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From: Adam Roach <adam@nostrum.com>
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Subject: [sipcore] SIPCORE Rechartering
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[as SIPCORE chair]

Back when we transitioned from the SIPPING/SIP working group structure 
to SIPCORE, we put relatively tight limits on the SIPCORE charter. This 
was a recognition of the fact that the volume of SIP-related work at the 
time was too large for any one working group to reasonably handle. 
Instead, the intention was that the DISPATCH model of creating small, 
short-lived working groups for specific topics would be a better way to 
manage the relatively heavy workload.

Fast forward seven years to today. SIP is a much more mature protocol, 
and the inflow of new work is significantly smaller than it was back 
then. At the same time, we have had several small, self-contained work 
items show up in DISPATCH that were deemed too small to create a new 
working group for, but precluded from being dispatched to SIPCORE by its 
charter. This has led to unnecessary delays as we determined the best 
disposition for these items.

We, the SIPCORE chairs and area director, seek to fix that. We would 
like to amend the SIPCORE charter to allow it to take on small, 
self-contained protocol extensions in addition to changes to the core 
SIP protocol. This is a relatively minor update to the existing charter, 
which expands the scope as described above, and adds back in two of the 
architectural principles from the SIP charter which are applicable only 
to protocol extensions.

Please provide feedback on the proposed charter by sending email to the 
SIPCORE mailing list. We will also discuss this briefly during the 
DISPATCH session in Seoul.

The proposed charter text follows.


> The Session Initiation Protocol Core (SIPCore) working group is
> chartered to maintain and continue the development of the SIP protocol,
> currently defined as proposed standard RFCs 3261, 3262, 3263, 3264, and
> 6665.
>
> The SIPCore working group will concentrate on specifications that update
> or replace the core SIP specifications named above as well as
> specifications pertaining to small, self-contained SIP protocol
> extensions. The process and requirements for new SIP extensions are
> documented in RFC5727, "Change Process for the Session Initiation
> Protocol".
>
> Throughout its work, the group will strive to maintain the basic model
> and architecture defined by SIP. In particular:
>
> 1. Services and features are provided end-to-end whenever possible.
>
> 2. Reuse of existing Internet protocols and architectures and
> integrating with other Internet applications is crucial.
>
> 3. Standards-track extensions and new features must be generally
> applicable, and not applicable only to a specific set of session
> types.
>
> 4. Simpler solutions that solve a given problem should be favored
> over more complex solutions.
>
> The primary source of change requirements to the core SIP specifications
> (enumerated above) will be a) interoperability problems that stem from
> ambiguous, or under-defined specification, and b) requirements from
> other working groups in the ART Area. The primary source of new protocol
> extensions is the DISPATCH working group, which will generally make the
> determination regarding whether new SIP-related work warrants a new
> working group or belongs in an existing one.


For ease of reference, the existing SIPCORE charter is at 
https://datatracker.ietf.org/doc/charter-ietf-sipcore/

Thanks!

/a


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    <p>[as SIPCORE chair]</p>
    <p>Back when we transitioned from the SIPPING/SIP working group
      structure to SIPCORE, we put relatively tight limits on the
      SIPCORE charter. This was a recognition of the fact that the
      volume of SIP-related work at the time was too large for any one
      working group to reasonably handle. Instead, the intention was
      that the DISPATCH model of creating small, short-lived working
      groups for specific topics would be a better way to manage the
      relatively heavy workload.</p>
    <p>Fast forward seven years to today. SIP is a much more mature
      protocol, and the inflow of new work is significantly smaller than
      it was back then. At the same time, we have had several small,
      self-contained work items show up in DISPATCH that were deemed too
      small to create a new working group for, but precluded from being
      dispatched to SIPCORE by its charter. This has led to unnecessary
      delays as we determined the best disposition for these items.<br>
    </p>
    <p>We, the SIPCORE chairs and area director, seek to fix that. We
      would like to amend the SIPCORE charter to allow it to take on
      small, self-contained protocol extensions in addition to changes
      to the core SIP protocol. This is a relatively minor update to the
      existing charter, which expands the scope as described above, and
      adds back in two of the architectural principles from the SIP
      charter which are applicable only to protocol extensions.</p>
    <p>Please provide feedback on the proposed charter by sending email
      to the SIPCORE mailing list. We will also discuss this briefly
      during the DISPATCH session in Seoul.</p>
    <p>The proposed charter text follows.</p>
    <p><br>
    </p>
    <p>
      <blockquote type="cite">
        <meta http-equiv="content-type" content="text/html;
          charset=utf-8">
        <div id="magicdomid2" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The
            Session Initiation Protocol Core (SIPCore) working group is</span></div>
        <div id="magicdomid3" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">chartered
            to maintain and continue the development of the SIP
            protocol,</span></div>
        <div id="magicdomid4" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">currently
            defined as proposed standard RFCs 3261, 3262, 3263, 3264,
            and</span></div>
        <div id="magicdomid5" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">6665.</span></div>
        <div id="magicdomid6" class=""><br>
        </div>
        <div id="magicdomid7" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The
            SIPCore working group will concentrate on specifications
            that update</span></div>
        <div id="magicdomid8" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">or
            replace the core SIP specifications named above as well as</span></div>
        <div id="magicdomid9" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">specifications
            pertaining to small, self-contained SIP protocol</span></div>
        <div id="magicdomid10" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">extensions. 
            The process and requirements for new SIP extensions are</span></div>
        <div id="magicdomid11" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">documented
            in RFC5727, "Change Process for the Session Initiation</span></div>
        <div id="magicdomid12" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Protocol".</span></div>
        <div id="magicdomid13" class=""><br>
        </div>
        <div id="magicdomid14" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Throughout
            its work, the group will strive to maintain the basic model</span></div>
        <div id="magicdomid15" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">and
            architecture defined by SIP. In particular:</span></div>
        <div id="magicdomid16" class=""><br>
        </div>
        <div id="magicdomid17" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">1.
            Services and features are provided end-to-end whenever
            possible.</span></div>
        <div id="magicdomid18" class=""><br>
        </div>
        <div id="magicdomid19" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">2.
            Reuse of existing Internet protocols and architectures and</span></div>
        <div id="magicdomid20" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">  
            integrating with other Internet applications is crucial.</span></div>
        <div id="magicdomid21" class=""><br>
        </div>
        <div id="magicdomid22" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">3.
            Standards-track extensions and new features must be
            generally</span></div>
        <div id="magicdomid23" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">  
            applicable, and not applicable only to a specific set of
            session</span></div>
        <div id="magicdomid24" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">  
            types.</span></div>
        <div id="magicdomid25" class=""><br>
        </div>
        <div id="magicdomid26" class=""><span
            class="author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">4.
            Simpler solutions that solve a given problem should be
            favored</span></div>
        <div id="magicdomid27" class=""><span
            class="author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">   
            over more complex solutions.</span></div>
        <div id="magicdomid28" class=""><br>
        </div>
        <div id="magicdomid29" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The
            primary source of change requirements to the core SIP
            specifications</span></div>
        <div id="magicdomid30" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">(enumerated
            above) will be a) interoperability problems that stem from</span></div>
        <div id="magicdomid31" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">ambiguous,
            or under-defined specification, and b) requirements from</span></div>
        <div id="magicdomid32" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">other
            working groups in the ART Area. The primary source of new
            protocol</span></div>
        <div id="magicdomid33" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">extensions
            is the DISPATCH working group, which will generally make the</span></div>
        <div id="magicdomid34" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">determination
            regarding whether new SIP-related work warrants a new</span></div>
        <div id="magicdomid35" class=""><span
            class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">working
            group or belongs in an existing one.</span></div>
      </blockquote>
    </p>
    <p><br>
    </p>
    <p>For ease of reference, the existing SIPCORE charter is at
      <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/charter-ietf-sipcore/">https://datatracker.ietf.org/doc/charter-ietf-sipcore/</a></p>
    <p>Thanks!<br>
    </p>
    <p>/a<br>
    </p>
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From: Dan York <york@isoc.org>
To: SIPCORE <sipcore@ietf.org>
Thread-Topic: [dispatch] SIPCORE Rechartering
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Subject: Re: [sipcore] [dispatch] SIPCORE Rechartering
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Adam,

On Nov 4, 2016, at 5:28 PM, Adam Roach <adam@nostrum.com<mailto:adam@nostru=
m.com>> wrote:
<snip>

We, the SIPCORE chairs and area director, seek to fix that. We would like t=
o amend the SIPCORE charter to allow it to take on small, self-contained pr=
otocol extensions in addition to changes to the core SIP protocol. This is =
a relatively minor update to the existing charter, which expands the scope =
as described above, and adds back in two of the architectural principles fr=
om the SIP charter which are applicable only to protocol extensions.

Please provide feedback on the proposed charter by sending email to the SIP=
CORE mailing list. We will also discuss this briefly during the DISPATCH se=
ssion in Seoul.

This makes sense to me. The proposed text looked fine to me.

Dan



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Adam,
<div class=3D""><br class=3D"">
<div>
<blockquote type=3D"cite" class=3D"">
<div class=3D"">On Nov 4, 2016, at 5:28 PM, Adam Roach &lt;<a href=3D"mailt=
o:adam@nostrum.com" class=3D"">adam@nostrum.com</a>&gt; wrote:</div>
<div class=3D"">
<div bgcolor=3D"#FFFFFF" text=3D"#000000" class=3D""></div>
</div>
</blockquote>
&lt;snip&gt;<br class=3D"">
<blockquote type=3D"cite" class=3D"">
<div class=3D"">
<div bgcolor=3D"#FFFFFF" text=3D"#000000" class=3D"">
<p class=3D"">We, the SIPCORE chairs and area director, seek to fix that. W=
e would like to amend the SIPCORE charter to allow it to take on small, sel=
f-contained protocol extensions in addition to changes to the core SIP prot=
ocol. This is a relatively minor update
 to the existing charter, which expands the scope as described above, and a=
dds back in two of the architectural principles from the SIP charter which =
are applicable only to protocol extensions.</p>
<p class=3D"">Please provide feedback on the proposed charter by sending em=
ail to the SIPCORE mailing list. We will also discuss this briefly during t=
he DISPATCH session in Seoul.</p>
</div>
</div>
</blockquote>
<br class=3D"">
</div>
<div>This makes sense to me. The proposed text looked fine to me.</div>
<div><br class=3D"">
</div>
<div>Dan</div>
<div><br class=3D"">
</div>
<br class=3D"">
</div>
</body>
</html>

--_000_BCC4B4E50CB4425DA6A0DA5CB8A8D8F2isocorg_--


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Will existing SIP-related drafts in DISPATCH be moved to SIPCORE?

On Fri, Nov 4, 2016 at 8:23 PM, Dan York <york@isoc.org> wrote:

> Adam,
>
> On Nov 4, 2016, at 5:28 PM, Adam Roach <adam@nostrum.com> wrote:
>
> <snip>
>
> We, the SIPCORE chairs and area director, seek to fix that. We would like
> to amend the SIPCORE charter to allow it to take on small, self-contained
> protocol extensions in addition to changes to the core SIP protocol. This
> is a relatively minor update to the existing charter, which expands the
> scope as described above, and adds back in two of the architectural
> principles from the SIP charter which are applicable only to protocol
> extensions.
>
> Please provide feedback on the proposed charter by sending email to the
> SIPCORE mailing list. We will also discuss this briefly during the DISPATCH
> session in Seoul.
>
>
> This makes sense to me. The proposed text looked fine to me.
>
> Dan
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>

--94eb2c11f30e060f91054083901f
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">Will existing SIP-related drafts in DISPATCH be moved to S=
IPCORE?</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On F=
ri, Nov 4, 2016 at 8:23 PM, Dan York <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:york@isoc.org" target=3D"_blank">york@isoc.org</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">



<div style=3D"word-wrap:break-word">
Adam,
<div><br>
<div>
<blockquote type=3D"cite">
<div>On Nov 4, 2016, at 5:28 PM, Adam Roach &lt;<a href=3D"mailto:adam@nost=
rum.com" target=3D"_blank">adam@nostrum.com</a>&gt; wrote:</div>
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000"></div>
</div>
</blockquote>
&lt;snip&gt;<br>
<blockquote type=3D"cite">
<div>
<div bgcolor=3D"#FFFFFF" text=3D"#000000">
<p>We, the SIPCORE chairs and area director, seek to fix that. We would lik=
e to amend the SIPCORE charter to allow it to take on small, self-contained=
 protocol extensions in addition to changes to the core SIP protocol. This =
is a relatively minor update
 to the existing charter, which expands the scope as described above, and a=
dds back in two of the architectural principles from the SIP charter which =
are applicable only to protocol extensions.</p>
<p>Please provide feedback on the proposed charter by sending email to the =
SIPCORE mailing list. We will also discuss this briefly during the DISPATCH=
 session in Seoul.</p>
</div>
</div>
</blockquote>
<br>
</div>
<div>This makes sense to me. The proposed text looked fine to me.</div><spa=
n class=3D"HOEnZb"><font color=3D"#888888">
<div><br>
</div>
<div>Dan</div>
<div><br>
</div>
<br>
</font></span></div>
</div>

<br>______________________________<wbr>_________________<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" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/sipcore</a><=
br>
<br></blockquote></div><br></div>

--94eb2c11f30e060f91054083901f--


From nobody Sun Nov  6 01:18:34 2016
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From: "Olle E. Johansson" <oej@edvina.net>
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Subject: [sipcore] Source address selection in SIP
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Hi!

Paul and I had a conversation about source address selection in SIP on =
the SIP Forum IPv6 list that I wanted to move here.

An IPv6 interface by default has multiple addresses. Today=E2=80=99s =
computers also quite frequently have multiple interfaces.
A modern SIP stack has many IP addresses to select from.

In many cases applications let the operating system handle address =
election and set the IP packets sender IP address to the
one that matches the interface and the selected destination address (by =
using DNS as an example).

In SIP we have multiple addresses in the SIP message. Much can be =
simplified by using fqdn in the messages
and a GRUU with a SIP domain in the contact. In fact most of the =
examples in RFC 3261 use FQDNs.

But it=E2=80=99s not how SIP is implemented out there in the wild=E2=80=A6=


I tried to find any words in RFC 3261 or RFC 3263 about selection of =
interfaces and addresses but found nothing.
Having said that, it=E2=80=99s not easy so if anyone on sipcore can =
point me to the right section I will be grateful.

If not, I think we may need a clarification on Transaction layer =
handling of IP address selection in
the SIP protocol. Given a specific target, how do we handle multiple =
interfaces and multiple addresses
per interface - which address do we put in the SIP messages - like in =
via=E2=80=99s, record-route, SDPs and contacts.

We have been testing at SIPit with multiple ULAs attached to a network =
and a proxy on one of the ULAs.
No one could connect to that proxy properly (sadly for me not even =
Kamailio itself, which brags about having
IPv6 support from day one).

IPv6 source address selection is covered here:
https://tools.ietf.org/html/rfc6724

Quoting RFC 6724:

"The IPv6 addressing architecture [RFC4291] allows multiple unicast
   addresses to be assigned to interfaces.  These addresses might have
   different reachability scopes (link-local, site-local, or global).
   These addresses might also be "preferred" or "deprecated" [RFC4862].
   Privacy considerations have introduced the concepts of "public
   addresses" and "temporary addresses" [RFC4941].  The mobility
   architecture introduces "home addresses" and "care-of addresses"
   [RFC6275].  In addition, multi-homing situations will result in more
   addresses per node.  For example, a node might have multiple
   interfaces, some of them tunnels or virtual interfaces, or a site
   might have multiple ISP attachments with a global prefix per ISP.

   The end result is that IPv6 implementations will very often be faced
   with multiple possible source and destination addresses when
   initiating communication.  It is desirable to have default
   algorithms, common across all implementations, for selecting source
   and destination addresses so that developers and administrators can
   reason about and predict the behavior of their systems.=E2=80=9D

Executive question summary: Does the current set of SIP standards cover =
IP address
selection for the *sender* when we have multiple IP interfaces (IPv4, =
IPv6)
or just one interface with multiple addresses?

If not, should we do something about it?

/O


From nobody Sun Nov  6 04:34:25 2016
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From: Jiri Kuthan <jiri@iptel.org>
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Subject: Re: [sipcore] Source address selection in SIP
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On 06/11/16 10:18, Olle E. Johansson wrote:
> Hi!
> 
> Paul and I had a conversation about source address selection in SIP on the SIP Forum IPv6 list that I wanted to move here.
> 
> An IPv6 interface by default has multiple addresses. Today’s computers also quite frequently have multiple interfaces.
> A modern SIP stack has many IP addresses to select from.
> 
> In many cases applications let the operating system handle address election and set the IP packets sender IP address to the
> one that matches the interface and the selected destination address (by using DNS as an example).
> 
> In SIP we have multiple addresses in the SIP message. Much can be simplified by using fqdn in the messages
> and a GRUU with a SIP domain in the contact. In fact most of the examples in RFC 3261 use FQDNs.
> 
> But it’s not how SIP is implemented out there in the wild…
> 
> I tried to find any words in RFC 3261 or RFC 3263 about selection of interfaces and addresses but found nothing.
> Having said that, it’s not easy so if anyone on sipcore can point me to the right section I will be grateful.
> 
> If not, I think we may need a clarification on Transaction layer handling of IP address selection in
> the SIP protocol. Given a specific target, how do we handle multiple interfaces and multiple addresses
> per interface - which address do we put in the SIP messages - like in via’s, record-route, SDPs and contacts.
> 
> We have been testing at SIPit with multiple ULAs attached to a network and a proxy on one of the ULAs.
> No one could connect to that proxy properly (sadly for me not even Kamailio itself, which brags about having
> IPv6 support from day one).
> 
> IPv6 source address selection is covered here:
> https://tools.ietf.org/html/rfc6724
> 
> Quoting RFC 6724:
> 
> "The IPv6 addressing architecture [RFC4291] allows multiple unicast
>    addresses to be assigned to interfaces.  These addresses might have
>    different reachability scopes (link-local, site-local, or global).
>    These addresses might also be "preferred" or "deprecated" [RFC4862].
>    Privacy considerations have introduced the concepts of "public
>    addresses" and "temporary addresses" [RFC4941].  The mobility
>    architecture introduces "home addresses" and "care-of addresses"
>    [RFC6275].  In addition, multi-homing situations will result in more
>    addresses per node.  For example, a node might have multiple
>    interfaces, some of them tunnels or virtual interfaces, or a site
>    might have multiple ISP attachments with a global prefix per ISP.
> 
>    The end result is that IPv6 implementations will very often be faced
>    with multiple possible source and destination addresses when
>    initiating communication.  It is desirable to have default
>    algorithms, common across all implementations, for selecting source
>    and destination addresses so that developers and administrators can
>    reason about and predict the behavior of their systems.”
> 
> Executive question summary: Does the current set of SIP standards cover IP address
> selection for the *sender* when we have multiple IP interfaces (IPv4, IPv6)
> or just one interface with multiple addresses?

