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From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: [sipcore] Fwd: [POSH] PKIX Over Secure HTTP (POSH)
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-----BEGIN PGP SIGNED MESSAGE-----
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Perhaps of interest here given recent discussions about server
identity checking in SIP...


- -------- Original Message --------
Subject: [POSH] PKIX Over Secure HTTP (POSH)
Date: Tue, 04 Jun 2013 17:24:15 -0600
From: Peter Saint-Andre <stpeter@stpeter.im>
To: posh@ietf.org

Matt Miller and I have been working on a specification for "PKIX Over
Secure HTTP" (POSH), which aims to make it easier to ensure proper TLS
server identity checking in multi-tenanted environments (where it's
basically impossible right now):

https://datatracker.ietf.org/doc/draft-miller-posh/

As the abstract says:

   This document defines two methods that make it easier to deploy
   certificates for proper server identity checking in application
   protocols.  The first method enables a TLS client to obtain a TLS
   server's end-entity certificate over secure HTTP as an alternative to
   standard Public Key Infrastructure using X.509 (PKIX) and DNS-Based
   Authentication of Named Entities (DANE).  The second method enables a
   source domain to securely delegate an application to a derived domain
   using HTTPS redirects.

We love PKIX (really!), we love DNSSEC, and we love DANE (which solves
some of the same problems for some application protocols as POSH
does). However, we want a technology that can be deployed more quickly
than DANE in order to solve pressing operational security issues with
standard PKIX in multi-tenanted environments.

This effort emerged from the XMPP community, but we have heard from
folks working on other application technologies that it might be
useful for things like IMAP and SMTP, thus the more generalized
version of POSH that we published today (superseding
draft-miller-xmpp-posh-prooftype).

We are planning to hold a BoF on this topic in Berlin, but in the
meantime comments are very much welcome. Please post your feedback to
the new posh@ietf.org list:

https://www.ietf.org/mailman/listinfo/posh

Thanks!

Peter

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


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Date: Wed, 12 Jun 2013 18:17:08 -0700
Cc: sipcore@ietf.org
Subject: [sipcore] I-D Action: draft-ietf-sipcore-sip-websocket-09.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           : The WebSocket Protocol as a Transport for the Session In=
itiation Protocol (SIP)
	Author(s)       : Inaki Baz Castillo
                          Jose Luis Millan Villegas
                          Victor Pascual
	Filename        : draft-ietf-sipcore-sip-websocket-09.txt
	Pages           : 25
	Date            : 2013-06-12

Abstract:
   The WebSocket protocol enables two-way realtime communication between
   clients and servers in web-based applications.  This document
   specifies a WebSocket sub-protocol as a reliable transport mechanism
   between SIP (Session Initiation Protocol) entities to enable usage of
   SIP in web-oriented deployments.  This document normatively updates
   RFC 3261.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-09

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-09


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


From ibc@aliax.net  Wed Jun 12 18:23:11 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 13 Jun 2013 03:22:43 +0200
Message-ID: <CALiegfkg-KU1bB01eLXuksZV1ehBY92uf+0+F3fQuha-WnOS1A@mail.gmail.com>
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Subject: Re: [sipcore] I-D Action: draft-ietf-sipcore-sip-websocket-09.txt
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Hi all,

This new revision addresses all the issues reported in the WG since
previous 08 revision. The changelog in revision 09 is the following:


- Improved section "Handshake" (under "The WebSocket SIP
Sub-Protocol") by mentioning the error handling when establishing a
WebSocket connection (thanks to Suresh Krishnan).

- Mention to RFC 4168 (SIP SCTP) removed from the "SIP URI Transport
Parameter" section (reported by Suresh Krishnan).

- "SIP Transport Implementation Requirements" section clarified by
removing the "MAY" keyword (reported by Suresh Krishnan), and text
from RFC 3261 section 18 amended (thanks to Barry Leiba).

- Text about certificate validation in secure WebSocket clarified in
section "Secure WebSocket Connection" within "Security Considerations"
(thanks to  Richard Barnes).

- Section "Authentication" made normative and text about SIP/Web
authentication requirement improved.

- New appendix "Authentication Use Cases".


Really thanks a lot for your help and comments.




2013/6/13  <internet-drafts@ietf.org>:
>
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>  This draft is a work item of the Session Initiation Protocol Core Workin=
g Group of the IETF.
>
>         Title           : The WebSocket Protocol as a Transport for the S=
ession Initiation Protocol (SIP)
>         Author(s)       : Inaki Baz Castillo
>                           Jose Luis Millan Villegas
>                           Victor Pascual
>         Filename        : draft-ietf-sipcore-sip-websocket-09.txt
>         Pages           : 25
>         Date            : 2013-06-12
>
> Abstract:
>    The WebSocket protocol enables two-way realtime communication between
>    clients and servers in web-based applications.  This document
>    specifies a WebSocket sub-protocol as a reliable transport mechanism
>    between SIP (Session Initiation Protocol) entities to enable usage of
>    SIP in web-oriented deployments.  This document normatively updates
>    RFC 3261.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-09
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-09
>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore



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

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 14 Jun 2013 09:59:03 +0200
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Subject: [sipcore] Open isues in draft-ietf-sipcore-sip-websocket-09
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Hi all,

Once  draft-ietf-sipcore-sip-websocket-09 has been submitted, and
after some comments in the WG, I consider that there are still the
following open issues (to be addressed in a new revision):



1) Remove "_This section is non-normative_" text at the top of
non-normative sections.

Those sections don't contain normative keywords (MUST, SHOULD) so they
are indeed non-normative. No need for such an statement.




2 Decide whether the draft updates 3261 or not.

After recent discussion in the WG my opinion is that if somebody wants
to build a SIP UDP and/or TCP device during 2013 year, he does not
need to read draft-ietf-sipcore-sip-websocket, and thus the draft does
not update RFC 3261.

The other point of view is that the draft makes implementation of both
UDP and TCP transports non mandatory if WS is implemented. This opens
the door to two solutions:

a) Stating that in the draft (as currently it does).

b) A new WG item for allowing other transports without requiring UDP/TCP.





In draft-ietf-sipcore-sip-websocket-09 we have tryed to address all
the other open issues reported by members of the WG after 08 was
published. I hope changes and additions in 09 satisfy their concerns.
Otherwise please let us know and we will address them in rev 10.



Thanks a lot to all.



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

From partha@parthasarathi.co.in  Sun Jun 16 09:36:57 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
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Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Hi Inaki,

Open Item 2 comment: The current statement states that=20

     " All SIP elements MUST implement at least one of the following:

      *  Both UDP and TCP transports.

      *  SIP WebSocket transport."

I could not understand how this update resolve the issue in the =
following scenario:

 SIP UA1 (WS only)----Proxy-----SIP UA2 (UDP/TCP)

Here, the dialog is established with contact details for SIP UA1 & SIP =
UA2 as follows:

1) contact: sip:sipua1@example.com;transport=3Dws
2) contact: sip:sipua2@example.com;transport=3Dtcp

In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, how does it =
possible now as per the current update in Sec 5.2.4 of =
draft-ietf-sipcore-sip-websocket-09.

Apart from this, I'm fine with stating in the draft or new draft wherein =
this draft is updated with new draft reference.

Thanks
Partha=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of I=C3=B1aki Baz Castillo
> Sent: Friday, June 14, 2013 1:29 PM
> To: SIPCORE (Session Initiation Protocol Core) WG
> Subject: [sipcore] Open isues in draft-ietf-sipcore-sip-websocket-09
>=20
> Hi all,
>=20
> Once  draft-ietf-sipcore-sip-websocket-09 has been submitted, and
> after some comments in the WG, I consider that there are still the
> following open issues (to be addressed in a new revision):
>=20
>=20
>=20
> 1) Remove "_This section is non-normative_" text at the top of
> non-normative sections.
>=20
> Those sections don't contain normative keywords (MUST, SHOULD) so they
> are indeed non-normative. No need for such an statement.
>=20
>=20
>=20
>=20
> 2 Decide whether the draft updates 3261 or not.
>=20
> After recent discussion in the WG my opinion is that if somebody wants
> to build a SIP UDP and/or TCP device during 2013 year, he does not
> need to read draft-ietf-sipcore-sip-websocket, and thus the draft does
> not update RFC 3261.
>=20
> The other point of view is that the draft makes implementation of both
> UDP and TCP transports non mandatory if WS is implemented. This opens
> the door to two solutions:
>=20
> a) Stating that in the draft (as currently it does).
>=20
> b) A new WG item for allowing other transports without requiring
> UDP/TCP.
>=20
>=20
>=20
>=20
>=20
> In draft-ietf-sipcore-sip-websocket-09 we have tryed to address all
> the other open issues reported by members of the WG after 08 was
> published. I hope changes and additions in 09 satisfy their concerns.
> Otherwise please let us know and we will address them in rev 10.
>=20
>=20
>=20
> Thanks a lot to all.
>=20
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sun, 16 Jun 2013 23:14:58 +0200
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Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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2013/6/16 Parthasarathi R <partha@parthasarathi.co.in>:
> Hi Inaki,
>
> Open Item 2 comment: The current statement states that
>
>      " All SIP elements MUST implement at least one of the following:
>
>       *  Both UDP and TCP transports.
>
>       *  SIP WebSocket transport."
>
> I could not understand how this update resolve the issue in the following=
 scenario:
>
>  SIP UA1 (WS only)----Proxy-----SIP UA2 (UDP/TCP)
>
> Here, the dialog is established with contact details for SIP UA1 & SIP UA=
2 as follows:
>
> 1) contact: sip:sipua1@example.com;transport=3Dws
> 2) contact: sip:sipua2@example.com;transport=3Dtcp
>
> In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, how does it pos=
sible now as per the current update in Sec 5.2.4 of draft-ietf-sipcore-sip-=
websocket-09.


