and 7.3=
.3:
>>
>>       "The format of the [global/SIP] tree is as described below:
>>
>>       Decimal      Name   Description   Reference
>>       ---------------------------------------------------
>>
>>       Decimal contains an integer value, incremented by IANA every time =
a new feature capability indicator is added to the table.
>
> Why? What value does this integer have?

It's just a "row number". Most media feature tag tables seem to have it, so=
 that's why I added it...

I didn't even check whether it's part of the associated registration templa=
tes, or whether IANA adds it by default.

>>> 4) Similarly, consider clarifying whether UPDATES nested within an INVI=
TE constrained to keep the same feature capability indicators as in the INV=
ITE?
>>
>> Again, as UPDATEs are target refresh requests (even when nested within a=
n INVITE), they can indicate a new set of supported capabilities.
>
>Consider:
>
>INVITE
>1xx INVITE
>UPDATE
>200 UPDATE
>200 INVITE
>
>(Regardless of the direction of the UPDATE)
>
>Do the P-F values in the 200 INVITE override those from the UPDATE or
>its response?
>
>In theory the 200 UPDATE and 200 INVITE can be received out of order.
>And proxies that are including P-F may not be stateful about this, so
>that they can only indicate their state at the time the message goes by.
>
>I don't see what could be done about it. In most cases it won't matter
>what the order is.

I agree, because I think the indicated features will be the same in these m=
essages.

Regards,

Christer=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Wed, 22 Aug 2012 20:36:14 +0200
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05
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Hi,

>>>> Minor Issues:
>>>>
>>>> Section 4.3.4: A SIP Stand-Alone Transaction is by definition a
>>>> non-INVITE transaction. RFC4320 made 18x to non-INVITES a MUST NOT. I
>>>> suggest s/18x or 2xx/2xx/ in this section.
>>>>
>>>> The two places where you talk about overriding registration policy in
>>>> section 5.2 isn't quite right. We should be specific that we're only
>>>> talking about SIP header field parameters. In the first case, we're
>>>> using IETF Consensus (this document) to delegate parameters starting
>>>> with a 'g.' for _this_ header field to a policy of Expert Review. In
>>>> the second case, there's no overriding at all - registrations use IETF
>>>> Consensus there.  Here's a stab at something more accurate. Maybe
>>>> others can come up with something better. I suggest replacing the
>>>> first instance with "Note that the general policy for registering SIP
>>>> header field parameters is 'IETF Consensus' as specified in [RFC3968].
>>>> This document delegates the registration of parameters to the
>>>> Feature-Caps header field starting with 'g.' to a policy of 'Expert
>>>> Review'", and deleting the second instance.
>>>
>>> You and I already talked about this, but I realize now that we have to =
work on the wording more. The issue is that from a sip
>>> header field perspective the parameter includes the "+", and its parame=
ters that begin with "+" that we are reserving in the header field paramete=
r registry.
>>>
>>> But the "+" isn't part of the name of the feature capability indicator.
>>>
>>> So we are delegating parameters that start with "+g." to the expert rev=
iew procedure.
>>>
>>> And while things in the sip tree (things beginning with "+sip.", while =
still requiring IETF consensus, will still go in the sip tree, not as indiv=
iduals in the sip header fields parameter registry. But maybe we don't need=
 to "note" this part.
>>
>> I am not sure I understand. The reviewer will check the feature capabili=
ty indicator. The "+" is a SIP coding issue. The draft even says that the l=
eading "+" is not part of the registration.
>
> I brought it up because when you look at the header as a whole, the
> *parameter* does include the "+". That is pertinent to the discussion of
> which parameters are being delegated. What do you think of:
>
> "Note that the general policy for registering SIP header field
> parameters is 'IETF Consensus' as specified in [RFC3968]. This document,
> itself subject to 'IETF Consensus' defines a new IANA procedure for
> registration of all parameters beginning with "+" of the "Proxy-Feature"
> sip header field. The registration of those parameters beginning with
> "+" followed by "g." are governed by 'Expert Review' policy, while
> others are 'IETF Consensus'.

Just to make sure I understand, where do you suggest to add this note, and =
what text would it replace? :)

Regards,

Christer=

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To: "Peterson, Jon" <jon.peterson@neustar.biz>, James Polk <jmpolk@cisco.com>,  Paul Kyzivat <pkyzivat@alum.mit.edu>, "Brian Rosen" <br@brianrosen.net>
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Subject: Re: [sipcore] Geolocation Header Question
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At 02:15 AM 8/22/2012, Peterson, Jon wrote:

>Um... why isn't sip:<tel>@domain valid? I see sip-uri as valid in 
>the BNF, and that implies tel in the user portion.

I'm basing my response on this from section 4.6

    If a location URI is included in a SIP request, the sending user
    agent MUST also include a Supported header field indicating which
    location profiles it supports.  Two option tags for location profiles
    are defined by this document: "geolocation-sip" and "geolocation-
    http".  Future specifications MAY define further location profiles
    per the IANA policy described in Section 8.3.

    The "geolocation-sip" option tag signals support for acquiring
    location information via the presence event package of SIP [RFC3856].
    A location recipient who supports this option can send a SUBSCRIBE
    request and parse a resulting NOTIFY containing a PIDF-LO object.
    The URI schemes supported by this option include "sip", "sips", and
    "pres".

I haven't read that this

    Geolocation: <tel-URI>; user=phone

will be subscribed to and expect to receive a PIDF-LO, which the RFC 
says earlier is the only location value format that's specified as acceptable.

I may have missed something in the discussion, but that's why I don't 
think the <tel> scenario is depicting what RFC 6442 is 
specifying.  Additionally, and I just may not know this - as I'm not 
an expert on <tel-URIs>, but I haven't ever heard of someone wanting 
to SUBSCRIBE to a <tel>. But again, I could be underinformed on this one.

James


>Jon Peterson
>NeuStar, Inc.
>________________________________________
>From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf 
>Of James Polk [jmpolk@cisco.com]
>Sent: Tuesday, August 21, 2012 5:05 PM
>To: Paul Kyzivat; Brian Rosen
>Cc: sipcore@ietf.org
>Subject: Re: [sipcore] Geolocation Header Question
>
>At 04:11 PM 8/14/2012, Paul Kyzivat wrote:
> >On 8/14/12 5:04 PM, Brian Rosen wrote:
> >>Yes, agree, but probably not relevant to the discussion.
> >>
> >>I went back and looked at the original email, and decided I either don't
> >>understand something, or there is some larger problem.
> >>
> >>Between what two elements is this Geolocation header running?
> >>
> >>UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?
> >>
> >>Regardless, it seems to be misusing the header, because it's not a sip or
> >>http LbyR.
> >>Better to get a real scheme somehow that does the right thing, use a
> >>parameter in another header, or create a new header.  Depends on what this
> >>actually represents.  It sounds like it really is LbyR, and you want to
> >>communicate a key.<scheme>:<tel>@domain is appropriate.  The issue is
> >>what is the right scheme?  It's not sip - it's not sip presence.  It's not
> >>HELD identity I take it.
> >
> >As I already stated, I think sip: is ok if this special lookup can
> >be viewed as an *optimization* over doing the presence lookup, and
> >doing it the presence way still works.
> >
> >Otherwise I agree with Brian.
>
><tel:> isn't an allowed URI in RFC 6442, neither is the use=phone
>tag, so IMO that makes this offering a no go as far as 'being in line
>with an IETF standards track RFC' (to be formal about it).
>
>James
>
>
> >         Thanks,
> >         Paul
> >
> >>Brian
> >>
> >>>I already commented to the original question.
> >>>Here I'm just commenting on the comments.
> >>>
> >>>On 8/14/12 1:55 PM, Brian Rosen wrote:
> >>>>I don't think so.
> >>>>
> >>>>A geolocation header with a sip uri has to lead to a presence server
> >>>>that
> >>>>can respond to a presence subscription which would return a PIDF with
> >>>>location.
> >>>>
> >>>>Routing a tel uri (which user=phone is supposed to be interchangeable
> >>>>with) to a presence service usually makes no sense.  I don't think you
> >>>>can
> >>>>send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If
> >>>>you
> >>>>can't, user=phone is not okay.
> >>>
> >>>I disagree that a tel uri is *equivalent* to a sip URI with user=phone!
> >>>
> >>>A sip URI with user=phone is supposed to be routed like any other sip
> >>>URI until it reaches a server responsible for the domain of the URI.
> >>>*Then* the user part is to be interpreted as representing a phone number.
> >>>
> >>>Conversely, a tel URI has no domain. (And no user=phone.) The semantics
> >>>of telephone numbers are public knowledge. So any server can decide
> >>>where to route it next, based on the number and whatever other context
> >>>it has.
> >>>
> >>>IMO the distinction is important. If you want to enable anybody to
> >>>decide how to reach the phone number (including going through a PSTN
> >>>GW), then use a tel URI. If you want the call handled via SIP at least
> >>>until it reaches a particular domain, then use a sip URI.
> >>>
> >>>         Thanks,
> >>>         Paul
> >>>
> >>>>It's perfectly reasonable to make the uri
> >>>>sip:+358222222222@presence.example.com
> >>>>that is, a real domain, and no user=phone
> >>>>
> >>>>Brian
> >>>>>
> >>>>>You could do this:
> >>>>>Geolocation:<sip:+358222222222@atis.invalid;user=phone>
> >>>>>(although why ATIS would be defining how calls from Finland work is
> >>>>>beyond
> >>>>>me ;)
> >>>>>
> >>>>>But I don't understand why they need it in a Geoloc header to begin
> >>>>>with.
> >>>>>I assumed that header was for devices/networks that use its mechanisms
> >>>>>(ie, HELD, SUB, pidf).  They're not going to do that, so why use that
> >>>>>header?
> >>>>>
> >>>>>You said they're doing the location lookup on the PSTN side -
> >>>>>presumably
> >>>>>then the location is being looked up based on the calling party info,
> >>>>>which should be in another existing header.  Can't they use one of them
> >>>>>as
> >>>>>the query key for the lookup? (we have plenty to choose from for that
> >>>>>already :)
> >>>>>
> >>>>>-hadriel
> >>>>>
> >>>>>
> >>>>>On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
> >>>>>
> >>>>>>I agree but the tel URI is not one of the supported location URIs in
> >>>>>>RFC
> >>>>>>6442.
> >>>>>>Hence, the guys just use a different way to convey the same
> >>>>>>information.
> >>>>>>
> >>>>>>On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
> >>>>>>
> >>>>>>>>From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
> >>>>>>>>
> >>>>>>>>In order to convey that phone number for a lookup they are thinking
> >>>>>>>>about using the Geolocation header in the following form:
> >>>>>>>>Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
> >>>>>>>
> >>>>>>>Naively, I'd say that since the goal is to carry an E.164 number, a
> >>>>>>>tel: URI would be the ideal choice.
> >>>>>>>
> >>>>>>>Dale
> >>>>>>
> >>>>>>_______________________________________________
> >>>>>>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
> >>>
> >>>_______________________________________________
> >>>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 pkyzivat@alum.mit.edu  Wed Aug 22 12:37:59 2012
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Subject: Re: [sipcore] Geolocation Header Question
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On 8/22/12 2:54 PM, James Polk wrote:
> At 02:15 AM 8/22/2012, Peterson, Jon wrote:
>
>> Um... why isn't sip:<tel>@domain valid? I see sip-uri as valid in the
>> BNF, and that implies tel in the user portion.
>
> I'm basing my response on this from section 4.6
>
>     If a location URI is included in a SIP request, the sending user
>     agent MUST also include a Supported header field indicating which
>     location profiles it supports.  Two option tags for location profiles
>     are defined by this document: "geolocation-sip" and "geolocation-
>     http".  Future specifications MAY define further location profiles
>     per the IANA policy described in Section 8.3.
>
>     The "geolocation-sip" option tag signals support for acquiring
>     location information via the presence event package of SIP [RFC3856].
>     A location recipient who supports this option can send a SUBSCRIBE
>     request and parse a resulting NOTIFY containing a PIDF-LO object.
>     The URI schemes supported by this option include "sip", "sips", and
>     "pres".
>
> I haven't read that this
>
>     Geolocation: <tel-URI>; user=phone

This discussion is mixing up two related but different things:
- a uri with a "tel:" scheme
- a sip uri with phone number syntax in the user part
   and a ";user=phone" parameter.

I guess this header doesn't all use of the tel: scheme, so that is off 
the table unless it is changed.

A sip uri with ";user=phone" is still a sip uri. The syntax you show 
above isn't right, because the parameter should be inside the <>. E.g.

    Geolocation: <sip:+12125551212@example.com;user=phone>

So *syntactically* it should be fine to do this. But then there are 
still the options to deal with. I assume geolocation-sip would apply 
here (since its hard to do a GET on a sip uri.) So that says the 
specified sip uri should be suitable for doing a subscribe for presence.

What I argued earlier is that IMO it might be ok in this case for the 
dereferencer to *try* some other proprietary mechanism, , and then fall 
back to subscription if that fails.

> will be subscribed to and expect to receive a PIDF-LO, which the RFC
> says earlier is the only location value format that's specified as
> acceptable.
>
> I may have missed something in the discussion, but that's why I don't
> think the <tel> scenario is depicting what RFC 6442 is specifying.
> Additionally, and I just may not know this - as I'm not an expert on
> <tel-URIs>, but I haven't ever heard of someone wanting to SUBSCRIBE to
> a <tel>. But again, I could be underinformed on this one.

Well, technically its ok to send a SUBSCRIBE with a tel: R-URI. That 
will be ok as long as it can be routed through the sip network. If it 
gets to a PSTN gateway its hard to say what will happen. Probably most 
gateways will have no idea what to do with a subscribe for presence.

My *impression* when I first saw this mentioned was that the intent was 
that the tel: URI would override geolocation-sip, and be an implicit 
indication to some telephony-specific mechanism. But that was just my guess.

	Thanks,
	Paul

> James
>
>
>> Jon Peterson
>> NeuStar, Inc.
>> ________________________________________
>> From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of
>> James Polk [jmpolk@cisco.com]
>> Sent: Tuesday, August 21, 2012 5:05 PM
>> To: Paul Kyzivat; Brian Rosen
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] Geolocation Header Question
>>
>> At 04:11 PM 8/14/2012, Paul Kyzivat wrote:
>> >On 8/14/12 5:04 PM, Brian Rosen wrote:
>> >>Yes, agree, but probably not relevant to the discussion.
>> >>
>> >>I went back and looked at the original email, and decided I either
>> don't
>> >>understand something, or there is some larger problem.
>> >>
>> >>Between what two elements is this Geolocation header running?
>> >>
>> >>UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?
>> >>
>> >>Regardless, it seems to be misusing the header, because it's not a
>> sip or
>> >>http LbyR.
>> >>Better to get a real scheme somehow that does the right thing, use a
>> >>parameter in another header, or create a new header.  Depends on
>> what this
>> >>actually represents.  It sounds like it really is LbyR, and you want to
>> >>communicate a key.<scheme>:<tel>@domain is appropriate.  The issue is
>> >>what is the right scheme?  It's not sip - it's not sip presence.
>> It's not
>> >>HELD identity I take it.
>> >
>> >As I already stated, I think sip: is ok if this special lookup can
>> >be viewed as an *optimization* over doing the presence lookup, and
>> >doing it the presence way still works.
>> >
>> >Otherwise I agree with Brian.
>>
>> <tel:> isn't an allowed URI in RFC 6442, neither is the use=phone
>> tag, so IMO that makes this offering a no go as far as 'being in line
>> with an IETF standards track RFC' (to be formal about it).
>>
>> James
>>
>>
>> >         Thanks,
>> >         Paul
>> >
>> >>Brian
>> >>
>> >>>I already commented to the original question.
>> >>>Here I'm just commenting on the comments.
>> >>>
>> >>>On 8/14/12 1:55 PM, Brian Rosen wrote:
>> >>>>I don't think so.
>> >>>>
>> >>>>A geolocation header with a sip uri has to lead to a presence server
>> >>>>that
>> >>>>can respond to a presence subscription which would return a PIDF with
>> >>>>location.
>> >>>>
>> >>>>Routing a tel uri (which user=phone is supposed to be interchangeable
>> >>>>with) to a presence service usually makes no sense.  I don't think
>> you
>> >>>>can
>> >>>>send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If
>> >>>>you
>> >>>>can't, user=phone is not okay.
>> >>>
>> >>>I disagree that a tel uri is *equivalent* to a sip URI with
>> user=phone!
>> >>>
>> >>>A sip URI with user=phone is supposed to be routed like any other sip
>> >>>URI until it reaches a server responsible for the domain of the URI.
>> >>>*Then* the user part is to be interpreted as representing a phone
>> number.
>> >>>
>> >>>Conversely, a tel URI has no domain. (And no user=phone.) The
>> semantics
>> >>>of telephone numbers are public knowledge. So any server can decide
>> >>>where to route it next, based on the number and whatever other context
>> >>>it has.
>> >>>
>> >>>IMO the distinction is important. If you want to enable anybody to
>> >>>decide how to reach the phone number (including going through a PSTN
>> >>>GW), then use a tel URI. If you want the call handled via SIP at least
>> >>>until it reaches a particular domain, then use a sip URI.
>> >>>
>> >>>         Thanks,
>> >>>         Paul
>> >>>
>> >>>>It's perfectly reasonable to make the uri
>> >>>>sip:+358222222222@presence.example.com
>> >>>>that is, a real domain, and no user=phone
>> >>>>
>> >>>>Brian
>> >>>>>
>> >>>>>You could do this:
>> >>>>>Geolocation:<sip:+358222222222@atis.invalid;user=phone>
>> >>>>>(although why ATIS would be defining how calls from Finland work is
>> >>>>>beyond
>> >>>>>me ;)
>> >>>>>
>> >>>>>But I don't understand why they need it in a Geoloc header to begin
>> >>>>>with.
>> >>>>>I assumed that header was for devices/networks that use its
>> mechanisms
>> >>>>>(ie, HELD, SUB, pidf).  They're not going to do that, so why use
>> that
>> >>>>>header?
>> >>>>>
>> >>>>>You said they're doing the location lookup on the PSTN side -
>> >>>>>presumably
>> >>>>>then the location is being looked up based on the calling party
>> info,
>> >>>>>which should be in another existing header.  Can't they use one
>> of them
>> >>>>>as
>> >>>>>the query key for the lookup? (we have plenty to choose from for
>> that
>> >>>>>already :)
>> >>>>>
>> >>>>>-hadriel
>> >>>>>
>> >>>>>
>> >>>>>On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>> >>>>>
>> >>>>>>I agree but the tel URI is not one of the supported location
>> URIs in
>> >>>>>>RFC
>> >>>>>>6442.
>> >>>>>>Hence, the guys just use a different way to convey the same
>> >>>>>>information.
>> >>>>>>
>> >>>>>>On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>> >>>>>>
>> >>>>>>>>From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>> >>>>>>>>
>> >>>>>>>>In order to convey that phone number for a lookup they are
>> thinking
>> >>>>>>>>about using the Geolocation header in the following form:
>> >>>>>>>>Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>> >>>>>>>
>> >>>>>>>Naively, I'd say that since the goal is to carry an E.164
>> number, a
>> >>>>>>>tel: URI would be the ideal choice.
>> >>>>>>>
>> >>>>>>>Dale
>> >>>>>>
>> >>>>>>_______________________________________________
>> >>>>>>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
>> >>>
>> >>>_______________________________________________
>> >>>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
>
>


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Subject: Re: [sipcore] Geolocation Header Question
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At 02:38 PM 8/22/2012, Paul Kyzivat wrote:
>On 8/22/12 2:54 PM, James Polk wrote:
>>At 02:15 AM 8/22/2012, Peterson, Jon wrote:
>>
>>>Um... why isn't sip:<tel>@domain valid? I see sip-uri as valid in the
>>>BNF, and that implies tel in the user portion.
>>
>>I'm basing my response on this from section 4.6
>>
>>     If a location URI is included in a SIP request, the sending user
>>     agent MUST also include a Supported header field indicating which
>>     location profiles it supports.  Two option tags for location profiles
>>     are defined by this document: "geolocation-sip" and "geolocation-
>>     http".  Future specifications MAY define further location profiles
>>     per the IANA policy described in Section 8.3.
>>
>>     The "geolocation-sip" option tag signals support for acquiring
>>     location information via the presence event package of SIP [RFC3856].
>>     A location recipient who supports this option can send a SUBSCRIBE
>>     request and parse a resulting NOTIFY containing a PIDF-LO object.
>>     The URI schemes supported by this option include "sip", "sips", and
>>     "pres".
>>
>>I haven't read that this
>>
>>     Geolocation: <tel-URI>; user=phone
>
>This discussion is mixing up two related but different things:
>- a uri with a "tel:" scheme
>- a sip uri with phone number syntax in the user part
>   and a ";user=phone" parameter.
>
>I guess this header doesn't all use of the tel: scheme, so that is 
>off the table unless it is changed.
>
>A sip uri with ";user=phone" is still a sip uri. The syntax you show 
>above isn't right, because the parameter should be inside the <>

sorry, I was writing quickly

>. E.g.
>
>    Geolocation: <sip:+12125551212@example.com;user=phone>
>
>So *syntactically* it should be fine to do this. But then there are 
>still the options to deal with. I assume geolocation-sip would apply 
>here (since its hard to do a GET on a sip uri.) So that says the 
>specified sip uri should be suitable for doing a subscribe for presence.
>
>What I argued earlier is that IMO it might be ok in this case for 
>the dereferencer to *try* some other proprietary mechanism, , and 
>then fall back to subscription if that fails.
>
>>will be subscribed to and expect to receive a PIDF-LO, which the RFC
>>says earlier is the only location value format that's specified as
>>acceptable.
>>
>>I may have missed something in the discussion, but that's why I don't
>>think the <tel> scenario is depicting what RFC 6442 is specifying.
>>Additionally, and I just may not know this - as I'm not an expert on
>><tel-URIs>, but I haven't ever heard of someone wanting to SUBSCRIBE to
>>a <tel>. But again, I could be underinformed on this one.
>
>Well, technically its ok to send a SUBSCRIBE with a tel: R-URI. That 
>will be ok as long as it can be routed through the sip network. If 
>it gets to a PSTN gateway its hard to say what will happen. Probably 
>most gateways will have no idea what to do with a subscribe for presence.