IMO, the OS makes the choice and the app just finds out what it is and print
it last-minute in the message being sent out. This seems more an implementor's
choice rather than protocol standardization, doesn't it.

-jiri

p.s. https://github.com/kamailio/kamailio/blob/master/forward.c
get_out_socket() does that as an example.


From nobody Sun Nov  6 05:47:52 2016
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "'SIPCORE'" <sipcore@ietf.org>
Thread-Topic: [dispatch] SIPCORE Rechartering
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Date: Sun, 6 Nov 2016 13:47:42 +0000
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Subject: Re: [sipcore] [dispatch] SIPCORE Rechartering
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "'SIPCORE'" <sipcore@ietf.org>, "dispatch@ietf.org" <dispatch@ietf.org>
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Subject: Re: [sipcore] [dispatch] SIPCORE Rechartering
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From nobody Mon Nov  7 00:59:52 2016
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From: "Olle E. Johansson" <oej@edvina.net>
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To: Jiri Kuthan <jiri@iptel.org>
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Cc: SIPCORE <sipcore@ietf.org>, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] Source address selection in SIP
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> On 06 Nov 2016, at 13:34, Jiri Kuthan <jiri@iptel.org> wrote:
>=20
> On 06/11/16 10:18, Olle E. Johansson wrote:
>> Hi!
>>=20
>> Paul and I had a conversation about source address selection in SIP =
on the SIP Forum IPv6 list that I wanted to move here.
>>=20
>> An IPv6 interface by default has multiple addresses. Today=E2=80=99s =
computers also quite frequently have multiple interfaces.
>> A modern SIP stack has many IP addresses to select from.
>>=20
>> In many cases applications let the operating system handle address =
election and set the IP packets sender IP address to the
>> one that matches the interface and the selected destination address =
(by using DNS as an example).
>>=20
>> In SIP we have multiple addresses in the SIP message. Much can be =
simplified by using fqdn in the messages
>> and a GRUU with a SIP domain in the contact. In fact most of the =
examples in RFC 3261 use FQDNs.
>>=20
>> But it=E2=80=99s not how SIP is implemented out there in the wild=E2=80=
=A6
>>=20
>> I tried to find any words in RFC 3261 or RFC 3263 about selection of =
interfaces and addresses but found nothing.
>> Having said that, it=E2=80=99s not easy so if anyone on sipcore can =
point me to the right section I will be grateful.
>>=20
>> If not, I think we may need a clarification on Transaction layer =
handling of IP address selection in
>> the SIP protocol. Given a specific target, how do we handle multiple =
interfaces and multiple addresses
>> per interface - which address do we put in the SIP messages - like in =
via=E2=80=99s, record-route, SDPs and contacts.
>>=20
>> We have been testing at SIPit with multiple ULAs attached to a =
network and a proxy on one of the ULAs.
>> No one could connect to that proxy properly (sadly for me not even =
Kamailio itself, which brags about having
>> IPv6 support from day one).
>>=20
>> IPv6 source address selection is covered here:
>> https://tools.ietf.org/html/rfc6724
>>=20
>> Quoting RFC 6724:
>>=20
>> "The IPv6 addressing architecture [RFC4291] allows multiple unicast
>>   addresses to be assigned to interfaces.  These addresses might have
>>   different reachability scopes (link-local, site-local, or global).
>>   These addresses might also be "preferred" or "deprecated" =
[RFC4862].
>>   Privacy considerations have introduced the concepts of "public
>>   addresses" and "temporary addresses" [RFC4941].  The mobility
>>   architecture introduces "home addresses" and "care-of addresses"
>>   [RFC6275].  In addition, multi-homing situations will result in =
more
>>   addresses per node.  For example, a node might have multiple
>>   interfaces, some of them tunnels or virtual interfaces, or a site
>>   might have multiple ISP attachments with a global prefix per ISP.
>>=20
>>   The end result is that IPv6 implementations will very often be =
faced
>>   with multiple possible source and destination addresses when
>>   initiating communication.  It is desirable to have default
>>   algorithms, common across all implementations, for selecting source
>>   and destination addresses so that developers and administrators can
>>   reason about and predict the behavior of their systems.=E2=80=9D
>>=20
>> Executive question summary: Does the current set of SIP standards =
cover IP address
>> selection for the *sender* when we have multiple IP interfaces (IPv4, =
IPv6)
>> or just one interface with multiple addresses?
>=20
> IMO, the OS makes the choice and the app just finds out what it is and =
print
> it last-minute in the message being sent out. This seems more an =
implementor's
> choice rather than protocol standardization, doesn't it.
Maybe. Like with other stuff, we may need a note to implementors so they
get it right. May note be an IETF document though.
>=20
> -jiri
>=20
> p.s. https://github.com/kamailio/kamailio/blob/master/forward.c
> get_out_socket() does that as an example.
Which is code that doesn=E2=80=99t do IPv6 source address selection =
right when we
tested ;-) Thanks for pointing me to the proper file!

/O


From nobody Mon Nov  7 01:49:56 2016
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From: Adnan Aziz <chaudhryadnan@gmail.com>
Date: Mon, 7 Nov 2016 10:49:47 +0100
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Subject: [sipcore] SIP-URI Without Userinfo
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Hello Everyone,

According to the rfc3261, the userinfo in the sip-uri is optional.
If there is no userinfo part then there should not be any "@" sign as well
in the sip-uri.


I have couple of questions

1) If I am receiving a sip request without userinfo part in the sip-usri,
starting with "@" then is it a malformed packet?

2) What should be the standard behavior of SIP server while responding to
such packets?

Regards,
Adnan Aziz
-- 
*Adnan Aziz*

--001a11c149a2c256c20540b2f182
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div><div><div>Hello Everyone,<br><br></div>According to t=
he rfc3261, the userinfo in the sip-uri is optional.<br></div>If there is n=
o userinfo part then there should not be any &quot;@&quot; sign as well in =
the sip-uri.<br><br><br></div><div>I have couple of questions<br><br></div>=
<div>1) If I am receiving a sip request without userinfo part in the sip-us=
ri, starting with &quot;@&quot; then is it a malformed packet?<br><br></div=
><div>2) What should be the standard behavior of SIP server while respondin=
g to such packets?<br><br></div><div>Regards,<br></div><div><div><div><div>=
Adnan Aziz<br></div><div>-- <br><div class=3D"gmail_signature" data-smartma=
il=3D"gmail_signature"><div dir=3D"ltr"><div><div dir=3D"ltr"><div><div dir=
=3D"ltr"><b>Adnan Aziz</b><br><br></div></div></div></div></div></div>
</div></div></div></div></div>

--001a11c149a2c256c20540b2f182--


From nobody Mon Nov  7 02:46:49 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adnan Aziz <chaudhryadnan@gmail.com>, "sip-implementors@lists.cs.columbia.edu" <sip-implementors@lists.cs.columbia.edu>
Thread-Topic: [sipcore] SIP-URI Without Userinfo
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Subject: Re: [sipcore] SIP-URI Without Userinfo
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--_000_D4461DE012ACEchristerholmbergericssoncom_
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Hi,

>According to the rfc3261, the userinfo in the sip-uri is optional.
>If there is no userinfo part then there should not be any "@" sign as well=
 in the sip-uri.

Correct.

>I have couple of questions
>
>1) If I am receiving a sip request without userinfo part in the sip-usri, =
starting with "@" then is it a malformed packet?

It=92s a malformed SIP URI.

>2) What should be the standard behavior of SIP server while responding to =
such packets?

Send a response with an error response code.

Regards,

Christer



--_000_D4461DE012ACEchristerholmbergericssoncom_
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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Hi,</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div>
<div>&gt;According to the rfc3261, the userinfo in the sip-uri is optional.=
<br>
</div>
&gt;If there is no userinfo part then there should not be any &quot;@&quot;=
 sign as well in the sip-uri.<br>
</div>
</div>
</div>
</div>
</span>
<div><br>
</div>
<div>Correct.</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div><br>
</div>
<div>&gt;I have couple of questions<br>
&gt;</div>
<div>&gt;1) If I am receiving a sip request without userinfo part in the si=
p-usri, starting with &quot;@&quot; then is it a malformed packet?<br>
</div>
</div>
</div>
</div>
</span>
<div>&nbsp;</div>
<div>It=92s a malformed SIP URI.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div>&gt;2) What should be the standard behavior of SIP server while respon=
ding to such packets?<br>
</div>
</div>
</div>
</div>
</span>
<div><br>
</div>
<div>Send a response with an error response code.</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Christer</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div><br>
</div>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_D4461DE012ACEchristerholmbergericssoncom_--


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To: "Olle E. Johansson" <oej@edvina.net>, SIPCORE <sipcore@ietf.org>
References: <165501DC-37DB-4EFD-B596-51CCF8FB6902@edvina.net>
From: Simon Perreault <sperreault@jive.com>
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Subject: Re: [sipcore] Source address selection in SIP
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Le 2016-11-06 à 04:18, Olle E. Johansson a écrit :
> Executive question summary: Does the current set of SIP standards cover IP address
> selection for the *sender* when we have multiple IP interfaces (IPv4, IPv6)
> or just one interface with multiple addresses?
> 
> If not, should we do something about it?

Even ignoring all implementation and real-world constraints, what do you
think would be the theoretically perfect UA behaviour? Let's start with
measuring how big is the gap between theory and practice?

-- 
Simon Perreault
Director of Engineering, Platform | Jive Communications, Inc.
https://jive.com | +1 418 478 0989 ext. 1241 | sperreault@jive.com


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Subject: Re: [sipcore] SIP-URI Without Userinfo
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To amplify Christer's replyP:

Christer Holmberg <christer.holmberg@ericsson.com> writes:
>>I have couple of questions
>>
>>1) If I am receiving a sip request without userinfo part in the
>>sip-usri, starting with "@" then is it a malformed packet?
>
> It's a malformed SIP URI.

In particular, it is syntactically incorrect, which you can see by
comparison with the ABNF in RFC 3261 section 25.1 production "SIP-URI".

>
>>2) What should be the standard behavior of SIP server while responding
>>to such packets?
>
> Send a response with an error response code.

Specifically, a 400 response:

   21.4.1 400 Bad Request

   The request could not be understood due to malformed syntax.  The
   Reason-Phrase SHOULD identify the syntax problem in more detail, for
   example, "Missing Call-ID header field".

However, there is a certain amount of freedom that SIP servers have
regarding precisely what 4xx response codes they give for particular
requests, because the errors in SIP requests cannot be precisely
categorized.

Dale


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 7 Nov 2016 09:50:55 -0500
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Subject: Re: [sipcore] Source address selection in SIP
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This is the same problem as selection of IPv4 vs IPv6 or UDP vs TCP when
running behind firewall.

What we typically do is run a STUN server on the same protocols, IPs and
ports as our SIP proxy. Client would enumerate local interfaces and send a
STUN BIND request to DNS results of the server name resolution. Once we
find the STUN request that received the response, we would start the SIP UA
on that interface and register. The whole process is very similar to ICE
candidate nomination. If keep-alive on the connection fails due to client
migration to another network or DHCP lease expiration, we would restart the
whole process with STUN BIND requests and re-registration.

Another alternative is to use SIP OPTIONS request for the same purpose as
STUN BIND. This uses more resources, but does not require STUN.

Final option is simply do the UDP connect from the unbound socket to the
server IP and then get the socket address. This would give you the local
address which has the route to the server. This does not guarantee that SIP
request will actually be delivered over that route, especially if client on
some kind of VPN, but it does give you the most likely local interface.
Even in the previous two cases we normally start with this interface when
sending connection verification requests.

Given that STUN BIND support is already required for UDP keep-alive
messages by sip-outbound, and that it is typically already supported by SIP
Proxies, we have preferred the STUN based implementation.

Bottom line is there are independent SIP UAs for each of the local
interfaces. Correct SIP UA to be used for connection to a specific server
is determined by sending a test STUN or SIP request.

Regards,

_____________
Roman Shpount

On Mon, Nov 7, 2016 at 9:21 AM, Simon Perreault <sperreault@jive.com> wrote=
:

> Le 2016-11-06 =C3=A0 04:18, Olle E. Johansson a =C3=A9crit :
> > Executive question summary: Does the current set of SIP standards cover
> IP address
> > selection for the *sender* when we have multiple IP interfaces (IPv4,
> IPv6)
> > or just one interface with multiple addresses?
> >
> > If not, should we do something about it?
>
> Even ignoring all implementation and real-world constraints, what do you
> think would be the theoretically perfect UA behaviour? Let's start with
> measuring how big is the gap between theory and practice?
>
> --
> Simon Perreault
> Director of Engineering, Platform | Jive Communications, Inc.
> https://jive.com | +1 418 478 0989 ext. 1241 | sperreault@jive.com
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr">This is the same problem as selection of IPv4 vs IPv6 or U=
DP vs TCP when running behind firewall.<div><br></div><div>What we typicall=
y do is run a STUN server on the same protocols, IPs and ports as our SIP p=
roxy. Client would enumerate local interfaces and send a STUN BIND request =
to DNS results of the server name resolution. Once we find the STUN request=
 that received the response, we would start the SIP UA on that interface an=
d register. The whole process is very similar to ICE candidate nomination. =
If keep-alive on the connection fails due to client migration to another ne=
twork or DHCP lease expiration, we would restart the whole process with STU=
N BIND requests and re-registration.</div><div><br></div><div>Another alter=
native is to use SIP OPTIONS request for the same purpose as STUN BIND. Thi=
s uses more resources, but does not require STUN.</div><div><br></div><div>=
Final option is simply do the UDP connect from the unbound socket to the se=
rver IP and then get the socket address. This would give you the local addr=
ess which has the route to the server. This does not guarantee that SIP req=
uest will actually be delivered over that route, especially if client on so=
me kind of VPN, but it does give you the most likely local interface. Even =
in the previous two cases we normally start with this interface when sendin=
g connection verification requests.</div><div><br></div><div>Given that STU=
N BIND support is already required for UDP keep-alive messages by sip-outbo=
und, and that it is typically already supported by SIP Proxies, we have pre=
ferred the STUN based implementation.</div><div><br></div><div>Bottom line =
is there are independent SIP UAs for each of the local interfaces. Correct =
SIP UA to be used for connection to a specific server is determined by send=
ing a test STUN or SIP request.</div><div>=C2=A0</div><div>Regards,</div></=
div><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_s=
ignature" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount=
</div></div>
<br><div class=3D"gmail_quote">On Mon, Nov 7, 2016 at 9:21 AM, Simon Perrea=
ult <span dir=3D"ltr">&lt;<a href=3D"mailto:sperreault@jive.com" target=3D"=
_blank">sperreault@jive.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex"><span class=3D"">Le 2016-11-06 =C3=A0 04:18, Olle E. Johansson a =
=C3=A9crit :<br>
&gt; Executive question summary: Does the current set of SIP standards cove=
r IP address<br>
&gt; selection for the *sender* when we have multiple IP interfaces (IPv4, =
IPv6)<br>
&gt; or just one interface with multiple addresses?<br>
&gt;<br>
&gt; If not, should we do something about it?<br>
<br>
</span>Even ignoring all implementation and real-world constraints, what do=
 you<br>
think would be the theoretically perfect UA behaviour? Let&#39;s start with=
<br>
measuring how big is the gap between theory and practice?<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
--<br>
Simon Perreault<br>
Director of Engineering, Platform | Jive Communications, Inc.<br>
<a href=3D"https://jive.com" rel=3D"noreferrer" target=3D"_blank">https://j=
ive.com</a> | <a href=3D"tel:%2B1%20418%20478%200989%20ext.%201241" value=
=3D"+14184780989">+1 418 478 0989 ext. 1241</a> | <a href=3D"mailto:sperrea=
ult@jive.com">sperreault@jive.com</a><br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
______________________________<wbr>_________________<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" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/sipcore</a><=
br>
</div></div></blockquote></div><br></div>

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Subject: Re: [sipcore] Source address selection in SIP
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> On 07 Nov 2016, at 15:50, Roman Shpount <roman@telurix.com> wrote:
>=20
> This is the same problem as selection of IPv4 vs IPv6 or UDP vs TCP =
when running behind firewall.
Yes and no. When you want to communicate with something in the same =
network you may have
multiple addresses available to use
IPv4
IPV6 global
IPv6 global temporary
IPv6 ULA

Is it crystal clear for a SIP developer which one to use?=20
I think this may be a generic issue for all IP applications, but wanted =
to open the discussion to
see if I missed something.