Hi Parthasarathi, this subject was heavily discussed during the past
year. It's clear that in 99% or 100% of cases, a SIP WebSocket Client
(i.e. a browser) will not be able to receive incoming WS connections
(it is not a SIP WebSocket Server) and thus, the proxy in your flow
has to perform record-routing, a *standard* mechanism defined in RFC
3261 for making your scenario feasible.

IMHO it is not so hard to assume that when SIP over WebSocket devices
are present, the proxy they connect to must perform record-routing.
The same is true for clients behind NAT.


Regards.


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

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Cc: "'SIPCORE \(Session Initiation Protocol Core\) WG'" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Hi Inaki,

As you mentioned, "record-route" is the right solution and not =
documented in the draft as of now. My comment is to explicitly add the =
statement for clarity.

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of I=C3=B1aki Baz Castillo
> Sent: Monday, June 17, 2013 2:45 AM
> To: Parthasarathi R
> Cc: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> 2013/6/16 Parthasarathi R <partha@parthasarathi.co.in>:
> > Hi Inaki,
> >
> > Open Item 2 comment: The current statement states that
> >
> >      " All SIP elements MUST implement at least one of the =
following:
> >
> >       *  Both UDP and TCP transports.
> >
> >       *  SIP WebSocket transport."
> >
> > I could not understand how this update resolve the issue in the
> following scenario:
> >
> >  SIP UA1 (WS only)----Proxy-----SIP UA2 (UDP/TCP)
> >
> > Here, the dialog is established with contact details for SIP UA1 &
> SIP UA2 as follows:
> >
> > 1) contact: sip:sipua1@example.com;transport=3Dws
> > 2) contact: sip:sipua2@example.com;transport=3Dtcp
> >
> > In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, how does =
it
> possible now as per the current update in Sec 5.2.4 of draft-ietf-
> sipcore-sip-websocket-09.
>=20
>=20
> Hi Parthasarathi, this subject was heavily discussed during the past
> year. It's clear that in 99% or 100% of cases, a SIP WebSocket Client
> (i.e. a browser) will not be able to receive incoming WS connections
> (it is not a SIP WebSocket Server) and thus, the proxy in your flow
> has to perform record-routing, a *standard* mechanism defined in RFC
> 3261 for making your scenario feasible.
>=20
> IMHO it is not so hard to assume that when SIP over WebSocket devices
> are present, the proxy they connect to must perform record-routing.
> The same is true for clients behind NAT.
>=20
>=20
> Regards.
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 17 Jun 2013 19:49:57 +0200
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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2013/6/17 Parthasarathi R <partha@parthasarathi.co.in>:
> As you mentioned, "record-route" is the right solution and not documented=
 in the draft as of now. My comment is to explicitly add the statement for =
clarity.

The draft describes a SIP transport, not a network topology.

For clients behind NAT, RFC 5626 (Outbound) already describes a
mechanism and requirements for clients and proxies to properly work.
RFC 5626 can perfectly be applied to scenarios in which SIP-WebSocket
is present. So no need to duplicate all the information provided by
RFC 5626 in this draft.

Anyhow, the Appendix "Implementation Guidelines" already suggests
using Record-Route within the "SIP WebSocket Server Considerations"
subsection:

   The SIP Outbound extension [RFC5626] seems an appropriate solution
   for this scenario.  Therefore these SIP WebSocket Clients and the SIP
   registrar implement both the Outbound and Path [RFC3327] extensions,
   and the SIP proxy acts as an Outbound Edge Proxy (as defined in
   [RFC5626] section 3.4).

And of course RFC 5626 already states that a Outbound Edge Proxy must
do record-routing. I hope this draft won't become a "general tutorial
for SIP and NAT".


Regards.

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

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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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2013/6/16 Parthasarathi R <partha@parthasarathi.co.in>:
> I could not understand how this update resolve the issue in the following=
 scenario:
>
>  SIP UA1 (WS only)----Proxy-----SIP UA2 (UDP/TCP)
>
> Here, the dialog is established with contact details for SIP UA1 & SIP UA=
2 as follows:
>
> 1) contact: sip:sipua1@example.com;transport=3Dws
> 2) contact: sip:sipua2@example.com;transport=3Dtcp
>
> In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, how does it pos=
sible now as per the current update in Sec 5.2.4 of draft-ietf-sipcore-sip-=
websocket-09.


Hi Parthasarathi,

  SIP UA1 (SCTP/UDP/TCP)----Proxy-----SIP UA2 (UDP/TCP)

Here, the dialog is established with contact details for SIP UA1 & SIP
UA2 as follows:

1) contact: sip:sipua1@example.com;transport=3Dsctp
2) contact: sip:sipua2@example.com;transport=3Dtcp

In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, please let
me know in which section of RFC 4168 (SIP STCP) it is stated that the
proxy must use Record-Route for the initial INVITE for this scenario
to work.

Regards.


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

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From: "Parthasarathi R" <partha@parthasarathi.co.in>
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Cc: "'SIPCORE \(Session Initiation Protocol Core\) WG'" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Hi Inaki,

I agree with you that RFC 4168 (SIP SCTP) does not explain the contact =
header usage. But it does not meant that it is the best practice. My =
comments is based on the later RFC namely RFC 5630 (The Use of the SIPS =
URI Scheme in the Session Initiation Protocol (SIP)) and look into the =
draft for the explanation and Normative statements around contact header =
w.r.t SIPS URI.=20

The current text in the draft is clear to explain the usage of contact =
header. It will confuse in case anybody implement by just looking into =
this draft. I think that it is required for SIP over WebSocket to =
explain the usage of contact header field with example and normative =
statement.=20

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of I=C3=B1aki Baz Castillo
> Sent: Tuesday, June 18, 2013 12:21 AM
> To: Parthasarathi R
> Cc: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> 2013/6/16 Parthasarathi R <partha@parthasarathi.co.in>:
> > I could not understand how this update resolve the issue in the
> following scenario:
> >
> >  SIP UA1 (WS only)----Proxy-----SIP UA2 (UDP/TCP)
> >
> > Here, the dialog is established with contact details for SIP UA1 &
> SIP UA2 as follows:
> >
> > 1) contact: sip:sipua1@example.com;transport=3Dws
> > 2) contact: sip:sipua2@example.com;transport=3Dtcp
> >
> > In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, how does =
it
> possible now as per the current update in Sec 5.2.4 of draft-ietf-
> sipcore-sip-websocket-09.
>=20
>=20
> Hi Parthasarathi,
>=20
>   SIP UA1 (SCTP/UDP/TCP)----Proxy-----SIP UA2 (UDP/TCP)
>=20
> Here, the dialog is established with contact details for SIP UA1 & SIP
> UA2 as follows:
>=20
> 1) contact: sip:sipua1@example.com;transport=3Dsctp
> 2) contact: sip:sipua2@example.com;transport=3Dtcp
>=20
> In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, please let
> me know in which section of RFC 4168 (SIP STCP) it is stated that the
> proxy must use Record-Route for the initial INVITE for this scenario
> to work.
>=20
> Regards.
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
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Cc: "'SIPCORE \(Session Initiation Protocol Core\) WG'" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Hi Inaki,

<snip> I hope this draft won't become a "general tutorial for SIP and =
NAT". </snip>

My reported "contact" header problem is related to SIP routing issue and =
not related to NAT.