That's my impression of this scenario also, unless the something 
within the <tel:> or <sip:> URI *doesn't* point that SUBSCRIBE 
request at the gateway, but rather to another presence server that 
can give out the PIDF-LO for that phone number. I'm just guessing 
here too. But at that point, why not use a <pres:> URI that points to 
the location record contained on that server?

James


>My *impression* when I first saw this mentioned was that the intent 
>was that the tel: URI would override geolocation-sip, and be an 
>implicit indication to some telephony-specific mechanism. But that 
>was just my guess.
>
>         Thanks,
>         Paul
>
>>James
>>
>>
>>>Jon Peterson
>>>NeuStar, Inc.
>>>________________________________________
>>>From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of
>>>James Polk [jmpolk@cisco.com]
>>>Sent: Tuesday, August 21, 2012 5:05 PM
>>>To: Paul Kyzivat; Brian Rosen
>>>Cc: sipcore@ietf.org
>>>Subject: Re: [sipcore] Geolocation Header Question
>>>
>>>At 04:11 PM 8/14/2012, Paul Kyzivat wrote:
>>> >On 8/14/12 5:04 PM, Brian Rosen wrote:
>>> >>Yes, agree, but probably not relevant to the discussion.
>>> >>
>>> >>I went back and looked at the original email, and decided I either
>>>don't
>>> >>understand something, or there is some larger problem.
>>> >>
>>> >>Between what two elements is this Geolocation header running?
>>> >>
>>> >>UAS to proxy server (UE to E-CSCF)?  Proxy to gateway (E-CSCF to MGCF)?
>>> >>
>>> >>Regardless, it seems to be misusing the header, because it's not a
>>>sip or
>>> >>http LbyR.
>>> >>Better to get a real scheme somehow that does the right thing, use a
>>> >>parameter in another header, or create a new header.  Depends on
>>>what this
>>> >>actually represents.  It sounds like it really is LbyR, and you want to
>>> >>communicate a key.<scheme>:<tel>@domain is appropriate.  The issue is
>>> >>what is the right scheme?  It's not sip - it's not sip presence.
>>>It's not
>>> >>HELD identity I take it.
>>> >
>>> >As I already stated, I think sip: is ok if this special lookup can
>>> >be viewed as an *optimization* over doing the presence lookup, and
>>> >doing it the presence way still works.
>>> >
>>> >Otherwise I agree with Brian.
>>>
>>><tel:> isn't an allowed URI in RFC 6442, neither is the use=phone
>>>tag, so IMO that makes this offering a no go as far as 'being in line
>>>with an IETF standards track RFC' (to be formal about it).
>>>
>>>James
>>>
>>>
>>> >         Thanks,
>>> >         Paul
>>> >
>>> >>Brian
>>> >>
>>> >>>I already commented to the original question.
>>> >>>Here I'm just commenting on the comments.
>>> >>>
>>> >>>On 8/14/12 1:55 PM, Brian Rosen wrote:
>>> >>>>I don't think so.
>>> >>>>
>>> >>>>A geolocation header with a sip uri has to lead to a presence server
>>> >>>>that
>>> >>>>can respond to a presence subscription which would return a PIDF with
>>> >>>>location.
>>> >>>>
>>> >>>>Routing a tel uri (which user=phone is supposed to be interchangeable
>>> >>>>with) to a presence service usually makes no sense.  I don't think
>>>you
>>> >>>>can
>>> >>>>send a SUBSCRIBE to a tel uri.  If you could, user=phone is okay.  If
>>> >>>>you
>>> >>>>can't, user=phone is not okay.
>>> >>>
>>> >>>I disagree that a tel uri is *equivalent* to a sip URI with
>>>user=phone!
>>> >>>
>>> >>>A sip URI with user=phone is supposed to be routed like any other sip
>>> >>>URI until it reaches a server responsible for the domain of the URI.
>>> >>>*Then* the user part is to be interpreted as representing a phone
>>>number.
>>> >>>
>>> >>>Conversely, a tel URI has no domain. (And no user=phone.) The
>>>semantics
>>> >>>of telephone numbers are public knowledge. So any server can decide
>>> >>>where to route it next, based on the number and whatever other context
>>> >>>it has.
>>> >>>
>>> >>>IMO the distinction is important. If you want to enable anybody to
>>> >>>decide how to reach the phone number (including going through a PSTN
>>> >>>GW), then use a tel URI. If you want the call handled via SIP at least
>>> >>>until it reaches a particular domain, then use a sip URI.
>>> >>>
>>> >>>         Thanks,
>>> >>>         Paul
>>> >>>
>>> >>>>It's perfectly reasonable to make the uri
>>> >>>>sip:+358222222222@presence.example.com
>>> >>>>that is, a real domain, and no user=phone
>>> >>>>
>>> >>>>Brian
>>> >>>>>
>>> >>>>>You could do this:
>>> >>>>>Geolocation:<sip:+358222222222@atis.invalid;user=phone>
>>> >>>>>(although why ATIS would be defining how calls from Finland work is
>>> >>>>>beyond
>>> >>>>>me ;)
>>> >>>>>
>>> >>>>>But I don't understand why they need it in a Geoloc header to begin
>>> >>>>>with.
>>> >>>>>I assumed that header was for devices/networks that use its
>>>mechanisms
>>> >>>>>(ie, HELD, SUB, pidf).  They're not going to do that, so why use
>>>that
>>> >>>>>header?
>>> >>>>>
>>> >>>>>You said they're doing the location lookup on the PSTN side -
>>> >>>>>presumably
>>> >>>>>then the location is being looked up based on the calling party
>>>info,
>>> >>>>>which should be in another existing header.  Can't they use one
>>>of them
>>> >>>>>as
>>> >>>>>the query key for the lookup? (we have plenty to choose from for
>>>that
>>> >>>>>already :)
>>> >>>>>
>>> >>>>>-hadriel
>>> >>>>>
>>> >>>>>
>>> >>>>>On Aug 7, 2012, at 11:31 AM, Hannes Tschofenig wrote:
>>> >>>>>
>>> >>>>>>I agree but the tel URI is not one of the supported location
>>>URIs in
>>> >>>>>>RFC
>>> >>>>>>6442.
>>> >>>>>>Hence, the guys just use a different way to convey the same
>>> >>>>>>information.
>>> >>>>>>
>>> >>>>>>On Aug 7, 2012, at 5:52 PM, Worley, Dale R (Dale) wrote:
>>> >>>>>>
>>> >>>>>>>>From: Hannes Tschofenig [hannes.tschofenig@gmx.net]
>>> >>>>>>>>
>>> >>>>>>>>In order to convey that phone number for a lookup they are
>>>thinking
>>> >>>>>>>>about using the Geolocation header in the following form:
>>> >>>>>>>>Geolocation:<SIP:+35822222222@mgcf.carrier.net;user=phone)
>>> >>>>>>>
>>> >>>>>>>Naively, I'd say that since the goal is to carry an E.164
>>>number, a
>>> >>>>>>>tel: URI would be the ideal choice.
>>> >>>>>>>
>>> >>>>>>>Dale
>>> >>>>>>
>>> >>>>>>_______________________________________________
>>> >>>>>>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
>>> >>>
>>> >>>_______________________________________________
>>> >>>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
>>


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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Hi Christer -

This all looks good to me (except for a further comment on the Security 
Considerations section inline).

On 8/22/12 7:09 AM, Christer Holmberg wrote:
> Hi Robert,
>
> This reply addresses your comment regarding the IANA registration template/procedure, and the major issues.
>
>> Summary: There are a few issues to address before progressing to IETF LC. I'm putting the document into revised ID needed to address at least the IANA points below.
>>
>> The primary issue is with the instructions to IANA on creating the new trees and the registration template used to put things in those trees.
>>
>> Sections 7.3.2 and 7.3.3 don't currently define the format of an entry in the trees in the new subregistry. This needs to be clearly defined, and the sections need to explicitly indicate that the trees are initially empty.
> I suggest to add the following text to the end of sections 7.3.2 and 7.3.3:
>
> 	"The format of the [global/SIP] tree is as described below:
>
> 	Decimal      Name   Description   Reference
> 	---------------------------------------------------
>
> 	Decimal contains an integer value, incremented by IANA every time a new feature capability indicator is added to the table.
>
> 	Name contains the Feature Capability Indicator Name, provided in the registration feature capability indication registration template.
>
> 	Description contains the Abstract, provided in the registration feature capability indication registration template.
>
> 	Reference contains the Feature Capability Indicator Specification Reference, provided in the registration feature capability indication registration template."
>
> ----------------
>
>> The template in section 8 was adapted from RFC2506, but it needs more tweaks and it needs to match whatever format is defined above.
>>
>>    * We don't need a separate list for reviewing these requests. We certainly don't need a list at apps.ietf.org. The normal process for submitting a request for the global tree should be submitting the template by email to iana@iana.org, and for the sip
>> tree, the normal process would be submitting an Internet-Draft to the IESG.
> I suggest the following change:
>
> OLD:
>
>     	"To: sip-feature-capability-indicators@apps.ietf.org
>     	(feature capability indicators mailing list)
>     	Subject: Registration of feature capability indicator XXXX"
>
> NEW:
>
>   	"Registration requests for the global tree are submitted by e-mail to iana@iana.org.
> 	Registration requests for the sip tree requires submitting an Internet-Draft to the IESG."
>
> ----------------
>
>>    * IANA is currently using a different for media feature tag registrations. It's available at <http://www.iana.org/protocols/media-feature-tag-template.txt>. If we really need to follow that registration format, please start from this template. But do we
>> need all this information in the registry?
> I did start from the media feature tag template, and then removed stuff :)
>
> However, I do agree that not all of the remaining stuff needs to be in the template. Unlike media feature tags, we do mandate a reference to a specification for feature capability indicators, so whatever information that IANA doesn't need can be put in that specification.
>
> So, my suggestion would be that the registration template would look like this:
>
>
>     	"| Instructions are preceded by '|'.  All fields are mandatory.
>
>     	Feature capability indicator name:
>
>     	Description:
>
>     	| The description should be no longer than 4 lines. More
>     	| detailed information can be provided in the feature
>     	| capability indicator specification.
>
>     	Feature capability indicator name:
>
>     	| The referenced specification MUST contain the information
>     	| listed in Section 5.3 of XXXX (IANA: Replace XXXX with
>     	| assigned RFC number of this specification.
>
>     	Contact:
>
>     	| Name(s) & email address(es) of person(s) to
>     	| contact for further information."
>
>
> The following information would be in the feature capability indication specification:
>
>
> 5.3.  Feature Capability Indicator Specification Requirements
>
> 5.3.1.  General
>
>     A feature capability indicator specification MUST address the issues
>     defined in the following subsections, or document why an issue is not
>     applicable for the specific feature capability indicator.  A
>     reference to the specification MUST be provided when the feature
>     capability indicator is registered with IANA (see Section 8).
>
>     It is bad practice for feature capability indicator specifications to
>     repeat procedures (e.g. general procedures on the usage of the
>     Feature-Caps header field and feature capability indicators) defined
>     in this specification, unless needed for clarification or emphasis
>     purpose.  A feature capability indicator specification MUST NOT
>     modify the Feature-Caps header field rules and semantics defined in
>     Section 4.
>
>     A feature capability indicator specification MUST NOT weaken any
>     behavior designated with "SHOULD" or "MUST" in this specification.
>     However, a specification MAY strengthen "SHOULD", "MAY", or
>     "RECOMMENDED" requirements to "MUST" strength if features and
>     capabilities associated with the SIP feature capability indicator
>
> 5.3.2.  Overall Description
>
>     The feature capability indicator specification MUST contain an
>     overall description of the feature capability indicator: how it is
>     used to indicate support of a feature, a description of the feature
>     associated with the SIP feature cap, a description of any additional
>     information (conveyed using one or more feature capability indicator
>     values) that can be conveyed together with the feature capability
>     indicator, and a description of how the associated feature may be
>     exercised/invoked.
>
> 5.3.3.  Feature Capability Indicator Values
>
>     A feature capability indicator can have an associated value, or a
>     list of values.
>
>     The feature capability indicator specification MUST define the syntax
>     and semantics of any value defined for the feature capability
>     indicator, including possible restrictions related to the usage of a
>     specific value.  The feature capability indicator specification MUST
>     define the value(s) in accordance with the ABNF defined in Section
>     6.3.2. The feature capability indicator specification MUST define
>     the default value, if applicable.
>
>     If not values are defined for the feature capability indicator, it MUST
>     be indicated in the feature capability indicator specification.
>
>     A feature capability indicator value is only applicable for the
>     feature capability indicator for which it has been defined.  For
>     other feature capability indicators, the value has to be defined
>     explicitly, even if the semantics are identical.
>
>     It is strongly RECOMMENDED to not re-use a value that already has
>     been defined for another feature capability indicator, unless the
>     semantics of the values are the same.
>
> 5.3.4.  Usage Restrictions
>
>     If there are restrictions on how SIP entities can insert a feature
>     capability indicator, the feature capability indicator specification MUST
>     document such restrictions.
>
>     There might be restrictions related to whether entities are allowed
>     to insert a feature capability indicator in registration related
>     messages, standalone transaction messages, or dialog related
>     messages, whether entities are allowed to insert a feature capability
>     indicator in requests or responses, whether entities also need to
>     support other features and capabilities in order to insert a feature
>     capability indicator, and whether entities are allowed to indicate
>     support of a feature in conjunction with another feature.
>
> 5.3.5.  Interoperability Considerations
>
>    If there are specific interoperability considerations that apply to the
>    feature capability indicator, the feature capability indicator specification
>    MUST document such considerations.
>
> 5.3.6.  Security Considerations
> 	
>    If there are specific security considerations that apply to the feature
>    capability indicator, the feature capability indicator specification
>    MUST document such considerations.
>
> 5.3.7.  Examples
>
>     It is RECOMMENDED that the feature capability indicator specification
>     provide demonstrative message flow diagrams, paired with complete
>     messages and message descriptions.
>
>     Note that example message flows are by definition informative, and do
>     not replace normative text.
>
> 5.3.8.  Other Information
>
>     If there is additional information about the feature capability indicator,
>     it is RECOMMENDED to describe such information. It can include e.g.
>     names of related feature capability indicators."
>     
>
>
> Other major issues:
> -----------------------------------------
>
>> 1) Should the expert reviewer be explicitly required to consider security issues as part of that review? (see section 4.6 of <https://datatracker.ietf.org/doc/draft-ietf-appsawg-media-type-regs/>
>> for example).
> Per my suggestion above, about moving 'Security considerations' from the template into the feature capability indicator specification, I guess the expert reviewer would also look at the Security Considerations. I don't think we need to explicitly say anything.
>
> ----------------
>
>> 2) The security considerations section claims that the aspects of the considerations in rfc3840 that apply to end user devices don't apply here since these are not typically used to convey information about end user devices. "Not typically" is weak. Can this > sentence be made more specific about what concerns do and don't apply? The concerns in 3840 about influencing application behavior, and in 2506 about providing information that can be used to exploit implementation weakeness at the elements that > do provide these tags should be reflected here.
> I suggest some additional security considerations text:
>
> "9.  Security Considerations
>
>     The security issues for feature capability indicators are similar to
>     the ones defined in RFC 3840 for media feature tags.  Media
>     feature tags can reveal information about end-users and end-user
>     equipment, which can be used for industrial espionage. The knowledge
>     about end-user equipment capabilities can also be used to influence
>     application behavior. As feature capability indicators are not intended
>     to convey  capability information of end-user devices, such end-user
>     security aspects of   RFC 3840 do not apply to feature capability indicators.
But they are conveying the capability information for intermediaries, 
and the
concerns about exploiting knowing about what features that intermediary 
supports,
particularly exploiting known implementation weaknesses in the 
implementation of
that feature do apply.
>
>     In addition, the RFC 3840 security issue regarding an attacker using
>     the SIP caller preferences extension [RFC3841] in order to affect
>     routing decisions does not apply, as the mechanism is not defined to
>     be used with feature capability indicators.
>
>     Feature capability indicators can, however, provide capability and
>     characteristics information about the SIP entity, some of which might
>     be sensitive.  The Feature- Caps header field does not convey address
>     information about SIP entities.  However, individual feature capability
>     indicators might provide address information as feature capability indicator
>     values. Therefore, mechanisms for guaranteeing confidentiality and
>     authenticity SHOULD be provided."
>
> ----------------
>
>> 3) Consider clarifying the behavior of SUBSCRIBE NOTIFY dialogs in section 4.3.2. Only one of the 200 OKs to a forked SUBSCRIBE
>> will get back to the UA that sent the subscribe. The information that creates the other dialogs for it must appear in the immediate
>> NOTIFYs. Is there a requirement for the end accepting a subscription to use the same feature capabilities in the 200 OK to the subscription
>> and in the immediate NOTIFY?
> I suggest the following new note to section 4.3.2.
>
>     	"NOTE: As it cannot be guaranteed that 2xx responses associated with
>     	SIP SUBSCRIBE requests will reach the UAC, due to forking of the request,
> 	entities need to indicate supported features and capabilities in the SIP NOTIFY
> 	request that will be sent for each of the created subscription dialogs."
>
> I don't think we need to mandate the capabilities in the 200 OK (if inserted in the first place) need to be identical to the NOTIFY. As the NOTIFY is a target refresh request, it is allowed to indicate different capabilities.
>
> ----------------
>
>> 4) Similarly, consider clarifying whether UPDATES nested within an INVITE constrained to keep the same feature capability indicators as in the INVITE?
> Again, as UPDATEs are target refresh requests (even when nested within an INVITE), they can indicate a new set of supported capabilities.
>
> Note that the indication of supported features is not "offer/answer" - the support is indicated independently in each direction.
>
> ----------------
>
> Thanks!
>
> Regards,
>
> Christer


From christer.holmberg@ericsson.com  Wed Aug 22 14:09:17 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Robert Sparks' <rjsparks@nostrum.com>
Date: Wed, 22 Aug 2012 23:09:13 +0200
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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=20
Hi,

> This all looks good to me (except for a further comment on the Security C=
onsiderations section inline).