We have seen problems with this at SIPit so this is something a lot of =
developers have not=20
implemented correctly in their SIP implementations.

>=20
> What we typically do is run a STUN server on the same protocols, IPs =
and ports as our SIP proxy. Client would enumerate local interfaces and =
send a STUN BIND request to DNS results of the server name resolution. =
Once we find the STUN request that received the response, we would start =
the SIP UA on that interface and register. The whole process is very =
similar to ICE candidate nomination. If keep-alive on the connection =
fails due to client migration to another network or DHCP lease =
expiration, we would restart the whole process with STUN BIND requests =
and re-registration.
STUN would not work in many of these cases.
>=20
> Another alternative is to use SIP OPTIONS request for the same purpose =
as STUN BIND. This uses more resources, but does not require STUN.
So who decides what to put in the VIA of the SIP Options??
>=20
> Final option is simply do the UDP connect from the unbound socket to =
the server IP and then get the socket address. This would give you the =
local address which has the route to the server. This does not guarantee =
that SIP request will actually be delivered over that route, especially =
if client on some kind of VPN, but it does give you the most likely =
local interface. Even in the previous two cases we normally start with =
this interface when sending connection verification requests.
I think that=E2=80=99s what the code that Jiri pointed to does. But we =
still have the issue with multiple ULAs. Maybe
the O/S have fixed that now.

>=20
> Given that STUN BIND support is already required for UDP keep-alive =
messages by sip-outbound, and that it is typically already supported by =
SIP Proxies, we have preferred the STUN based implementation.
You assume we have an inside and an outside, based on the IPv4 NAT =
model. Think IPv6 for a while :-)
>=20
> Bottom line is there are independent SIP UAs for each of the local =
interfaces. Correct SIP UA to be used for connection to a specific =
server is determined by sending a test STUN or SIP request.
Food for thought.

/O
> =20
> Regards,
>=20
> _____________
> Roman Shpount
>=20
> On Mon, Nov 7, 2016 at 9:21 AM, Simon Perreault <sperreault@jive.com =
<mailto:sperreault@jive.com>> wrote:
> Le 2016-11-06 =C3=A0 04:18, Olle E. Johansson a =C3=A9crit :
> > Executive question summary: Does the current set of SIP standards =
cover IP address
> > selection for the *sender* when we have multiple IP interfaces =
(IPv4, IPv6)
> > or just one interface with multiple addresses?
> >
> > If not, should we do something about it?
>=20
> Even ignoring all implementation and real-world constraints, what do =
you
> think would be the theoretically perfect UA behaviour? Let's start =
with
> measuring how big is the gap between theory and practice?
>=20
> --
> Simon Perreault
> Director of Engineering, Platform | Jive Communications, Inc.
> https://jive.com <https://jive.com/> | +1 418 478 0989 ext. 1241 =
<tel:%2B1%20418%20478%200989%20ext.%201241> | sperreault@jive.com =
<mailto:sperreault@jive.com>
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org <mailto:sipcore@ietf.org>
> https://www.ietf.org/mailman/listinfo/sipcore =
<https://www.ietf.org/mailman/listinfo/sipcore>
>=20


--Apple-Mail=_4150EF5C-D98C-4D51-A71A-9E45A5293DB3
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 07 Nov 2016, at 15:50, Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com" class=3D"">roman@telurix.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
dir=3D"ltr" class=3D"">This is the same problem as selection of IPv4 vs =
IPv6 or UDP vs TCP when running behind =
firewall.</div></div></blockquote>Yes and no. When you want to =
communicate with something in the same network you may =
have</div><div>multiple addresses available to =
use</div><div>IPv4</div><div>IPV6 global</div><div>IPv6 global =
temporary</div><div>IPv6 ULA</div><div><br class=3D""></div><div>Is it =
crystal clear for a SIP developer which one to use?&nbsp;</div><div>I =
think this may be a generic issue for all IP applications, but wanted to =
open the discussion to</div><div>see if I missed =
something.</div><div><br class=3D""></div><div>We have seen problems =
with this at SIPit so this is something a lot of developers have =
not&nbsp;</div><div>implemented correctly in their SIP =
implementations.</div><div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div class=3D""><br=
 class=3D""></div><div class=3D"">What we typically do is run a STUN =
server on the same protocols, IPs and ports as our SIP proxy. Client =
would enumerate local interfaces and send a STUN BIND request to DNS =
results of the server name resolution. Once we find the STUN request =
that received the response, we would start the SIP UA on that interface =
and register. The whole process is very similar to ICE candidate =
nomination. If keep-alive on the connection fails due to client =
migration to another network or DHCP lease expiration, we would restart =
the whole process with STUN BIND requests and =
re-registration.</div></div></div></blockquote>STUN would not work in =
many of these cases.<br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div class=3D""><br=
 class=3D""></div><div class=3D"">Another alternative is to use SIP =
OPTIONS request for the same purpose as STUN BIND. This uses more =
resources, but does not require STUN.</div></div></div></blockquote>So =
who decides what to put in the VIA of the SIP Options??<br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
dir=3D"ltr" class=3D""><div class=3D""><br class=3D""></div><div =
class=3D"">Final option is simply do the UDP connect from the unbound =
socket to the server IP and then get the socket address. This would give =
you the local address which has the route to the server. This does not =
guarantee that SIP request will actually be delivered over that route, =
especially if client on some kind of VPN, but it does give you the most =
likely local interface. Even in the previous two cases we normally start =
with this interface when sending connection verification =
requests.</div></div></div></blockquote>I think that=E2=80=99s what the =
code that Jiri pointed to does. But we still have the issue with =
multiple ULAs. Maybe</div><div>the O/S have fixed that =
now.</div><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><div dir=3D"ltr" class=3D""><div class=3D""><br =
class=3D""></div><div class=3D"">Given that STUN BIND support is already =
required for UDP keep-alive messages by sip-outbound, and that it is =
typically already supported by SIP Proxies, we have preferred the STUN =
based implementation.</div></div></div></blockquote>You assume we have =
an inside and an outside, based on the IPv4 NAT model. Think IPv6 for a =
while :-)<br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><div dir=3D"ltr" class=3D""><div class=3D""><br =
class=3D""></div><div class=3D"">Bottom line is there are independent =
SIP UAs for each of the local interfaces. Correct SIP UA to be used for =
connection to a specific server is determined by sending a test STUN or =
SIP request.</div></div></div></blockquote>Food for =
thought.</div><div><br class=3D""></div><div>/O<br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div =
class=3D"">&nbsp;</div><div class=3D"">Regards,</div></div><div =
class=3D"gmail_extra"><br clear=3D"all" class=3D""><div class=3D""><div =
class=3D"gmail_signature" =
data-smartmail=3D"gmail_signature">_____________<br class=3D"">Roman =
Shpount</div></div>
<br class=3D""><div class=3D"gmail_quote">On Mon, Nov 7, 2016 at 9:21 =
AM, Simon Perreault <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:sperreault@jive.com" target=3D"_blank" =
class=3D"">sperreault@jive.com</a>&gt;</span> wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=3D"">Le =
2016-11-06 =C3=A0 04:18, Olle E. Johansson a =C3=A9crit :<br class=3D"">
&gt; Executive question summary: Does the current set of SIP standards =
cover IP address<br class=3D"">
&gt; selection for the *sender* when we have multiple IP interfaces =
(IPv4, IPv6)<br class=3D"">
&gt; or just one interface with multiple addresses?<br class=3D"">
&gt;<br class=3D"">
&gt; If not, should we do something about it?<br class=3D"">
<br class=3D"">
</span>Even ignoring all implementation and real-world constraints, what =
do you<br class=3D"">
think would be the theoretically perfect UA behaviour? Let's start =
with<br class=3D"">
measuring how big is the gap between theory and practice?<br class=3D"">
<span class=3D"HOEnZb"><font color=3D"#888888" class=3D""><br class=3D"">
--<br class=3D"">
Simon Perreault<br class=3D"">
Director of Engineering, Platform | Jive Communications, Inc.<br =
class=3D"">
<a href=3D"https://jive.com/" rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://jive.com</a> | <a =
href=3D"tel:%2B1%20418%20478%200989%20ext.%201241" value=3D"+14184780989" =
class=3D"">+1 418 478 0989 ext. 1241</a> | <a =
href=3D"mailto:sperreault@jive.com" class=3D"">sperreault@jive.com</a><br =
class=3D"">
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br class=3D"">
______________________________<wbr class=3D"">_________________<br =
class=3D"">
sipcore mailing list<br class=3D"">
<a href=3D"mailto:sipcore@ietf.org" class=3D"">sipcore@ietf.org</a><br =
class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/<wbr =
class=3D"">listinfo/sipcore</a><br class=3D"">
</div></div></blockquote></div><br class=3D""></div>
</div></blockquote></div><br class=3D""></body></html>=

--Apple-Mail=_4150EF5C-D98C-4D51-A71A-9E45A5293DB3--


From nobody Mon Nov  7 09:03:09 2016
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
To: sipcore@ietf.org
References: <165501DC-37DB-4EFD-B596-51CCF8FB6902@edvina.net> <80999829-c6de-0cd0-589d-0dfb0bf5cd8a@jive.com>
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Subject: Re: [sipcore] Source address selection in SIP
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On 11/7/16 9:21 AM, Simon Perreault wrote:
> Le 2016-11-06 à 04:18, Olle E. Johansson a écrit :
>> Executive question summary: Does the current set of SIP standards cover IP address
>> selection for the *sender* when we have multiple IP interfaces (IPv4, IPv6)
>> or just one interface with multiple addresses?
>>
>> If not, should we do something about it?
>
> Even ignoring all implementation and real-world constraints, what do you
> think would be the theoretically perfect UA behaviour? Let's start with
> measuring how big is the gap between theory and practice?

I think we need to first clarify what we mean here.

Does this choice necessarily impact *anything* in the outgoing sip 
message? If so, what?

The most likely thing that it would affect is Via. It is reasonable to 
assume that the Via header has the same address that goes into the 
sender field of the IP packet. 3261 doesn't exactly say that, but 
section 8.1.1.7 does say:

    The Via header field indicates the transport used for the transaction
    and identifies the location where the response is to be sent.  A Via
    header field value is added only after the transport that will be
    used to reach the next hop has been selected (which may involve the
    usage of the procedures in [4]).

which is pretty close.

But that only needs to work for routing of the transaction response. So 
as long as the next hop can get back this way all should be good. It 
isn't necessary for the UAS to be able to route to this address.

Is there some thought that this choice affects anything else in the 
outgoing message? (What to put in the Contact header is a different 
question. I trust that isn't part of the current question.)

	Thanks,
	Paul


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 7 Nov 2016 12:31:59 -0500
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Subject: Re: [sipcore] Source address selection in SIP
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On Mon, Nov 7, 2016 at 10:17 AM, Olle E. Johansson <oej@edvina.net> wrote:

>
> On 07 Nov 2016, at 15:50, Roman Shpount <roman@telurix.com> wrote:
>
> This is the same problem as selection of IPv4 vs IPv6 or UDP vs TCP when
> running behind firewall.
>
> Yes and no. When you want to communicate with something in the same
> network you may have
> multiple addresses available to use
> IPv4
> IPV6 global
> IPv6 global temporary
> IPv6 ULA
>

My suggestion was to use the same procedures as the ones defined when
collecting base candidates for ICE. These should be fairly well defined and
tested with things like WebRTC.



> Is it crystal clear for a SIP developer which one to use?
> I think this may be a generic issue for all IP applications, but wanted t=
o
> open the discussion to
> see if I missed something.
>
> We have seen problems with this at SIPit so this is something a lot of
> developers have not
> implemented correctly in their SIP implementations.
>
>
> What we typically do is run a STUN server on the same protocols, IPs and
> ports as our SIP proxy. Client would enumerate local interfaces and send =
a
> STUN BIND request to DNS results of the server name resolution. Once we
> find the STUN request that received the response, we would start the SIP =
UA
> on that interface and register. The whole process is very similar to ICE
> candidate nomination. If keep-alive on the connection fails due to client
> migration to another network or DHCP lease expiration, we would restart t=
he
> whole process with STUN BIND requests and re-registration.
>
> STUN would not work in many of these cases.
>

Why would not it work? This is the same procedure as ICE -- send a STUN
bind message over UDP or TCP from each potential client address (gathered
through interface enumeration) to each potential server address (gathered
through DNS resolution). If the server is running STUN on each protocol
(including TCP and TLS), address and port it exposes for external calls,
this check will work if communications are possible.

> Another alternative is to use SIP OPTIONS request for the same purpose as
> STUN BIND. This uses more resources, but does not require STUN.
>
> So who decides what to put in the VIA of the SIP Options??
>

 In this case client starts a SIP UA on each potential local address and
sends a SIP OPTIONS request to each potential server address from each SIP
UA. Each SIP UA is bound to the specific address, so it should be clear
which address to put in the VIA.

> Final option is simply do the UDP connect from the unbound socket to the
> server IP and then get the socket address. This would give you the local
> address which has the route to the server. This does not guarantee that S=
IP
> request will actually be delivered over that route, especially if client =
on
> some kind of VPN, but it does give you the most likely local interface.
> Even in the previous two cases we normally start with this interface when
> sending connection verification requests.
>
> I think that=E2=80=99s what the code that Jiri pointed to does. But we st=
ill have
> the issue with multiple ULAs. Maybe
> the O/S have fixed that now.
>

This code works often enough. Issues that we saw with just doing this are
related to clients who are connected via a VPN to a corporate environment
that block SIP. Another issue is deciding when to use UDP vs TCP when
behind firewall. This code provides no indication that UDP is blocked.

> Given that STUN BIND support is already required for UDP keep-alive
> messages by sip-outbound, and that it is typically already supported by S=
IP
> Proxies, we have preferred the STUN based implementation.
>
> You assume we have an inside and an outside, based on the IPv4 NAT model.
> Think IPv6 for a while :-)
>

Once again, what I am proposing is not that different then full ICE with
ICE lite communication. This is known to work with IPv6 :-).

> Bottom line is there are independent SIP UAs for each of the local
> interfaces. Correct SIP UA to be used for connection to a specific server
> is determined by sending a test STUN or SIP request.
>
> Food for thought.
>
>
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Nov 7, 2016 at 10:17 AM, Olle E. Johansson <span dir=3D"ltr">&=
lt;<a href=3D"mailto:oej@edvina.net" target=3D"_blank">oej@edvina.net</a>&g=
t;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid r=
gb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-word"><br><=
div><span class=3D"gmail-"><blockquote type=3D"cite"><div>On 07 Nov 2016, a=
t 15:50, Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D"_=
blank">roman@telurix.com</a>&gt; wrote:</div><br class=3D"gmail-m_620976944=
0662907870Apple-interchange-newline"><div><div dir=3D"ltr">This is the same=
 problem as selection of IPv4 vs IPv6 or UDP vs TCP when running behind fir=
ewall.</div></div></blockquote></span>Yes and no. When you want to communic=
ate with something in the same network you may have</div><div>multiple addr=
esses available to use</div><div>IPv4</div><div>IPV6 global</div><div>IPv6 =
global temporary</div><div>IPv6 ULA</div></div></blockquote><div><br></div>=
<div>My suggestion was to use the same procedures as the ones defined when =
collecting base candidates for ICE. These should be fairly well defined and=
 tested with things like WebRTC.</div><div><br></div><div>=C2=A0</div><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:=
1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-=
word"><div>Is it crystal clear for a SIP developer which one to use?=C2=A0<=
br></div><div>I think this may be a generic issue for all IP applications, =
but wanted to open the discussion to</div><div>see if I missed something.</=
div><div><br></div><div>We have seen problems with this at SIPit so this is=
 something a lot of developers have not=C2=A0</div><div>implemented correct=
ly in their SIP implementations.</div><div><span class=3D"gmail-"><br><bloc=
kquote type=3D"cite"><div><div dir=3D"ltr"><div><br></div><div>What we typi=
cally do is run a STUN server on the same protocols, IPs and ports as our S=
IP proxy. Client would enumerate local interfaces and send a STUN BIND requ=
est to DNS results of the server name resolution. Once we find the STUN req=
uest that received the response, we would start the SIP UA on that interfac=
e and register. The whole process is very similar to ICE candidate nominati=
on. If keep-alive on the connection fails due to client migration to anothe=
r network or DHCP lease expiration, we would restart the whole process with=
 STUN BIND requests and re-registration.</div></div></div></blockquote></sp=
an>STUN would not work in many of these cases.</div></div></blockquote><div=
>=C2=A0</div><div>Why would not it work? This is the same procedure as ICE =
-- send a STUN bind message over UDP or TCP from each potential client addr=
ess (gathered through interface enumeration) to each potential server addre=
ss (gathered through DNS resolution). If the server is running STUN on each=
 protocol (including TCP and TLS), address and port it exposes for external=
 calls, this check will work if communications are possible.</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px =
solid rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-word=
"><div><span class=3D"gmail-"><blockquote type=3D"cite"><div dir=3D"ltr"><d=
iv>Another alternative is to use SIP OPTIONS request for the same purpose a=
s STUN BIND. This uses more resources, but does not require STUN.<br></div>=
</div></blockquote></span>So who decides what to put in the VIA of the SIP =
Options??</div></div></blockquote><div><br></div><div>=C2=A0In this case cl=
ient starts a SIP UA on each potential local address and sends a SIP OPTION=
S request to each potential server address from each SIP UA. Each SIP UA is=
 bound to the specific address, so it should be clear which address to put =
in the VIA.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0p=
x 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><d=
iv style=3D"word-wrap:break-word"><div><span class=3D"gmail-"><blockquote t=
ype=3D"cite"><div dir=3D"ltr"><div>Final option is simply do the UDP connec=
t from the unbound socket to the server IP and then get the socket address.=
 This would give you the local address which has the route to the server. T=
his does not guarantee that SIP request will actually be delivered over tha=
t route, especially if client on some kind of VPN, but it does give you the=
 most likely local interface. Even in the previous two cases we normally st=
art with this interface when sending connection verification requests.<br><=
/div></div></blockquote></span>I think that=E2=80=99s what the code that Ji=
ri pointed to does. But we still have the issue with multiple ULAs. Maybe</=
div><div>the O/S have fixed that now.</div></div></blockquote><div>=C2=A0</=
div><div>This code works often enough. Issues that we saw with just doing t=
his are related to clients who are connected via a VPN to a corporate envir=
onment that block SIP. Another issue is deciding when to use UDP vs TCP whe=
n behind firewall. This code provides no indication that UDP is blocked.=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"wo=
rd-wrap:break-word"><div><span class=3D"gmail-"><blockquote type=3D"cite"><=
div dir=3D"ltr"><div>Given that STUN BIND support is already required for U=
DP keep-alive messages by sip-outbound, and that it is typically already su=
pported by SIP Proxies, we have preferred the STUN based implementation.<br=
></div></div></blockquote></span>You assume we have an inside and an outsid=
e, based on the IPv4 NAT model. Think IPv6 for a while :-)</div></div></blo=
ckquote><div>=C2=A0</div><div>Once again, what I am proposing is not that d=
ifferent then full ICE with ICE lite communication. This is known to work w=
ith IPv6 :-).</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div s=
tyle=3D"word-wrap:break-word"><div><span class=3D"gmail-"><blockquote type=
=3D"cite"><div dir=3D"ltr"><div>Bottom line is there are independent SIP UA=
s for each of the local interfaces. Correct SIP UA to be used for connectio=
n to a specific server is determined by sending a test STUN or SIP request.=
<br></div></div></blockquote></span>Food for thought.</div><span class=3D"g=
mail-HOEnZb"><font color=3D"#888888"><div><br></div></font></span></div></b=
lockquote><div><br></div><div><div class=3D"gmail_signature">_____________<=
br>Roman Shpount</div></div><br><div>=C2=A0</div></div></div></div>

--001a114a89ccbacc050540b966b3--


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Date: Mon, 7 Nov 2016 12:35:18 -0500
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Subject: Re: [sipcore] Source address selection in SIP
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On Mon, Nov 7, 2016 at 12:03 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Is there some thought that this choice affects anything else in the
> outgoing message?