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of I=C3=B1aki Baz Castillo
> Sent: Monday, June 17, 2013 11:20 PM
> To: Parthasarathi R
> Cc: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> 2013/6/17 Parthasarathi R <partha@parthasarathi.co.in>:
> > As you mentioned, "record-route" is the right solution and not
> documented in the draft as of now. My comment is to explicitly add the
> statement for clarity.
>=20
> The draft describes a SIP transport, not a network topology.
>=20
> For clients behind NAT, RFC 5626 (Outbound) already describes a
> mechanism and requirements for clients and proxies to properly work.
> RFC 5626 can perfectly be applied to scenarios in which SIP-WebSocket
> is present. So no need to duplicate all the information provided by
> RFC 5626 in this draft.
>=20
> Anyhow, the Appendix "Implementation Guidelines" already suggests
> using Record-Route within the "SIP WebSocket Server Considerations"
> subsection:
>=20
>    The SIP Outbound extension [RFC5626] seems an appropriate solution
>    for this scenario.  Therefore these SIP WebSocket Clients and the
> SIP
>    registrar implement both the Outbound and Path [RFC3327] =
extensions,
>    and the SIP proxy acts as an Outbound Edge Proxy (as defined in
>    [RFC5626] section 3.4).
>=20
> And of course RFC 5626 already states that a Outbound Edge Proxy must
> do record-routing. I hope this draft won't become a "general tutorial
> for SIP and NAT".
>=20
>=20
> Regards.
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From partha@parthasarathi.co.in  Tue Jun 18 11:03:48 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: =?UTF-8?Q?'I=C3=B1aki_Baz_Castillo'?= <ibc@aliax.net>, "'SIPCORE \(Session Initiation Protocol Core\) WG'" <sipcore@ietf.org>
References: <20130613011708.18316.28106.idtracker@ietfa.amsl.com> <CALiegfkg-KU1bB01eLXuksZV1ehBY92uf+0+F3fQuha-WnOS1A@mail.gmail.com>
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Subject: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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Hi Inaki & all,

IIUC, The below text in Sec 7 of the draft:

"If no authentication is done at WebSocket level then SIP Digest
   authentication is required for every SIP request coming over the
   WebSocket connection."

has to be changed as Normative statement as follows:

"If no authentication is done at WebSocket level then SIP Digest
   authentication MUST be done for every SIP request coming over the
   WebSocket connection."

Please let me know in case I'm missing something.

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of I=C3=B1aki Baz Castillo
> Sent: Thursday, June 13, 2013 6:53 AM
> To: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] I-D Action: draft-ietf-sipcore-sip-websocket-
> 09.txt
>=20
> Hi all,
>=20
> This new revision addresses all the issues reported in the WG since
> previous 08 revision. The changelog in revision 09 is the following:
>=20
>=20
> - Improved section "Handshake" (under "The WebSocket SIP
> Sub-Protocol") by mentioning the error handling when establishing a
> WebSocket connection (thanks to Suresh Krishnan).
>=20
> - Mention to RFC 4168 (SIP SCTP) removed from the "SIP URI Transport
> Parameter" section (reported by Suresh Krishnan).
>=20
> - "SIP Transport Implementation Requirements" section clarified by
> removing the "MAY" keyword (reported by Suresh Krishnan), and text
> from RFC 3261 section 18 amended (thanks to Barry Leiba).
>=20
> - Text about certificate validation in secure WebSocket clarified in
> section "Secure WebSocket Connection" within "Security Considerations"
> (thanks to  Richard Barnes).
>=20
> - Section "Authentication" made normative and text about SIP/Web
> authentication requirement improved.
>=20
> - New appendix "Authentication Use Cases".
>=20
>=20
> Really thanks a lot for your help and comments.
>=20
>=20
>=20
>=20
> 2013/6/13  <internet-drafts@ietf.org>:
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >  This draft is a work item of the Session Initiation Protocol Core
> Working Group of the IETF.
> >
> >         Title           : The WebSocket Protocol as a Transport for
> the Session Initiation Protocol (SIP)
> >         Author(s)       : Inaki Baz Castillo
> >                           Jose Luis Millan Villegas
> >                           Victor Pascual
> >         Filename        : draft-ietf-sipcore-sip-websocket-09.txt
> >         Pages           : 25
> >         Date            : 2013-06-12
> >
> > Abstract:
> >    The WebSocket protocol enables two-way realtime communication
> between
> >    clients and servers in web-based applications.  This document
> >    specifies a WebSocket sub-protocol as a reliable transport
> mechanism
> >    between SIP (Session Initiation Protocol) entities to enable =
usage
> of
> >    SIP in web-oriented deployments.  This document normatively
> updates
> >    RFC 3261.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-09
> >
> > A diff from the previous version is available at:
> > =
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-09
> >
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
>=20
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From ibc@aliax.net  Tue Jun 18 15:05:25 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 19 Jun 2013 00:04:53 +0200
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Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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Right. I will modify it. Anyhow, I don't know if more changes are
required in that "Authentication" section (due to pending issues in
another mail thread).

Thanks a lot.

2013/6/18 Parthasarathi R <partha@parthasarathi.co.in>:
> Hi Inaki & all,
>
> IIUC, The below text in Sec 7 of the draft:
>
> "If no authentication is done at WebSocket level then SIP Digest
>    authentication is required for every SIP request coming over the
>    WebSocket connection."
>
> has to be changed as Normative statement as follows:
>
> "If no authentication is done at WebSocket level then SIP Digest
>    authentication MUST be done for every SIP request coming over the
>    WebSocket connection."
>
> Please let me know in case I'm missing something.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf Of I=C3=B1aki Baz Castillo
>> Sent: Thursday, June 13, 2013 6:53 AM
>> To: SIPCORE (Session Initiation Protocol Core) WG
>> Subject: Re: [sipcore] I-D Action: draft-ietf-sipcore-sip-websocket-
>> 09.txt
>>
>> Hi all,
>>
>> This new revision addresses all the issues reported in the WG since
>> previous 08 revision. The changelog in revision 09 is the following:
>>
>>
>> - Improved section "Handshake" (under "The WebSocket SIP
>> Sub-Protocol") by mentioning the error handling when establishing a
>> WebSocket connection (thanks to Suresh Krishnan).
>>
>> - Mention to RFC 4168 (SIP SCTP) removed from the "SIP URI Transport
>> Parameter" section (reported by Suresh Krishnan).
>>
>> - "SIP Transport Implementation Requirements" section clarified by
>> removing the "MAY" keyword (reported by Suresh Krishnan), and text
>> from RFC 3261 section 18 amended (thanks to Barry Leiba).
>>
>> - Text about certificate validation in secure WebSocket clarified in
>> section "Secure WebSocket Connection" within "Security Considerations"
>> (thanks to  Richard Barnes).
>>
>> - Section "Authentication" made normative and text about SIP/Web
>> authentication requirement improved.
>>
>> - New appendix "Authentication Use Cases".
>>
>>
>> Really thanks a lot for your help and comments.
>>
>>
>>
>>
>> 2013/6/13  <internet-drafts@ietf.org>:
>> >
>> > A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>> >  This draft is a work item of the Session Initiation Protocol Core
>> Working Group of the IETF.
>> >
>> >         Title           : The WebSocket Protocol as a Transport for
>> the Session Initiation Protocol (SIP)
>> >         Author(s)       : Inaki Baz Castillo
>> >                           Jose Luis Millan Villegas
>> >                           Victor Pascual
>> >         Filename        : draft-ietf-sipcore-sip-websocket-09.txt
>> >         Pages           : 25
>> >         Date            : 2013-06-12
>> >
>> > Abstract:
>> >    The WebSocket protocol enables two-way realtime communication
>> between
>> >    clients and servers in web-based applications.  This document
>> >    specifies a WebSocket sub-protocol as a reliable transport
>> mechanism
>> >    between SIP (Session Initiation Protocol) entities to enable usage
>> of
>> >    SIP in web-oriented deployments.  This document normatively
>> updates
>> >    RFC 3261.
>> >
>> >
>> > The IETF datatracker status page for this draft is:
>> > https://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket
>> >
>> > There's also a htmlized version available at:
>> > http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-09
>> >
>> > A diff from the previous version is available at:
>> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-09
>> >
>> >
>> > Internet-Drafts are also available by anonymous FTP at:
>> > ftp://ftp.ietf.org/internet-drafts/
>> >
>> > _______________________________________________
>> > sipcore mailing list
>> > sipcore@ietf.org
>> > https://www.ietf.org/mailman/listinfo/sipcore
>>
>>
>>
>> --
>> I=C3=B1aki Baz Castillo
>> <ibc@aliax.net>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>



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

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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2013/6/18 Parthasarathi R <partha@parthasarathi.co.in>:
> I think that it is required for SIP over WebSocket to explain the usage o=
f contact header field with example and normative statement.