>>> 2) The security considerations section claims that the aspects of the c=
onsiderations in rfc3840=20
>>> that apply to end user devices don't apply here since these are not typ=
ically used to convey=20
>>> information about end user devices. "Not typically" is weak. Can this s=
entence be made more=20
>>> specific about what concerns do and don't apply? The concerns in 3840 a=
bout influencing application=20
>>> behavior, and in 2506 about providing information that can be used to e=
xploit implementation weakeness=20
>>> at the elements that do provide these tags should be reflected here.
>> I suggest some additional security considerations text:
>>
>> "9.  Security Considerations
>>
>>     The security issues for feature capability indicators are similar to
>>     the ones defined in RFC 3840 for media feature tags.  Media
>>     feature tags can reveal information about end-users and end-user
>>     equipment, which can be used for industrial espionage. The knowledge
>>     about end-user equipment capabilities can also be used to influence
>>     application behavior. As feature capability indicators are not inten=
ded
>>     to convey  capability information of end-user devices, such end-user
>>     security aspects of   RFC 3840 do not apply to feature capability in=
dicators.
>>
> But they are conveying the capability information for intermediaries, and=
 the concerns about=20
> exploiting knowing about what features that intermediary supports, partic=
ularly exploiting=20
> known implementation weaknesses in the implementation of that feature do =
apply.

I am trying to cover that in the 3rd paragraph (below).

I guess I could move that text up, if it helps.

>>     In addition, the RFC 3840 security issue regarding an attacker using
>>     the SIP caller preferences extension [RFC3841] in order to affect
>>     routing decisions does not apply, as the mechanism is not defined to
>>     be used with feature capability indicators.
>>
>>     Feature capability indicators can, however, provide capability and
>>     characteristics information about the SIP entity, some of which migh=
t
>>     be sensitive.  The Feature- Caps header field does not convey addres=
s
>>     information about SIP entities.  However, individual feature capabil=
ity
>>     indicators might provide address information as feature capability i=
ndicator
>>     values. Therefore, mechanisms for guaranteeing confidentiality and
>>     authenticity SHOULD be provided."

Regards,

Christer

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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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On 8/22/12 4:09 PM, Christer Holmberg wrote:
>   
> Hi,
>
>> This all looks good to me (except for a further comment on the Security Considerations section inline).
>>>> 2) The security considerations section claims that the aspects of the considerations in rfc3840
>>>> that apply to end user devices don't apply here since these are not typically used to convey
>>>> information about end user devices. "Not typically" is weak. Can this sentence be made more
>>>> specific about what concerns do and don't apply? The concerns in 3840 about influencing application
>>>> behavior, and in 2506 about providing information that can be used to exploit implementation weakeness
>>>> at the elements that do provide these tags should be reflected here.
>>> I suggest some additional security considerations text:
>>>
>>> "9.  Security Considerations
>>>
>>>      The security issues for feature capability indicators are similar to
>>>      the ones defined in RFC 3840 for media feature tags.  Media
>>>      feature tags can reveal information about end-users and end-user
>>>      equipment, which can be used for industrial espionage. The knowledge
>>>      about end-user equipment capabilities can also be used to influence
>>>      application behavior. As feature capability indicators are not intended
>>>      to convey  capability information of end-user devices, such end-user
>>>      security aspects of   RFC 3840 do not apply to feature capability indicators.
>>>
>> But they are conveying the capability information for intermediaries, and the concerns about
>> exploiting knowing about what features that intermediary supports, particularly exploiting
>> known implementation weaknesses in the implementation of that feature do apply.
> I am trying to cover that in the 3rd paragraph (below).
>
> I guess I could move that text up, if it helps.
I don't think "might be sensitive" raises the reader's level of 
awareness sufficiently.

How about adding "Malicious elements viewing the indicators may be able 
to discern
application deployment details or identify elements with exploitable feature
implementation weaknesses." after that first sentence, leaving the 
paragraph where it is?
>
>>>      In addition, the RFC 3840 security issue regarding an attacker using
>>>      the SIP caller preferences extension [RFC3841] in order to affect
>>>      routing decisions does not apply, as the mechanism is not defined to
>>>      be used with feature capability indicators.
>>>
>>>      Feature capability indicators can, however, provide capability and
>>>      characteristics information about the SIP entity, some of which might
>>>      be sensitive.  The Feature- Caps header field does not convey address
>>>      information about SIP entities.  However, individual feature capability
>>>      indicators might provide address information as feature capability indicator
>>>      values. Therefore, mechanisms for guaranteeing confidentiality and
>>>      authenticity SHOULD be provided."
> Regards,
>
> Christer


From christer.holmberg@ericsson.com  Wed Aug 22 23:04:19 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>
Date: Thu, 23 Aug 2012 08:04:16 +0200
Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Hi,

>>>>> 2) The security considerations section claims that the aspects of the=
 considerations in rfc3840
>>>>> that apply to end user devices don't apply here since these are not t=
ypically used to convey
>>>>> information about end user devices. "Not typically" is weak. Can this=
 sentence be made more
>>>>> specific about what concerns do and don't apply? The concerns in 3840=
 about influencing application
>>>>> behavior, and in 2506 about providing information that can be used to=
 exploit implementation weakeness
>>>>> at the elements that do provide these tags should be reflected here.
>>>> I suggest some additional security considerations text:
>>>>
>>>> "9.  Security Considerations
>>>>
>>>>      The security issues for feature capability indicators are similar=
 to
>>>>      the ones defined in RFC 3840 for media feature tags.  Media
>>>>      feature tags can reveal information about end-users and end-user
>>>>      equipment, which can be used for industrial espionage. The knowle=
dge
>>>>      about end-user equipment capabilities can also be used to influen=
ce
>>>>      application behavior. As feature capability indicators are not in=
tended
>>>>      to convey  capability information of end-user devices, such end-u=
ser
>>>>      security aspects of   RFC 3840 do not apply to feature capability=
 indicators.
>>>>
>>> But they are conveying the capability information for intermediaries, a=
nd the concerns about
>>> exploiting knowing about what features that intermediary supports, part=
icularly exploiting
>>> known implementation weaknesses in the implementation of that feature d=
o apply.
>> I am trying to cover that in the 3rd paragraph (below).
>>
>> I guess I could move that text up, if it helps.
>
>I don't think "might be sensitive" raises the reader's level of awareness =
sufficiently.
>
> How about adding "Malicious elements viewing the indicators may be able
> to discern application deployment details or identify elements with explo=
itable feature
> implementation weaknesses." after that first sentence, leaving the paragr=
aph where it is?

Sounds good. So, something like:

 "9.  Security Considerations

     The security issues for feature capability indicators are similar to
     the ones defined in RFC 3840 for media feature tags.  Media
     feature tags can reveal information about end-users and end-user
     equipment, which can be used for industrial espionage. The knowledge
     about end-user equipment capabilities can also be used to influence
     application behavior. As feature capability indicators are not intende=
d
     to convey  capability information of end-user devices, such end-user
     security aspects of   RFC 3840 do not apply to feature capability indi=
cators.

     In addition, the RFC 3840 security issue regarding an attacker using
     the SIP caller preferences extension [RFC3841] in order to affect
     routing decisions does not apply, as the mechanism is not defined to
     be used with feature capability indicators.

     Feature capability indicators can, however, provide capability and
     characteristics information about the SIP entity, some of which might
     be sensitive. Malicious elements viewing the indicators may be able
     to discern application deployment details or identify elements with=20
     exploitable feature implementation weaknesses. The Feature-Caps=20
     header field does not convey address information about SIP entities. =
=20
     However, individual feature capability indicators might provide addres=
s=20
     information as feature capability indicator values. Therefore, mechani=
sms=20
     for guaranteeing confidentiality and authenticity SHOULD be provided."

Regards,

Christer=

From christer.holmberg@ericsson.com  Wed Aug 22 23:23:52 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-proxy-feature@tools.ietf.org" <draft-ietf-sipcore-proxy-feature@tools.ietf.org>
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Thread-Topic: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-proxy-feature-05 - Robert's comments - IANA issues and major issues
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Hi,

>So, my suggestion would be that the registration template would look like =
this:
>
>
>        "| Instructions are preceded by '|'.  All fields are mandatory.
>
>        Feature capability indicator name:
>
>        Description:
>
>        | The description should be no longer than 4 lines. More
>        | detailed information can be provided in the feature
>        | capability indicator specification.
>
>        Feature capability indicator name:
>
>        | The referenced specification MUST contain the information
>        | listed in Section 5.3 of XXXX (IANA: Replace XXXX with
>        | assigned RFC number of this specification.
>
>        Contact:
>
>        | Name(s) & email address(es) of person(s) to
>        | contact for further information."

The fci specification reference will of course also be part of the template=
:

     "Feature capability indicator specification reference:  =20

     | The referenced specification MUST contain the information=20
     | listed in Section 5.3 of XXXX (IANA: Replace XXXX with=20
     | assigned RFC number of this specification."

Regards,

Christer=

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On Sat, Aug 11, 2012 at 1:20 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> In addition to this issue, there is the one about the current 3261
> requirement to support TCP and UDP transports, which many implementations of
> this draft will have difficulty complying with.

This transport can be implemented also on User Agents other than web
browsers (e.g. smartphones). What would be your suggestion, relax the
"MUST implement UDP and TCP" requirement (1)only for web browsers,
(2)for any device implementing the websockets transport, (3)in
general?

Cheers,
-Victor

From partha@parthasarathi.co.in  Thu Aug 23 05:53:31 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: <sipcore@ietf.org>
Date: Thu, 23 Aug 2012 18:23:18 +0530
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Subject: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Hi all,

 

As I discussed during IETF-84 meeting,  the following text is provided for
the starting the discussion on deviation of RFC 3261 mandatory to implement
(UDP/TCP) transport in this draft. 

 

"In case SIP WebSocket Client is a web browser and SIP Websocket Server is a
SIP proxy, then record-route header MUST be included by SIP Websocket server
in all SIP messages. This requirement assures that SIP WebSocket Client is
able to contact with UAS through SIP Websocket Server during the mid-dialog
irrespective of the contact SIP URI received from UAS "

 

The above normative statement will be update to  RFC 3261.

 

Thanks

Partha


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<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times =
New Roman","serif"'>Hi
all,<o:p></o:p></span></p>

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

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times =
New Roman","serif"'>As
I discussed during IETF-84 meeting, &nbsp;the following text is provided =
for
the starting the discussion on deviation of RFC 3261 mandatory to =
implement (UDP/TCP)
transport in this draft. <o:p></o:p></span></p>

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

<p class=3DMsoNormal><span lang=3DEN =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>&#8220;In
case SIP WebSocket Client is a web browser and SIP Websocket Server is a =
SIP
proxy, then record-route header MUST be included by SIP Websocket server =
in all
SIP messages. This requirement assures that SIP WebSocket Client is able =
to
contact with UAS through SIP Websocket Server during the mid-dialog =
irrespective
of the contact SIP URI received from UAS &#8221;<o:p></o:p></span></p>

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

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times =
New Roman","serif"'>The
above normative statement will be update to&nbsp; RFC =
3261.<o:p></o:p></span></p>

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

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times =
New Roman","serif"'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times =
New Roman","serif"'>Partha<o:p></o:p></span></p>

</div>

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Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Hi,

2012/8/23 Parthasarathi R <partha@parthasarathi.co.in>

>  Hi all,****
>
> ** **
>
> As I discussed during IETF-84 meeting,  the following text is provided fo=
r
> the starting the discussion on deviation of RFC 3261 mandatory to impleme=
nt
> (UDP/TCP) transport in this draft. ****
>
> ** **
>
> =93In case SIP WebSocket Client is a web browser and SIP Websocket Server=
 is
> a SIP proxy, then record-route header MUST be included by SIP Websocket
> server in all SIP messages. This requirement assures that SIP WebSocket
> Client is able to contact with UAS through SIP Websocket Server during th=
e
> mid-dialog irrespective of the contact SIP URI received from UAS =94
>

Record-Route header field is useful for a proxy in order to remain in the
dialog path. It is not necessary for all SIP messages. This is defined in
RFC 3261 as 'loose routing' and draft-ibc-sipcore-sip-websocket-02 does not
change it at all.

Section 2 of Apendix A (Implementation guidelines) already mentions it:

"
 The proxy performs Loose Routing and remains in dialogs path as

   specified in [RFC3261 <http://tools.ietf.org/html/rfc3261>].
Otherwise in-dialog requests would fail
   since SIP WebSocket Clients make use of their SIP WebSocket Server in
   order to send and receive SIP requests and responses.

"

Regards,

>  **
>
> The above normative statement will be update to  RFC 3261.****
>
> ** **
>
> Thanks****
>
> Partha****
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>

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Hi,<br><br>2012/8/23 Parthasarathi R <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:partha@parthasarathi.co.in" target=3D"_blank">partha@parthasarathi.co.in<=
/a>&gt;</span><br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>









<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;">Hi
all,<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;"><u></u>=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;">As
I discussed during IETF-84 meeting, =A0the following text is provided for
the starting the discussion on deviation of RFC 3261 mandatory to implement=
 (UDP/TCP)
transport in this draft. <u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;"><u></u>=A0<u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN" style=3D"font-size:12.0pt;font-fam=
ily:&quot;Times New Roman&quot;,&quot;serif&quot;">=93In
case SIP WebSocket Client is a web browser and SIP Websocket Server is a SI=
P
proxy, then record-route header MUST be included by SIP Websocket server in=
 all
SIP messages. This requirement assures that SIP WebSocket Client is able to
contact with UAS through SIP Websocket Server during the mid-dialog irrespe=
ctive
of the contact SIP URI received from UAS =94</span></p></div></div></blockq=
uote><div><br></div><div>Record-Route header field is useful for a proxy in=
 order to remain in the dialog path. It is not necessary for all SIP messag=
es.=A0This is defined in RFC 3261 as &#39;loose routing&#39; and=A0draft-ib=
c-sipcore-sip-websocket-02 does not change it at all.</div>
<div><br></div><div>Section 2 of Apendix A (Implementation guidelines) alre=
ady=A0mentions=A0it:</div><div><br></div><div>&quot;</div><div><font face=
=3D"arial, helvetica, sans-serif">=A0The proxy performs Loose Routing and r=
emains in dialogs path as</font></div>
<pre class=3D"newpage" style=3D"margin-top:0px;margin-bottom:0px"><font fac=
e=3D"arial, helvetica, sans-serif">   specified in [<a href=3D"http://tools=
.ietf.org/html/rfc3261" title=3D"&quot;SIP: Session Initiation Protocol&quo=
t;">RFC3261</a>].  Otherwise in-dialog requests would fail
   since SIP WebSocket Clients make use of their SIP WebSocket Server in
   order to send and receive SIP requests and responses.</font></pre><h1 cl=
ass=3D"ha" style=3D"margin:12px 1px 9px 0px;padding:0px 0px 0px 8px;color:r=
gb(34,34,34);background:inherit;border-right:inherit;background-color:rgb(2=
55,255,255);font-weight:normal">
<font face=3D"arial, helvetica, sans-serif">&quot;</font></h1><div><br></di=
v><div>Regards,</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"EN-US" lin=
k=3D"blue" vlink=3D"purple">
<div><p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&qu=
ot;Times New Roman&quot;,&quot;serif&quot;">=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;">The
above normative statement will be update to=A0 RFC 3261.<u></u><u></u></spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;"><u></u>=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;">Thanks<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;">Partha<u></u><u></u></span></p>

</div>

</div>


<br>_______________________________________________<br>
sipcore mailing list<br>
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<br></blockquote></div><br>

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I think this document is well written overall and I have no
comments other than one nit.

Referring to page 18, the paragraph that begins "If a REFER
request is sent to a third SIP user agent ..." -- I think this
paragraph should be rewritten.  The first sentence,
being in passive voice, does not provide a clue as to who sends
the REFER.  Also, it's not clear who "a third SIP user agent" is
supposed to be, and it seems to imply there is a "first SIP user
agent" and a "second SIP user agent" (who are not identified).  I
found this paragraph to be confusing.

--
Doug Sauder

On 2012-08-10 13:38 , Adam Roach - SIPCORE Chair wrote:
> [as chair]
>
> Based on feedback received from the authors and working group, we 
> believe that the document "The WebSocket Protocol as a Transport for 
> the Session Initiation Protocol (SIP)" is ready for working group last 
> call.
>
> Working group participants -- and, in particular, those participants 
> who indicated a willingness to perform a dedicated review of the 
> document -- are asked to read the current document and provide 
> feedback. Even short statements along the lines of "I have carefully 
> read the document and believe it is ready for publication" are useful.
>
> The current version of the document is available here:
>
> http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/
>
> This WGLC runs until Friday, August 24th. Thank you.
>
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From internet-drafts@ietf.org  Fri Aug 24 02:44:43 2012
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-proxy-feature-06.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : Mechanism to indicate support of features and capabiliti=
es in the Session Initiation Protocol (SIP)
	Author(s)       : Christer Holmberg
                          Ivo Sedlacek
                          Hadriel Kaplan
	Filename        : draft-ietf-sipcore-proxy-feature-06.txt
	Pages           : 21
	Date            : 2012-08-24

Abstract:
   This specification defines a new SIP header field, Feature-Caps, to
   convey feature capability indicators, which are used by SIP entities
   not represented by the URI of the Contact header field to indicate
   support of features and capabilities, where media feature tags cannot
   be used to indicate the support.

   This specification also defines feature capability indicators, and
   creates a new IANA registry, "Proxy-Feature Feature Capability
   Indicator Trees", for registering feature capability indicators.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-proxy-feature

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-proxy-feature-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-proxy-feature-06


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


From christer.holmberg@ericsson.com  Fri Aug 24 02:46:24 2012
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Subject: [sipcore] Draft new version: proxy-feature-06
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Hi,

Based on the comments from Robert, I've updated and submitted a new version=
 (-06) of the proxy-feature draft.

Regards,

Christer

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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-micr=
osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:=
//www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV=3D"Content-Type" CONTENT=
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	{mso-style-type:export-only;
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	{size:612.0pt 792.0pt;
	margin:72.0pt 72.0pt 72.0pt 72.0pt;}
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--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Hi,<o:p></o:p></=
p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Based on t=
he comments from Robert, I&#8217;ve updated and submitted a new version (-0=
6) of the proxy-feature draft.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbs=
p;</o:p></p><p class=3DMsoNormal>Regards,<o:p></o:p></p><p class=3DMsoNorma=
l><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Christer<o:p></o:p></p></div></=
body></html>=

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From brett@broadsoft.com  Fri Aug 24 05:51:54 2012
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From: Brett Tate <brett@broadsoft.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-sip-websocket@tools.ietf.org" <draft-ietf-sipcore-sip-websocket@tools.ietf.org>
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Howdy,

The following are some minor comments concerning draft-ietf-sipcore-sip-web=
socket-02.