The only other issue this affects is delivering SIP response to a message
when client IP changes in the middle of the SIP transaction. You either
need to accept that the transaction can fail or come up with a mechanism to
recover a SIP transaction on the new client local IP or use something like
TURN to always tunnel SIP messages through the same IP/port in the cloud
(resource intensive).

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Nov 7, 2016 at 12:03 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.=
edu</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockq=
uote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1p=
x solid rgb(204,204,204);padding-left:1ex">Is there some thought that this =
choice affects anything else in the outgoing message?</blockquote><div><br>=
</div><div>The only other issue this affects is delivering SIP response to =
a message when client IP changes in the middle of the SIP transaction. You =
either need to accept that the transaction can fail or come up with a mecha=
nism to recover a SIP transaction on the new client local IP or use somethi=
ng like TURN to always tunnel SIP messages through the same IP/port in the =
cloud (resource intensive).</div><div><br></div><div>Regards,</div><div><di=
v class=3D"gmail_signature">_____________<br>Roman Shpount</div></div><div>=
=C2=A0</div></div></div></div>

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On 11/04/2016 04:28 PM, Adam Roach wrote:
> Please provide feedback on the proposed charter by sending email to the
> SIPCORE mailing list. We will also discuss this briefly during the
> DISPATCH session in Seoul.

The change to the charter sounds reasonable.

Having been a beneficiary of one of the first mini-WGs to be
dispatch'ed (sipclf), the model of dispatch'ing has worked.  However, in
its trajectory, as Adam states, SIP is now a more mature protocol and I
suspect the itches that need to be scratched have more to do with
operational issues with SIP than core modifications to the protocol
itself.  Seems reasonable to perform such tweaks in sipcore.

Cheers,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Nokia Networks
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
Email: vkg@bell-labs.com / vijay.gurbani@nokia-bell-labs.com
Web: http://ect.bell-labs.com/who/vkg/  | Calendar: http://goo.gl/x3Ogq


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 7 Nov 2016 15:14:34 -0500
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Source address selection in SIP
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--001a11c01b1e30dd520540bbac9c
Content-Type: text/plain; charset=UTF-8
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Just a side note: received and rport are generally useless for TCP or TLS
based transactions. Responses are supposed to be sent over the same TCP/TLS
connection.

For UDP, when sending a message you can safely put 127.0.0.1 in the VIA
address and include rport. The response will get to the right place. I
agree that always adding rport by the receiver is a good idea.

Not implementing transaction fail-over when devices migrate from one IP to
another typically causes about 1-2% call drops. If you are running a
service with long average call duration, such as call conferencing service,
this percentage becomes much higher. You can typically implement some sort
of call recovery by using INVITE with Replaces to work around such dropped
calls.

Regards,
_____________
Roman Shpount

On Mon, Nov 7, 2016 at 3:00 PM, Asveren, Tolga <tasveren@sonusnet.com>
wrote:

> i- For UDP, shouldn=E2=80=99t the reply sent based on =E2=80=9Creceived=
=E2=80=9D parameter?
> (RFC3261 18.2.2)
>
> This should take care of the IP Address issue IMHO.
>
>
>
> ii- For the port, =E2=80=9Crport=E2=80=9D would provide a solution but it=
 is not defined
> in RFC3261 but in RFC3581 (IMHO it is a good idea for the sender always a=
dd
> the =E2=80=9Crport=E2=80=9D, and even in its absence I think there is no =
harm for the
> receiver to assume that it has been received and hence use the port value
> in the packet header for response routing).
>
>
>
> iii- I don=E2=80=99t think =E2=80=9Crouting responses during failover wit=
hout failing the
> transaction=E2=80=9D is a widely required/implemented strategy. By and la=
rge, such
> transaction fail and are retried.
>
>
>
> I am not saying that populating Via header with the actual packet source
> IP/port is a bad idea but I think there is nothing much broken from
> protocol mechanics perspective if it is not done. In practice, it may cau=
se
> =E2=80=9Cfalse alarms=E2=80=9D as far as presence of a NAT device is conc=
erned.
>
>
>
> Thanks,
>
> Tolga
>
>
>
> *From:* sipcore [mailto:sipcore-bounces@ietf.org] *On Behalf Of *Roman
> Shpount
> *Sent:* Monday, November 07, 2016 12:35 PM
> *To:* Paul Kyzivat <pkyzivat@alum.mit.edu>
> *Cc:* SIPCORE <sipcore@ietf.org>
> *Subject:* Re: [sipcore] Source address selection in SIP
>
>
>
> On Mon, Nov 7, 2016 at 12:03 PM, Paul Kyzivat <pkyzivat@alum.mit.edu>
> wrote:
>
> Is there some thought that this choice affects anything else in the
> outgoing message?
>
>
>
> The only other issue this affects is delivering SIP response to a message
> when client IP changes in the middle of the SIP transaction. You either
> need to accept that the transaction can fail or come up with a mechanism =
to
> recover a SIP transaction on the new client local IP or use something lik=
e
> TURN to always tunnel SIP messages through the same IP/port in the cloud
> (resource intensive).
>
>
>
> Regards,
>
> _____________
> Roman Shpount
>
>
>

--001a11c01b1e30dd520540bbac9c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Just a side note: received and rport are generally useless=
 for TCP or TLS based transactions. Responses are supposed to be sent over =
the same TCP/TLS connection.<div><br></div><div>For UDP, when sending a mes=
sage you can safely put 127.0.0.1 in the VIA address and include rport. The=
 response will get to the right place. I agree that always adding rport by =
the receiver is a good idea.</div><div><br></div><div>Not implementing tran=
saction fail-over when devices migrate from one IP to another typically cau=
ses about 1-2% call drops. If you are running a service with long average c=
all duration, such as call conferencing service, this percentage becomes mu=
ch higher. You can typically implement some sort of call recovery by using =
INVITE with Replaces to work around such dropped calls.<br><div class=3D"gm=
ail_extra"><br></div><div class=3D"gmail_extra">Regards,<br clear=3D"all"><=
div><div class=3D"gmail_signature" data-smartmail=3D"gmail_signature">_____=
________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Mon, Nov 7, 2016 at 3:00 PM, Asveren, Tol=
ga <span dir=3D"ltr">&lt;<a href=3D"mailto:tasveren@sonusnet.com" target=3D=
"_blank">tasveren@sonusnet.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"m_2445297143837035421WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">i- For UDP, shouldn=E2=80=99t the rep=
ly sent based on =E2=80=9Creceived=E2=80=9D parameter? (RFC3261 18.2.2)
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">This should take care of the IP Addre=
ss issue IMHO.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">ii- For the port, =E2=80=9Crport=E2=
=80=9D would provide a solution but it is not defined in RFC3261 but in RFC=
3581 (IMHO it is a good idea for the sender always add the =E2=80=9Crport=
=E2=80=9D,
 and even in its absence I think there is no harm for the receiver to assum=
e that it has been received and hence use the port value in the packet head=
er for response routing).<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">iii- I don=E2=80=99t think =E2=80=9Cr=
outing responses during failover without failing the transaction=E2=80=9D i=
s a widely required/implemented strategy. By and large, such transaction
 fail and are retried.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">I am not saying that populating Via h=
eader with the actual packet source IP/port is a bad idea but I think there=
 is nothing much broken from protocol mechanics
 perspective if it is not done. In practice, it may cause =E2=80=9Cfalse al=
arms=E2=80=9D as far as presence of a NAT device is concerned.<u></u><u></u=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Thanks,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Tolga<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d"><u></u>=C2=A0<u></u></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #e1e1e1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,sans-serif">From:</span></b><span style=3D"font-size:11.0pt;=
font-family:&quot;Calibri&quot;,sans-serif"> sipcore [mailto:<a href=3D"mai=
lto:sipcore-bounces@ietf.org" target=3D"_blank">sipcore-bounces@ietf.<wbr>o=
rg</a>]
<b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> Monday, November 07, 2016 12:35 PM<br>
<b>To:</b> Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alum.mit.edu" target=
=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;<br>
<b>Cc:</b> SIPCORE &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank=
">sipcore@ietf.org</a>&gt;<br>
<b>Subject:</b> Re: [sipcore] Source address selection in SIP<u></u><u></u>=
</span></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Mon, Nov 7, 2016 at 12:03 PM, Paul Kyzivat &lt;<a=
 href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.=
edu</a>&gt; wrote:<u></u><u></u></p>
</div>
</div>
<div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal">Is there some thought that this choice affects anyth=
ing else in the outgoing message?<u></u><u></u></p>
</blockquote>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">The only other issue this affects is delivering SIP =
response to a message when client IP changes in the middle of the SIP trans=
action. You either need to accept that the transaction can fail or come up =
with a mechanism to recover a SIP
 transaction on the new client local IP or use something like TURN to alway=
s tunnel SIP messages through the same IP/port in the cloud (resource inten=
sive).<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<u></u><u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">_____________<br>
Roman Shpount<u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div></div></div>
</div>
</div>

</blockquote></div><br></div></div></div>

--001a11c01b1e30dd520540bbac9c--


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Subject: Re: [sipcore] [dispatch] SIPCORE Rechartering
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--=_MailMate_D0419DDD-97D5-4FFF-92FD-557E0D4BDDB2_=
Content-Type: text/plain; format=flowed

The proposed new SIPCORE charter text is now available at 
https://datatracker.ietf.org/doc/charter-ietf-sipcore/ . It's identical 
to the text from Adam's email, other than some potential formatting 
differences.

Thanks!

Ben.

On 4 Nov 2016, at 16:28, Adam Roach wrote:

> [as SIPCORE chair]
>
> Back when we transitioned from the SIPPING/SIP working group structure 
> to SIPCORE, we put relatively tight limits on the SIPCORE charter. 
> This was a recognition of the fact that the volume of SIP-related work 
> at the time was too large for any one working group to reasonably 
> handle. Instead, the intention was that the DISPATCH model of creating 
> small, short-lived working groups for specific topics would be a 
> better way to manage the relatively heavy workload.
>
> Fast forward seven years to today. SIP is a much more mature protocol, 
> and the inflow of new work is significantly smaller than it was back 
> then. At the same time, we have had several small, self-contained work 
> items show up in DISPATCH that were deemed too small to create a new 
> working group for, but precluded from being dispatched to SIPCORE by 
> its charter. This has led to unnecessary delays as we determined the 
> best disposition for these items.
>
> We, the SIPCORE chairs and area director, seek to fix that. We would 
> like to amend the SIPCORE charter to allow it to take on small, 
> self-contained protocol extensions in addition to changes to the core 
> SIP protocol. This is a relatively minor update to the existing 
> charter, which expands the scope as described above, and adds back in 
> two of the architectural principles from the SIP charter which are 
> applicable only to protocol extensions.
>
> Please provide feedback on the proposed charter by sending email to 
> the SIPCORE mailing list. We will also discuss this briefly during the 
> DISPATCH session in Seoul.
>
> The proposed charter text follows.
>
>> The Session Initiation Protocol Core (SIPCore) working group is
>> chartered to maintain and continue the development of the SIP 
>> protocol,
>> currently defined as proposed standard RFCs 3261, 3262, 3263, 3264, 
>> and
>> 6665.
>>
>> The SIPCore working group will concentrate on specifications that 
>> update
>> or replace the core SIP specifications named above as well as
>> specifications pertaining to small, self-contained SIP protocol
>> extensions. The process and requirements for new SIP extensions are
>> documented in RFC5727, "Change Process for the Session Initiation
>> Protocol".
>>
>> Throughout its work, the group will strive to maintain the basic 
>> model
>> and architecture defined by SIP. In particular:
>>
>> 1. Services and features are provided end-to-end whenever possible.
>>
>> 2. Reuse of existing Internet protocols and architectures and
>> integrating with other Internet applications is crucial.
>>
>> 3. Standards-track extensions and new features must be generally
>> applicable, and not applicable only to a specific set of session
>> types.
>>
>> 4. Simpler solutions that solve a given problem should be favored
>> over more complex solutions.
>>
>> The primary source of change requirements to the core SIP 
>> specifications
>> (enumerated above) will be a) interoperability problems that stem 
>> from
>> ambiguous, or under-defined specification, and b) requirements from
>> other working groups in the ART Area. The primary source of new 
>> protocol
>> extensions is the DISPATCH working group, which will generally make 
>> the
>> determination regarding whether new SIP-related work warrants a new
>> working group or belongs in an existing one.
>
> For ease of reference, the existing SIPCORE charter is at 
> https://datatracker.ietf.org/doc/charter-ietf-sipcore/
>
> Thanks!
>
> /a


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

--=_MailMate_D0419DDD-97D5-4FFF-92FD-557E0D4BDDB2_=
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/xhtml; charset=3Dutf-8"=
>
</head>
<body>
<div style=3D"font-family:sans-serif"><div style=3D"white-space:pre-wrap"=
><div dir=3D"auto">The proposed new SIPCORE charter text is now available=
 at <a href=3D"https://datatracker.ietf.org/doc/charter-ietf-sipcore/" st=
yle=3D"color:#3983C4">https://datatracker.ietf.org/doc/charter-ietf-sipco=
re/</a> . It's identical to the text from Adam's email, other than some p=
otential formatting differences.
</div><div dir=3D"auto">
</div><div dir=3D"auto">Thanks!
</div><div dir=3D"auto">
</div><div dir=3D"auto">Ben.
</div><div dir=3D"auto">
</div><div dir=3D"auto">On 4 Nov 2016, at 16:28, Adam Roach wrote:
</div><div dir=3D"auto">
</div></div>
<blockquote style=3D"border-left:2px solid #777; color:#777; margin:0 0 5=
px; padding-left:5px"><div id=3D"1600C0CA-5C8A-43AB-B0D1-5C23687E04E7">
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <p>[as SIPCORE chair]</p>
    <p>Back when we transitioned from the SIPPING/SIP working group
      structure to SIPCORE, we put relatively tight limits on the
      SIPCORE charter. This was a recognition of the fact that the
      volume of SIP-related work at the time was too large for any one
      working group to reasonably handle. Instead, the intention was
      that the DISPATCH model of creating small, short-lived working
      groups for specific topics would be a better way to manage the
      relatively heavy workload.</p>
    <p>Fast forward seven years to today. SIP is a much more mature
      protocol, and the inflow of new work is significantly smaller than
      it was back then. At the same time, we have had several small,
      self-contained work items show up in DISPATCH that were deemed too
      small to create a new working group for, but precluded from being
      dispatched to SIPCORE by its charter. This has led to unnecessary
      delays as we determined the best disposition for these items.<br>
    </p>
    <p>We, the SIPCORE chairs and area director, seek to fix that. We
      would like to amend the SIPCORE charter to allow it to take on
      small, self-contained protocol extensions in addition to changes
      to the core SIP protocol. This is a relatively minor update to the
      existing charter, which expands the scope as described above, and
      adds back in two of the architectural principles from the SIP
      charter which are applicable only to protocol extensions.</p>
    <p>Please provide feedback on the proposed charter by sending email
      to the SIPCORE mailing list. We will also discuss this briefly
      during the DISPATCH session in Seoul.</p>
    <p>The proposed charter text follows.</p>
    <p><br>
    </p>
    <p>
      <blockquote type=3D"cite">
        <meta http-equiv=3D"content-type" content=3D"text/html;
          charset=3Dutf-8">
        <div id=3D"magicdomid2" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The=