Honestly I have no idea of what you mean.
draft-ietf-sipcore-sip-websocket must explain how to use the SIP
Contact header?

Sorry but I don't understand your concern. Could you please describe it aga=
in?

Thanks a lot.


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

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From: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
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From saul@ag-projects.com  Wed Jun 19 00:14:29 2013
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, Parthasarathi R <partha@parthasarathi.co.in>
Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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Hi,

After reading those 2 paragraphs a few times I have a question about =
some corner case (below)

On Jun 19, 2013, at 12:04 AM, I=F1aki Baz Castillo wrote:

> Right. I will modify it. Anyhow, I don't know if more changes are
> required in that "Authentication" section (due to pending issues in
> another mail thread).
>=20
> Thanks a lot.
>=20
> 2013/6/18 Parthasarathi R <partha@parthasarathi.co.in>:
>> Hi Inaki & all,
>>=20
>> IIUC, The below text in Sec 7 of the draft:
>>=20
>> "If no authentication is done at WebSocket level then SIP Digest
>>   authentication is required for every SIP request coming over the
>>   WebSocket connection."
>>=20
>> has to be changed as Normative statement as follows:
>>=20
>> "If no authentication is done at WebSocket level then SIP Digest
>>   authentication MUST be done for every SIP request coming over the
>>   WebSocket connection."
>>=20
>> Please let me know in case I'm missing something.
>>=20

Why is authentication a MUST? Lets assume that I'm using UDP and my =
proxy establishes a WS connection with a foreign domain's proxy because =
of NAPTR and my proxy supports acting as a WS client. It obviously won't =
be able to authenticate. If this scenario supposed to be covered?


Thanks,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From ibc@aliax.net  Wed Jun 19 04:32:36 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 19 Jun 2013 13:32:09 +0200
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, Parthasarathi R <partha@parthasarathi.co.in>
Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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2013/6/19 Sa=C3=BAl Ibarra Corretg=C3=A9 <saul@ag-projects.com>:
> Why is authentication a MUST? Lets assume that I'm using UDP and my proxy=
 establishes a WS connection with a foreign domain's proxy because of NAPTR=
 and my proxy supports acting as a WS client. It obviously won't be able to=
 authenticate. If this scenario supposed to be covered?

Honestly I agree. I cannot find in RFC 3261 (or other RFCs) a
normative statement mandating authentication, regardless the request
comes from a UA.

In another thread we are discussing about MTI authentication
mechanisms that must be implemented by SIP WS Clients and Servers.
IMHO that is correct, but mandating SIP authentication or WWW
authentication for ALL the scenarios seem innapropriate for me. I come
back to an use case:

A website (a shop) offers a widget in which the visitor can click and
made a SIP call (+WebRTC) that will end in the callcenter of the
company, answered by an agent that will inform the user about the
product he is interested in. Why do we require WWW or SIP
authentication in this scenario?

If WG agrees with this, I will remove the normative statements in
"Authentication" section, and instead address the MTI authentication
mechanisms.


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

From keith.drage@alcatel-lucent.com  Wed Jun 19 05:06:47 2013
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Thread-Topic: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, Parthasarathi R <partha@parthasarathi.co.in>
Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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I would not use RFC 3261 as justification for what should, or should not, b=
e said about authentication. The current RFC 3261 would probably fail a sec=
urity directorate review if it was attempted to be approved as an RFC now.

(I'd also point out that for any security consideration of RFC 3261, one sh=
ould also read RFC 5630.)

So I would suggest you conduct an independent security evaluation of what i=
s needed.

For the use case you give:

> A website (a shop) offers a widget in which the visitor can click and
> made a SIP call (+WebRTC) that will end in the callcenter of the
> company, answered by an agent that will inform the user about the
> product he is interested in. Why do we require WWW or SIP
> authentication in this scenario?

I'd suggest that the issue to be discussed is what happens when the action =
described results in a transaction of some form to a third party (in the SI=
P case a call). The visitor then includes information that will be relayed =
to the third party. Who does the third party rely on to ensure that informa=
tion is authentically given by the visitor.

Regards

Keith



> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of I=F1aki Baz Castillo
> Sent: 19 June 2013 12:32
> To: Sa=FAl Ibarra Corretg=E9
> Cc: SIPCORE (Session Initiation Protocol Core) WG; Parthasarathi R
> Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE=
:
> I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
>=20
> 2013/6/19 Sa=FAl Ibarra Corretg=E9 <saul@ag-projects.com>:
> > Why is authentication a MUST? Lets assume that I'm using UDP and my
> proxy establishes a WS connection with a foreign domain's proxy because o=
f
> NAPTR and my proxy supports acting as a WS client. It obviously won't be
> able to authenticate. If this scenario supposed to be covered?
>=20
> Honestly I agree. I cannot find in RFC 3261 (or other RFCs) a
> normative statement mandating authentication, regardless the request
> comes from a UA.
>=20
> In another thread we are discussing about MTI authentication
> mechanisms that must be implemented by SIP WS Clients and Servers.
> IMHO that is correct, but mandating SIP authentication or WWW
> authentication for ALL the scenarios seem innapropriate for me. I come
> back to an use case:
>=20
> A website (a shop) offers a widget in which the visitor can click and
> made a SIP call (+WebRTC) that will end in the callcenter of the
> company, answered by an agent that will inform the user about the
> product he is interested in. Why do we require WWW or SIP
> authentication in this scenario?
>=20
> If WG agrees with this, I will remove the normative statements in
> "Authentication" section, and instead address the MTI authentication
> mechanisms.
>=20
>=20
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From stpeter@stpeter.im  Wed Jun 19 10:11:45 2013
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Subject: [sipcore] Fwd: [Stox] STOX WG officially approved - charter and work plan
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FYI. Please join the stox@ietf.org list to participate:

https://www.ietf.org/mailman/listinfo/stox

Peter


-------- Original Message --------
Subject: [Stox] STOX WG officially approved - charter and work plan
Date: Wed, 19 Jun 2013 09:57:28 +0000
From: <Markus.Isomaki@nokia.com>
To: <stox@ietf.org>
CC: yana@jitsi.org

Hi all,

The SIP-to-XMPP (STOX) WG has now been officially approved by the IESG.
Our charter can be found at
<http://datatracker.ietf.org/wg/stox/charter/>. Myself
(markus.isomaki@nokia.com) and Yana Stamcheva (yana@jitsi.org) will act
as co-chairs.

The plan outlined in the charter is that this will be a focused and
relatively straight-forward effort that we aim to finish in just a few
months. The charter has these milestones:

Jun 2013  Accept starting-point mapping specifications as WG items
Aug 2013  Start Working Group Last Call on mapping specifications
Oct 2013  Submit mapping specifications to the IESG

Yana and I have additionally outlined a bit more detailed plan:

- June: One week consensus call on adopting the current drafts as WG items.
- Latest by July 8: Submit current drafts as -00 WG drafts.
- July: Ask people to comment and raise open issues about the drafts,
ask co-authors to prepare presentations about open issues for IETF 87.
Start recruiting reviewers and if needed, co-authors.
- IETF 87: Meet  for 1 hour. Discuss open issues. Agree on the deadline
for reviews and WG last call on the documents. Decide if all documents
can proceed in the same timeline, or whether some phasing will be needed.
- By end of August: Start the official WG last call.
- September - October: Process last call comments and finalize the
documents for the IESG review.
- IETF 88: Meet if there is interest and energy to discuss about
rechartering, i.e. adding new WG items.

Let us know if you have any feedback to this.

I will next send a few mails to the list about specific topics such as
our first milestone, i.e. the acceptance of the current drafts as WG items.