Thanks,
Brett

----

Section 8.2:

- The response examples should likely indicate the Via "received" parameter=
 when Via transport is UDP.

- Max-Forwards should likely be removed from response examples.

Section A.1: "hostpart" potentially should be "hostport".


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From partha@parthasarathi.co.in  Fri Aug 24 09:22:19 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
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Subject: Re: [sipcore] Mandatory to implement transport comment on	draft-ibc-sipcore-sip-websocket-02
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Hi Jose,

=20

Let us assume draft-ibc-sipcore-sip-websocket-02 does not change proxy
behavior in SIP websocket server then

the dialog creation in the following topology breaks as follows:

=20

Topolgy:

=20

  UAC SIP Websocket client (Browser)------Proxy SIP Websocket
Server------UAS (RFC 3261 UA)=20

=20

Callflow steps:

=20

1)    UAC SIP Websocket client initiates INVITE dialog towards SIP =
Websocket
server

2)    INVITE is forwarded towards UAS by SIP Websocket server

3)    UAS receives INVITE and sent 200 OK response with contact address =
with
UDP transport

towards SIP Websocket server

4)    SIP Websocket server forwards 200 OK towards SIP Websocket client
without record-route

5)    Whether SIP Websocket Client(browser) will be able to send ACK =
towards
UAS contact

       address in UDP transport? No.

=20

In case normal RFC 3261 application or application based on SIP =
websocket
client, ACK will send towards=20

UAS contact address with UDP transport as per mandatory to implement
transport mechanism in RFC 3261.=20

As browser acting as SIP Websocket client violates the mandatory to
implement transport mechanism,

the callflow breaks in this specification.=20

=20

In case you wish to put some other text, Please let us discuss.

=20

Thanks

Partha

=20

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On =
Behalf
Of Jos=E9 Luis Mill=E1n
Sent: Thursday, August 23, 2012 9:46 PM
To: Parthasarathi R
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on
draft-ibc-sipcore-sip-websocket-02

=20

Hi,

2012/8/23 Parthasarathi R <partha@parthasarathi.co.in>

Hi all,

=20

As I discussed during IETF-84 meeting,  the following text is provided =
for
the starting the discussion on deviation of RFC 3261 mandatory to =
implement
(UDP/TCP) transport in this draft.=20

=20

=93In case SIP WebSocket Client is a web browser and SIP Websocket =
Server is a
SIP proxy, then record-route header MUST be included by SIP Websocket =
server
in all SIP messages. This requirement assures that SIP WebSocket Client =
is
able to contact with UAS through SIP Websocket Server during the =
mid-dialog
irrespective of the contact SIP URI received from UAS =94

=20

Record-Route header field is useful for a proxy in order to remain in =
the
dialog path. It is not necessary for all SIP messages. This is defined =
in
RFC 3261 as 'loose routing' and draft-ibc-sipcore-sip-websocket-02 does =
not
change it at all.

=20

Section 2 of Apendix A (Implementation guidelines) already mentions it:

=20

"

 The proxy performs Loose Routing and remains in dialogs path as

   specified in [RFC3261 <http://tools.ietf.org/html/rfc3261> ].  =
Otherwise
in-dialog requests would fail
   since SIP WebSocket Clients make use of their SIP WebSocket Server in
   order to send and receive SIP requests and responses.

"


=20

Regards,

=20

The above normative statement will be update to  RFC 3261.

=20

Thanks

Partha


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

=20


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

<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Jose,<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'>Let us assume </span><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>draft-ibc-sipcore-sip-websocket-02 does not =
change proxy
behavior in SIP websocket server then<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>the
dialog creation in the following topology breaks as =
follows:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p>&nbsp;<=
/o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Topolgy:<o:p=
></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p>&nbsp;<=
/o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>=A0
UAC SIP Websocket client (Browser)------Proxy SIP Websocket =
Server------UAS
(RFC 3261 UA) <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p>&nbsp;<=
/o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Callflow
steps:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p>&nbsp;<=
/o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><span
style=3D'mso-list:Ignore'>1)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>UAC SIP Websocket client initiates INVITE dialog towards =
SIP
Websocket server<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><span
style=3D'mso-list:Ignore'>2)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>INVITE is forwarded towards UAS by SIP Websocket =
server<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><span
style=3D'mso-list:Ignore'>3)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>UAS receives INVITE and sent 200 OK response with contact =
address
with UDP transport<o:p></o:p></span></p>

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

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><span
style=3D'mso-list:Ignore'>4)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>SIP Websocket server forwards 200 OK towards SIP =
Websocket
client without record-route<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><span
style=3D'mso-list:Ignore'>5)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Whether SIP Websocket Client(browser) will be able to =
send ACK
towards UAS contact<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-left:.25in'><span =
style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>=A0=A0=A0=A0=A0 =
=A0address in UDP
transport? No.<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'>In case normal RFC 3261 application or application based =
on SIP
websocket client, ACK will send towards <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>UAS contact address with UDP transport as per mandatory =
to
implement transport mechanism in RFC 3261. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>As browser acting as SIP Websocket client violates the =
mandatory
to implement transport mechanism,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>the callflow breaks in this specification. =
<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'>In case you wish to put some other text, Please let us =
discuss.<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'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<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-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>Jos=E9
Luis Mill=E1n<br>
<b>Sent:</b> Thursday, August 23, 2012 9:46 PM<br>
<b>To:</b> Parthasarathi R<br>
<b>Cc:</b> sipcore@ietf.org<br>
<b>Subject:</b> Re: [sipcore] Mandatory to implement transport comment =
on
draft-ibc-sipcore-sip-websocket-02<o:p></o:p></span></p>

</div>

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

<p class=3DMsoNormal>Hi,<br>
<br>
2012/8/23 Parthasarathi R &lt;<a =
href=3D"mailto:partha@parthasarathi.co.in"
target=3D"_blank">partha@parthasarathi.co.in</a>&gt;<o:p></o:p></p>

<div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<div>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Hi
all,<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>As
I discussed during IETF-84 meeting, &nbsp;the following text is provided =
for the
starting the discussion on deviation of RFC 3261 mandatory to implement
(UDP/TCP) transport in this draft. <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
lang=3DEN>&#8220;In case SIP WebSocket Client is a web browser and SIP =
Websocket
Server is a SIP proxy, then record-route header MUST be included by SIP =
Websocket
server in all SIP messages. This requirement assures that SIP WebSocket =
Client
is able to contact with UAS through SIP Websocket Server during the =
mid-dialog
irrespective of the contact SIP URI received from UAS =
&#8221;</span><o:p></o:p></p>

</div>

</div>

</blockquote>

<div>

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

</div>

<div>

<p class=3DMsoNormal>Record-Route header field is useful for a proxy in =
order to
remain in the dialog path. It is not necessary for all SIP =
messages.&nbsp;This
is defined in RFC 3261 as 'loose routing'
and&nbsp;draft-ibc-sipcore-sip-websocket-02 does not change it at =
all.<o:p></o:p></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal>Section 2 of Apendix A (Implementation guidelines)
already&nbsp;mentions&nbsp;it:<o:p></o:p></p>

</div>

<div>

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

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>&nbsp;The
proxy performs Loose Routing and remains in dialogs path =
as</span><o:p></o:p></p>

</div>

<pre><span style=3D'font-family:"Arial","sans-serif"'>=A0=A0 specified =
in [<a
href=3D"http://tools.ietf.org/html/rfc3261"
title=3D"&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].=A0 =
Otherwise in-dialog requests would =
fail<o:p></o:p></span></pre><pre><span
style=3D'font-family:"Arial","sans-serif"'> =A0=A0since SIP WebSocket =
Clients make use of their SIP WebSocket Server =
in<o:p></o:p></span></pre><pre><span
style=3D'font-family:"Arial","sans-serif"'>=A0=A0 order to send and =
receive SIP requests and responses.</span><o:p></o:p></pre>

<h1 =
style=3D'mso-margin-top-alt:9.0pt;margin-right:.75pt;margin-bottom:6.75pt=
;
margin-left:0in;background:white inherit;border-right:inherit'><span
style=3D'font-family:"Arial","sans-serif";color:#222222;font-weight:norma=
l'>&quot;</span><span
style=3D'color:#222222;font-weight:normal'><o:p></o:p></span></h1>

<div>

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

</div>

<div>

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

</div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<div>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>The
above normative statement will be update to&nbsp; RFC =
3261.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Thanks<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Partha<o:p><=
/o:p></p>

</div>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
_______________________________________________<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><o:p><=
/o:p></p>

</blockquote>

</div>

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

</div>

</body>

</html>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 24 Aug 2012 18:39:12 +0200
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Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> In case normal RFC 3261 application or application based on SIP websocket
> client, ACK will send towards
>
> UAS contact address with UDP transport as per mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism,
>
> the callflow breaks in this specification.

Hi Partha. As said many times this draft is not *just* about web
browsers. Please imagine that new versions of iOS and Android come
with a WebSocket Client stack (so native apps can use it). Should we
also add them to the draft?:

=E2=80=9CIn case SIP WebSocket Client is a web browser or a native iOS or
Android application, and SIP Websocket Server is a SIP proxy, then..."

Ugly, agree?


Anyhow, for the case of a web browser (which indeed cannot send a UDP
datagram) the draft already describes a full guideline in the Appendix
section:

http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02#appendix-A

IMHO that's enough. The mandatory sections of the draft should not
focus just on web browsers.


BTW your text is incomplete:


=E2=80=9CIn case SIP WebSocket Client is a web browser and SIP Websocket
Server is a SIP proxy, then record-route header MUST be included by
SIP Websocket server in all SIP messages. This requirement assures
that SIP WebSocket Client is able to contact with UAS through SIP
Websocket Server during the mid-dialog irrespective of the contact SIP
URI received from UAS =E2=80=9D

That's not enough. If you want to also *receive* incoming requests you
need to mandate Outbound RFC 5626 (since the proxy cannot establish a
WebSocket connection with a web browser). But we don't want to mandate
Outbound (it was discussed and voted in Paris) since the draft just
describes a SIP transport and does not assume specific topologies or
UA devices.

IMHO the Appendix guideline (which was already discussed in this WG)
is the appropriate section for talking about specific common
scenarios. The draft should not "update" RFC 3261 just for the case of
web browsers.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From jmillan@aliax.net  Fri Aug 24 09:48:50 2012
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Hi Partha,

Please, read inline.

2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> Hi Jose,
>
>
>
> Let us assume draft-ibc-sipcore-sip-websocket-02 does not change proxy
> behavior in SIP websocket server then
>
> the dialog creation in the following topology breaks as follows:
>
>
>
> Topolgy:
>
>
>
>   UAC SIP Websocket client (Browser)------Proxy SIP Websocket
> Server------UAS (RFC 3261 UA)
>
>
>
> Callflow steps:
>
>
>
> 1)    UAC SIP Websocket client initiates INVITE dialog towards SIP Websoc=
ket
> server
>
> 2)    INVITE is forwarded towards UAS by SIP Websocket server

.. which adds a Record-Route header field because it wants to stay in
the signaling path during the dialog ;-)

>
> 3)    UAS receives INVITE and sent 200 OK response with contact address w=
ith
> UDP transport

it builds the 200 OK response taking the Record-Route headers from the
received INVITE request.

>
> towards SIP Websocket server
>
> 4)    SIP Websocket server forwards 200 OK towards SIP Websocket client
> without record-route

It receives 200 OK with the Record-Route headers

>
> 5)    Whether SIP Websocket Client(browser) will be able to send ACK towa=
rds
> UAS contact
>
>        address in UDP transport? No.
>

ACK request will be formed with the dialog`s route set, being the
first hop the SIP WebSocket Server.

>
>
> In case normal RFC 3261 application or application based on SIP websocket
> client, ACK will send towards
>
> UAS contact address with UDP transport as per mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism,
>
> the callflow breaks in this specification.
>

Why didn't you add a Record-Route header in the point '2' ( INVITE is
forwarded towards UAS by SIP Websocket server) ?

You should have added if you wanted to stay in the signaling path
during the dialog. This is stated in RFC 3261. The reason your ACK
does not reach the destination is a configuration problem more that
specification definition issue.

>
>
> In case you wish to put some other text, Please let us discuss.
>
>
>
> Thanks
>
> Partha
>

Regards.

>
>
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Jos=E9 Luis Mill=E1n
> Sent: Thursday, August 23, 2012 9:46 PM
> To: Parthasarathi R
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Mandatory to implement transport comment on
> draft-ibc-sipcore-sip-websocket-02
>
>
>
> Hi,
>
> 2012/8/23 Parthasarathi R <partha@parthasarathi.co.in>
>
> Hi all,
>
>
>
> As I discussed during IETF-84 meeting,  the following text is provided fo=
r
> the starting the discussion on deviation of RFC 3261 mandatory to impleme=
nt
> (UDP/TCP) transport in this draft.
>
>
>
> =93In case SIP WebSocket Client is a web browser and SIP Websocket Server=
 is a
> SIP proxy, then record-route header MUST be included by SIP Websocket ser=
ver
> in all SIP messages. This requirement assures that SIP WebSocket Client i=
s
> able to contact with UAS through SIP Websocket Server during the mid-dial=
og
> irrespective of the contact SIP URI received from UAS =94
>
>
>
> Record-Route header field is useful for a proxy in order to remain in the
> dialog path. It is not necessary for all SIP messages. This is defined in
> RFC 3261 as 'loose routing' and draft-ibc-sipcore-sip-websocket-02 does n=
ot
> change it at all.
>
>
>
> Section 2 of Apendix A (Implementation guidelines) already mentions it:
>
>
>
> "
>
>  The proxy performs Loose Routing and remains in dialogs path as
>
>    specified in [RFC3261].  Otherwise in-dialog requests would fail
>
>    since SIP WebSocket Clients make use of their SIP WebSocket Server in
>
>    order to send and receive SIP requests and responses.
>
> "
>
>
>
> Regards,
>
>
>
> The above normative statement will be update to  RFC 3261.
>
>
>
> Thanks
>
> Partha
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>

From jmillan@aliax.net  Fri Aug 24 10:15:24 2012
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From: =?ISO-8859-1?Q?Jos=E9_Luis_Mill=E1n?= <jmillan@aliax.net>
To: Brett Tate <brett@broadsoft.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-sip-websocket@tools.ietf.org" <draft-ietf-sipcore-sip-websocket@tools.ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-sip-websocket-02: comments
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Hi Brett,

Thanks for your comments. We will address them.

Regards,

2012/8/24 Brett Tate <brett@broadsoft.com>:
> Howdy,
>
> The following are some minor comments concerning draft-ietf-sipcore-sip-w=
ebsocket-02.
>
> Thanks,
> Brett
>
> ----
>
> Section 8.2:
>
> - The response examples should likely indicate the Via "received" paramet=
er when Via transport is UDP.
>
> - Max-Forwards should likely be removed from response examples.
>
> Section A.1: "hostpart" potentially should be "hostport".
>
>
> This email is intended solely for the person or entity to which it is add=
ressed and may contain confidential and/or privileged information.  If you =
are not the intended recipient and have received this email in error, pleas=
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> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/sipcore

From partha@parthasarathi.co.in  Fri Aug 24 10:15:28 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: =?iso-8859-1?Q?'Jos=E9_Luis_Mill=E1n'?= <jmillan@aliax.net>
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Subject: Re: [sipcore] Mandatory to implement transport comment on	draft-ibc-sipcore-sip-websocket-02
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Jose,

I haven't added "record-route" in point 2 as RFC 3261 does not mandates =
to
add :-)=20
In case of RFC 3261 SIP transports, the dialog creation happens without
"record-route"=20
whereas SIP over WebSocket transport in browser is different.

Thanks
Partha

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On =
Behalf
Of Jos=E9 Luis Mill=E1n
Sent: Friday, August 24, 2012 10:19 PM
To: Parthasarathi R
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on
draft-ibc-sipcore-sip-websocket-02

Hi Partha,

Please, read inline.

2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> Hi Jose,
>
>
>
> Let us assume draft-ibc-sipcore-sip-websocket-02 does not change proxy
> behavior in SIP websocket server then
>
> the dialog creation in the following topology breaks as follows:
>
>
>
> Topolgy:
>
>
>
>   UAC SIP Websocket client (Browser)------Proxy SIP Websocket
> Server------UAS (RFC 3261 UA)
>
>
>
> Callflow steps:
>
>
>
> 1)    UAC SIP Websocket client initiates INVITE dialog towards SIP
Websocket
> server
>
> 2)    INVITE is forwarded towards UAS by SIP Websocket server

.. which adds a Record-Route header field because it wants to stay in
the signaling path during the dialog ;-)

>
> 3)    UAS receives INVITE and sent 200 OK response with contact =
address
with
> UDP transport

it builds the 200 OK response taking the Record-Route headers from the
received INVITE request.

>
> towards SIP Websocket server
>
> 4)    SIP Websocket server forwards 200 OK towards SIP Websocket =
client
> without record-route

It receives 200 OK with the Record-Route headers

>
> 5)    Whether SIP Websocket Client(browser) will be able to send ACK
towards
> UAS contact
>
>        address in UDP transport? No.
>

ACK request will be formed with the dialog`s route set, being the
first hop the SIP WebSocket Server.

>
>
> In case normal RFC 3261 application or application based on SIP =
websocket
> client, ACK will send towards
>
> UAS contact address with UDP transport as per mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism,
>
> the callflow breaks in this specification.
>

Why didn't you add a Record-Route header in the point '2' ( INVITE is
forwarded towards UAS by SIP Websocket server) ?

You should have added if you wanted to stay in the signaling path
during the dialog. This is stated in RFC 3261. The reason your ACK
does not reach the destination is a configuration problem more that
specification definition issue.

>
>
> In case you wish to put some other text, Please let us discuss.
>
>
>
> Thanks
>
> Partha
>

Regards.

>
>
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On =
Behalf
> Of Jos=E9 Luis Mill=E1n
> Sent: Thursday, August 23, 2012 9:46 PM
> To: Parthasarathi R
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Mandatory to implement transport comment on
> draft-ibc-sipcore-sip-websocket-02
>
>
>
> Hi,
>
> 2012/8/23 Parthasarathi R <partha@parthasarathi.co.in>
>
> Hi all,
>
>
>
> As I discussed during IETF-84 meeting,  the following text is provided =
for
> the starting the discussion on deviation of RFC 3261 mandatory to
implement
> (UDP/TCP) transport in this draft.
>
>
>
> =93In case SIP WebSocket Client is a web browser and SIP Websocket =
Server is
a
> SIP proxy, then record-route header MUST be included by SIP Websocket
server
> in all SIP messages. This requirement assures that SIP WebSocket =
Client is
> able to contact with UAS through SIP Websocket Server during the
mid-dialog
> irrespective of the contact SIP URI received from UAS =94
>
>
>
> Record-Route header field is useful for a proxy in order to remain in =
the
> dialog path. It is not necessary for all SIP messages. This is defined =
in
> RFC 3261 as 'loose routing' and draft-ibc-sipcore-sip-websocket-02 =
does
not
> change it at all.
>
>
>
> Section 2 of Apendix A (Implementation guidelines) already mentions =
it:
>
>
>
> "
>
>  The proxy performs Loose Routing and remains in dialogs path as
>
>    specified in [RFC3261].  Otherwise in-dialog requests would fail
>
>    since SIP WebSocket Clients make use of their SIP WebSocket Server =
in
>
>    order to send and receive SIP requests and responses.
>
> "
>
>
>
> Regards,
>
>
>
> The above normative statement will be update to  RFC 3261.
>
>
>
> Thanks
>
> Partha
>
>
> _______________________________________________
> 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 partha@parthasarathi.co.in  Fri Aug 24 10:41:19 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: =?UTF-8?Q?'I=C3=B1aki_Baz_Castillo'?= <ibc@aliax.net>
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Date: Fri, 24 Aug 2012 23:11:02 +0530
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Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Inaki,

Paul started the relaxing the criteria for mandatory to implement =
transport mail thread during WG adoption of SIP over websocket draft and =
one of the mail is =
http://www.ietf.org/mail-archive/web/sipcore/current/msg04930.html. My =
understanding of the mail thread conclusion was that narrow approach of =
providing leniency for mandatory to implement transport in SIP over =
Websocket draft itself.  I noticed in the current document that there is =
no relevant text for this leniency and so raised the issue in IETF-84 =
SIPCore meeting. As per IETF-84 SIPCore meeting discussion, I started =
this mail thread.