            Session Initiation Protocol Core (SIPCore) working group is</=
span></div>
        <div id=3D"magicdomid3" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">cha=
rtered
            to maintain and continue the development of the SIP
            protocol,</span></div>
        <div id=3D"magicdomid4" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">cur=
rently
            defined as proposed standard RFCs 3261, 3262, 3263, 3264,
            and</span></div>
        <div id=3D"magicdomid5" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">666=
5.</span></div>
        <div id=3D"magicdomid6" class=3D""><br>
        </div>
        <div id=3D"magicdomid7" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The=

            SIPCore working group will concentrate on specifications
            that update</span></div>
        <div id=3D"magicdomid8" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">or
            replace the core SIP specifications named above as well as</s=
pan></div>
        <div id=3D"magicdomid9" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">spe=
cifications
            pertaining to small, self-contained SIP protocol</span></div>=

        <div id=3D"magicdomid10" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">ext=
ensions.=C2=A0
            The process and requirements for new SIP extensions are</span=
></div>
        <div id=3D"magicdomid11" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">doc=
umented
            in RFC5727, "Change Process for the Session Initiation</span>=
</div>
        <div id=3D"magicdomid12" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Pro=
tocol".</span></div>
        <div id=3D"magicdomid13" class=3D""><br>
        </div>
        <div id=3D"magicdomid14" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Thr=
oughout
            its work, the group will strive to maintain the basic model</=
span></div>
        <div id=3D"magicdomid15" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">and=

            architecture defined by SIP. In particular:</span></div>
        <div id=3D"magicdomid16" class=3D""><br>
        </div>
        <div id=3D"magicdomid17" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">1.
            Services and features are provided end-to-end whenever
            possible.</span></div>
        <div id=3D"magicdomid18" class=3D""><br>
        </div>
        <div id=3D"magicdomid19" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">2.
            Reuse of existing Internet protocols and architectures and</s=
pan></div>
        <div id=3D"magicdomid20" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">=C2=
=A0=C2=A0
            integrating with other Internet applications is crucial.</spa=
n></div>
        <div id=3D"magicdomid21" class=3D""><br>
        </div>
        <div id=3D"magicdomid22" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">3.
            Standards-track extensions and new features must be
            generally</span></div>
        <div id=3D"magicdomid23" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">=C2=
=A0=C2=A0
            applicable, and not applicable only to a specific set of
            session</span></div>
        <div id=3D"magicdomid24" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">=C2=
=A0=C2=A0
            types.</span></div>
        <div id=3D"magicdomid25" class=3D""><br>
        </div>
        <div id=3D"magicdomid26" class=3D""><span
            class=3D"author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">4.
            Simpler solutions that solve a given problem should be
            favored</span></div>
        <div id=3D"magicdomid27" class=3D""><span
            class=3D"author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">=C2=A0=
=C2=A0=C2=A0
            over more complex solutions.</span></div>
        <div id=3D"magicdomid28" class=3D""><br>
        </div>
        <div id=3D"magicdomid29" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The=

            primary source of change requirements to the core SIP
            specifications</span></div>
        <div id=3D"magicdomid30" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">(en=
umerated
            above) will be a) interoperability problems that stem from</s=
pan></div>
        <div id=3D"magicdomid31" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">amb=
iguous,
            or under-defined specification, and b) requirements from</spa=
n></div>
        <div id=3D"magicdomid32" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">oth=
er
            working groups in the ART Area. The primary source of new
            protocol</span></div>
        <div id=3D"magicdomid33" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">ext=
ensions
            is the DISPATCH working group, which will generally make the<=
/span></div>
        <div id=3D"magicdomid34" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">det=
ermination
            regarding whether new SIP-related work warrants a new</span><=
/div>
        <div id=3D"magicdomid35" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">wor=
king
            group or belongs in an existing one.</span></div>
      </blockquote>
    </p>
    <p><br>
    </p>
    <p>For ease of reference, the existing SIPCORE charter is at
      <a class=3D"moz-txt-link-freetext" href=3D"https://datatracker.ietf=
=2Eorg/doc/charter-ietf-sipcore/">https://datatracker.ietf.org/doc/charte=
r-ietf-sipcore/</a></p>
    <p>Thanks!<br>
    </p>
    <p>/a<br>
    </p>
  </div></div></blockquote>
<div style=3D"white-space:pre-wrap"><div dir=3D"auto">
</div><blockquote style=3D"border-left:2px solid #777; color:#777; margin=
:0 0 5px; padding-left:5px"><div dir=3D"auto">___________________________=
____________________
</div><div dir=3D"auto">dispatch mailing list
</div><div dir=3D"auto">dispatch@ietf.org
</div><div dir=3D"auto"><a href=3D"https://www.ietf.org/mailman/listinfo/=
dispatch" style=3D"color:#777">https://www.ietf.org/mailman/listinfo/disp=
atch</a>
</div></blockquote></div>
</div>
</body>
</html>

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From nobody Mon Nov  7 15:22:38 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>
Date: Mon, 07 Nov 2016 17:22:26 -0600
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Subject: Re: [sipcore] [dispatch] SIPCORE Rechartering
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On 4 Nov 2016, at 20:17, Henning Schulzrinne wrote:

> Will existing SIP-related drafts in DISPATCH be moved to SIPCORE?

To my knowledge, there are no SIP related drafts that are actual 
milestones for DISPATCH. It would seem reasonable to "dispatch" drafts 
currently in discussion that fit the proposed new SIPCORE charter to 
SIPCORE.

There have been some drafts that DISPATCH recommended to be AD 
sponsored. I wouldn't expect a wholesale move of active AD sponsored 
drafts to SIPCORE, but I think it is an option we could consider on a 
case by case basis.

In any case, we should not hold up progress on active drafts pending the 
SIPCORE recharter.
>
> On Fri, Nov 4, 2016 at 8:23 PM, Dan York <york@isoc.org> wrote:
>
>> Adam,
>>
>> On Nov 4, 2016, at 5:28 PM, Adam Roach <adam@nostrum.com> wrote:
>>
>> <snip>
>>
>> We, the SIPCORE chairs and area director, seek to fix that. We would 
>> like
>> to amend the SIPCORE charter to allow it to take on small, 
>> self-contained
>> protocol extensions in addition to changes to the core SIP protocol. 
>> This
>> is a relatively minor update to the existing charter, which expands 
>> the
>> scope as described above, and adds back in two of the architectural
>> principles from the SIP charter which are applicable only to protocol
>> extensions.
>>
>> Please provide feedback on the proposed charter by sending email to 
>> the
>> SIPCORE mailing list. We will also discuss this briefly during the 
>> DISPATCH
>> session in Seoul.
>>
>>
>> This makes sense to me. The proposed text looked fine to me.
>>
>> Dan
>>
>>
>>
>> _______________________________________________
>> 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 nobody Mon Nov  7 17:28:43 2016
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From: worley@ariadne.com (Dale R. Worley)
To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [sipcore] Source address selection in SIP
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"Olle E. Johansson" <oej@edvina.net> writes:
> An IPv6 interface by default has multiple addresses. Today's computers
> also quite frequently have multiple interfaces.  A modern SIP stack
> has many IP addresses to select from.

> I tried to find any words in RFC 3261 or RFC 3263 about selection of
> interfaces and addresses but found nothing.  Having said that, it's
> not easy so if anyone on sipcore can point me to the right section I
> will be grateful.
>
> If not, I think we may need a clarification on Transaction layer
> handling of IP address selection in the SIP protocol. Given a specific
> target, how do we handle multiple interfaces and multiple addresses
> per interface - which address do we put in the SIP messages - like in
> via's, record-route, SDPs and contacts.
>
> We have been testing at SIPit with multiple ULAs attached to a network
> and a proxy on one of the ULAs.  No one could connect to that proxy
> properly (sadly for me not even Kamailio itself, which brags about
> having IPv6 support from day one).

It seems to me that the relevant texts are:

RFC 7984 section 4, "Clarification of Interaction with RFC 6724"
RFC 6157 section 5, "Contacting Servers: Interaction of RFC 3263 and RFC
3484"
RFC 6724 section 5, "Source Address Selection"

In the case where the client's one IPv6 interface has a number of
global-scope addresses, only one of which can talk to an outgoing
router, and the destination has one IPv6 address (that can only be
reached in the global scope), processing seems to progress as follows:

Destination address selection is trivial, as there's only one
destination address.

Source address selection starts with all of the client's IPv6 addresses.
RFC 6724 section 5's source address prioritization rules are performed
as follows:

Rule 1 makes no choice.
Rule 2 makes no choice because all of the interface addresses have
global scope.
Rule 3 makes no choice because (I assume) no addresses are deprecated.
Rule 4 makes no choice because (I assume) we are not using IPv6
mobility.
Rule 5 makes no choice because the destination address is not one of the
client's addresses.

Rule 5.5 is:

   Rule 5.5: Prefer addresses in a prefix advertised by the next-hop.
   If SA or SA's prefix is assigned by the selected next-hop that will
   be used to send to D and SB or SB's prefix is assigned by a different
   next-hop, then prefer SA.  Similarly, if SB or SB's prefix is
   assigned by the next-hop that will be used to send to D and SA or
   SA's prefix is assigned by a different next-hop, then prefer SB.

Now since there is only one next-hop router, the next-hop is
well-defined.  And that router has assigned the prefix of exactly one of
the interface's addresses.  Rule 5.5 prioritizes that source address
above all the other addresses.

Since that step prefers one address over all others, the remaining rules
are not significant.

Thus, the client should be using the source address which links to the
proxy.  Which is what you would expect.

Interestingly, this logic is supposed to be incorporated into the
getaddrinfo() function, so the SIP implementation shouldn't need to
contain specific logic to handle it if the implementation invokes
getaddrinfo().

However, that section has this qualifier:

      Discussion: An IPv6 implementation is not required to remember
      which next-hops advertised which prefixes.  The conceptual models
      of IPv6 hosts in Section 5 of [RFC4861] and Section 3 of [RFC4191]
      have no such requirement.  Hence, Rule 5.5 is only applicable to
      implementations that track this information.

Interestingly, I've been having problems with my browser (an old edition
of Firefox) when I got IPv6 service, and it is exactly this problem!

> Executive question summary: Does the current set of SIP standards cover
> IP address selection for the *sender* when we have multiple IP
> interfaces (IPv4, IPv6) or just one interface with multiple addresses?

RFC 6724 section 5 clearly deals with all addresses of all interfaces at
the same time, so I think that issue is properly handled.

Dale


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From: Roman Shpount <roman@telurix.com>
Date: Tue, 8 Nov 2016 14:31:28 -0500
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Subject: [sipcore]  Source address cahnges in SIP
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I am changing the subject since this is not about the SIP source address
selection any more.

On Tue, Nov 8, 2016 at 3:38 AM, Asveren, Tolga <tasveren@sonusnet.com>
wrote:

> I am surprised that calls fail because of IP address migration. This seems
> to be more of an application issue rather than related with the SIP
> protocol. SIP has retransmissions for transactions. I think applications
> need to be updated/corrected for the scenarios you mention.
>
>
>
SIP has re-transmission for transactions but a lot of state full SIP
transactions implementations will not update the via header between
re-transmissions. So, if original message is sent from one IP and
re-transmission from another, response to the re-transmission will go the
the original IP/port.

Another interesting error case is when SIP over TCP or TLS is used and IP
address changes when SIP message is in flight. TCP connection is reset, but
it is unclear if SIP message was delivered to the server. Also, server has
no active TCP connection to send back the response.

Both of those situations in combination with SIP Session Timers can cause
calls to fail.

Finally, there is a situation when server needs to send a SIP Session timer
message to the client, but client IP have changed and client have not
detected this change yet. In such cases, server need to lookup client IP/port
based on GRUU in contact on every re-transmit.
_____________
Roman Shpount

--001a1149a18cdd32e60540cf2f3e
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">I am changing the subject since this is not about the SIP source addre=
ss selection any more.</div><div class=3D"gmail_signature"><br></div><div c=
lass=3D"gmail_signature">On Tue, Nov 8, 2016 at 3:38 AM, <span class=3D"" i=
d=3D":6s3.1" tabindex=3D"-1">Asveren</span>, <span class=3D"" id=3D":6s3.2"=
 tabindex=3D"-1">Tolga</span> <span dir=3D"ltr">&lt;<a href=3D"mailto:tasve=
ren@sonusnet.com" target=3D"_blank"><span class=3D"" id=3D":6s3.3" tabindex=
=3D"-1">tasveren</span>@<span class=3D"" id=3D":6s3.4" tabindex=3D"-1">sonu=
snet</span>.com</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_qu=
ote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left:1px solid rgb(204,204,204);padding-left:1ex">





<div lang=3D"EN-US">
<div class=3D"gmail-m_-1385903308383310173WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">I am surprised that calls fail because of IP=
 address migration. This seems to be more of an application issue rather th=
an related with the SIP protocol. SIP has
 retransmissions for transactions. I think applications need to be updated/=
corrected for the scenarios you mention.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>SIP has re-transmission for transactions but a lot of state full SIP trans=
actions implementations will not update the via header between re-transmiss=
ions. So, if original message is sent from one <span class=3D"" id=3D":6s3.=
6" tabindex=3D"-1">IP</span> and re-transmission from another, response to =
the re-transmission will go the the original <span class=3D"" id=3D":6s3.7"=
 tabindex=3D"-1">IP</span>/port.</div><div><br></div><div>Another interesti=
ng error case is when SIP over <span class=3D"" id=3D":6s3.8" tabindex=3D"-=
1">TCP</span> or <span class=3D"" id=3D":6s3.9" tabindex=3D"-1">TLS</span> =
is used and <span class=3D"" id=3D":6s3.10" tabindex=3D"-1">IP</span> addre=
ss changes when SIP message is in flight. <span class=3D"" id=3D":6s3.11" t=
abindex=3D"-1">TCP</span> connection is reset, but it is unclear if SIP mes=
sage was delivered to the server. Also, server has no active <span class=3D=
"" id=3D":6s3.12" tabindex=3D"-1">TCP</span> connection to send back the re=
sponse.</div><div><br></div><div>Both of those situations in combination wi=
th SIP Session Timers can cause calls to fail.</div><div><br></div><div>Fin=
ally, there is a situation when server needs to send a SIP Session timer me=
ssage to the client, but client <span class=3D"" id=3D":6s3.13" tabindex=3D=
"-1">IP</span> have changed and client have not detected this change yet. I=
n such cases, server need to <span class=3D"" id=3D":6s3.14" tabindex=3D"-1=
">lookup</span> client <span class=3D"" id=3D":6s3.15" tabindex=3D"-1">IP</=
span>/port based on <span class=3D"" id=3D":6s3.16" tabindex=3D"-1">GRUU</s=
pan> in contact on every re-transmit.</div><div><div class=3D"gmail_signatu=
re">_____________<br>Roman <span class=3D"" id=3D":6s3.17" tabindex=3D"-1">=
Shpount</span></div></div><div>=C2=A0</div></div></div></div>

--001a1149a18cdd32e60540cf2f3e--


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To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [sipcore] Source address selection in SIP
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"Olle E. Johansson" <oej@edvina.net> writes:
> So who decides what to put in the VIA of the SIP Options??

And also, how does it get put there?  In many cases, it seems that the
underlying network stack will send the request with the "right" source
address, but the SIP client first has to extract from the network stack
what the source address will be, then insert it into the Via header, and
then send the request...  I suppose this isn't an interesting question
of protocol design, but it is probably an annoyance to implementers.

Dale


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Subject: Re: [sipcore] Source address selection in SIP
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"Olle E. Johansson" <oej@edvina.net> writes:
> An IPv6 interface by default has multiple addresses. Today's computers
> also quite frequently have multiple interfaces.  A modern SIP stack
> has many IP addresses to select from.

> We have been testing at SIPit with multiple ULAs attached to a network
> and a proxy on one of the ULAs.  No one could connect to that proxy
> properly (sadly for me not even Kamailio itself, which brags about
> having IPv6 support from day one).

Thinking more about this, it seems that a very similar problem occurs in
IPv4:  Consider a SIP client with two IPv4 interfaces, only one of which
is on a network with a gateway router to the global Internet.  When the
client sends a message, it specifies the destination IPv4 address.  The
client's network stack consults the "route table" to find the IPv4
address of the global gateway and the interface with which to contact
the gateway.  The network stack uses the source address of that
interface.  Everything works as you would like.  The only detail that is
not covered is arranging for the Via to contain an address, host name,
or rport such that the response can route back to the client.

Approximately the same thing should happen in the IPv6 case that Olle
cites:  The underlying network stack will, by default, execute the IPv6
source selection algorithm based on the destination IPv6 address, and it
will identify the IPv6 interface and the source address that can contact
the gateway.

However, Olle reports that the requests fail.  I do not see how that
could happen -- the default behavior of the network stack should cause
the request to be sent in a way that will succeed.

So I hope that Olle can supply more information regarding exactly how
this failure happens.  I suspect that the failure is more complicated
that it initially appears.

Dale


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[as individual]

Can you give a little more detail about the nature of the failures? It's not=
 immediately clear what would fail here. As was pointed out elsewhere, for U=
DP, responses are routed per received address; and for TCP, they're sent bac=
k on the connection from which the request arrived.