Markus


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



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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Avasarala, Ranjit \(NSN - IN/Bangalore\)'" <ranjit.avasarala@nsn.com>, =?UTF-8?Q?'ext_I=C3=B1aki_Baz_Castillo'?= <ibc@aliax.net>
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Date: Thu, 20 Jun 2013 01:08:15 +0530
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Cc: "'SIPCORE \(Session Initiation Protocol Core\) WG'" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Hi Inaki,

Just adding to Ranjit mentioned. The text similar to the following has =
to be added. "Record-route header MUST be added by SIP WebSocket Server =
in case WebSocket is the only supported SIP transport between SIP =
WebSocket client & server and SIP WebSocket server acts as SIP proxy"=20

Thanks
Partha


> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Avasarala, Ranjit (NSN - IN/Bangalore)
> Sent: Wednesday, June 19, 2013 11:24 AM
> To: ext I=C3=B1aki Baz Castillo; Parthasarathi R
> Cc: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> Hi Inaki
>=20
> Usually contact header contains the address on which the callee can
> reach caller. But when a SIP message is sent in websocket, what is the
> use of putting contact header?
>=20
> Should Contact header contain browser's address, since SIP UA is =
anyway
> part of browser and the callee cannot anyway reach SIP UA directly.
>=20
> So I think there needs to be text in the draft explaining this
> behavior.
>=20
> Regards
> Ranjit
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of ext I=C3=B1aki Baz Castillo
> Sent: Wednesday, June 19, 2013 4:26 AM
> To: Parthasarathi R
> Cc: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> 2013/6/18 Parthasarathi R <partha@parthasarathi.co.in>:
> > I think that it is required for SIP over WebSocket to explain the
> usage of contact header field with example and normative statement.
>=20
> Honestly I have no idea of what you mean.
> draft-ietf-sipcore-sip-websocket must explain how to use the SIP
> Contact header?
>=20
> Sorry but I don't understand your concern. Could you please describe =
it
> again?
>=20
> Thanks a lot.
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/sipcore


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 20 Jun 2013 00:05:06 +0200
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Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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2013/6/19 Parthasarathi R <partha@parthasarathi.co.in>:
>
> Just adding to Ranjit mentioned. The text similar to the following has to=
 be added. "Record-route header MUST be added by SIP WebSocket Server in ca=
se WebSocket is the only supported SIP transport between SIP WebSocket clie=
nt & server and SIP WebSocket server acts as SIP proxy"

That is just true in case the client is a web browser (a SIP WS Client
but not a SIP WS Server so it can not receive incoming WS
connections). So the text would be:

"Record-route header MUST be added by SIP WebSocket Server in case
WebSocket is the only supported SIP transport between client and
server, the server acts as SIP proxy, and the client is a SIP
WebSocket Client (but not a SIP WebSocket Server so it can not accept
incoming WebSocket connections)."

Does it sound ok?

If the WG agrees with this I will add such an statement in the draft.
BTW, in which section?


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

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 20 Jun 2013 00:09:23 +0200
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, Parthasarathi R <partha@parthasarathi.co.in>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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2013/6/19 Avasarala, Ranjit (NSN - IN/Bangalore) <ranjit.avasarala@nsn.com>=
:
> Hi Inaki
>
> Usually contact header contains the address on which the callee can reach=
 caller. But when a SIP message is sent in websocket, what is the use of pu=
tting contact header?


As said in my previous mail, that is just in case the client is a SIP
WebSocket Client but not a SIP WebSocket Server. Anyhow, the Contact
header MUST be added anyway as it can contain parameters other than
just the UA's address and transport. And it must be added in an INVITE
so the remote peer can create the "dialog target" and use its URI
value as Request-URI for in-dialog requests.

Those are SIP rules, they must be respected.


> Should Contact header contain browser's address, since SIP UA is anyway p=
art of browser and the callee cannot anyway reach SIP UA directly.

Please take a look to the appendix B.1 of the draft:



B.1.  SIP WebSocket Client Considerations

   The JavaScript stack in web browsers does not have the ability to
   discover the local transport address used for originating WebSocket
   connections.  A SIP WebSocket client running in such an environment
   can construct a domain name consisting of a random token followed by
   the ".invalid" top-level domain name, as stated in [RFC2606], and
   uses it within its Via and Contact headers.

      The Contact URI provided by SIP UAs requesting (and receiving)
      Outbound support is not used for routing requests to those UAs,
      thus it is safe to set a random domain in the Contact URI
      hostport.


Clear? :)

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

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 20 Jun 2013 00:10:43 +0200
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Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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Thanks a lot Keith, it's clear. I will address it in next revision.

2013/6/19 DRAGE, Keith (Keith) <keith.drage@alcatel-lucent.com>:
> I would not use RFC 3261 as justification for what should, or should not,=
 be said about authentication. The current RFC 3261 would probably fail a s=
ecurity directorate review if it was attempted to be approved as an RFC now=
.
>
> (I'd also point out that for any security consideration of RFC 3261, one =
should also read RFC 5630.)
>
> So I would suggest you conduct an independent security evaluation of what=
 is needed.
>
> For the use case you give:
>
>> A website (a shop) offers a widget in which the visitor can click and
>> made a SIP call (+WebRTC) that will end in the callcenter of the
>> company, answered by an agent that will inform the user about the
>> product he is interested in. Why do we require WWW or SIP
>> authentication in this scenario?
>
> I'd suggest that the issue to be discussed is what happens when the actio=
n described results in a transaction of some form to a third party (in the =
SIP case a call). The visitor then includes information that will be relaye=
d to the third party. Who does the third party rely on to ensure that infor=
mation is authentically given by the visitor.
>
> Regards
>
> Keith
>
>
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Beha=
lf
>> Of I=C3=B1aki Baz Castillo
>> Sent: 19 June 2013 12:32
>> To: Sa=C3=BAl Ibarra Corretg=C3=A9
>> Cc: SIPCORE (Session Initiation Protocol Core) WG; Parthasarathi R
>> Subject: Re: [sipcore] No WebSocket level authentication scenario [was R=
E:
>> I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
>>
>> 2013/6/19 Sa=C3=BAl Ibarra Corretg=C3=A9 <saul@ag-projects.com>:
>> > Why is authentication a MUST? Lets assume that I'm using UDP and my
>> proxy establishes a WS connection with a foreign domain's proxy because =
of
>> NAPTR and my proxy supports acting as a WS client. It obviously won't be
>> able to authenticate. If this scenario supposed to be covered?
>>
>> Honestly I agree. I cannot find in RFC 3261 (or other RFCs) a
>> normative statement mandating authentication, regardless the request
>> comes from a UA.
>>
>> In another thread we are discussing about MTI authentication
>> mechanisms that must be implemented by SIP WS Clients and Servers.
>> IMHO that is correct, but mandating SIP authentication or WWW
>> authentication for ALL the scenarios seem innapropriate for me. I come
>> back to an use case:
>>
>> A website (a shop) offers a widget in which the visitor can click and
>> made a SIP call (+WebRTC) that will end in the callcenter of the
>> company, answered by an agent that will inform the user about the
>> product he is interested in. Why do we require WWW or SIP
>> authentication in this scenario?
>>
>> If WG agrees with this, I will remove the normative statements in
>> "Authentication" section, and instead address the MTI authentication
>> mechanisms.
>>
>>
>> --
>> I=C3=B1aki Baz Castillo
>> <ibc@aliax.net>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore



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

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Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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On 6/19/13 6:05 PM, IÃ±aki Baz Castillo wrote:
> 2013/6/19 Parthasarathi R <partha@parthasarathi.co.in>:
>>
>> Just adding to Ranjit mentioned. The text similar to the following has to be added. "Record-route header MUST be added by SIP WebSocket Server in case WebSocket is the only supported SIP transport between SIP WebSocket client & server and SIP WebSocket server acts as SIP proxy"
>
> That is just true in case the client is a web browser (a SIP WS Client
> but not a SIP WS Server so it can not receive incoming WS
> connections). So the text would be:
>
> "Record-route header MUST be added by SIP WebSocket Server in case
> WebSocket is the only supported SIP transport between client and
> server, the server acts as SIP proxy, and the client is a SIP
> WebSocket Client (but not a SIP WebSocket Server so it can not accept
> incoming WebSocket connections)."

How will the server know that the client is not capable of being a 
WebSocket Server?

I guess in many cases it might be able to know that because the client 
is a browser being controlled by scripting from the server. But in other 
cases ISTM that it just won't know.

	Thanks,
	Paul


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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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On 6/19/13 8:06 AM, DRAGE, Keith (Keith) wrote:
> I would not use RFC 3261 as justification for what should, or should not, be said about authentication. The current RFC 3261 would probably fail a security directorate review if it was attempted to be approved as an RFC now.
>
> (I'd also point out that for any security consideration of RFC 3261, one should also read RFC 5630.)
>
> So I would suggest you conduct an independent security evaluation of what is needed.

I think we are in the midst of one with Stephen Farrell.

> For the use case you give:
>
>> A website (a shop) offers a widget in which the visitor can click and
>> made a SIP call (+WebRTC) that will end in the callcenter of the
>> company, answered by an agent that will inform the user about the
>> product he is interested in. Why do we require WWW or SIP
>> authentication in this scenario?
>
> I'd suggest that the issue to be discussed is what happens when the action described results in a transaction of some form to a third party (in the SIP case a call). The visitor then includes information that will be relayed to the third party. Who does the third party rely on to ensure that information is authentically given by the visitor.

I'm inclined to support Iñaki, that authentication of any sort shouldn't 
be Mandatory to *Use*.  Individual applications can decide when they 
have uses that require authentication and when they don't.