In case you are not in favor of narrow approach proposed by Paul, then =
it is a separate discussion. Also you have indicated that the narrow =
approach needs more text, then please suggest the text and let us =
discuss.

Thanks
Partha=20

-----Original Message-----
From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]=20
Sent: Friday, August 24, 2012 10:09 PM
To: Parthasarathi R
Cc: Jos=C3=A9 Luis Mill=C3=A1n; sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on =
draft-ibc-sipcore-sip-websocket-02

2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> In case normal RFC 3261 application or application based on SIP =
websocket
> client, ACK will send towards
>
> UAS contact address with UDP transport as per mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism,
>
> the callflow breaks in this specification.

Hi Partha. As said many times this draft is not *just* about web
browsers. Please imagine that new versions of iOS and Android come
with a WebSocket Client stack (so native apps can use it). Should we
also add them to the draft?:

=E2=80=9CIn case SIP WebSocket Client is a web browser or a native iOS =
or
Android application, and SIP Websocket Server is a SIP proxy, then..."

Ugly, agree?


Anyhow, for the case of a web browser (which indeed cannot send a UDP
datagram) the draft already describes a full guideline in the Appendix
section:

http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02#appendix-A

IMHO that's enough. The mandatory sections of the draft should not
focus just on web browsers.


BTW your text is incomplete:


=E2=80=9CIn case SIP WebSocket Client is a web browser and SIP Websocket
Server is a SIP proxy, then record-route header MUST be included by
SIP Websocket server in all SIP messages. This requirement assures
that SIP WebSocket Client is able to contact with UAS through SIP
Websocket Server during the mid-dialog irrespective of the contact SIP
URI received from UAS =E2=80=9D

That's not enough. If you want to also *receive* incoming requests you
need to mandate Outbound RFC 5626 (since the proxy cannot establish a
WebSocket connection with a web browser). But we don't want to mandate
Outbound (it was discussed and voted in Paris) since the draft just
describes a SIP transport and does not assume specific topologies or
UA devices.

IMHO the Appendix guideline (which was already discussed in this WG)
is the appropriate section for talking about specific common
scenarios. The draft should not "update" RFC 3261 just for the case of
web browsers.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>


From jmillan@aliax.net  Fri Aug 24 10:44:28 2012
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Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> Jose,
>
> I haven't added "record-route" in point 2 as RFC 3261 does not mandates t=
o
> add :-)

It mandates to insert a Record-Route header field if the proxy wishes
to remain on the path of future requests in a dialog.

"
16.6 Request Forwarding

If this proxy wishes to remain on the path of future requests
in a dialog created by this request (assuming the request
creates a dialog), it MUST insert a Record-Route header field
value into the copy before any existing Record-Route header
field values, even if a Route header field is already present.
"

Addition of Record-Route header field is already commented in
draft-ibc-sipcore-sip-websocket-02, in section 2 of Appendix A:  SIP
WebSocket Server Considerations.

"
   The proxy performs Loose Routing and remains in dialogs path as
   specified in [RFC3261].  Otherwise in-dialog requests would fail
   since SIP WebSocket Clients make use of their SIP WebSocket Server in
   order to send and receive SIP requests and responses.
"

May the following (extracted from the previous text) not be clear?

"
SIP WebSocket Clients make use of their SIP WebSocket Server in
order to send and receive SIP requests and responses
"

Considering the previous text, isn't it clear that the SIP WebSocket
Server is used by the SIP WebSocket Client for sending and receiving
SIP requests and responses, which involves SIP WebSocket Server
staying in the signaling path?


> In case of RFC 3261 SIP transports, the dialog creation happens without
> "record-route"
> whereas SIP over WebSocket transport in browser is different.
>
> Thanks
> Partha

Regards.

>
> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Jos=E9 Luis Mill=E1n
> Sent: Friday, August 24, 2012 10:19 PM
> To: Parthasarathi R
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Mandatory to implement transport comment on
> draft-ibc-sipcore-sip-websocket-02
>
> Hi Partha,
>
> Please, read inline.
>
> 2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
>> Hi Jose,
>>
>>
>>
>> Let us assume draft-ibc-sipcore-sip-websocket-02 does not change proxy
>> behavior in SIP websocket server then
>>
>> the dialog creation in the following topology breaks as follows:
>>
>>
>>
>> Topolgy:
>>
>>
>>
>>   UAC SIP Websocket client (Browser)------Proxy SIP Websocket
>> Server------UAS (RFC 3261 UA)
>>
>>
>>
>> Callflow steps:
>>
>>
>>
>> 1)    UAC SIP Websocket client initiates INVITE dialog towards SIP
> Websocket
>> server
>>
>> 2)    INVITE is forwarded towards UAS by SIP Websocket server
>
> .. which adds a Record-Route header field because it wants to stay in
> the signaling path during the dialog ;-)
>
>>
>> 3)    UAS receives INVITE and sent 200 OK response with contact address
> with
>> UDP transport
>
> it builds the 200 OK response taking the Record-Route headers from the
> received INVITE request.
>
>>
>> towards SIP Websocket server
>>
>> 4)    SIP Websocket server forwards 200 OK towards SIP Websocket client
>> without record-route
>
> It receives 200 OK with the Record-Route headers
>
>>
>> 5)    Whether SIP Websocket Client(browser) will be able to send ACK
> towards
>> UAS contact
>>
>>        address in UDP transport? No.
>>
>
> ACK request will be formed with the dialog`s route set, being the
> first hop the SIP WebSocket Server.
>
>>
>>
>> In case normal RFC 3261 application or application based on SIP websocke=
t
>> client, ACK will send towards
>>
>> UAS contact address with UDP transport as per mandatory to implement
>> transport mechanism in RFC 3261.
>>
>> As browser acting as SIP Websocket client violates the mandatory to
>> implement transport mechanism,
>>
>> the callflow breaks in this specification.
>>
>
> Why didn't you add a Record-Route header in the point '2' ( INVITE is
> forwarded towards UAS by SIP Websocket server) ?
>
> You should have added if you wanted to stay in the signaling path
> during the dialog. This is stated in RFC 3261. The reason your ACK
> does not reach the destination is a configuration problem more that
> specification definition issue.
>
>>
>>
>> In case you wish to put some other text, Please let us discuss.
>>
>>
>>
>> Thanks
>>
>> Partha
>>
>
> Regards.
>
>>
>>
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Beha=
lf
>> Of Jos=E9 Luis Mill=E1n
>> Sent: Thursday, August 23, 2012 9:46 PM
>> To: Parthasarathi R
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] Mandatory to implement transport comment on
>> draft-ibc-sipcore-sip-websocket-02
>>
>>
>>
>> Hi,
>>
>> 2012/8/23 Parthasarathi R <partha@parthasarathi.co.in>
>>
>> Hi all,
>>
>>
>>
>> As I discussed during IETF-84 meeting,  the following text is provided f=
or
>> the starting the discussion on deviation of RFC 3261 mandatory to
> implement
>> (UDP/TCP) transport in this draft.
>>
>>
>>
>> =93In case SIP WebSocket Client is a web browser and SIP Websocket Serve=
r is
> a
>> SIP proxy, then record-route header MUST be included by SIP Websocket
> server
>> in all SIP messages. This requirement assures that SIP WebSocket Client =
is
>> able to contact with UAS through SIP Websocket Server during the
> mid-dialog
>> irrespective of the contact SIP URI received from UAS =94
>>
>>
>>
>> Record-Route header field is useful for a proxy in order to remain in th=
e
>> dialog path. It is not necessary for all SIP messages. This is defined i=
n
>> RFC 3261 as 'loose routing' and draft-ibc-sipcore-sip-websocket-02 does
> not
>> change it at all.
>>
>>
>>
>> Section 2 of Apendix A (Implementation guidelines) already mentions it:
>>
>>
>>
>> "
>>
>>  The proxy performs Loose Routing and remains in dialogs path as
>>
>>    specified in [RFC3261].  Otherwise in-dialog requests would fail
>>
>>    since SIP WebSocket Clients make use of their SIP WebSocket Server in
>>
>>    order to send and receive SIP requests and responses.
>>
>> "
>>
>>
>>
>> Regards,
>>
>>
>>
>> The above normative statement will be update to  RFC 3261.
>>
>>
>>
>> Thanks
>>
>> Partha
>>
>>
>> _______________________________________________
>> 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 jmillan@aliax.net  Fri Aug 24 11:15:35 2012
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Date: Fri, 24 Aug 2012 20:15:28 +0200
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From: =?ISO-8859-1?Q?Jos=E9_Luis_Mill=E1n?= <jmillan@aliax.net>
To: Parthasarathi R <partha@parthasarathi.co.in>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Hi Partha,

I honestly think this thread is not a good starting point for
discussing the subject related to 3261 requirement for UDP and TCP
transports.

Best regards,

2012/8/24 Jos=E9 Luis Mill=E1n <jmillan@aliax.net>:
> 2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
>> Jose,
>>
>> I haven't added "record-route" in point 2 as RFC 3261 does not mandates =
to
>> add :-)
>
> It mandates to insert a Record-Route header field if the proxy wishes
> to remain on the path of future requests in a dialog.
>
> "
> 16.6 Request Forwarding
>
> If this proxy wishes to remain on the path of future requests
> in a dialog created by this request (assuming the request
> creates a dialog), it MUST insert a Record-Route header field
> value into the copy before any existing Record-Route header
> field values, even if a Route header field is already present.
> "
>
> Addition of Record-Route header field is already commented in
> draft-ibc-sipcore-sip-websocket-02, in section 2 of Appendix A:  SIP
> WebSocket Server Considerations.
>
> "
>    The proxy performs Loose Routing and remains in dialogs path as
>    specified in [RFC3261].  Otherwise in-dialog requests would fail
>    since SIP WebSocket Clients make use of their SIP WebSocket Server in
>    order to send and receive SIP requests and responses.
> "
>
> May the following (extracted from the previous text) not be clear?
>
> "
> SIP WebSocket Clients make use of their SIP WebSocket Server in
> order to send and receive SIP requests and responses
> "
>
> Considering the previous text, isn't it clear that the SIP WebSocket
> Server is used by the SIP WebSocket Client for sending and receiving
> SIP requests and responses, which involves SIP WebSocket Server
> staying in the signaling path?
>
>
>> In case of RFC 3261 SIP transports, the dialog creation happens without
>> "record-route"
>> whereas SIP over WebSocket transport in browser is different.
>>
>> Thanks
>> Partha
>
> Regards.
>
>>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Beha=
lf
>> Of Jos=E9 Luis Mill=E1n
>> Sent: Friday, August 24, 2012 10:19 PM
>> To: Parthasarathi R
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] Mandatory to implement transport comment on
>> draft-ibc-sipcore-sip-websocket-02
>>
>> Hi Partha,
>>
>> Please, read inline.
>>
>> 2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
>>> Hi Jose,
>>>
>>>
>>>
>>> Let us assume draft-ibc-sipcore-sip-websocket-02 does not change proxy
>>> behavior in SIP websocket server then
>>>
>>> the dialog creation in the following topology breaks as follows:
>>>
>>>
>>>
>>> Topolgy:
>>>
>>>
>>>
>>>   UAC SIP Websocket client (Browser)------Proxy SIP Websocket
>>> Server------UAS (RFC 3261 UA)
>>>
>>>
>>>
>>> Callflow steps:
>>>
>>>
>>>
>>> 1)    UAC SIP Websocket client initiates INVITE dialog towards SIP
>> Websocket
>>> server
>>>
>>> 2)    INVITE is forwarded towards UAS by SIP Websocket server
>>
>> .. which adds a Record-Route header field because it wants to stay in
>> the signaling path during the dialog ;-)
>>
>>>
>>> 3)    UAS receives INVITE and sent 200 OK response with contact address
>> with
>>> UDP transport
>>
>> it builds the 200 OK response taking the Record-Route headers from the
>> received INVITE request.
>>
>>>
>>> towards SIP Websocket server
>>>
>>> 4)    SIP Websocket server forwards 200 OK towards SIP Websocket client
>>> without record-route
>>
>> It receives 200 OK with the Record-Route headers
>>
>>>
>>> 5)    Whether SIP Websocket Client(browser) will be able to send ACK
>> towards
>>> UAS contact
>>>
>>>        address in UDP transport? No.
>>>
>>
>> ACK request will be formed with the dialog`s route set, being the
>> first hop the SIP WebSocket Server.
>>
>>>
>>>
>>> In case normal RFC 3261 application or application based on SIP websock=
et
>>> client, ACK will send towards
>>>
>>> UAS contact address with UDP transport as per mandatory to implement
>>> transport mechanism in RFC 3261.
>>>
>>> As browser acting as SIP Websocket client violates the mandatory to
>>> implement transport mechanism,
>>>
>>> the callflow breaks in this specification.
>>>
>>
>> Why didn't you add a Record-Route header in the point '2' ( INVITE is
>> forwarded towards UAS by SIP Websocket server) ?
>>
>> You should have added if you wanted to stay in the signaling path
>> during the dialog. This is stated in RFC 3261. The reason your ACK
>> does not reach the destination is a configuration problem more that
>> specification definition issue.
>>
>>>
>>>
>>> In case you wish to put some other text, Please let us discuss.
>>>
>>>
>>>
>>> Thanks
>>>
>>> Partha
>>>
>>
>> Regards.
>>
>>>
>>>
>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Beh=
alf
>>> Of Jos=E9 Luis Mill=E1n
>>> Sent: Thursday, August 23, 2012 9:46 PM
>>> To: Parthasarathi R
>>> Cc: sipcore@ietf.org
>>> Subject: Re: [sipcore] Mandatory to implement transport comment on
>>> draft-ibc-sipcore-sip-websocket-02
>>>
>>>
>>>
>>> Hi,
>>>
>>> 2012/8/23 Parthasarathi R <partha@parthasarathi.co.in>
>>>
>>> Hi all,
>>>
>>>
>>>
>>> As I discussed during IETF-84 meeting,  the following text is provided =
for
>>> the starting the discussion on deviation of RFC 3261 mandatory to
>> implement
>>> (UDP/TCP) transport in this draft.
>>>
>>>
>>>
>>> =93In case SIP WebSocket Client is a web browser and SIP Websocket Serv=
er is
>> a
>>> SIP proxy, then record-route header MUST be included by SIP Websocket
>> server
>>> in all SIP messages. This requirement assures that SIP WebSocket Client=
 is
>>> able to contact with UAS through SIP Websocket Server during the
>> mid-dialog
>>> irrespective of the contact SIP URI received from UAS =94
>>>
>>>
>>>
>>> Record-Route header field is useful for a proxy in order to remain in t=
he
>>> dialog path. It is not necessary for all SIP messages. This is defined =
in
>>> RFC 3261 as 'loose routing' and draft-ibc-sipcore-sip-websocket-02 does
>> not
>>> change it at all.
>>>
>>>
>>>
>>> Section 2 of Apendix A (Implementation guidelines) already mentions it:
>>>
>>>
>>>
>>> "
>>>
>>>  The proxy performs Loose Routing and remains in dialogs path as
>>>
>>>    specified in [RFC3261].  Otherwise in-dialog requests would fail
>>>
>>>    since SIP WebSocket Clients make use of their SIP WebSocket Server i=
n
>>>
>>>    order to send and receive SIP requests and responses.
>>>
>>> "
>>>
>>>
>>>
>>> Regards,
>>>
>>>
>>>
>>> The above normative statement will be update to  RFC 3261.
>>>
>>>
>>>
>>> Thanks
>>>
>>> Partha
>>>
>>>
>>> _______________________________________________
>>> 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 ibc@aliax.net  Fri Aug 24 11:42:35 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 24 Aug 2012 20:42:12 +0200
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Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> Topolgy:
>
>   UAC SIP Websocket client (Browser)------Proxy SIP Websocket
> Server------UAS (RFC 3261 UA)
>
> Callflow steps:
>
> 1)    UAC SIP Websocket client initiates INVITE dialog towards SIP Websoc=
ket
> server
>
> 2)    INVITE is forwarded towards UAS by SIP Websocket server
>
> 3)    UAS receives INVITE and sent 200 OK response with contact address w=
ith
> UDP transport
>
> towards SIP Websocket server
>
> 4)    SIP Websocket server forwards 200 OK towards SIP Websocket client
> without record-route
>
> 5)    Whether SIP Websocket Client(browser) will be able to send ACK towa=
rds
> UAS contact address in UDP transport? No.
>
> In case normal RFC 3261 application or application based on SIP websocket
> client, ACK will send towards UAS contact address with UDP transport as p=
er mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism, the callflow breaks in this specification.


Hi Partha, please read the following scenario:


UAC SIP UDP/TCP client ------ Proxy SIP ----- UAS SIP UDP/TCP/SCTP

Callflow steps:

1) UAC resolves via RFC3263 the transport and destination of the Proxy
(UDP is the first NAPTR option) and uses UDP to send an INVITE to the
AoR of UAS through the Proxy. The INVITE has a Contact URI with
;transport=3Dudp.


2) INVITE arrives to the Proxy which does NOT add Record-Route,
performs RFC3263 with the Request-URI (SCTP is the preferred
transport) and opens an SCTP connection with UAS for forwarding the
INVITE.

3) UAS receives INVITE and sent 200 OK response with Contact address
with ;transport=3Dsctp.

4) Proxy forwards the 200 OK to UAC (no Record-Route).

5) UAC should now send the ACK by opening an SCTP connection with UAS.

Can UAC do that? not, because it does NOT implement SCTP transport. So
the only solution here is for the Proxy to add Record-Route to the
initial INVITE.

Please let me know where RFC 4168 (SIP SCTP transport) talks about
this scenario and where the RFC mentions the words "Record-Route" or
"Proxy" (it does not). So, shouldn't RFC 4168 also update RFC 3261 by
stating that "If an INVITE arrives to a proxy through an SCTP
connection and the proxy forwards it via UDP or TCP then it
MUST/SHOULD add Record-Route headers"? It does not so, why should the
sip-websocket draft mandate it?

Please, reply. Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From rjsparks@nostrum.com  Fri Aug 24 11:46:03 2012
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Subject: [sipcore] AD review: draft-ietf-sipcore-rfc4244bis-08
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Summary: There are a couple of questions to discuss before progressing 
to IETF LC.

This document is much easier to follow than RFC4244. It is still 
complex, but the rewrite effort improved its readability significantly.

There are a couple of things I would like the WG to consider before 
moving into IETF LC.

1) Since the WG adopted the call-flows document, does this document 
still need Appendix A, or should any of the examples there that are not 
already in the call-flows document be moved to that document?

2) The document requires that redirect servers include an 
hi-target-param in the Contact header fields it provides, and requires 
elements recursing on redirect responses to only include 
hi-target-params that it finds in those Contacts. They are specifically 
forbidden from providing the param if there are no params in the 
Contact. Please consider saying why this is required. If I recall the 
conversation correctly, it's because the redirect server is the only 
element that can tell whether mp or rc would be appropriate. For np, 
however, it's not clear why this requirement exists.  Is it just to keep 
things simple?