/a

> On Nov 6, 2016, at 18:18, Olle E. Johansson <oej@edvina.net> wrote:
>=20
> We have been testing at SIPit with multiple ULAs attached to a network and=
 a proxy on one of the ULAs.
> No one could connect to that proxy properly (sadly for me not even Kamaili=
o itself, which brags about having
> IPv6 support from day one).


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To: Roman Shpount <roman@telurix.com>
Thread-Topic: [sipcore] Source address cahnges in SIP
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Subject: Re: [sipcore] Source address cahnges in SIP
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From nobody Wed Nov  9 01:23:08 2016
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To: Adam Roach <adam@nostrum.com>
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Cc: SIPCORE <sipcore@ietf.org>, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] Source address selection in SIP
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> On 09 Nov 2016, at 09:08, Adam Roach <adam@nostrum.com> wrote:
>=20
> [as individual]
>=20
> Can you give a little more detail about the nature of the failures? =
It's not immediately clear what would fail here. As was pointed out =
elsewhere, for UDP, responses are routed per received address; and for =
TCP, they're sent back on the connection from which the request arrived.

1. DNS for a server has ULA address
2. Client with a hole set of IPv6 addresses - link local, global and =
multiple ULA resolves DNS
3. Client fails to select ULA address as source to reach server

/O
>=20
> /a
>=20
>> On Nov 6, 2016, at 18:18, Olle E. Johansson <oej@edvina.net> wrote:
>>=20
>> We have been testing at SIPit with multiple ULAs attached to a =
network and a proxy on one of the ULAs.
>> No one could connect to that proxy properly (sadly for me not even =
Kamailio itself, which brags about having
>> IPv6 support from day one).
>=20


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: "Olle E. Johansson" <oej@edvina.net>, Adam Roach <adam@nostrum.com>
Thread-Topic: [sipcore] Source address selection in SIP
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/B96E7_f2ivu8dU0YJxH_Dr7x98Y>
Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Source address selection in SIP
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Isn't this more an issue on OS-level, that it fails to select ULA as the so=
urce when destination is ULA?

Thanks,
Tolga

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E.
> Johansson
> Sent: Wednesday, November 09, 2016 4:23 AM
> To: Adam Roach <adam@nostrum.com>
> Cc: SIPCORE <sipcore@ietf.org>; Olle E Johansson <oej@edvina.net>
> Subject: Re: [sipcore] Source address selection in SIP
>=20
>=20
> > On 09 Nov 2016, at 09:08, Adam Roach <adam@nostrum.com> wrote:
> >
> > [as individual]
> >
> > Can you give a little more detail about the nature of the failures? It'=
s not
> immediately clear what would fail here. As was pointed out elsewhere, for
> UDP, responses are routed per received address; and for TCP, they're sent
> back on the connection from which the request arrived.
>=20
> 1. DNS for a server has ULA address
> 2. Client with a hole set of IPv6 addresses - link local, global and mult=
iple ULA
> resolves DNS 3. Client fails to select ULA address as source to reach ser=
ver
>=20
> /O
> >
> > /a
> >
> >> On Nov 6, 2016, at 18:18, Olle E. Johansson <oej@edvina.net> wrote:
> >>
> >> We have been testing at SIPit with multiple ULAs attached to a network
> and a proxy on one of the ULAs.
> >> No one could connect to that proxy properly (sadly for me not even
> >> Kamailio itself, which brags about having
> >> IPv6 support from day one).
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Wed Nov  9 02:35:20 2016
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To: "Asveren, Tolga" <tasveren@sonusnet.com>
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Subject: Re: [sipcore] Source address selection in SIP
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> On 09 Nov 2016, at 11:25, Asveren, Tolga <tasveren@sonusnet.com> =
wrote:
>=20
> Isn't this more an issue on OS-level, that it fails to select ULA as =
the source when destination is ULA?
Could be an O/S level issue, but the report indicated that most O/S has =
sorted out these problems in regards to IPv6.

Which means that we need to make sure that implementations do the right =
thing and use the right
source address in the SIP message. Easy to test in the SIPit network - =
maybe some of you can set something
up for the IETF hackathon coming up?

This failed miserably when we tested with many implementations.=20

I think the question for this group really is whether this is something =
that needs some extra specification,
as the current set of RFCs (as far as I can see) is pretty silent about =
selecting IP address from a set=20
to choose from. Seems like most of the text is based on one single IP =
per device.

In addition we have the issue with changing IP during a call where =
there=E2=80=99s no guidance for developers
in the current set of documents. Outbound doesn=E2=80=99t really handle =
in-dialog situations and there seems
to be a set of different solutions out there.

I am still not sure there=E2=80=99s a problem that requires additional =
RFCs or updates from this wg or if it
just something we should publish another SIP Forum developer document =
about...

/O


>=20
> Thanks,
> Tolga
>=20
>> -----Original Message-----
>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E.
>> Johansson
>> Sent: Wednesday, November 09, 2016 4:23 AM
>> To: Adam Roach <adam@nostrum.com>
>> Cc: SIPCORE <sipcore@ietf.org>; Olle E Johansson <oej@edvina.net>
>> Subject: Re: [sipcore] Source address selection in SIP
>>=20
>>=20
>>> On 09 Nov 2016, at 09:08, Adam Roach <adam@nostrum.com> wrote:
>>>=20
>>> [as individual]
>>>=20
>>> Can you give a little more detail about the nature of the failures? =
It's not
>> immediately clear what would fail here. As was pointed out elsewhere, =
for
>> UDP, responses are routed per received address; and for TCP, they're =
sent
>> back on the connection from which the request arrived.
>>=20
>> 1. DNS for a server has ULA address
>> 2. Client with a hole set of IPv6 addresses - link local, global and =
multiple ULA
>> resolves DNS 3. Client fails to select ULA address as source to reach =
server
>>=20
>> /O
>>>=20
>>> /a
>>>=20
>>>> On Nov 6, 2016, at 18:18, Olle E. Johansson <oej@edvina.net> wrote:
>>>>=20
>>>> We have been testing at SIPit with multiple ULAs attached to a =
network
>> and a proxy on one of the ULAs.
>>>> No one could connect to that proxy properly (sadly for me not even
>>>> Kamailio itself, which brags about having
>>>> IPv6 support from day one).
>>>=20
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] Source address selection in SIP
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"Olle E. Johansson" <oej@edvina.net> writes:
> I think the question for this group really is whether this is
> something that needs some extra specification, as the current set of
> RFCs (as far as I can see) is pretty silent about selecting IP address
> from a set to choose from. Seems like most of the text is based on one
> single IP per device.

It seems to me that the existing RFCs provide sufficient guidance, as my
long-winded analysis a few days ago detailed.

Dale


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"Olle E. Johansson" <oej@edvina.net> writes:
>> Can you give a little more detail about the nature of the failures?
>> It's not immediately clear what would fail here. As was pointed out
>> elsewhere, for UDP, responses are routed per received address; and
>> for TCP, they're sent back on the connection from which the request
>> arrived.
>
> 1. DNS for a server has ULA address
> 2. Client with a hole set of IPv6 addresses - link local, global and
> multiple ULA resolves DNS
> 3. Client fails to select ULA address as source to reach server

It would help if you listed all the addresses involved, or at least gave
a generic description of what source address the client did select.

Also, what version of Kamailio(sp?) was involved, and what was the OS
that it was running on?  Does Kamailio do source address selection
itself or does it allow the OS to choose the source address?

Dale


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From: Roman Shpount <roman@telurix.com>
Date: Wed, 9 Nov 2016 19:58:24 -0500
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Subject: Re: [sipcore] Source address changes in SIP
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On Wed, Nov 9, 2016 at 4:21 AM, Asveren, Tolga <tasveren@sonusnet.com>
wrote:

> i- If a transaction fails, it is application=E2=80=99s responsibility to =
do the
> right thing =E2=80=93whatever that is based on the expected outcome-, e.g=
. sending
> a new request.
>

What happens on most of the clients if SIP Session timer refresh
transaction fails? Form what I know, typically the call is torn down.


> ii- For the TCP scenario you mentioned, it is again up to the application
> to send a new request just to be sure that it has been received.
>

For TCP sending the same request for an existing transaction over a new
flow can produce "interesting" results. In most cases response will be sent
to via address and port from the original transaction, which are ephemeral.
Plus the RFC says that message sent over TCP should be re-transmitted.


> iii-  For SIP Session Timers: They usually are set on a value much higher
> than transaction timeout values.
>

It does not matter what SIP Session timer value is. If SIP Session timer
refresh transaction fails, the whole call normally fails.


> Overall, I don=E2=80=99t think there is a serious SIP protocol level issu=
e here,
> which is causing pain in practice.
>
>
>
Actually, as far as I know, there are at least two SIP protocol level
issues here:

1. Handling of client IP change in the middle of transaction is not
specified anywhere.
2. Handling of TCP connection tear down in the middle of transaction is not
specified anywhere. Handling of SIP responses when client TCP connection
was torn down in RFC 3621 (sending to received/port from VIA over TCP) does
not work unless client is on public IP.

Regards,
_____________
Roman Shpount

--001a114a89cceba83f0540e7de29
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<div dir=3D"ltr"><div>On Wed, Nov 9, 2016 at 4:21 AM, Asveren, Tolga <span =
dir=3D"ltr">&lt;<a href=3D"mailto:tasveren@sonusnet.com" target=3D"_blank">=
tasveren@sonusnet.com</a>&gt;</span> wrote:<br></div><div><div class=3D"gma=
il_extra"><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);paddin=
g-left:1ex">





<div lang=3D"EN-US">
<div class=3D"gmail-m_6057315346717934809WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">i- If a transaction fails, it is application=
=E2=80=99s responsibility to do the right thing =E2=80=93whatever that is b=
ased on the expected outcome-, e.g. sending a new request.</span></p></div>=
</div></blockquote><div class=3D"gmail_quote"><br></div>What happens on mos=
t of the clients if SIP Session timer refresh transaction fails? Form what =
I know, typically the call is torn down.</div><div class=3D"gmail_quote"><d=
iv>=C2=A0<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div l=
ang=3D"EN-US"><div class=3D"gmail-m_6057315346717934809WordSection1"><p cla=
ss=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sans-ser=
if;color:rgb(31,73,125)"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">ii- For the TCP scenario you mentioned, it i=
s again up to the application to send a new request just to be sure that it=
 has been received.</span></p></div></div></blockquote><div><br></div><div>=
For TCP sending the same request for an existing transaction over a new flo=
w can produce &quot;interesting&quot; results. In most cases response will =
be sent to via address and port from the original transaction, which are ep=
hemeral. Plus the RFC says that message sent over TCP should be re-transmit=
ted.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex=
"><div lang=3D"EN-US"><div class=3D"gmail-m_6057315346717934809WordSection1=
"><p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,=
sans-serif;color:rgb(31,73,125)"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:calibri,sa=
ns-serif;color:rgb(31,73,125)">iii-=C2=A0 For SIP Session Timers: They usua=
lly are set on a value much higher than transaction timeout values.</span><=
/p></div></div></blockquote><div><br></div><div>It does not matter what SIP=
 Session timer value is. If SIP Session timer refresh transaction fails, th=
e whole call normally fails.<br></div><div>=C2=A0</div><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(2=
04,204,204);padding-left:1ex"><div lang=3D"EN-US"><div class=3D"gmail-m_605=
7315346717934809WordSection1"><p class=3D"MsoNormal"><span style=3D"font-si=
ze:11pt;font-family:calibri,sans-serif;color:rgb(31,73,125)">
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125);font-family:cali=
bri,sans-serif;font-size:11pt">Overall, I don=E2=80=99t think there is a se=
rious SIP protocol level issue here, which is causing pain in practice.</sp=
an><br></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>Actually, as far as I know, there are at least two SIP protocol level issu=
es here:</div><div><br></div><div>1. Handling of client IP change in the mi=
ddle of transaction is not specified anywhere.=C2=A0</div><div>2. Handling =
of TCP connection tear down in the middle of transaction is not specified a=
nywhere. Handling of SIP responses when client TCP connection was torn down=
 in RFC 3621 (sending to received/port from VIA over TCP) does not work unl=
ess client is on public IP.=C2=A0</div><div><br></div>Regards,<div class=3D=
"gmail_extra"><div class=3D"gmail_signature">_____________<br>Roman Shpount=
</div></div><div>=C2=A0</div></div></div></div></div>

--001a114a89cceba83f0540e7de29--


From nobody Fri Nov 11 01:20:03 2016
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Cc: sipcore@ietf.org, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] Source address selection in SIP
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> On 09 Nov 2016, at 17:20, Dale R. Worley <worley@ariadne.com> wrote:
>=20
> "Olle E. Johansson" <oej@edvina.net> writes:
>>> Can you give a little more detail about the nature of the failures?
>>> It's not immediately clear what would fail here. As was pointed out
>>> elsewhere, for UDP, responses are routed per received address; and
>>> for TCP, they're sent back on the connection from which the request
>>> arrived.
>>=20
>> 1. DNS for a server has ULA address
>> 2. Client with a hole set of IPv6 addresses - link local, global and
>> multiple ULA resolves DNS
>> 3. Client fails to select ULA address as source to reach server
>=20
> It would help if you listed all the addresses involved, or at least =
gave
> a generic description of what source address the client did select.
I have no idea what address the clients selected - I just got no =
messages
to my server.=20
>=20
> Also, what version of Kamailio(sp?) was involved, and what was the OS
> that it was running on?  Does Kamailio do source address selection
> itself or does it allow the OS to choose the source address?

I guess at that time it was Kamailio 4.0 - I test the dev code in github
so there=E2=80=99s no version number really.  This was SIPit 31 in Nice, =
France.

There was a link to Kamailio=E2=80=99s code for doing this earlier in =
this thread.
I haven=E2=80=99t gone through the details, but I think we=E2=80=99re =
opening a socket
to the target and check what address the O/S selects.=20

Will try to set up more tests so I can file a bug report. :-)

/O=


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Subject: Re: [sipcore] Source address selection in SIP
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"Olle E. Johansson" <oej@edvina.net> writes:
> I have no idea what address the clients selected - I just got no messages
> to my server. 

> Will try to set up more tests so I can file a bug report. :-)

It seems to me that there are two groups of possible causes.  One group
is fairly simple code failures, the OS is choosing the link address as
the source address, Kamailio is choosing the wrong destination address,
or something like that.  That group has the property that you can point
to some piece of code that is behaving incorrectly (out of
specification).

The other group comes from this clause in RFC 6724:

   Rule 5.5: Prefer addresses in a prefix advertised by the next-hop.
   If SA or SA's prefix is assigned by the selected next-hop that will
   be used to send to D and SB or SB's prefix is assigned by a different
   next-hop, then prefer SA.  Similarly, if SB or SB's prefix is
   assigned by the next-hop that will be used to send to D and SA or
   SA's prefix is assigned by a different next-hop, then prefer SB.

      Discussion: An IPv6 implementation is not required to remember
      which next-hops advertised which prefixes.  The conceptual models
      of IPv6 hosts in Section 5 of [RFC4861] and Section 3 of [RFC4191]
      have no such requirement.  Hence, Rule 5.5 is only applicable to
      implementations that track this information.

It's possible that the OS is selecting a global source address but not
one that the gateway router sees, because the source address has a
prefix that the router isn't using.  And it seems from "Discussion" that
the OS is *not* required by existing RFCs to pick the global address
with the prefix that will actually get gatewayed globally.

How to go about fixing that is an interesting question.

Dale


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From: Roman Shpount <roman@telurix.com>
Date: Fri, 11 Nov 2016 12:44:48 -0500
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To: "Dale R. Worley" <worley@ariadne.com>
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Cc: SIPCORE <sipcore@ietf.org>, "Olle E. Johansson" <oej@edvina.net>
Subject: Re: [sipcore] Source address selection in SIP
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One issue we saw was clients using VPN, with VPN taking them to a business
network which blocks SIP.

This is why we added addresses enumeration and STUN checks similar to ICE
to verify which address actually has internet access.