	Thanks,
	Paul

> Regards
>
> Keith
>
>
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
>> Of Iñaki Baz Castillo
>> Sent: 19 June 2013 12:32
>> To: Saúl Ibarra Corretgé
>> Cc: SIPCORE (Session Initiation Protocol Core) WG; Parthasarathi R
>> Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE:
>> I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
>>
>> 2013/6/19 Saúl Ibarra Corretgé <saul@ag-projects.com>:
>>> Why is authentication a MUST? Lets assume that I'm using UDP and my
>> proxy establishes a WS connection with a foreign domain's proxy because of
>> NAPTR and my proxy supports acting as a WS client. It obviously won't be
>> able to authenticate. If this scenario supposed to be covered?
>>
>> Honestly I agree. I cannot find in RFC 3261 (or other RFCs) a
>> normative statement mandating authentication, regardless the request
>> comes from a UA.
>>
>> In another thread we are discussing about MTI authentication
>> mechanisms that must be implemented by SIP WS Clients and Servers.
>> IMHO that is correct, but mandating SIP authentication or WWW
>> authentication for ALL the scenarios seem innapropriate for me. I come
>> back to an use case:
>>
>> A website (a shop) offers a widget in which the visitor can click and
>> made a SIP call (+WebRTC) that will end in the callcenter of the
>> company, answered by an agent that will inform the user about the
>> product he is interested in. Why do we require WWW or SIP
>> authentication in this scenario?
>>
>> If WG agrees with this, I will remove the normative statements in
>> "Authentication" section, and instead address the MTI authentication
>> mechanisms.
>>
>>
>> --
>> Iñaki Baz Castillo
>> <ibc@aliax.net>
>> _______________________________________________
>> 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: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>, <sipcore@ietf.org>
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Date: Tue, 25 Jun 2013 06:11:35 +0530
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Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Paul,

I think that we are talking about two problems here

1) SIP routing issue because of SIP WebSocket Client has only WebSocket =
as a transport (Browser scenario)
	a) Solution: Record-route header has to be added by SIP WebSocket =
Server.
=20
2) How will SIP WebSocket Server know that SIP WebSocket client supports =
only SIP WebSocket as a transport? (Non-browser scenario)
	a) AFAIK, there is no other environment other than browser has the =
limitation of not supporting UDP & TCP. In case any other environment =
exists, Please let me know.
	b) The beauty of RFC 3261 is that there is no need to negotiate the set =
of "SIP" transport protocol as it mandates for UDP, TCP. I agree that it =
is an open item how to represent the list of transport supported by a =
SIP UA in case of supporting WebSocket only transport scenario. But I'm =
not sure whether this WG is interested in solving this issue. We need to =
here from more folks and its relevant usecases.

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Thursday, June 20, 2013 9:28 PM
> To: sipcore@ietf.org
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> On 6/19/13 6:05 PM, I=C3=B1aki Baz Castillo wrote:
> > 2013/6/19 Parthasarathi R <partha@parthasarathi.co.in>:
> >>
> >> Just adding to Ranjit mentioned. The text similar to the following
> has to be added. "Record-route header MUST be added by SIP WebSocket
> Server in case WebSocket is the only supported SIP transport between
> SIP WebSocket client & server and SIP WebSocket server acts as SIP
> proxy"
> >
> > That is just true in case the client is a web browser (a SIP WS
> Client
> > but not a SIP WS Server so it can not receive incoming WS
> > connections). So the text would be:
> >
> > "Record-route header MUST be added by SIP WebSocket Server in case
> > WebSocket is the only supported SIP transport between client and
> > server, the server acts as SIP proxy, and the client is a SIP
> > WebSocket Client (but not a SIP WebSocket Server so it can not =
accept
> > incoming WebSocket connections)."
>=20
> How will the server know that the client is not capable of being a
> WebSocket Server?
>=20
> I guess in many cases it might be able to know that because the client
> is a browser being controlled by scripting from the server. But in
> other
> cases ISTM that it just won't know.
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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Paul,

In case it is not mandatory to use authentication, the following =
statement
in Sec 7 of the draft is not correct:

"  If no authentication is done at WebSocket level then SIP Digest
   authentication is required for every SIP request coming over the
   WebSocket connection."

Please let me know your comment on the same.

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Thursday, June 20, 2013 9:37 PM
> To: sipcore@ietf.org
> Subject: Re: [sipcore] No WebSocket level authentication scenario [was
> RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
>=20
> On 6/19/13 8:06 AM, DRAGE, Keith (Keith) wrote:
> > I would not use RFC 3261 as justification for what should, or should
> not, be said about authentication. The current RFC 3261 would probably
> fail a security directorate review if it was attempted to be approved
> as an RFC now.
> >
> > (I'd also point out that for any security consideration of RFC 3261,
> one should also read RFC 5630.)
> >
> > So I would suggest you conduct an independent security evaluation of
> what is needed.
>=20
> I think we are in the midst of one with Stephen Farrell.
>=20
> > For the use case you give:
> >
> >> A website (a shop) offers a widget in which the visitor can click
> and
> >> made a SIP call (+WebRTC) that will end in the callcenter of the
> >> company, answered by an agent that will inform the user about the
> >> product he is interested in. Why do we require WWW or SIP
> >> authentication in this scenario?
> >
> > I'd suggest that the issue to be discussed is what happens when the
> action described results in a transaction of some form to a third =
party
> (in the SIP case a call). The visitor then includes information that
> will be relayed to the third party. Who does the third party rely on =
to
> ensure that information is authentically given by the visitor.
>=20
> I'm inclined to support I=F1aki, that authentication of any sort
> shouldn't
> be Mandatory to *Use*.  Individual applications can decide when they
> have uses that require authentication and when they don't.
>=20
> 	Thanks,
> 	Paul
>=20
> > Regards
> >
> > Keith
> >
> >
> >
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf
> >> Of I=F1aki Baz Castillo
> >> Sent: 19 June 2013 12:32
> >> To: Sa=FAl Ibarra Corretg=E9
> >> Cc: SIPCORE (Session Initiation Protocol Core) WG; Parthasarathi R
> >> Subject: Re: [sipcore] No WebSocket level authentication scenario
> [was RE:
> >> I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
> >>
> >> 2013/6/19 Sa=FAl Ibarra Corretg=E9 <saul@ag-projects.com>:
> >>> Why is authentication a MUST? Lets assume that I'm using UDP and =
my
> >> proxy establishes a WS connection with a foreign domain's proxy
> because of
> >> NAPTR and my proxy supports acting as a WS client. It obviously
> won't be
> >> able to authenticate. If this scenario supposed to be covered?
> >>
> >> Honestly I agree. I cannot find in RFC 3261 (or other RFCs) a
> >> normative statement mandating authentication, regardless the =
request
> >> comes from a UA.
> >>
> >> In another thread we are discussing about MTI authentication
> >> mechanisms that must be implemented by SIP WS Clients and Servers.
> >> IMHO that is correct, but mandating SIP authentication or WWW
> >> authentication for ALL the scenarios seem innapropriate for me. I
> come
> >> back to an use case:
> >>
> >> A website (a shop) offers a widget in which the visitor can click
> and
> >> made a SIP call (+WebRTC) that will end in the callcenter of the
> >> company, answered by an agent that will inform the user about the
> >> product he is interested in. Why do we require WWW or SIP
> >> authentication in this scenario?
> >>
> >> If WG agrees with this, I will remove the normative statements in
> >> "Authentication" section, and instead address the MTI =
authentication
> >> mechanisms.
> >>
> >>
> >> --
> >> I=F1aki Baz Castillo
> >> <ibc@aliax.net>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From pkyzivat@alum.mit.edu  Wed Jun 26 08:14:04 2013
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Partha,

You addressed your reply to me. But I just posed the question, not the 
answer. I'm waiting for those who proposed the language to clarify it.

The real issue is: how can a websocket server ever decide that it 
*doesn't* need to R-R?