3) Section 103 step 6 says "If the request clearly has a gap", but I 
don't see any description of what an element is supposed to look for to 
determine there is a gap. From a discussion with Mary, it should be as 
simple as comparing the last HI entry to the Request-URI. If that's 
right, please add that to this description, and make it clear that you 
would only add one .0. when identifying a gap, not try to figure out how 
many elements may have not added history (that is, you would never 
produce something that looked like 1.0.0.1).

4) The Changes from RFC4244 section doesn't call out the new .0. gap 
detection/representation mechanic.

5) The backwards compatibility consideration focuses on exploring how 
RFC4244 elements will deal with messages created following this 
specification. There should be some additional exploration of how 
elements implementing this specification will deal with messages created 
by RFC4244 elements. Those messages could be missing things this 
specification says MUST be there. Can we make it more likely that an 
implementer won't create code that rejects such messages as malformed?


Nits:

At the definition of hi-index, the document talks of "nesting of 
requests". It's the only place the word nesting occurs in the document. 
Could you provide a description/definition or a pointer to one?

The description of using an internal cache to describe how an element 
will construct the history-info values it emits is likely to draw 
discussion during IETF/IESG review. It uses 2119 language to describe a 
possible implementation. Please consider changing the phrasing to make 
it clear that the normative constraints are on what's emitted, not on 
they way you store the information internally.

First paragraph of 10.4 still says "two header field parameters" and 
"Both", but talks about three.

"As for the behavior of the entity followings have changed" does not parse.

In the first example in appendix A, the Contact in F3 has no parameters, 
but the corresponding H-I value in F4 has an mp=1. The version of this 
example in the callflow doc has the mp=1 in the Contact in the 302.


From ibc@aliax.net  Fri Aug 24 11:52:04 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 24 Aug 2012 20:51:42 +0200
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To: Brett Tate <brett@broadsoft.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-sip-websocket@tools.ietf.org" <draft-ietf-sipcore-sip-websocket@tools.ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-sip-websocket-02: comments
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2012/8/24 Brett Tate <brett@broadsoft.com>:
> The following are some minor comments concerning draft-ietf-sipcore-sip-w=
ebsocket-02.

Hi Brett, thanks a lot. Please read inline:



> Section 8.2:
>
> - The response examples should likely indicate the Via "received" paramet=
er when Via transport is UDP.

Adding Via "received" is just mandatory in case the source IP of the
request does not match the Via sentby-host component.


> - Max-Forwards should likely be removed from response examples.

Done in our working copy for the next version.


> Section A.1: "hostpart" potentially should be "hostport".

Done in our working copy for the next version.


Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Fri, 24 Aug 2012 12:13:39 -0700
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 24 Aug 2012 21:27:54 +0200
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To: Brett Tate <brett@broadsoft.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "draft-ietf-sipcore-sip-websocket@tools.ietf.org" <draft-ietf-sipcore-sip-websocket@tools.ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-sip-websocket-02: comments
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2012/8/24 Brett Tate <brett@broadsoft.com>:
> Oops, sorry for resend.  I just noticed the ip-addresses within F9, F10, =
F11; only F4 should be changed.

You are right, I missed that. Fixed in our local copy.

Thanks a lot.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: [sipcore] I-D Action: draft-ietf-sipcore-proxy-feature-07.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : Mechanism to indicate support of features and capabiliti=
es in the Session Initiation Protocol (SIP)
	Author(s)       : Christer Holmberg
                          Ivo Sedlacek
                          Hadriel Kaplan
	Filename        : draft-ietf-sipcore-proxy-feature-07.txt
	Pages           : 21
	Date            : 2012-08-24

Abstract:
   This specification defines a new SIP header field, Feature-Caps, to
   convey feature capability indicators, which are used by SIP entities
   not represented by the URI of the Contact header field to indicate
   support of features and capabilities, where media feature tags cannot
   be used to indicate the support.

   This specification also defines feature capability indicators, and
   creates a new IANA registry, "Proxy-Feature Feature Capability
   Indicator Trees", for registering feature capability indicators.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-proxy-feature

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-proxy-feature-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-proxy-feature-07


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


From christer.holmberg@ericsson.com  Fri Aug 24 15:13:00 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: [sipcore] Draft new version: proxy-feature-07
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Hi,

I've submitted a new version (-07) of proxy-feature, fixing the remaining i=
ssues.

Regards,

Christer

From rjsparks@nostrum.com  Fri Aug 24 15:23:10 2012
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Subject: Re: [sipcore] Draft new version: proxy-feature-07
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SIPCORE -

There is quite a bit of new text here. It addresses my review comments.

Please make sure you are all OK with these changes:

<https://www.ietf.org/rfcdiff?url1=draft-ietf-sipcore-proxy-feature-05&difftype=--html&submit=Go!&url2=draft-ietf-sipcore-proxy-feature-07>

RjS

On 8/24/12 5:12 PM, Christer Holmberg wrote:
> Hi,
>
> I've submitted a new version (-07) of proxy-feature, fixing the remaining issues.
>
> Regards,
>
> Christer


From internet-drafts@ietf.org  Fri Aug 24 16:47:39 2012
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-proxy-feature-08.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : Mechanism to indicate support of features and capabiliti=
es in the Session Initiation Protocol (SIP)
	Author(s)       : Christer Holmberg
                          Ivo Sedlacek
                          Hadriel Kaplan
	Filename        : draft-ietf-sipcore-proxy-feature-08.txt
	Pages           : 21
	Date            : 2012-08-24

Abstract:
   This specification defines a new SIP header field, Feature-Caps, to
   convey feature capability indicators, which are used by SIP entities
   not represented by the URI of the Contact header field to indicate
   support of features and capabilities, where media feature tags cannot
   be used to indicate the support.

   This specification also defines feature capability indicators, and
   creates a new IANA registry, "Proxy-Feature Feature Capability
   Indicator Trees", for registering feature capability indicators.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-proxy-feature

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-proxy-feature-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-proxy-feature-08


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


From christer.holmberg@ericsson.com  Fri Aug 24 16:50:04 2012
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Subject: [sipcore] Draft new version: proxy-feature-08
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Hi,

Yet another new version (-08) of proxy-feature, due to some xml2rfc feature=
s that didn't work as expected :)

The only difference between -07 and -08 is that I manually had to add "XXXX=
" (which will be replaced by the assigned RFC number later) in a few places=
.

Regards,

Christer


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Yet another new version (-08) of proxy-feature, due t=
o some xml2rfc features that didn't work as expected :)</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">The only difference between -07 and -08 is that I man=
ually had to add &quot;XXXX&quot; (which will be replaced by the assigned R=
FC number later) in a few places.</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>
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From partha@parthasarathi.co.in  Sat Aug 25 08:51:50 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: =?utf-8?Q?'I=C3=B1aki_Baz_Castillo'?= <ibc@aliax.net>
References: <000d01cd812e$46bb4850$d431d8f0$@co.in> <CABw3bnPMKtWFq7eGQC6fNKym-6YCnaCer07iR=DrwTk05Yn66g@mail.gmail.com> <000f01cd8214$9d255a00$d7700e00$@co.in> <CALiegf=ZTtpdgFuwuUpQ7c_dyokHO4rpqYPtPqWn=jmX_JbyWQ@mail.gmail.com>
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Date: Sat, 25 Aug 2012 21:21:37 +0530
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Cc: sipcore@ietf.org, =?utf-8?Q?'Jos=C3=A9_Luis_Mill=C3=A1n'?= <jmillan@aliax.net>
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Inaki,

I could not see the relation between UAC SIP Websocket client violates =
RFC 3261 normative statement and your callflow. In your callflow, the =
dialog fails because of the extension is not supported by UAC.

Thanks
Partha=20

-----Original Message-----
From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]=20
Sent: Saturday, August 25, 2012 12:12 AM
To: Parthasarathi R
Cc: Jos=C3=A9 Luis Mill=C3=A1n; sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on =
draft-ibc-sipcore-sip-websocket-02

2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> Topolgy:
>
>   UAC SIP Websocket client (Browser)------Proxy SIP Websocket
> Server------UAS (RFC 3261 UA)
>
> Callflow steps:
>
> 1)    UAC SIP Websocket client initiates INVITE dialog towards SIP =
Websocket
> server
>
> 2)    INVITE is forwarded towards UAS by SIP Websocket server
>
> 3)    UAS receives INVITE and sent 200 OK response with contact =
address with
> UDP transport
>
> towards SIP Websocket server
>
> 4)    SIP Websocket server forwards 200 OK towards SIP Websocket =
client
> without record-route
>
> 5)    Whether SIP Websocket Client(browser) will be able to send ACK =
towards
> UAS contact address in UDP transport? No.
>
> In case normal RFC 3261 application or application based on SIP =
websocket
> client, ACK will send towards UAS contact address with UDP transport =
as per mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism, the callflow breaks in this =
specification.


Hi Partha, please read the following scenario:


UAC SIP UDP/TCP client ------ Proxy SIP ----- UAS SIP UDP/TCP/SCTP

Callflow steps:

1) UAC resolves via RFC3263 the transport and destination of the Proxy
(UDP is the first NAPTR option) and uses UDP to send an INVITE to the
AoR of UAS through the Proxy. The INVITE has a Contact URI with
;transport=3Dudp.


2) INVITE arrives to the Proxy which does NOT add Record-Route,
performs RFC3263 with the Request-URI (SCTP is the preferred
transport) and opens an SCTP connection with UAS for forwarding the
INVITE.

3) UAS receives INVITE and sent 200 OK response with Contact address
with ;transport=3Dsctp.

4) Proxy forwards the 200 OK to UAC (no Record-Route).

5) UAC should now send the ACK by opening an SCTP connection with UAS.

Can UAC do that? not, because it does NOT implement SCTP transport. So
the only solution here is for the Proxy to add Record-Route to the
initial INVITE.

Please let me know where RFC 4168 (SIP SCTP transport) talks about
this scenario and where the RFC mentions the words "Record-Route" or
"Proxy" (it does not). So, shouldn't RFC 4168 also update RFC 3261 by
stating that "If an INVITE arrives to a proxy through an SCTP
connection and the proxy forwards it via UDP or TCP then it
MUST/SHOULD add Record-Route headers"? It does not so, why should the
sip-websocket draft mandate it?

Please, reply. Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>


From partha@parthasarathi.co.in  Sat Aug 25 09:07:11 2012
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Inaki/Jose,

Your current argument in the different mail of this same mail thread =
argue for not to close this open item which allows to relaxing the =
criteria for mandatory to implement (UDP/TCP) transport. If so, SIP over =
Websocket client MUST support UDP & TCP as SIP transport to meet RFC =
3261 normative statement. Please note that SIP over Websocket =
implementation with UDP & TCP as SIP transport is possible by =
Linux/Windows application but not by Browser as of now.

I'll like to hear more opinion from different folks before closing this =
open item.

Thanks
Partha

-----Original Message-----
From: Parthasarathi R [mailto:partha@parthasarathi.co.in]=20
Sent: Friday, August 24, 2012 11:11 PM
To: 'I=C3=B1aki Baz Castillo'
Cc: 'Jos=C3=A9 Luis Mill=C3=A1n'; 'sipcore@ietf.org'
Subject: RE: [sipcore] Mandatory to implement transport comment on =
draft-ibc-sipcore-sip-websocket-02

Inaki,

Paul started the relaxing the criteria for mandatory to implement =
transport mail thread during WG adoption of SIP over websocket draft and =
one of the mail is =
http://www.ietf.org/mail-archive/web/sipcore/current/msg04930.html. My =
understanding of the mail thread conclusion was that narrow approach of =
providing leniency for mandatory to implement transport in SIP over =
Websocket draft itself.  I noticed in the current document that there is =
no relevant text for this leniency and so raised the issue in IETF-84 =
SIPCore meeting. As per IETF-84 SIPCore meeting discussion, I started =
this mail thread.

In case you are not in favor of narrow approach proposed by Paul, then =
it is a separate discussion. Also you have indicated that the narrow =
approach needs more text, then please suggest the text and let us =
discuss.

Thanks
Partha=20

-----Original Message-----
From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]=20
Sent: Friday, August 24, 2012 10:09 PM
To: Parthasarathi R
Cc: Jos=C3=A9 Luis Mill=C3=A1n; sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on =
draft-ibc-sipcore-sip-websocket-02

2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> In case normal RFC 3261 application or application based on SIP =
websocket
> client, ACK will send towards
>
> UAS contact address with UDP transport as per mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism,
>
> the callflow breaks in this specification.

Hi Partha. As said many times this draft is not *just* about web
browsers. Please imagine that new versions of iOS and Android come
with a WebSocket Client stack (so native apps can use it). Should we
also add them to the draft?:

=E2=80=9CIn case SIP WebSocket Client is a web browser or a native iOS =
or
Android application, and SIP Websocket Server is a SIP proxy, then..."

Ugly, agree?


Anyhow, for the case of a web browser (which indeed cannot send a UDP
datagram) the draft already describes a full guideline in the Appendix
section:

http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02#appendix-A

IMHO that's enough. The mandatory sections of the draft should not
focus just on web browsers.


BTW your text is incomplete:


=E2=80=9CIn case SIP WebSocket Client is a web browser and SIP Websocket
Server is a SIP proxy, then record-route header MUST be included by
SIP Websocket server in all SIP messages. This requirement assures
that SIP WebSocket Client is able to contact with UAS through SIP
Websocket Server during the mid-dialog irrespective of the contact SIP
URI received from UAS =E2=80=9D

That's not enough. If you want to also *receive* incoming requests you
need to mandate Outbound RFC 5626 (since the proxy cannot establish a
WebSocket connection with a web browser). But we don't want to mandate
Outbound (it was discussed and voted in Paris) since the draft just
describes a SIP transport and does not assume specific topologies or
UA devices.

IMHO the Appendix guideline (which was already discussed in this WG)
is the appropriate section for talking about specific common
scenarios. The draft should not "update" RFC 3261 just for the case of
web browsers.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>


From christer.holmberg@ericsson.com  Sat Aug 25 11:50:41 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, =?Windows-1252?Q?=27I=F1aki_Baz_Castillo=27?= <ibc@aliax.net>, =?Windows-1252?Q?=27Jos=E9_Luis_Mill=E1n=27?= <jmillan@aliax.net>
Date: Sat, 25 Aug 2012 20:46:07 +0200
Thread-Topic: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Mandatory to implement transport comment on	draft-ibc-sipcore-sip-websocket-02
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Hi,

I am not sure who I agree/disagree with, but my opinion is that a WebSocket=
 UA shall not be mandated to implement UDP and TCP.

A WebSocket UA might work in an environement where it can't use UDP or TCP =
anyway. For example, if a JavaScript SIP UA in a browser technically can on=
ly use WebSocket for SIP, why mandate it to implement something that it can=
't use???

As far as the WebServer SIP proxy it concerned, it MUST Record-Route.

Regards,

Christer

________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Part=
hasarathi R [partha@parthasarathi.co.in]
Sent: Saturday, August 25, 2012 7:06 PM
To: 'I=F1aki Baz Castillo'; 'Jos=E9 Luis Mill=E1n'
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on      dra=
ft-ibc-sipcore-sip-websocket-02

Inaki/Jose,

Your current argument in the different mail of this same mail thread argue =
for not to close this open item which allows to relaxing the criteria for m=
andatory to implement (UDP/TCP) transport. If so, SIP over Websocket client=
 MUST support UDP & TCP as SIP transport to meet RFC 3261 normative stateme=
nt. Please note that SIP over Websocket implementation with UDP & TCP as SI=
P transport is possible by Linux/Windows application but not by Browser as =
of now.

I'll like to hear more opinion from different folks before closing this ope=
n item.

Thanks
Partha

-----Original Message-----
From: Parthasarathi R [mailto:partha@parthasarathi.co.in]
Sent: Friday, August 24, 2012 11:11 PM
To: 'I=F1aki Baz Castillo'
Cc: 'Jos=E9 Luis Mill=E1n'; 'sipcore@ietf.org'
Subject: RE: [sipcore] Mandatory to implement transport comment on draft-ib=
c-sipcore-sip-websocket-02

Inaki,

Paul started the relaxing the criteria for mandatory to implement transport=
 mail thread during WG adoption of SIP over websocket draft and one of the =
mail is http://www.ietf.org/mail-archive/web/sipcore/current/msg04930.html.=
 My understanding of the mail thread conclusion was that narrow approach of=
 providing leniency for mandatory to implement transport in SIP over Websoc=
ket draft itself.  I noticed in the current document that there is no relev=
ant text for this leniency and so raised the issue in IETF-84 SIPCore meeti=
ng. As per IETF-84 SIPCore meeting discussion, I started this mail thread.

In case you are not in favor of narrow approach proposed by Paul, then it i=
s a separate discussion. Also you have indicated that the narrow approach n=
eeds more text, then please suggest the text and let us discuss.

Thanks
Partha

-----Original Message-----
From: I=F1aki Baz Castillo [mailto:ibc@aliax.net]
Sent: Friday, August 24, 2012 10:09 PM
To: Parthasarathi R
Cc: Jos=E9 Luis Mill=E1n; sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ib=
c-sipcore-sip-websocket-02

2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
> In case normal RFC 3261 application or application based on SIP websocket
> client, ACK will send towards
>
> UAS contact address with UDP transport as per mandatory to implement
> transport mechanism in RFC 3261.
>
> As browser acting as SIP Websocket client violates the mandatory to
> implement transport mechanism,
>
> the callflow breaks in this specification.

Hi Partha. As said many times this draft is not *just* about web
browsers. Please imagine that new versions of iOS and Android come
with a WebSocket Client stack (so native apps can use it). Should we
also add them to the draft?:

=93In case SIP WebSocket Client is a web browser or a native iOS or
Android application, and SIP Websocket Server is a SIP proxy, then..."

Ugly, agree?


Anyhow, for the case of a web browser (which indeed cannot send a UDP
datagram) the draft already describes a full guideline in the Appendix
section:

http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02#appendix-A

IMHO that's enough. The mandatory sections of the draft should not
focus just on web browsers.


BTW your text is incomplete:


=93In case SIP WebSocket Client is a web browser and SIP Websocket
Server is a SIP proxy, then record-route header MUST be included by
SIP Websocket server in all SIP messages. This requirement assures
that SIP WebSocket Client is able to contact with UAS through SIP
Websocket Server during the mid-dialog irrespective of the contact SIP
URI received from UAS =94

That's not enough. If you want to also *receive* incoming requests you
need to mandate Outbound RFC 5626 (since the proxy cannot establish a
WebSocket connection with a web browser). But we don't want to mandate
Outbound (it was discussed and voted in Paris) since the draft just
describes a SIP transport and does not assume specific topologies or
UA devices.

IMHO the Appendix guideline (which was already discussed in this WG)
is the appropriate section for talking about specific common
scenarios. The draft should not "update" RFC 3261 just for the case of
web browsers.


Regards.



--
I=F1aki Baz Castillo
<ibc@aliax.net>

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

From jmillan@aliax.net  Sun Aug 26 03:09:40 2012
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Date: Sun, 26 Aug 2012 12:09:37 +0200
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From: =?ISO-8859-1?Q?Jos=E9_Luis_Mill=E1n?= <jmillan@aliax.net>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "sipcore@ietf.org" <sipcore@ietf.org>, Parthasarathi R <partha@parthasarathi.co.in>
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Hi,

Most WebSocket implementations do not allow (and there is no evidence
they will in the near future) receiving incoming connections. They can
only establish outgoing connections with WebSocket server
implementations.