Regards,
_____________
Roman Shpount

On Fri, Nov 11, 2016 at 10:30 AM, Dale R. Worley <worley@ariadne.com> wrote:

> "Olle E. Johansson" <oej@edvina.net> writes:
> > I have no idea what address the clients selected - I just got no messages
> > to my server.
>
> > Will try to set up more tests so I can file a bug report. :-)
>
> It seems to me that there are two groups of possible causes.  One group
> is fairly simple code failures, the OS is choosing the link address as
> the source address, Kamailio is choosing the wrong destination address,
> or something like that.  That group has the property that you can point
> to some piece of code that is behaving incorrectly (out of
> specification).
>
> The other group comes from this clause in RFC 6724:
>
>    Rule 5.5: Prefer addresses in a prefix advertised by the next-hop.
>    If SA or SA's prefix is assigned by the selected next-hop that will
>    be used to send to D and SB or SB's prefix is assigned by a different
>    next-hop, then prefer SA.  Similarly, if SB or SB's prefix is
>    assigned by the next-hop that will be used to send to D and SA or
>    SA's prefix is assigned by a different next-hop, then prefer SB.
>
>       Discussion: An IPv6 implementation is not required to remember
>       which next-hops advertised which prefixes.  The conceptual models
>       of IPv6 hosts in Section 5 of [RFC4861] and Section 3 of [RFC4191]
>       have no such requirement.  Hence, Rule 5.5 is only applicable to
>       implementations that track this information.
>
> It's possible that the OS is selecting a global source address but not
> one that the gateway router sees, because the source address has a
> prefix that the router isn't using.  And it seems from "Discussion" that
> the OS is *not* required by existing RFCs to pick the global address
> with the prefix that will actually get gatewayed globally.
>
> How to go about fixing that is an interesting question.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr">One issue we saw was clients using VPN, with VPN taking th=
em to a business network which blocks SIP.<div><br></div><div>This is why w=
e added addresses enumeration and STUN checks similar to ICE to verify whic=
h address actually has internet access.</div><div><br></div><div>Regards,<d=
iv class=3D"gmail_extra"><div><div class=3D"gmail_signature" data-smartmail=
=3D"gmail_signature">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Fri, Nov 11, 2016 at 10:30 AM, Dale R. Wo=
rley <span dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com" target=3D"=
_blank">worley@ariadne.com</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex"><span class=3D"">&quot;Olle E. Johansson&quot; &lt;<a href=3D"mailt=
o:oej@edvina.net">oej@edvina.net</a>&gt; writes:<br>
</span><span class=3D"">&gt; I have no idea what address the clients select=
ed - I just got no messages<br>
&gt; to my server.<br>
<br>
</span><span class=3D"">&gt; Will try to set up more tests so I can file a =
bug report. :-)<br>
<br>
</span>It seems to me that there are two groups of possible causes.=C2=A0 O=
ne group<br>
is fairly simple code failures, the OS is choosing the link address as<br>
the source address, Kamailio is choosing the wrong destination address,<br>
or something like that.=C2=A0 That group has the property that you can poin=
t<br>
to some piece of code that is behaving incorrectly (out of<br>
specification).<br>
<br>
The other group comes from this clause in RFC 6724:<br>
<br>
=C2=A0 =C2=A0Rule 5.5: Prefer addresses in a prefix advertised by the next-=
hop.<br>
=C2=A0 =C2=A0If SA or SA&#39;s prefix is assigned by the selected next-hop =
that will<br>
=C2=A0 =C2=A0be used to send to D and SB or SB&#39;s prefix is assigned by =
a different<br>
=C2=A0 =C2=A0next-hop, then prefer SA.=C2=A0 Similarly, if SB or SB&#39;s p=
refix is<br>
=C2=A0 =C2=A0assigned by the next-hop that will be used to send to D and SA=
 or<br>
=C2=A0 =C2=A0SA&#39;s prefix is assigned by a different next-hop, then pref=
er SB.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 Discussion: An IPv6 implementation is not required to =
remember<br>
=C2=A0 =C2=A0 =C2=A0 which next-hops advertised which prefixes.=C2=A0 The c=
onceptual models<br>
=C2=A0 =C2=A0 =C2=A0 of IPv6 hosts in Section 5 of [RFC4861] and Section 3 =
of [RFC4191]<br>
=C2=A0 =C2=A0 =C2=A0 have no such requirement.=C2=A0 Hence, Rule 5.5 is onl=
y applicable to<br>
=C2=A0 =C2=A0 =C2=A0 implementations that track this information.<br>
<br>
It&#39;s possible that the OS is selecting a global source address but not<=
br>
one that the gateway router sees, because the source address has a<br>
prefix that the router isn&#39;t using.=C2=A0 And it seems from &quot;Discu=
ssion&quot; that<br>
the OS is *not* required by existing RFCs to pick the global address<br>
with the prefix that will actually get gatewayed globally.<br>
<br>
How to go about fixing that is an interesting question.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Dale<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
______________________________<wbr>_________________<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" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/sipcore</a><=
br>
</div></div></blockquote></div><br></div></div></div>

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Date: Mon, 14 Nov 2016 14:38:07 +0900
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Subject: Re: [sipcore] [dispatch] SIPCORE Rechartering
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Hi,

The proposed charter (at the link below) will be reviewed by the IESG at 
the Dec 1 Telechat. If people have further comments, please make them as 
soon as possible.

Thanks!

Ben.

On 8 Nov 2016, at 8:05, Ben Campbell wrote:

> The proposed new SIPCORE charter text is now available at 
> https://datatracker.ietf.org/doc/charter-ietf-sipcore/ . It's 
> identical to the text from Adam's email, other than some potential 
> formatting differences.
>
> Thanks!
>
> Ben.
>
> On 4 Nov 2016, at 16:28, Adam Roach wrote:
>
>> [as SIPCORE chair]
>>
>> Back when we transitioned from the SIPPING/SIP working group 
>> structure to SIPCORE, we put relatively tight limits on the SIPCORE 
>> charter. This was a recognition of the fact that the volume of 
>> SIP-related work at the time was too large for any one working group 
>> to reasonably handle. Instead, the intention was that the DISPATCH 
>> model of creating small, short-lived working groups for specific 
>> topics would be a better way to manage the relatively heavy workload.
>>
>> Fast forward seven years to today. SIP is a much more mature 
>> protocol, and the inflow of new work is significantly smaller than it 
>> was back then. At the same time, we have had several small, 
>> self-contained work items show up in DISPATCH that were deemed too 
>> small to create a new working group for, but precluded from being 
>> dispatched to SIPCORE by its charter. This has led to unnecessary 
>> delays as we determined the best disposition for these items.
>>
>> We, the SIPCORE chairs and area director, seek to fix that. We would 
>> like to amend the SIPCORE charter to allow it to take on small, 
>> self-contained protocol extensions in addition to changes to the core 
>> SIP protocol. This is a relatively minor update to the existing 
>> charter, which expands the scope as described above, and adds back in 
>> two of the architectural principles from the SIP charter which are 
>> applicable only to protocol extensions.
>>
>> Please provide feedback on the proposed charter by sending email to 
>> the SIPCORE mailing list. We will also discuss this briefly during 
>> the DISPATCH session in Seoul.
>>
>> The proposed charter text follows.
>>
>>> The Session Initiation Protocol Core (SIPCore) working group is
>>> chartered to maintain and continue the development of the SIP 
>>> protocol,
>>> currently defined as proposed standard RFCs 3261, 3262, 3263, 3264, 
>>> and
>>> 6665.
>>>
>>> The SIPCore working group will concentrate on specifications that 
>>> update
>>> or replace the core SIP specifications named above as well as
>>> specifications pertaining to small, self-contained SIP protocol
>>> extensions. The process and requirements for new SIP extensions are
>>> documented in RFC5727, "Change Process for the Session Initiation
>>> Protocol".
>>>
>>> Throughout its work, the group will strive to maintain the basic 
>>> model
>>> and architecture defined by SIP. In particular:
>>>
>>> 1. Services and features are provided end-to-end whenever possible.
>>>
>>> 2. Reuse of existing Internet protocols and architectures and
>>> integrating with other Internet applications is crucial.
>>>
>>> 3. Standards-track extensions and new features must be generally
>>> applicable, and not applicable only to a specific set of session
>>> types.
>>>
>>> 4. Simpler solutions that solve a given problem should be favored
>>> over more complex solutions.
>>>
>>> The primary source of change requirements to the core SIP 
>>> specifications
>>> (enumerated above) will be a) interoperability problems that stem 
>>> from
>>> ambiguous, or under-defined specification, and b) requirements from
>>> other working groups in the ART Area. The primary source of new 
>>> protocol
>>> extensions is the DISPATCH working group, which will generally make 
>>> the
>>> determination regarding whether new SIP-related work warrants a new
>>> working group or belongs in an existing one.
>>
>> For ease of reference, the existing SIPCORE charter is at 
>> https://datatracker.ietf.org/doc/charter-ietf-sipcore/
>>
>> Thanks!
>>
>> /a
>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch


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

--=_MailMate_500C35E0-72AC-40F0-BB2C-4C325947C2D5_=
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/xhtml; charset=3Dutf-8"=
>
</head>
<body>
<div style=3D"font-family:sans-serif"><div style=3D"white-space:pre-wrap"=
><div dir=3D"auto">Hi,
</div><div dir=3D"auto">
</div><div dir=3D"auto">The proposed charter (at the link below) will be =
reviewed by the IESG at the Dec 1 Telechat. If people have further commen=
ts, please make them as soon as possible.
</div><div dir=3D"auto">
</div><div dir=3D"auto">Thanks!
</div><div dir=3D"auto">
</div><div dir=3D"auto">Ben.
</div><div dir=3D"auto">
</div><div dir=3D"auto">On 8 Nov 2016, at 8:05, Ben Campbell wrote:
</div><div dir=3D"auto">
</div></div>
<blockquote style=3D"border-left:2px solid #777; color:#777; margin:0 0 5=
px; padding-left:5px"><div id=3D"07E4E0D5-73EC-47BF-84B4-3EABD719FFF2">

<div style=3D"font-family:sans-serif"><div style=3D"white-space:pre-wrap"=
><div dir=3D"auto">The proposed new SIPCORE charter text is now available=
 at <a href=3D"https://datatracker.ietf.org/doc/charter-ietf-sipcore/" st=
yle=3D"color:#3983C4">https://datatracker.ietf.org/doc/charter-ietf-sipco=
re/</a> . It's identical to the text from Adam's email, other than some p=
otential formatting differences.
</div><div dir=3D"auto">
</div><div dir=3D"auto">Thanks!
</div><div dir=3D"auto">
</div><div dir=3D"auto">Ben.
</div><div dir=3D"auto">
</div><div dir=3D"auto">On 4 Nov 2016, at 16:28, Adam Roach wrote:
</div><div dir=3D"auto">
</div></div>
<blockquote style=3D"border-left:2px solid #777; color:#777; margin:0 0 5=
px; padding-left:5px"><div id=3D"1600C0CA-5C8A-43AB-B0D1-5C23687E04E7">
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <p>[as SIPCORE chair]</p>
    <p>Back when we transitioned from the SIPPING/SIP working group
      structure to SIPCORE, we put relatively tight limits on the
      SIPCORE charter. This was a recognition of the fact that the
      volume of SIP-related work at the time was too large for any one
      working group to reasonably handle. Instead, the intention was
      that the DISPATCH model of creating small, short-lived working
      groups for specific topics would be a better way to manage the
      relatively heavy workload.</p>
    <p>Fast forward seven years to today. SIP is a much more mature
      protocol, and the inflow of new work is significantly smaller than
      it was back then. At the same time, we have had several small,
      self-contained work items show up in DISPATCH that were deemed too
      small to create a new working group for, but precluded from being
      dispatched to SIPCORE by its charter. This has led to unnecessary
      delays as we determined the best disposition for these items.<br>
    </p>
    <p>We, the SIPCORE chairs and area director, seek to fix that. We
      would like to amend the SIPCORE charter to allow it to take on
      small, self-contained protocol extensions in addition to changes
      to the core SIP protocol. This is a relatively minor update to the
      existing charter, which expands the scope as described above, and
      adds back in two of the architectural principles from the SIP
      charter which are applicable only to protocol extensions.</p>
    <p>Please provide feedback on the proposed charter by sending email
      to the SIPCORE mailing list. We will also discuss this briefly
      during the DISPATCH session in Seoul.</p>
    <p>The proposed charter text follows.</p>
    <p><br>
    </p>
    <p>
      <blockquote type=3D"cite">
        <meta http-equiv=3D"content-type" content=3D"text/html;
          charset=3Dutf-8">
        <div id=3D"magicdomid2" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The=

            Session Initiation Protocol Core (SIPCore) working group is</=
span></div>
        <div id=3D"magicdomid3" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">cha=
rtered
            to maintain and continue the development of the SIP
            protocol,</span></div>
        <div id=3D"magicdomid4" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">cur=
rently
            defined as proposed standard RFCs 3261, 3262, 3263, 3264,
            and</span></div>
        <div id=3D"magicdomid5" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">666=
5.</span></div>
        <div id=3D"magicdomid6" class=3D""><br>
        </div>
        <div id=3D"magicdomid7" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The=

            SIPCore working group will concentrate on specifications
            that update</span></div>
        <div id=3D"magicdomid8" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">or
            replace the core SIP specifications named above as well as</s=
pan></div>
        <div id=3D"magicdomid9" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">spe=
cifications
            pertaining to small, self-contained SIP protocol</span></div>=

        <div id=3D"magicdomid10" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">ext=
ensions.=C2=A0
            The process and requirements for new SIP extensions are</span=
></div>
        <div id=3D"magicdomid11" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">doc=
umented
            in RFC5727, "Change Process for the Session Initiation</span>=
</div>
        <div id=3D"magicdomid12" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Pro=
tocol".</span></div>
        <div id=3D"magicdomid13" class=3D""><br>
        </div>
        <div id=3D"magicdomid14" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Thr=
oughout
            its work, the group will strive to maintain the basic model</=
span></div>
        <div id=3D"magicdomid15" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">and=

            architecture defined by SIP. In particular:</span></div>
        <div id=3D"magicdomid16" class=3D""><br>
        </div>
        <div id=3D"magicdomid17" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">1.
            Services and features are provided end-to-end whenever
            possible.</span></div>
        <div id=3D"magicdomid18" class=3D""><br>
        </div>
        <div id=3D"magicdomid19" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">2.
            Reuse of existing Internet protocols and architectures and</s=
pan></div>
        <div id=3D"magicdomid20" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">=C2=
=A0=C2=A0
            integrating with other Internet applications is crucial.</spa=
n></div>
        <div id=3D"magicdomid21" class=3D""><br>
        </div>
        <div id=3D"magicdomid22" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">3.
            Standards-track extensions and new features must be
            generally</span></div>
        <div id=3D"magicdomid23" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">=C2=
=A0=C2=A0
            applicable, and not applicable only to a specific set of
            session</span></div>
        <div id=3D"magicdomid24" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">=C2=
=A0=C2=A0
            types.</span></div>
        <div id=3D"magicdomid25" class=3D""><br>
        </div>
        <div id=3D"magicdomid26" class=3D""><span
            class=3D"author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">4.
            Simpler solutions that solve a given problem should be
            favored</span></div>
        <div id=3D"magicdomid27" class=3D""><span
            class=3D"author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">=C2=A0=
=C2=A0=C2=A0
            over more complex solutions.</span></div>
        <div id=3D"magicdomid28" class=3D""><br>
        </div>
        <div id=3D"magicdomid29" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The=

            primary source of change requirements to the core SIP
            specifications</span></div>
        <div id=3D"magicdomid30" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">(en=
umerated
            above) will be a) interoperability problems that stem from</s=
pan></div>
        <div id=3D"magicdomid31" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">amb=
iguous,
            or under-defined specification, and b) requirements from</spa=
n></div>
        <div id=3D"magicdomid32" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">oth=
er
            working groups in the ART Area. The primary source of new
            protocol</span></div>
        <div id=3D"magicdomid33" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">ext=
ensions
            is the DISPATCH working group, which will generally make the<=
/span></div>
        <div id=3D"magicdomid34" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">det=
ermination
            regarding whether new SIP-related work warrants a new</span><=
/div>
        <div id=3D"magicdomid35" class=3D""><span
            class=3D"author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">wor=
king
            group or belongs in an existing one.</span></div>
      </blockquote>
    </p>
    <p><br>
    </p>
    <p>For ease of reference, the existing SIPCORE charter is at
      <a class=3D"moz-txt-link-freetext" href=3D"https://datatracker.ietf=
=2Eorg/doc/charter-ietf-sipcore/">https://datatracker.ietf.org/doc/charte=
r-ietf-sipcore/</a></p>
    <p>Thanks!<br>
    </p>
    <p>/a<br>
    </p>
  </div></div></blockquote>
<div style=3D"white-space:pre-wrap"><div dir=3D"auto">
</div><blockquote style=3D"border-left:2px solid #777; color:#777; margin=
:0 0 5px; padding-left:5px"><div dir=3D"auto">___________________________=
____________________
</div><div dir=3D"auto">dispatch mailing list
</div><div dir=3D"auto">dispatch@ietf.org
</div><div dir=3D"auto"><a href=3D"https://www.ietf.org/mailman/listinfo/=
dispatch" style=3D"color:#777">https://www.ietf.org/mailman/listinfo/disp=
atch</a>
</div></blockquote></div>
</div></div></blockquote>
<div style=3D"white-space:pre-wrap"><div dir=3D"auto">
</div><blockquote style=3D"border-left:2px solid #777; color:#777; margin=
:0 0 5px; padding-left:5px"><div dir=3D"auto">___________________________=
____________________
</div><div dir=3D"auto">sipcore mailing list
</div><div dir=3D"auto">sipcore@ietf.org
</div><div dir=3D"auto"><a href=3D"https://www.ietf.org/mailman/listinfo/=
sipcore" style=3D"color:#777">https://www.ietf.org/mailman/listinfo/sipco=
re</a>
</div></blockquote></div>
</div>
</body>
</html>

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From nobody Tue Nov 15 18:56:48 2016
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From: worley@ariadne.com (Dale R. Worley)
To: roman@telurix.com, oej@edvina.net, sipcore@ietf.org
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/jUGb65v8vC6mJdy8sn_1SWG-2SU>
Subject: Re: [sipcore] Source address selection in SIP
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worley@ariadne.com (Dale R. Worley) writes:
> Part of the trouble is that you can send to a single destination address
> through more than one interface.  I wonder if the address selection
> system provides for a way for an application to say "send all global
> traffic through interface X"?

Looking at RFC 6724, it looks like the "policy table" (section 2.1) can
be used to instruct the implementation to send SIP traffic destined for
certain prefixes using certain source addresses (which map to
interfaces).  And it looks like the glibc version of getaddrinfo() uses
a file /etc/gai.conf to provide policy table services.

Dale


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References: <87fumsz8az.fsf@hobgoblin.ariadne.com> <87d1hwyvx2.fsf@hobgoblin.ariadne.com>
From: Roman Shpount <roman@telurix.com>
Date: Tue, 15 Nov 2016 22:17:23 -0500
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To: "Dale R. Worley" <worley@ariadne.com>
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Cc: SIPCORE <sipcore@ietf.org>, "Olle E. Johansson" <oej@edvina.net>
Subject: Re: [sipcore] Source address selection in SIP
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--001a114a89cc07a389054162835c
Content-Type: text/plain; charset=UTF-8

Typically doing a UDP connect with unbound local socket and then looking at
the socket address works quite well for determining local policy.
Unfortunately local policy can lie. For instance end point can be behind
the VPN that sets the default route to something (office network) that
blocks SIP. There is an option to be aggressive and send connectivity
checks from all available interfaces or, alternatively, implementation can
follow the local policy and fail to connect. The implementation needs to
decide what is more important -- following the policy or establishing the
connection.