	Thanks,
	Paul

On 6/24/13 8:41 PM, Parthasarathi R wrote:
> Paul,
>
> I think that we are talking about two problems here
>
> 1) SIP routing issue because of SIP WebSocket Client has only WebSocket as a transport (Browser scenario)
> 	a) Solution: Record-route header has to be added by SIP WebSocket Server.
>
> 2) How will SIP WebSocket Server know that SIP WebSocket client supports only SIP WebSocket as a transport? (Non-browser scenario)
> 	a) AFAIK, there is no other environment other than browser has the limitation of not supporting UDP & TCP. In case any other environment exists, Please let me know.
> 	b) The beauty of RFC 3261 is that there is no need to negotiate the set of "SIP" transport protocol as it mandates for UDP, TCP. I agree that it is an open item how to represent the list of transport supported by a SIP UA in case of supporting WebSocket only transport scenario. But I'm not sure whether this WG is interested in solving this issue. We need to here from more folks and its relevant usecases.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: Thursday, June 20, 2013 9:28 PM
>> To: sipcore@ietf.org
>> Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-
>> 09
>>
>> On 6/19/13 6:05 PM, IÃ±aki Baz Castillo wrote:
>>> 2013/6/19 Parthasarathi R <partha@parthasarathi.co.in>:
>>>>
>>>> Just adding to Ranjit mentioned. The text similar to the following
>> has to be added. "Record-route header MUST be added by SIP WebSocket
>> Server in case WebSocket is the only supported SIP transport between
>> SIP WebSocket client & server and SIP WebSocket server acts as SIP
>> proxy"
>>>
>>> That is just true in case the client is a web browser (a SIP WS
>> Client
>>> but not a SIP WS Server so it can not receive incoming WS
>>> connections). So the text would be:
>>>
>>> "Record-route header MUST be added by SIP WebSocket Server in case
>>> WebSocket is the only supported SIP transport between client and
>>> server, the server acts as SIP proxy, and the client is a SIP
>>> WebSocket Client (but not a SIP WebSocket Server so it can not accept
>>> incoming WebSocket connections)."
>>
>> How will the server know that the client is not capable of being a
>> WebSocket Server?
>>
>> I guess in many cases it might be able to know that because the client
>> is a browser being controlled by scripting from the server. But in
>> other
>> cases ISTM that it just won't know.
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>
>


From pkyzivat@alum.mit.edu  Wed Jun 26 08:16:25 2013
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] No WebSocket level authentication scenario [was RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
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Partha,

Again, I'm not the one to answer. I'm just the chair, not the author.

	Thanks,
	Paul

On 6/24/13 8:47 PM, Parthasarathi R wrote:
> Paul,
>
> In case it is not mandatory to use authentication, the following statement
> in Sec 7 of the draft is not correct:
>
> "  If no authentication is done at WebSocket level then SIP Digest
>     authentication is required for every SIP request coming over the
>     WebSocket connection."
>
> Please let me know your comment on the same.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: Thursday, June 20, 2013 9:37 PM
>> To: sipcore@ietf.org
>> Subject: Re: [sipcore] No WebSocket level authentication scenario [was
>> RE: I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
>>
>> On 6/19/13 8:06 AM, DRAGE, Keith (Keith) wrote:
>>> I would not use RFC 3261 as justification for what should, or should
>> not, be said about authentication. The current RFC 3261 would probably
>> fail a security directorate review if it was attempted to be approved
>> as an RFC now.
>>>
>>> (I'd also point out that for any security consideration of RFC 3261,
>> one should also read RFC 5630.)
>>>
>>> So I would suggest you conduct an independent security evaluation of
>> what is needed.
>>
>> I think we are in the midst of one with Stephen Farrell.
>>
>>> For the use case you give:
>>>
>>>> A website (a shop) offers a widget in which the visitor can click
>> and
>>>> made a SIP call (+WebRTC) that will end in the callcenter of the
>>>> company, answered by an agent that will inform the user about the
>>>> product he is interested in. Why do we require WWW or SIP
>>>> authentication in this scenario?
>>>
>>> I'd suggest that the issue to be discussed is what happens when the
>> action described results in a transaction of some form to a third party
>> (in the SIP case a call). The visitor then includes information that
>> will be relayed to the third party. Who does the third party rely on to
>> ensure that information is authentically given by the visitor.
>>
>> I'm inclined to support Iñaki, that authentication of any sort
>> shouldn't
>> be Mandatory to *Use*.  Individual applications can decide when they
>> have uses that require authentication and when they don't.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Regards
>>>
>>> Keith
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf
>>>> Of Iñaki Baz Castillo
>>>> Sent: 19 June 2013 12:32
>>>> To: Saúl Ibarra Corretgé
>>>> Cc: SIPCORE (Session Initiation Protocol Core) WG; Parthasarathi R
>>>> Subject: Re: [sipcore] No WebSocket level authentication scenario
>> [was RE:
>>>> I-D Action: draft-ietf-sipcore-sip-websocket-09.txt]
>>>>
>>>> 2013/6/19 Saúl Ibarra Corretgé <saul@ag-projects.com>:
>>>>> Why is authentication a MUST? Lets assume that I'm using UDP and my
>>>> proxy establishes a WS connection with a foreign domain's proxy
>> because of
>>>> NAPTR and my proxy supports acting as a WS client. It obviously
>> won't be
>>>> able to authenticate. If this scenario supposed to be covered?
>>>>
>>>> Honestly I agree. I cannot find in RFC 3261 (or other RFCs) a
>>>> normative statement mandating authentication, regardless the request
>>>> comes from a UA.
>>>>
>>>> In another thread we are discussing about MTI authentication
>>>> mechanisms that must be implemented by SIP WS Clients and Servers.
>>>> IMHO that is correct, but mandating SIP authentication or WWW
>>>> authentication for ALL the scenarios seem innapropriate for me. I
>> come
>>>> back to an use case:
>>>>
>>>> A website (a shop) offers a widget in which the visitor can click
>> and
>>>> made a SIP call (+WebRTC) that will end in the callcenter of the
>>>> company, answered by an agent that will inform the user about the
>>>> product he is interested in. Why do we require WWW or SIP
>>>> authentication in this scenario?
>>>>
>>>> If WG agrees with this, I will remove the normative statements in
>>>> "Authentication" section, and instead address the MTI authentication
>>>> mechanisms.
>>>>
>>>>
>>>> --
>>>> Iñaki Baz Castillo
>>>> <ibc@aliax.net>
>>>> _______________________________________________
>>>> 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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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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On 26 June 2013 16:13, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> The real issue is: how can a websocket server ever decide that it *doesn'=
t*
> need to R-R?

The websocket server shouldn't R-R a request if it is a non-dialog-establis=
hing
request (e.g. REGISTER, PUBLISH) or an in-dialog non-target-refresh request
(e.g. INFO).  It is probably fair to say that INVITEs should be, but
that specific
case is already covered by RFC5626, as I=F1aki has pointed out.

Regards,

Michael

From pkyzivat@alum.mit.edu  Thu Jun 27 07:58:18 2013
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To: Michael Procter <michael@voip.co.uk>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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On 6/27/13 9:57 AM, Michael Procter wrote:
> On 26 June 2013 16:13, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> The real issue is: how can a websocket server ever decide that it *doesn't*
>> need to R-R?
>
> The websocket server shouldn't R-R a request if it is a non-dialog-establishing
> request (e.g. REGISTER, PUBLISH) or an in-dialog non-target-refresh request
> (e.g. INFO).  It is probably fair to say that INVITEs should be, but
> that specific
> case is already covered by RFC5626, as Iñaki has pointed out.

You miss my point. The cases you describe above are cases where it never 
makes sense to R-R. The only place the question is relevant is on dialog 
establishing requests.

The simple answer would be that websocket proxy handling a dialog 
establishing request should *always* R-R. But there has been an 
assertion that there may be some sip-websocket UACs that are also 
capable of being sip-websocket servers. In that case, it would be 
desirable for the proxy to *not* R-R, so that it won't be in the 
signaling path for the entire dialog. But how would the proxy know that 
it is safe to not R-R?

One thing that comes to mind for me is that the UAC could decide whether 
or not to tag its contact URI with ";gr". If it were so tagged, then 
that says it is globally routable, so it would be safe for the proxy to 
bypass the R-R. If the contact isn't a gruu, then the proxy would decide 
that it must R-R.

I think Iñaki said he was on vacation for a couple of weeks, so we may 
not hear from him on this soon.

	Thanks,
	Paul


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From: Michael Procter <michael@voip.co.uk>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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On 27 June 2013 15:57, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> You miss my point. The cases you describe above are cases where it never
> makes sense to R-R. The only place the question is relevant is on dialog
> establishing requests.

Apologies.  Partha seemed to be edging towards a more general
statement than simply for dialog establishing requests and your
question was similarly unrestricted.

> One thing that comes to mind for me is that the UAC could decide whether or
> not to tag its contact URI with ";gr". If it were so tagged, then that says
> it is globally routable, so it would be safe for the proxy to bypass the
> R-R.

That sounds like a step forward.  I think this problem, specifically
for the non-browser scenario, has overlaps with the scenario RFC5923
addresses.  Maybe the 'alias' via parameter would also be a clue that
R-R is not necessarily required.

Regards,

Michael

From partha@parthasarathi.co.in  Thu Jun 27 09:09:38 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Paul,

Thanks for the clarification. Let us wait for the clarification on R-R =
issue.