The specification of this new transport allows SIP to be used as a
session establishment protocol in the World Wide Web, in scenarios
where non of other transport protocols defined for SIP can be used.

SIP is the one entering in the WWW within this draft, allowing
massively used applications to use our loving protocol. Most of those
applications do not implement any existing SIP transport but
WebSocket, the one we are adopting in this document.

This specification defines the way SIP WebSocket Client and Server
communicate and the mechanisms for the SIP element acting as WebSocket
Server to interact with other non WebSocket SIP elements.

We authors propose the following statements to be used in this
specification according to this task:

"
SIP WebSocket Servers implementing this specification MUST implement
UDP and TCP as stated in [RFC 3261].

SIP WebSocket Clients implementing this specification MAY implement
UDP and TCP protocols, being their interoperability guaranteed by SIP
WebSocket Servers which MUST be used as the first hop in the signaling
path.

"

Of course, by definition, a SIP WebSocket server MUST Record-Route.


Any comment about the proposed statements is appreciated.

Regards,

2012/8/25 Christer Holmberg <christer.holmberg@ericsson.com>:
> Hi,
>
> I am not sure who I agree/disagree with, but my opinion is that a WebSock=
et UA shall not be mandated to implement UDP and TCP.
>
> A WebSocket UA might work in an environement where it can't use UDP or TC=
P anyway. For example, if a JavaScript SIP UA in a browser technically can =
only use WebSocket for SIP, why mandate it to implement something that it c=
an't use???
>
> As far as the WebServer SIP proxy it concerned, it MUST Record-Route.
>
> Regards,
>
> Christer
>
> ________________________________________
> From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Pa=
rthasarathi R [partha@parthasarathi.co.in]
> Sent: Saturday, August 25, 2012 7:06 PM
> To: 'I=F1aki Baz Castillo'; 'Jos=E9 Luis Mill=E1n'
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Mandatory to implement transport comment on      d=
raft-ibc-sipcore-sip-websocket-02
>
> Inaki/Jose,
>
> Your current argument in the different mail of this same mail thread argu=
e for not to close this open item which allows to relaxing the criteria for=
 mandatory to implement (UDP/TCP) transport. If so, SIP over Websocket clie=
nt MUST support UDP & TCP as SIP transport to meet RFC 3261 normative state=
ment. Please note that SIP over Websocket implementation with UDP & TCP as =
SIP transport is possible by Linux/Windows application but not by Browser a=
s of now.
>
> I'll like to hear more opinion from different folks before closing this o=
pen item.
>
> Thanks
> Partha
>
> -----Original Message-----
> From: Parthasarathi R [mailto:partha@parthasarathi.co.in]
> Sent: Friday, August 24, 2012 11:11 PM
> To: 'I=F1aki Baz Castillo'
> Cc: 'Jos=E9 Luis Mill=E1n'; 'sipcore@ietf.org'
> Subject: RE: [sipcore] Mandatory to implement transport comment on draft-=
ibc-sipcore-sip-websocket-02
>
> Inaki,
>
> Paul started the relaxing the criteria for mandatory to implement transpo=
rt mail thread during WG adoption of SIP over websocket draft and one of th=
e mail is http://www.ietf.org/mail-archive/web/sipcore/current/msg04930.htm=
l. My understanding of the mail thread conclusion was that narrow approach =
of providing leniency for mandatory to implement transport in SIP over Webs=
ocket draft itself.  I noticed in the current document that there is no rel=
evant text for this leniency and so raised the issue in IETF-84 SIPCore mee=
ting. As per IETF-84 SIPCore meeting discussion, I started this mail thread=
.
>
> In case you are not in favor of narrow approach proposed by Paul, then it=
 is a separate discussion. Also you have indicated that the narrow approach=
 needs more text, then please suggest the text and let us discuss.
>
> Thanks
> Partha
>
> -----Original Message-----
> From: I=F1aki Baz Castillo [mailto:ibc@aliax.net]
> Sent: Friday, August 24, 2012 10:09 PM
> To: Parthasarathi R
> Cc: Jos=E9 Luis Mill=E1n; sipcore@ietf.org
> Subject: Re: [sipcore] Mandatory to implement transport comment on draft-=
ibc-sipcore-sip-websocket-02
>
> 2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
>> In case normal RFC 3261 application or application based on SIP websocke=
t
>> client, ACK will send towards
>>
>> UAS contact address with UDP transport as per mandatory to implement
>> transport mechanism in RFC 3261.
>>
>> As browser acting as SIP Websocket client violates the mandatory to
>> implement transport mechanism,
>>
>> the callflow breaks in this specification.
>
> Hi Partha. As said many times this draft is not *just* about web
> browsers. Please imagine that new versions of iOS and Android come
> with a WebSocket Client stack (so native apps can use it). Should we
> also add them to the draft?:
>
> =93In case SIP WebSocket Client is a web browser or a native iOS or
> Android application, and SIP Websocket Server is a SIP proxy, then..."
>
> Ugly, agree?
>
>
> Anyhow, for the case of a web browser (which indeed cannot send a UDP
> datagram) the draft already describes a full guideline in the Appendix
> section:
>
> http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02#appendix-A
>
> IMHO that's enough. The mandatory sections of the draft should not
> focus just on web browsers.
>
>
> BTW your text is incomplete:
>
>
> =93In case SIP WebSocket Client is a web browser and SIP Websocket
> Server is a SIP proxy, then record-route header MUST be included by
> SIP Websocket server in all SIP messages. This requirement assures
> that SIP WebSocket Client is able to contact with UAS through SIP
> Websocket Server during the mid-dialog irrespective of the contact SIP
> URI received from UAS =94
>
> That's not enough. If you want to also *receive* incoming requests you
> need to mandate Outbound RFC 5626 (since the proxy cannot establish a
> WebSocket connection with a web browser). But we don't want to mandate
> Outbound (it was discussed and voted in Paris) since the draft just
> describes a SIP transport and does not assume specific topologies or
> UA devices.
>
> IMHO the Appendix guideline (which was already discussed in this WG)
> is the appropriate section for talking about specific common
> scenarios. The draft should not "update" RFC 3261 just for the case of
> web browsers.
>
>
> Regards.
>
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From christer.holmberg@ericsson.com  Sun Aug 26 04:35:53 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?Windows-1252?Q?Jos=E9_Luis_Mill=E1n?= <jmillan@aliax.net>
Date: Sun, 26 Aug 2012 13:35:39 +0200
Thread-Topic: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Cc: =?Windows-1252?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "sipcore@ietf.org" <sipcore@ietf.org>, Parthasarathi R <partha@parthasarathi.co.in>
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Hi,

>Most WebSocket implementations do not allow (and there is no evidence
>they will in the near future) receiving incoming connections. They can
>only establish outgoing connections with WebSocket server
>implementations.

Exactly.

>The specification of this new transport allows SIP to be used as a
>session establishment protocol in the World Wide Web, in scenarios
>where non of other transport protocols defined for SIP can be used.

Exactly. It's a completely new environment.

>SIP is the one entering in the WWW within this draft, allowing
>massively used applications to use our loving protocol. Most of those
>applications do not implement any existing SIP transport but
>WebSocket, the one we are adopting in this document.

Exactly.

>This specification defines the way SIP WebSocket Client and Server
>communicate and the mechanisms for the SIP element acting as WebSocket
>Server to interact with other non WebSocket SIP elements.

We authors propose the following statements to be used in this
specification according to this task:

>"
>SIP WebSocket Servers implementing this specification MUST implement
>UDP and TCP as stated in [RFC 3261].

I would suggest to say that the MUST applies if such server also communicat=
e SIP using non-WS interfaces.

Because, there could be a SIP WebSocket Server that *only* provides service=
s to WS clients.

Maybe something like:

"SIP WebSocket Servers implementing this specification, and that also suppo=
rt SIP using the transport mechanisms defined in [RFC 3261], MUST implement=
 both UDP and TCP, as stated in the RFC"

>>SIP WebSocket Clients implementing this specification MAY implement
>>UDP and TCP protocols, being their interoperability guaranteed by SIP
>>WebSocket Servers which MUST be used as the first hop in the signaling
>>path."

I can't parse the sentence.

Maybe something like:

"As a SIP WebSocket Client will always communicate with a SIP WebSocket Ser=
ver, and since some environments do not enable SIP WebSocket Clients to use=
 other UDP and TCP as SIP transport protocols for SIP, a SIP WebSocket Clie=
nt is not mandated to implement support of UDP and TCP."

Regards,

Christer






>Of course, by definition, a SIP WebSocket server MUST Record-Route.
>
>Any comment about the proposed statements is appreciated.


2012/8/25 Christer Holmberg <christer.holmberg@ericsson.com>:
> Hi,
>
> I am not sure who I agree/disagree with, but my opinion is that a WebSock=
et UA shall not be mandated to implement UDP and TCP.
>
> A WebSocket UA might work in an environement where it can't use UDP or TC=
P anyway. For example, if a JavaScript SIP UA in a browser technically can =
only use WebSocket for SIP, why mandate it to implement something that it c=
an't use???
>
> As far as the WebServer SIP proxy it concerned, it MUST Record-Route.
>
> Regards,
>
> Christer
>
> ________________________________________
> From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Pa=
rthasarathi R [partha@parthasarathi.co.in]
> Sent: Saturday, August 25, 2012 7:06 PM
> To: 'I=F1aki Baz Castillo'; 'Jos=E9 Luis Mill=E1n'
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Mandatory to implement transport comment on      d=
raft-ibc-sipcore-sip-websocket-02
>
> Inaki/Jose,
>
> Your current argument in the different mail of this same mail thread argu=
e for not to close this open item which allows to relaxing the criteria for=
 mandatory to implement (UDP/TCP) transport. If so, SIP over Websocket clie=
nt MUST support UDP & TCP as SIP transport to meet RFC 3261 normative state=
ment. Please note that SIP over Websocket implementation with UDP & TCP as =
SIP transport is possible by Linux/Windows application but not by Browser a=
s of now.
>
> I'll like to hear more opinion from different folks before closing this o=
pen item.
>
> Thanks
> Partha
>
> -----Original Message-----
> From: Parthasarathi R [mailto:partha@parthasarathi.co.in]
> Sent: Friday, August 24, 2012 11:11 PM
> To: 'I=F1aki Baz Castillo'
> Cc: 'Jos=E9 Luis Mill=E1n'; 'sipcore@ietf.org'
> Subject: RE: [sipcore] Mandatory to implement transport comment on draft-=
ibc-sipcore-sip-websocket-02
>
> Inaki,
>
> Paul started the relaxing the criteria for mandatory to implement transpo=
rt mail thread during WG adoption of SIP over websocket draft and one of th=
e mail is http://www.ietf.org/mail-archive/web/sipcore/current/msg04930.htm=
l. My understanding of the mail thread conclusion was that narrow approach =
of providing leniency for mandatory to implement transport in SIP over Webs=
ocket draft itself.  I noticed in the current document that there is no rel=
evant text for this leniency and so raised the issue in IETF-84 SIPCore mee=
ting. As per IETF-84 SIPCore meeting discussion, I started this mail thread=
.
>
> In case you are not in favor of narrow approach proposed by Paul, then it=
 is a separate discussion. Also you have indicated that the narrow approach=
 needs more text, then please suggest the text and let us discuss.
>
> Thanks
> Partha
>
> -----Original Message-----
> From: I=F1aki Baz Castillo [mailto:ibc@aliax.net]
> Sent: Friday, August 24, 2012 10:09 PM
> To: Parthasarathi R
> Cc: Jos=E9 Luis Mill=E1n; sipcore@ietf.org
> Subject: Re: [sipcore] Mandatory to implement transport comment on draft-=
ibc-sipcore-sip-websocket-02
>
> 2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
>> In case normal RFC 3261 application or application based on SIP websocke=
t
>> client, ACK will send towards
>>
>> UAS contact address with UDP transport as per mandatory to implement
>> transport mechanism in RFC 3261.
>>
>> As browser acting as SIP Websocket client violates the mandatory to
>> implement transport mechanism,
>>
>> the callflow breaks in this specification.
>
> Hi Partha. As said many times this draft is not *just* about web
> browsers. Please imagine that new versions of iOS and Android come
> with a WebSocket Client stack (so native apps can use it). Should we
> also add them to the draft?:
>
> =93In case SIP WebSocket Client is a web browser or a native iOS or
> Android application, and SIP Websocket Server is a SIP proxy, then..."
>
> Ugly, agree?
>
>
> Anyhow, for the case of a web browser (which indeed cannot send a UDP
> datagram) the draft already describes a full guideline in the Appendix
> section:
>
> http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02#appendix-A
>
> IMHO that's enough. The mandatory sections of the draft should not
> focus just on web browsers.
>
>
> BTW your text is incomplete:
>
>
> =93In case SIP WebSocket Client is a web browser and SIP Websocket
> Server is a SIP proxy, then record-route header MUST be included by
> SIP Websocket server in all SIP messages. This requirement assures
> that SIP WebSocket Client is able to contact with UAS through SIP
> Websocket Server during the mid-dialog irrespective of the contact SIP
> URI received from UAS =94
>
> That's not enough. If you want to also *receive* incoming requests you
> need to mandate Outbound RFC 5626 (since the proxy cannot establish a
> WebSocket connection with a web browser). But we don't want to mandate
> Outbound (it was discussed and voted in Paris) since the draft just
> describes a SIP transport and does not assume specific topologies or
> UA devices.
>
> IMHO the Appendix guideline (which was already discussed in this WG)
> is the appropriate section for talking about specific common
> scenarios. The draft should not "update" RFC 3261 just for the case of
> web browsers.
>
>
> Regards.
>
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore=

From pkyzivat@alum.mit.edu  Mon Aug 27 13:01:52 2012
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Date: Mon, 27 Aug 2012 16:01:49 -0400
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Subject: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-08
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Sorry to do this again, but we need to do another WGLC on 
draft-ietf-sipcore-proxy-feature-08.

The WGLC will run for a week. It will conclude at 23:59 UTC on Monday, 
September 3, 2012. (A week should be enough considering the nature of 
the changes since the last WGLC.)

This covers changes since the last WGLC, which was for the -02 version. 
Those changes were to cover things I found while writing the document up 
for submission to the iesg, and things found by Robert Sparks in his AD 
review. Most of those things were editorial, but some of them were quite 
heavy editorial, such as rearranging sections of the document. (This 
causes the diff to version -02 to be largely worthless.)

There was one significant change that isn't editorial. We changed the 
review policy for registering new feature capability indicators in the 
global tree from Expert Reveiw to Specification Required. This isn't a 
big change, since we were always requiring a specification. But it is a 
better fit. (This doesn't require an RFC, only a specification that can 
be referenced permanently.)

Please look at this and then indicate you acceptance or issues here on 
the mailing list. We really need responses to indicate that it has been 
reviewed.

Some helpful links:

The draft version being reviewed:
http://tools.ietf.org/html/draft-ietf-sipcore-proxy-feature-08

Diff from version -06 (recent changes):
http://tools.ietf.org/rfcdiff?url2=draft-ietf-sipcore-proxy-feature-08.txt;url1=draft-ietf-sipcore-proxy-feature-06.txt

Diff from version -02 (the subject of the last WGLC):
http://tools.ietf.org/rfcdiff?url2=draft-ietf-sipcore-proxy-feature-08.txt;url1=draft-ietf-sipcore-proxy-feature-02.txt

	Thanks,
	Paul (as co-chair)
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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2012/8/26 Christer Holmberg <christer.holmberg@ericsson.com>:
> Hi,
>
>>Most WebSocket implementations do not allow (and there is no evidence
>>they will in the near future) receiving incoming connections. They can
>>only establish outgoing connections with WebSocket server
>>implementations.
>
> Exactly.
>
>>The specification of this new transport allows SIP to be used as a
>>session establishment protocol in the World Wide Web, in scenarios
>>where non of other transport protocols defined for SIP can be used.
>
> Exactly. It's a completely new environment.
>
>>SIP is the one entering in the WWW within this draft, allowing
>>massively used applications to use our loving protocol. Most of those
>>applications do not implement any existing SIP transport but
>>WebSocket, the one we are adopting in this document.
>
> Exactly.
>
>>This specification defines the way SIP WebSocket Client and Server
>>communicate and the mechanisms for the SIP element acting as WebSocket
>>Server to interact with other non WebSocket SIP elements.
>
> We authors propose the following statements to be used in this
> specification according to this task:
>
>>"
>>SIP WebSocket Servers implementing this specification MUST implement
>>UDP and TCP as stated in [RFC 3261].
>
> I would suggest to say that the MUST applies if such server also communic=
ate SIP using non-WS interfaces.
>
> Because, there could be a SIP WebSocket Server that *only* provides servi=
ces to WS clients.
>
> Maybe something like:
>
> "SIP WebSocket Servers implementing this specification, and that also sup=
port SIP using the transport mechanisms defined in [RFC 3261], MUST impleme=
nt both UDP and TCP, as stated in the RFC"
>

Although it could seem logical at a first glance, such a statement
breaks the basic level of protocol interoperability. This limitation
is not imposed by the new transport and hence it seems reasonable that
the SIP WebSocket Server MUST implement UDP and TCP as stated in [RFC
3261].

do you agree?

>>>SIP WebSocket Clients implementing this specification MAY implement
>>>UDP and TCP protocols, being their interoperability guaranteed by SIP
>>>WebSocket Servers which MUST be used as the first hop in the signaling
>>>path."
>
> I can't parse the sentence.
>
> Maybe something like:
>
> "As a SIP WebSocket Client will always communicate with a SIP WebSocket S=
erver, and since some environments do not enable SIP WebSocket Clients to u=
se other UDP and TCP as SIP transport protocols for SIP, a SIP WebSocket Cl=
ient is not mandated to implement support of UDP and TCP."
>

Agree with this.

> Regards,
>
> Christer
>
>

Thank you for your comments Christer,

Regards

>
>
>
>
>>Of course, by definition, a SIP WebSocket server MUST Record-Route.
>>
>>Any comment about the proposed statements is appreciated.
>
>
> 2012/8/25 Christer Holmberg <christer.holmberg@ericsson.com>:
>> Hi,
>>
>> I am not sure who I agree/disagree with, but my opinion is that a WebSoc=
ket UA shall not be mandated to implement UDP and TCP.
>>
>> A WebSocket UA might work in an environement where it can't use UDP or T=
CP anyway. For example, if a JavaScript SIP UA in a browser technically can=
 only use WebSocket for SIP, why mandate it to implement something that it =
can't use???
>>
>> As far as the WebServer SIP proxy it concerned, it MUST Record-Route.
>>
>> Regards,
>>
>> Christer
>>
>> ________________________________________
>> From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of P=
arthasarathi R [partha@parthasarathi.co.in]
>> Sent: Saturday, August 25, 2012 7:06 PM
>> To: 'I=F1aki Baz Castillo'; 'Jos=E9 Luis Mill=E1n'
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] Mandatory to implement transport comment on      =
draft-ibc-sipcore-sip-websocket-02
>>
>> Inaki/Jose,
>>
>> Your current argument in the different mail of this same mail thread arg=
ue for not to close this open item which allows to relaxing the criteria fo=
r mandatory to implement (UDP/TCP) transport. If so, SIP over Websocket cli=
ent MUST support UDP & TCP as SIP transport to meet RFC 3261 normative stat=
ement. Please note that SIP over Websocket implementation with UDP & TCP as=
 SIP transport is possible by Linux/Windows application but not by Browser =
as of now.
>>
>> I'll like to hear more opinion from different folks before closing this =
open item.
>>
>> Thanks
>> Partha
>>
>> -----Original Message-----
>> From: Parthasarathi R [mailto:partha@parthasarathi.co.in]
>> Sent: Friday, August 24, 2012 11:11 PM
>> To: 'I=F1aki Baz Castillo'
>> Cc: 'Jos=E9 Luis Mill=E1n'; 'sipcore@ietf.org'
>> Subject: RE: [sipcore] Mandatory to implement transport comment on draft=
-ibc-sipcore-sip-websocket-02
>>
>> Inaki,
>>
>> Paul started the relaxing the criteria for mandatory to implement transp=
ort mail thread during WG adoption of SIP over websocket draft and one of t=
he mail is http://www.ietf.org/mail-archive/web/sipcore/current/msg04930.ht=
ml. My understanding of the mail thread conclusion was that narrow approach=
 of providing leniency for mandatory to implement transport in SIP over Web=
socket draft itself.  I noticed in the current document that there is no re=
levant text for this leniency and so raised the issue in IETF-84 SIPCore me=
eting. As per IETF-84 SIPCore meeting discussion, I started this mail threa=
d.
>>
>> In case you are not in favor of narrow approach proposed by Paul, then i=
t is a separate discussion. Also you have indicated that the narrow approac=
h needs more text, then please suggest the text and let us discuss.
>>
>> Thanks
>> Partha
>>
>> -----Original Message-----
>> From: I=F1aki Baz Castillo [mailto:ibc@aliax.net]
>> Sent: Friday, August 24, 2012 10:09 PM
>> To: Parthasarathi R
>> Cc: Jos=E9 Luis Mill=E1n; sipcore@ietf.org
>> Subject: Re: [sipcore] Mandatory to implement transport comment on draft=
-ibc-sipcore-sip-websocket-02
>>
>> 2012/8/24 Parthasarathi R <partha@parthasarathi.co.in>:
>>> In case normal RFC 3261 application or application based on SIP websock=
et
>>> client, ACK will send towards
>>>
>>> UAS contact address with UDP transport as per mandatory to implement
>>> transport mechanism in RFC 3261.
>>>
>>> As browser acting as SIP Websocket client violates the mandatory to
>>> implement transport mechanism,
>>>
>>> the callflow breaks in this specification.
>>
>> Hi Partha. As said many times this draft is not *just* about web
>> browsers. Please imagine that new versions of iOS and Android come
>> with a WebSocket Client stack (so native apps can use it). Should we
>> also add them to the draft?:
>>
>> =93In case SIP WebSocket Client is a web browser or a native iOS or
>> Android application, and SIP Websocket Server is a SIP proxy, then..."
>>
>> Ugly, agree?
>>
>>
>> Anyhow, for the case of a web browser (which indeed cannot send a UDP
>> datagram) the draft already describes a full guideline in the Appendix
>> section:
>>
>> http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02#appendix-A
>>
>> IMHO that's enough. The mandatory sections of the draft should not
>> focus just on web browsers.
>>
>>
>> BTW your text is incomplete:
>>
>>
>> =93In case SIP WebSocket Client is a web browser and SIP Websocket
>> Server is a SIP proxy, then record-route header MUST be included by
>> SIP Websocket server in all SIP messages. This requirement assures
>> that SIP WebSocket Client is able to contact with UAS through SIP
>> Websocket Server during the mid-dialog irrespective of the contact SIP
>> URI received from UAS =94
>>
>> That's not enough. If you want to also *receive* incoming requests you
>> need to mandate Outbound RFC 5626 (since the proxy cannot establish a
>> WebSocket connection with a web browser). But we don't want to mandate
>> Outbound (it was discussed and voted in Paris) since the draft just
>> describes a SIP transport and does not assume specific topologies or
>> UA devices.
>>
>> IMHO the Appendix guideline (which was already discussed in this WG)
>> is the appropriate section for talking about specific common
>> scenarios. The draft should not "update" RFC 3261 just for the case of
>> web browsers.
>>
>>
>> Regards.
>>
>>
>>
>> --
>> I=F1aki Baz Castillo
>> <ibc@aliax.net>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore

From atle.monrad@ericsson.com  Mon Aug 27 22:45:24 2012
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From: Atle Monrad <atle.monrad@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, SIPCORE <sipcore@ietf.org>
Date: Tue, 28 Aug 2012 07:45:20 +0200
Thread-Topic: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-08
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Subject: Re: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-08
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Paul, all

I've read through draft-ietf-sipcore-proxy-feature-08 and provided Christer=
 et.al. with a number of proposed nits plus a couple of other wordings that=
 seems to need corrections aswell.=20

Generally I'm fine with this version of the draft.

It seems to me that the draft has progressed as indicated by the previous c=
omments given, thus I hope that the draft -08 can be accepted and with some=
 polishing can progress towards completion rather soon.

We have features in 3GPP Rel-10 and 3GPP Rel-11 that are waiting for comple=
tion of this draft.

Thanks
/atle


________________________________=20


Atle Monrad
3GPP CT Chairman
Standardization and Regulation,
Group Function Technology and Portfolio Management=20
Ericsson


-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Paul Kyzivat
Sent: 27. august 2012 22:02
To: SIPCORE
Subject: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-=
08

Sorry to do this again, but we need to do another WGLC on draft-ietf-sipcor=
e-proxy-feature-08.

The WGLC will run for a week. It will conclude at 23:59 UTC on Monday, Sept=
ember 3, 2012. (A week should be enough considering the nature of the chang=
es since the last WGLC.)

This covers changes since the last WGLC, which was for the -02 version.=20
Those changes were to cover things I found while writing the document up fo=
r submission to the iesg, and things found by Robert Sparks in his AD revie=
w. Most of those things were editorial, but some of them were quite heavy e=
ditorial, such as rearranging sections of the document. (This causes the di=
ff to version -02 to be largely worthless.)

There was one significant change that isn't editorial. We changed the revie=
w policy for registering new feature capability indicators in the global tr=
ee from Expert Reveiw to Specification Required. This isn't a big change, s=
ince we were always requiring a specification. But it is a better fit. (Thi=
s doesn't require an RFC, only a specification that can be referenced perma=
nently.)

Please look at this and then indicate you acceptance or issues here on the =
mailing list. We really need responses to indicate that it has been reviewe=
d.

Some helpful links:

The draft version being reviewed:
http://tools.ietf.org/html/draft-ietf-sipcore-proxy-feature-08

Diff from version -06 (recent changes):
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-proxy-feature-08.tx=
t;url1=3Ddraft-ietf-sipcore-proxy-feature-06.txt

Diff from version -02 (the subject of the last WGLC):
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-proxy-feature-08.tx=
t;url1=3Ddraft-ietf-sipcore-proxy-feature-02.txt

	Thanks,
	Paul (as co-chair)
_______________________________________________
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  Mon Aug 27 23:50:59 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?Jos=E9_Luis_Mill=E1n?= <jmillan@aliax.net>
Date: Tue, 28 Aug 2012 08:50:55 +0200
Thread-Topic: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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Hi,

>>>This specification defines the way SIP WebSocket Client and Server=20
>>>communicate and the mechanisms for the SIP element acting as WebSocket=20
>>>Server to interact with other non WebSocket SIP elements.
>>
>> We authors propose the following statements to be used in this=20
>> specification according to this task:
>>
>>>"
>>>SIP WebSocket Servers implementing this specification MUST implement=20
>>>UDP and TCP as stated in [RFC 3261].
>>
>>> I would suggest to say that the MUST applies if such server also commun=
icate SIP using non-WS interfaces.
>>
>> Because, there could be a SIP WebSocket Server that *only* provides serv=
ices to WS clients.
>>
>> Maybe something like:
>>
>> "SIP WebSocket Servers implementing this specification, and that also su=
pport SIP using the transport mechanisms defined in [RFC 3261], MUST implem=
ent both UDP and TCP, as stated in the RFC"
>>
>
> Although it could seem logical at a first glance, such a statement breaks=
 the basic level of protocol interoperability. This limitation is not impos=
ed by the new transport and hence it seems reasonable that the SIP WebSocke=
t Server MUST implement UDP=20
> and TCP as stated in [RFC 3261].
>
> do you agree?

I'm fine with your suggestion.

Regards,

Christer


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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Mandatory to implement transport comment on draft-ibc-sipcore-sip-websocket-02
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On Tue, Aug 28, 2012 at 8:50 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> Hi,
>
>>>>This specification defines the way SIP WebSocket Client and Server
>>>>communicate and the mechanisms for the SIP element acting as WebSocket
>>>>Server to interact with other non WebSocket SIP elements.
>>>
>>> We authors propose the following statements to be used in this
>>> specification according to this task:
>>>
>>>>"
>>>>SIP WebSocket Servers implementing this specification MUST implement
>>>>UDP and TCP as stated in [RFC 3261].
>>>
>>>> I would suggest to say that the MUST applies if such server also communicate SIP using non-WS interfaces.
>>>
>>> Because, there could be a SIP WebSocket Server that *only* provides services to WS clients.
>>>
>>> Maybe something like:
>>>
>>> "SIP WebSocket Servers implementing this specification, and that also support SIP using the transport mechanisms defined in [RFC 3261], MUST implement both UDP and TCP, as stated in the RFC"
>>>
>>
>> Although it could seem logical at a first glance, such a statement breaks the basic level of protocol interoperability. This limitation is not imposed by the new transport and hence it seems reasonable that the SIP WebSocket Server MUST implement UDP
>> and TCP as stated in [RFC 3261].
>>
>> do you agree?
>
> I'm fine with your suggestion.

Great -- we'll update the draft accordingly

Cheers,
-Victor

From pkyzivat@alum.mit.edu  Tue Aug 28 06:10:20 2012
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Date: Tue, 28 Aug 2012 09:10:18 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-08
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On 8/28/12 1:45 AM, Atle Monrad wrote:
> Paul, all
>
> I've read through draft-ietf-sipcore-proxy-feature-08 and provided Christer et.al. with a number of proposed nits plus a couple of other wordings that seems to need corrections aswell.
>
> Generally I'm fine with this version of the draft.
>
> It seems to me that the draft has progressed as indicated by the previous comments given, thus I hope that the draft -08 can be accepted and with some polishing can progress towards completion rather soon.
>
> We have features in 3GPP Rel-10 and 3GPP Rel-11 that are waiting for completion of this draft.

The only "polishing" I anticipate will be
- to respond to any iesg comments
- editorial fixes requested by the editor

	Thanks,
	Paul


From partha@parthasarathi.co.in  Tue Aug 28 09:37:49 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Victor Pascual Avila'" <victor.pascual.avila@gmail.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
References: <000d01cd812e$46bb4850$d431d8f0$@co.in>	<CABw3bnPMKtWFq7eGQC6fNKym-6YCnaCer07iR=DrwTk05Yn66g@mail.gmail.com>	<000f01cd8214$9d255a00$d7700e00$@co.in>	<CALiegfnPNcqy9F-UjoJ8eHfgNPYufCXE4L4sWaiDx6SM6B+R8g@mail.gmail.com>	<000101cd82db$ab93a0c0$02bae240$@co.in>	<7F2072F1E0DE894DA4B517B93C6A05853409FF2F25@ESESSCMS0356.eemea.ericsson.se>	<CABw3bnMDkM2nhynbYed=JZ+hAjL0Q7EP+sPJzgTGMyyfXw76Wg@mail.gmail.com>	<7F2072F1E0DE894DA4B517B93C6A05853409FF2F28@ESESSCMS0356.eemea.ericsson.se>	<CABw3bnO5Dpb5TthKia34Sd90Rsg4hJAhUxayT=ETk-fG=A=BkQ@mail.gmail.com>	<7F2072F1E0DE894DA4B517B93C6A0585340A356BF8@ESESSCMS0356.eemea.ericsson.se> <CAGTXFp_hMkfdv+VnyEZwJkOhHz-9UekPtEEkDH1K0Nq+uSL_=A@mail.gmail.com>
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Date: Tue, 28 Aug 2012 22:07:36 +0530
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on	draft-ibc-sipcore-sip-websocket-02
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Hi,

It is good to see the addition of text around this open issue. As there is
no normative usage in the proposed text, The minor correction shall be as
follows:

"As a SIP WebSocket Client will always communicate with a SIP WebSocket
Server, and since some environments do not enable SIP WebSocket Clients to
use other UDP and TCP as SIP transport protocols for SIP, a SIP WebSocket
Client MAY not implement UDP and TCP as SIP transport"

Thanks
Partha

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
Of Victor Pascual Avila
Sent: Tuesday, August 28, 2012 2:33 PM
To: Christer Holmberg
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Mandatory to implement transport comment on
draft-ibc-sipcore-sip-websocket-02

On Tue, Aug 28, 2012 at 8:50 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> Hi,
>
>>>>This specification defines the way SIP WebSocket Client and Server
>>>>communicate and the mechanisms for the SIP element acting as WebSocket
>>>>Server to interact with other non WebSocket SIP elements.
>>>
>>> We authors propose the following statements to be used in this
>>> specification according to this task:
>>>
>>>>"
>>>>SIP WebSocket Servers implementing this specification MUST implement
>>>>UDP and TCP as stated in [RFC 3261].
>>>
>>>> I would suggest to say that the MUST applies if such server also
communicate SIP using non-WS interfaces.
>>>
>>> Because, there could be a SIP WebSocket Server that *only* provides
services to WS clients.
>>>
>>> Maybe something like:
>>>
>>> "SIP WebSocket Servers implementing this specification, and that also
support SIP using the transport mechanisms defined in [RFC 3261], MUST
implement both UDP and TCP, as stated in the RFC"
>>>
>>
>> Although it could seem logical at a first glance, such a statement breaks
the basic level of protocol interoperability. This limitation is not imposed
by the new transport and hence it seems reasonable that the SIP WebSocket
Server MUST implement UDP
>> and TCP as stated in [RFC 3261].
>>
>> do you agree?
>
> I'm fine with your suggestion.

Great -- we'll update the draft accordingly

Cheers,
-Victor
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore


From jmillan@aliax.net  Tue Aug 28 15:47:51 2012
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Subject: [sipcore] Via 'received' parameter (was Re: WGLC: draft-ietf-sipcore-sip-websocket-02)
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Hi,

Is it convenient for a SIP WebSocket Server to strictly satisfy 18.2.1
and  18.2.2 sections in [RFC 3261] ?

The WebSocket connection  may traverse one or more HTTP elements
before reaching the SIP WebSocket Server. In this case, the Via
'received' parameter will point to a non SIP element.

Additionally, many SIP WebSocket Clients won't act as SIP WebSocket
Servers. They won't allow receiving WebSocket connections.

Trying to establish a new connection to the address represented in the
Via 'received' parameter,  as stated in section 18.2.2 of  [RFC 3261],
won't succeed in any of the scenarios defined above. For this reason,
it may be reasonable to relax 18.2.1 and 18.2.2 sections of [RFC 3261]
by stating something like:

"
SIP WebSocket Server implementations MAY decide not to add the Via
'received' parameter to received SIP requests, in which case the
server transaction MUST NOT attempt to open a connection for sending a
response.
"

It might be worth saying that:

"
It is RECOMMENDED that SIP WebSocket Server implementations do not add
the Via 'received' parameter to SIP requests when acting as edge
proxies.
"


Regards,

From christer.holmberg@ericsson.com  Wed Aug 29 01:10:38 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Paul Kyzivat' <pkyzivat@alum.mit.edu>, Atle Monrad <atle.monrad@ericsson.com>
Date: Wed, 29 Aug 2012 10:10:32 +0200
Thread-Topic: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-08
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Subject: Re: [sipcore] Another quick WGLC for	draft-ietf-sipcore-proxy-feature-08
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Hi,

Based on Atle's findings, there are a few editorial nits that I think we sh=
all fix:


Section 4.3.2:
----------------

The text says:

	"Unless a feature capability indicator is inserted in a Feature-Caps
   	header field or a target refresh request,.."

Replace "or" with "of".


Section 5.3.4:
-----------------

The text says:

	"There might be restrictions related to whether entities are allowed
  	 to insert a feature capability indicator in registration related
   	messages, standalone transaction messages, or dialog related
  	messages, whether entities are..."

Remove "or".



Sections 7.3.2 and 7.3.3:
-----------------------------

In both sections we have the following text:

   	"Description contains the Abstract, provided in the registration
   	feature capability indication registration template."

However, there is no "Abstract" in the template. There is a "Description", =
though.

So, I suggest to simply say:

   	"Description, provided in the registration feature capability=20
	indication registration template."


Regards,

Christer

=20

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Paul Kyzivat
Sent: 28. elokuuta 2012 16:10
To: Atle Monrad
Cc: SIPCORE
Subject: Re: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feat=
ure-08

On 8/28/12 1:45 AM, Atle Monrad wrote:
> Paul, all
>
> I've read through draft-ietf-sipcore-proxy-feature-08 and provided Christ=
er et.al. with a number of proposed nits plus a couple of other wordings th=
at seems to need corrections aswell.
>
> Generally I'm fine with this version of the draft.
>
> It seems to me that the draft has progressed as indicated by the previous=
 comments given, thus I hope that the draft -08 can be accepted and with so=
me polishing can progress towards completion rather soon.
>
> We have features in 3GPP Rel-10 and 3GPP Rel-11 that are waiting for comp=
letion of this draft.

The only "polishing" I anticipate will be
- to respond to any iesg comments
- editorial fixes requested by the editor

	Thanks,
	Paul

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From pkyzivat@alum.mit.edu  Wed Aug 29 07:53:06 2012
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Date: Wed, 29 Aug 2012 10:53:03 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: Atle Monrad <atle.monrad@ericsson.com>, SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-08
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These are very editorial in nature. (Hopefully the RFC editor would have 
caught them too.) They should be fixed, but I think you can save them 
for the next required edit.

	Thanks,
	Paul

On 8/29/12 4:10 AM, Christer Holmberg wrote:
> Hi,
>
> Based on Atle's findings, there are a few editorial nits that I think we shall fix:
>
>
> Section 4.3.2:
> ----------------
>
> The text says:
>
> 	"Unless a feature capability indicator is inserted in a Feature-Caps
>     	header field or a target refresh request,.."
>
> Replace "or" with "of".
>
>
> Section 5.3.4:
> -----------------
>
> The text says:
>
> 	"There might be restrictions related to whether entities are allowed
>    	 to insert a feature capability indicator in registration related
>     	messages, standalone transaction messages, or dialog related
>    	messages, whether entities are..."
>
> Remove "or".
>
>
>
> Sections 7.3.2 and 7.3.3:
> -----------------------------
>
> In both sections we have the following text:
>
>     	"Description contains the Abstract, provided in the registration
>     	feature capability indication registration template."
>
> However, there is no "Abstract" in the template. There is a "Description", though.
>
> So, I suggest to simply say:
>
>     	"Description, provided in the registration feature capability
> 	indication registration template."
>
>
> Regards,
>
> Christer
>
>
>
> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 28. elokuuta 2012 16:10
> To: Atle Monrad
> Cc: SIPCORE
> Subject: Re: [sipcore] Another quick WGLC for draft-ietf-sipcore-proxy-feature-08
>
> On 8/28/12 1:45 AM, Atle Monrad wrote:
>> Paul, all
>>
>> I've read through draft-ietf-sipcore-proxy-feature-08 and provided Christer et.al. with a number of proposed nits plus a couple of other wordings that seems to need corrections aswell.
>>
>> Generally I'm fine with this version of the draft.
>>
>> It seems to me that the draft has progressed as indicated by the previous comments given, thus I hope that the draft -08 can be accepted and with some polishing can progress towards completion rather soon.
>>
>> We have features in 3GPP Rel-10 and 3GPP Rel-11 that are waiting for completion of this draft.
>
> The only "polishing" I anticipate will be
> - to respond to any iesg comments
> - editorial fixes requested by the editor
>
> 	Thanks,
> 	Paul
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