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Wednesday, June 26, 2013 8:44 PM
> To: Parthasarathi R
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Open issues in =
draft-ietf-sipcore-sip-websocket-
> 09
>=20
> Partha,
>=20
> You addressed your reply to me. But I just posed the question, not the
> answer. I'm waiting for those who proposed the language to clarify it.
>=20
> The real issue is: how can a websocket server ever decide that it
> *doesn't* need to R-R?
>=20
> 	Thanks,
> 	Paul
>=20
> On 6/24/13 8:41 PM, Parthasarathi R wrote:
> > Paul,
> >
> > I think that we are talking about two problems here
> >
> > 1) SIP routing issue because of SIP WebSocket Client has only
> WebSocket as a transport (Browser scenario)
> > 	a) Solution: Record-route header has to be added by SIP WebSocket
> Server.
> >
> > 2) How will SIP WebSocket Server know that SIP WebSocket client
> supports only SIP WebSocket as a transport? (Non-browser scenario)
> > 	a) AFAIK, there is no other environment other than browser has
> the limitation of not supporting UDP & TCP. In case any other
> environment exists, Please let me know.
> > 	b) The beauty of RFC 3261 is that there is no need to negotiate
> the set of "SIP" transport protocol as it mandates for UDP, TCP. I
> agree that it is an open item how to represent the list of transport
> supported by a SIP UA in case of supporting WebSocket only transport
> scenario. But I'm not sure whether this WG is interested in solving
> this issue. We need to here from more folks and its relevant usecases.
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> >> Behalf Of Paul Kyzivat
> >> Sent: Thursday, June 20, 2013 9:28 PM
> >> To: sipcore@ietf.org
> >> Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-
> websocket-
> >> 09
> >>
> >> On 6/19/13 6:05 PM, I=C3=B1aki Baz Castillo wrote:
> >>> 2013/6/19 Parthasarathi R <partha@parthasarathi.co.in>:
> >>>>
> >>>> Just adding to Ranjit mentioned. The text similar to the =
following
> >> has to be added. "Record-route header MUST be added by SIP =
WebSocket
> >> Server in case WebSocket is the only supported SIP transport =
between
> >> SIP WebSocket client & server and SIP WebSocket server acts as SIP
> >> proxy"
> >>>
> >>> That is just true in case the client is a web browser (a SIP WS
> >> Client
> >>> but not a SIP WS Server so it can not receive incoming WS
> >>> connections). So the text would be:
> >>>
> >>> "Record-route header MUST be added by SIP WebSocket Server in case
> >>> WebSocket is the only supported SIP transport between client and
> >>> server, the server acts as SIP proxy, and the client is a SIP
> >>> WebSocket Client (but not a SIP WebSocket Server so it can not
> accept
> >>> incoming WebSocket connections)."
> >>
> >> How will the server know that the client is not capable of being a
> >> WebSocket Server?
> >>
> >> I guess in many cases it might be able to know that because the
> client
> >> is a browser being controlled by scripting from the server. But in
> >> other
> >> cases ISTM that it just won't know.
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> >
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From jmillan@aliax.net  Fri Jun 28 02:50:01 2013
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From: =?UTF-8?B?Sm9zw6kgTHVpcyBNaWxsw6Fu?= <jmillan@aliax.net>
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Subject: [sipcore] Outbound clarification
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Hi,

May someone help me parsing this phrase from Outbound specification?

The prhase I don't understand is:
'''
that was instantiated using a Contact header field value that included an
"ob" parameter
'''

It is taken from RFC 5626 section 4.3. "Sending Non-REGISTER Requests":

'''
If the UAC is using a Globally
Routable UA URI (GRUU) [RFC5627] that was instantiated using a
Contact header field value that included an "ob" parameter, the UAC
sends the request over the flow used for registration, and subsequent
requests will arrive over that same flow.  If the UAC is not using
such a GRUU, then the UAC adds an "ob" parameter to its Contact
header field value.
'''

Thanks a lot

--047d7bd74cf883edf304e033c7f1
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<div dir=3D"ltr"><div>Hi,</div><div><br></div><div>May someone help me pars=
ing this phrase from Outbound specification?</div><div><br></div><div>The p=
rhase I don&#39;t understand is:</div><div>&#39;&#39;&#39;</div><div>that w=
as instantiated using a=C2=A0Contact header field value that included an &q=
uot;ob&quot; parameter<br>
</div><div>&#39;&#39;&#39;</div><div><br></div><div>It is taken from RFC 56=
26 section 4.3. &quot;Sending Non-REGISTER Requests&quot;:</div><div><br></=
div>&#39;&#39;&#39;<br>If the UAC is using a Globally<br>Routable UA URI (G=
RUU) [RFC5627] that was instantiated using a<br>
Contact header field value that included an &quot;ob&quot; parameter, the U=
AC<br>sends the request over the flow used for registration, and subsequent=
<br>requests will arrive over that same flow. =C2=A0If the UAC is not using=
<br>
such a GRUU, then the UAC adds an &quot;ob&quot; parameter to its Contact<b=
r>header field value.<div>&#39;&#39;&#39;</div><div><br></div><div>Thanks a=
 lot</div></div>

--047d7bd74cf883edf304e033c7f1--

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Subject: Re: [sipcore] Outbound clarification
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May it mean that the GRUU was obtained from a REGISTER request which
Contact header contained the ;ob parameter?


2013/6/28 Jos=C3=A9 Luis Mill=C3=A1n <jmillan@aliax.net>

> Hi,
>
> May someone help me parsing this phrase from Outbound specification?
>
> The prhase I don't understand is:
> '''
> that was instantiated using a Contact header field value that included an
> "ob" parameter
> '''
>
> It is taken from RFC 5626 section 4.3. "Sending Non-REGISTER Requests":
>
> '''
> If the UAC is using a Globally
> Routable UA URI (GRUU) [RFC5627] that was instantiated using a
> Contact header field value that included an "ob" parameter, the UAC
> sends the request over the flow used for registration, and subsequent
> requests will arrive over that same flow.  If the UAC is not using
> such a GRUU, then the UAC adds an "ob" parameter to its Contact
> header field value.
> '''
>
> Thanks a lot
>



--=20
Jos=C3=A9 Luis Mill=C3=A1n

--047d7bd750b282088a04e0345b91
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">May it mean that the GRUU was obtained from a REGISTER req=
uest which Contact header contained the ;ob parameter?</div><div class=3D"g=
mail_extra"><br><br><div class=3D"gmail_quote">2013/6/28 Jos=C3=A9 Luis Mil=
l=C3=A1n <span dir=3D"ltr">&lt;<a href=3D"mailto:jmillan@aliax.net" target=
=3D"_blank">jmillan@aliax.net</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div>Hi,</div><div><br></di=
v><div>May someone help me parsing this phrase from Outbound specification?=
</div>
<div><br></div><div>The prhase I don&#39;t understand is:</div><div>&#39;&#=
39;&#39;</div><div>that was instantiated using a=C2=A0Contact header field =
value that included an &quot;ob&quot; parameter<br>
</div><div>&#39;&#39;&#39;</div><div><br></div><div>It is taken from RFC 56=
26 section 4.3. &quot;Sending Non-REGISTER Requests&quot;:</div><div><br></=
div>&#39;&#39;&#39;<br>If the UAC is using a Globally<br>Routable UA URI (G=
RUU) [RFC5627] that was instantiated using a<br>

Contact header field value that included an &quot;ob&quot; parameter, the U=
AC<br>sends the request over the flow used for registration, and subsequent=
<br>requests will arrive over that same flow. =C2=A0If the UAC is not using=
<br>

such a GRUU, then the UAC adds an &quot;ob&quot; parameter to its Contact<b=
r>header field value.<div>&#39;&#39;&#39;</div><div><br></div><div>Thanks a=
 lot</div></div>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>Jos=C3=A9 Lu=
is Mill=C3=A1n
</div>

--047d7bd750b282088a04e0345b91--

From worley@shell01.TheWorld.com  Fri Jun 28 13:35:33 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: Jose Luis Millan <jmillan@aliax.net>
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Subject: Re: [sipcore] Outbound clarification
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> From: JosC) Luis MillC!n <jmillan@aliax.net>
> 
> May it mean that the GRUU was obtained from a REGISTER request which
> Contact header contained the ;ob parameter?

Yes, I believe that is what it means.  Every GRUU refers to an
underlying Contact address, which is where requests to that GRUU will
be routed to.  That Contact address is what was used in the REGISTER
request to obtain the GRUU.  The question seems to be, "Does this GRUU
refer to a Contact address that includes an "ob" parameter?"

Dale