Regards,
_____________
Roman Shpount

On Tue, Nov 15, 2016 at 9:56 PM, Dale R. Worley <worley@ariadne.com> wrote:

> worley@ariadne.com (Dale R. Worley) writes:
> > Part of the trouble is that you can send to a single destination address
> > through more than one interface.  I wonder if the address selection
> > system provides for a way for an application to say "send all global
> > traffic through interface X"?
>
> Looking at RFC 6724, it looks like the "policy table" (section 2.1) can
> be used to instruct the implementation to send SIP traffic destined for
> certain prefixes using certain source addresses (which map to
> interfaces).  And it looks like the glibc version of getaddrinfo() uses
> a file /etc/gai.conf to provide policy table services.
>
> Dale
>

--001a114a89cc07a389054162835c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Typically doing a UDP connect with unbound local socket an=
d then looking at the socket address works quite well for determining local=
 policy. Unfortunately local policy can lie. For instance end point can be =
behind the VPN that sets the default route to something (office network) th=
at blocks SIP. There is an option to be aggressive and send connectivity ch=
ecks from all available interfaces or, alternatively, implementation can fo=
llow the local policy and fail to connect. The implementation needs to deci=
de what is more important -- following the policy or establishing the conne=
ction.<div><br></div><div>Regards,<div class=3D"gmail_extra"><div><div clas=
s=3D"gmail_signature" data-smartmail=3D"gmail_signature">_____________<br>R=
oman Shpount</div></div>
<br><div class=3D"gmail_quote">On Tue, Nov 15, 2016 at 9:56 PM, Dale R. Wor=
ley <span dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com" target=3D"_=
blank">worley@ariadne.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><a href=3D"mailto:worley@ariadne.com">worley@ariadne.com</a> (Dale R=
. Worley) writes:<br>
&gt; Part of the trouble is that you can send to a single destination addre=
ss<br>
&gt; through more than one interface.=C2=A0 I wonder if the address selecti=
on<br>
&gt; system provides for a way for an application to say &quot;send all glo=
bal<br>
&gt; traffic through interface X&quot;?<br>
<br>
Looking at RFC 6724, it looks like the &quot;policy table&quot; (section 2.=
1) can<br>
be used to instruct the implementation to send SIP traffic destined for<br>
certain prefixes using certain source addresses (which map to<br>
interfaces).=C2=A0 And it looks like the glibc version of getaddrinfo() use=
s<br>
a file /etc/gai.conf to provide policy table services.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Dale<br>
</font></span></blockquote></div><br></div></div></div>

--001a114a89cc07a389054162835c--


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Subject: Re: [sipcore] Source address selection in SIP
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Roman Shpount <roman@telurix.com> writes:
> One issue we saw was clients using VPN, with VPN taking them to a business
> network which blocks SIP.
>
> This is why we added addresses enumeration and STUN checks similar to ICE
> to verify which address actually has internet access.

Interesting... that's not fixable as an ordinary source address
selection problem.

It's really a sort of Happy Eyeballs problem, where one interface has
global SIP access and one doesn't but the client has no way to determine
the right interface without trial because both interfaces have outgoing
routers.

We're working on a "Happy Eyeballs for SIP" draft, but the current
version explicitly doesn't address this sort of problem:

   A dual-stack host normally has one physical interface, and all
   network access is done via IPv4 and IPv6 addresses assigned to that
   interface.  However, a dual-stack host might have additional physical
   interfaces or additional logical interfaces (e.g., because of a VPN).
   Additional Happy EarBalls considerations for optimal operation with
   additional physical or logical interfaces is for further study and is
   outside the scope of this document.

Part of the trouble is that you can send to a single destination address
through more than one interface.  I wonder if the address selection
system provides for a way for an application to say "send all global
traffic through interface X"?

Perhaps we should add some guidance in this area, either in this cycle
of work or in a future one.

Dale


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Subject: Re: [sipcore] Source address selection in SIP
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Roman Shpount <roman@telurix.com> writes:
> There is an option to be aggressive and send connectivity
> checks from all available interfaces or, alternatively, implementation can
> follow the local policy and fail to connect. The implementation needs to
> decide what is more important -- following the policy or establishing the
> connection.

Within the conceptual framework of Happy Eyeballs, it's clear that the
optimal strategy is to consider the alternative source interfaces (or
perhaps source addresses) as connectivity alternatives in much the same
way that destination addresses are.  Then, use the local policy to
determine the order in which to check the possibilties, but (in general)
use the most policy-preferred alternative that actually works.  The
details to do this well are probably messy, though.

As we expand the scope of the Happy Eyeballs draft, please remind us to
address this issue at some point.

Dale


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Subject: [sipcore] Discussion venue for 666 (was SHAKEN but not STIRred means 666 can cause collateral damage)
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[as SIPCORE chair]

On 11/17/16 10:14, Brian Rosen wrote:
> I’ve sent this to stir, even though it has not been decided where the 666 draft will land

My understanding coming out of DISPATCH is that 666 is appropriate for 
SIPCORE, assuming the new charter is approved by the IESG. I expect the 
proposed charter changes to be uncontroversial.

Based on the foregoing, we (the SIPCORE chairs) think it would be 
appropriate to hold general discussion of both of the
draft-schulzrinne-dispatch-callinfo-spam and 
draft-schulzrinne-dispatch-status-unwanted documents on the SIPCORE list.

I've copied DISPATCH and SIPCORE on this message, and set followups to 
SIPCORE.

/a


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Subject: [sipcore] Fwd: Re: [dispatch] Discussion venue for 666 (was SHAKEN but not STIRred means 666 can cause collateral damage)
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--------------BE1EC0B521888C9EC1D72E8B
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Reply-all betrayed me. Forwarding for completeness.

Sorry for the extra noise.



-------- Forwarded Message --------
Subject: 	Re: [dispatch] Discussion venue for 666 (was SHAKEN but not 
STIRred means 666 can cause collateral damage)
Date: 	Thu, 17 Nov 2016 17:51:52 +0900
From: 	Robert Sparks <rjsparks@nostrum.com>
To: 	dispatch@ietf.org



[as a STIR co-chair]

I agree with this.

Later in the discussion, I expect we will recommend splitting the
passport extension work out and running that through STIR, but that
doesn't need to happen immediately.

If we discover that there's frequent unnatural tension in needing to
talk about stir things while figuring out the SIP mechanics in these
drafts, we can revisit.

RjS


On 11/17/16 5:27 PM, Adam Roach wrote:
> [as SIPCORE chair]
>
> On 11/17/16 10:14, Brian Rosen wrote:
>> I’ve sent this to stir, even though it has not been decided where the
>> 666 draft will land
>
> My understanding coming out of DISPATCH is that 666 is appropriate for
> SIPCORE, assuming the new charter is approved by the IESG. I expect
> the proposed charter changes to be uncontroversial.
>
> Based on the foregoing, we (the SIPCORE chairs) think it would be
> appropriate to hold general discussion of both of the
> draft-schulzrinne-dispatch-callinfo-spam and
> draft-schulzrinne-dispatch-status-unwanted documents on the SIPCORE list.
>
> I've copied DISPATCH and SIPCORE on this message, and set followups to
> SIPCORE.
>
> /a
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


--------------BE1EC0B521888C9EC1D72E8B
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<html>
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    <meta http-equiv="content-type" content="text/html; charset=utf-8">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <p>Reply-all betrayed me. Forwarding for completeness.</p>
    <p>Sorry for the extra noise.<br>
    </p>
    <div class="moz-forward-container"><br>
      <br>
      -------- Forwarded Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>Re: [dispatch] Discussion venue for 666 (was SHAKEN but
              not STIRred means 666 can cause collateral damage)</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Thu, 17 Nov 2016 17:51:52 +0900</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td>Robert Sparks <a class="moz-txt-link-rfc2396E" href="mailto:rjsparks@nostrum.com">&lt;rjsparks@nostrum.com&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>[as a STIR co-chair]

I agree with this.

Later in the discussion, I expect we will recommend splitting the 
passport extension work out and running that through STIR, but that 
doesn't need to happen immediately.

If we discover that there's frequent unnatural tension in needing to 
talk about stir things while figuring out the SIP mechanics in these 
drafts, we can revisit.

RjS


On 11/17/16 5:27 PM, Adam Roach wrote:
&gt; [as SIPCORE chair]
&gt;
&gt; On 11/17/16 10:14, Brian Rosen wrote:
&gt;&gt; I’ve sent this to stir, even though it has not been decided where the 
&gt;&gt; 666 draft will land
&gt;
&gt; My understanding coming out of DISPATCH is that 666 is appropriate for 
&gt; SIPCORE, assuming the new charter is approved by the IESG. I expect 
&gt; the proposed charter changes to be uncontroversial.
&gt;
&gt; Based on the foregoing, we (the SIPCORE chairs) think it would be 
&gt; appropriate to hold general discussion of both of the
&gt; draft-schulzrinne-dispatch-callinfo-spam and 
&gt; draft-schulzrinne-dispatch-status-unwanted documents on the SIPCORE list.
&gt;
&gt; I've copied DISPATCH and SIPCORE on this message, and set followups to 
&gt; SIPCORE.
&gt;
&gt; /a
&gt;
&gt; _______________________________________________
&gt; dispatch mailing list
&gt; <a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a>
&gt; <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a>

</pre>
    </div>
  </body>
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--------------BE1EC0B521888C9EC1D72E8B--


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From: "Petersen, Jan 1. (Nokia - DE/Munich)" <jan.1.petersen@nokia.com>
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Subject: [sipcore] Question regarding RFC 3262
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Hello SIP Experts and RFC 3262 authors,


I am wondering how the race condition between provisional and final respons=
es to the INVITE request that may occur based on this rfc 3262 is to be sol=
ved.

I have checked Errata and subsequent rfcs (such as RFC 5407), but was not a=
ble to find any solution to this specific race condition.


The issue occurs if the UAS is responding with an (subsequent) provisional =
response to the INVITE request, using 100rel, while the shortly afterwards =
the UAS is actually sending the final response to the INVITE, before the PR=
ACK to the provisional response has been received by the UAS:

The UAS MAY send a final response to the initial request before
having received PRACKs for all unacknowledged reliable provisional
responses, unless the final response is 2xx and any of the
unacknowledged reliable provisional responses contained a session
description.  In that case, it MUST NOT send a final response until
those provisional responses are acknowledged.  If the UAS does send a
final response when reliable responses are still unacknowledged, it
SHOULD NOT continue to retransmit the unacknowledged reliable
provisional responses, but it MUST be prepared to process PRACK
requests for those outstanding responses.


So as a result, the UAC may receive the provisional response to the INVITE,=
 immediately followed by the final response 2xx to the INVITE. However, the=
 UAC may still send the PRACK for the provisional response here, but the UA=
S has already terminated the transaction when this "late" PRACK is received=
. Consequently the UAS responds to this "late" PRACK with a 481 Call/Transa=
ction Does Not Exist, which in turn causes the UAC to terminate the already=
 established and ongoing SIP session.

Can you please clarify how SIP UAs (UAC and UAS and even SIP Proxies) shoul=
d handle and solve such a race condition? Especially, is the PRACK transact=
ion directly affecting the corresponding SIP dialog, such that a late negat=
ive response to the PRACK received on UAC side (after the 2xx final respons=
e to the INVITE has been received already before) is justifying to execute =
the release of the already successfully established session by the UAC?

What exactly is meant by "... but[the UAS] MUST be prepared to process PRAC=
K requests for those outstanding responses"? Does "process" refers to respo=
nd with a 2xx, or to provide any final response, i.e. even a negative respo=
nse - compared to just dropping it without any response?


Many thanks in advance!

Jan Petersen, NOKIA

email: jan.1.petersen@nokia.com<mailto:jan.1.petersen@nokia.com>





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<font face=3D"Calibri" size=3D"2"><span style=3D"font-size:11pt;">
<div>Hello SIP Experts and RFC 3262 authors,</div>
<div><font color=3D"#1F497D">&nbsp;</font></div>
<div>&nbsp;</div>
<div>I am wondering how the race condition between provisional and final re=
sponses to the INVITE request that may occur based on this rfc 3262 is to b=
e solved.</div>
<div>&nbsp;</div>
<div>I have checked Errata and subsequent rfcs (such as RFC 5407), but was =
not able to find any solution to this specific race condition.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>The issue occurs if the UAS is responding with an (subsequent) provisi=
onal response to the INVITE request, using 100rel, while the shortly afterw=
ards the UAS is actually sending the final response to the INVITE, before t=
he PRACK to the provisional response
has been received by the UAS:</div>
<div>&nbsp;</div>
<div><font face=3D"Courier New" color=3D"#002060">The UAS MAY send a final =
response to the initial request before</font></div>
<div><font face=3D"Courier New" color=3D"#002060">having received PRACKs fo=
r all unacknowledged reliable provisional</font></div>
<div><font face=3D"Courier New" color=3D"#002060">responses, unless the fin=
al response is 2xx and any of the</font></div>
<div><font face=3D"Courier New" color=3D"#002060">unacknowledged reliable p=
rovisional responses contained a session</font></div>
<div><font face=3D"Courier New" color=3D"#002060">description.&nbsp; In tha=
t case, it MUST NOT send a final response until</font></div>
<div><font face=3D"Courier New" color=3D"#002060">those provisional respons=
es are acknowledged.&nbsp; If the UAS does send a</font></div>
<div><font face=3D"Courier New" color=3D"#002060">final response when relia=
ble responses are still unacknowledged, it</font></div>
<div><font face=3D"Courier New" color=3D"#002060">SHOULD NOT continue to re=
transmit the unacknowledged reliable</font></div>
<div><font face=3D"Courier New" color=3D"#002060">provisional responses, bu=
t it MUST be prepared to process PRACK</font></div>
<div><font face=3D"Courier New" color=3D"#002060">requests for those outsta=
nding responses.</font></div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>So as a result, the UAC may receive the provisional response to the IN=
VITE, immediately followed by the final response 2xx to the INVITE. However=
, the UAC may still send the PRACK for the provisional response here, but t=
he UAS has already terminated the
transaction when this &#8220;late&#8221; PRACK is received. Consequently th=
e UAS responds to this &#8220;late&#8221; PRACK with a 481 Call/Transaction=
 Does Not Exist, which in turn causes the UAC to terminate the already esta=
blished and ongoing SIP session.</div>
<div>&nbsp;</div>
<div>Can you please clarify how SIP UAs (UAC and UAS and even SIP Proxies) =
should handle and solve such a race condition? Especially, is the PRACK tra=
nsaction directly affecting the corresponding SIP dialog, such that a late =
negative response to the PRACK received
on UAC side (after the 2xx final response to the INVITE has been received a=
lready before) is justifying to execute the release of the already successf=
ully established session by the UAC?</div>
<div>&nbsp;</div>
<div>What exactly is meant by <font color=3D"#002060">&#8220;</font><font f=
ace=3D"Courier New" color=3D"#C00000">&#8230;</font><font face=3D"Courier N=
ew" color=3D"#002060"> </font><font face=3D"Courier New" color=3D"#C00000">=
but[the UAS] MUST be prepared to process PRACK requests for
those outstanding responses</font>&#8221;? Does &#8220;<font color=3D"#C000=
00">process</font>&#8221; refers to respond with a 2xx, or to provide any f=
inal response, i.e. even a negative response &#8211; compared to just dropp=
ing it without any response?</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>Many thanks in advance!</div>
<div>&nbsp;</div>
<div>Jan Petersen, NOKIA<br>

</div>
<div><font face=3D"Nokia Pure Headline" size=3D"2"><span style=3D"font-size=
:9pt;">email: <a href=3D"mailto:jan.1.petersen@nokia.com"><font color=3D"#0=
563C1"><u>jan.1.petersen@nokia.com</u></font></a>
<br>

&nbsp;</span></font></div>
<div style=3D"margin-bottom:5pt;"><font color=3D"#7F7F7F">&nbsp;</font></di=
v>
<div>&nbsp;</div>
<div>&nbsp;</div>
</span></font>
</body>
</html>

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Subject: Re: [sipcore] Question regarding RFC 3262
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On 11/21/16 12:38 PM, Petersen, Jan 1. (Nokia - DE/Munich) wrote:
> Hello SIP Experts and RFC 3262 authors,
> I am wondering how the race condition between provisional and final 
> responses to the INVITE request that may occur based on this rfc 3262 
> is to be solved.
> I have checked Errata and subsequent rfcs (such as RFC 5407), but was 
> not able to find any solution to this specific race condition.
> The issue occurs if the UAS is responding with an (subsequent) 
> provisional response to the INVITE request, using 100rel, while the 
> shortly afterwards the UAS is actually sending the final response to 
> the INVITE, before the PRACK to the provisional response has been 
> received by the UAS:
> The UAS MAY send a final response to the initial request before
> having received PRACKs for all unacknowledged reliable provisional
> responses, unless the final response is 2xx and any of the
> unacknowledged reliable provisional responses contained a session
> description.  In that case, it MUST NOT send a final response until
> those provisional responses are acknowledged.  If the UAS does send a
> final response when reliable responses are still unacknowledged, it
> SHOULD NOT continue to retransmit the unacknowledged reliable
> provisional responses, but it MUST be prepared to process PRACK
> requests for those outstanding responses.
> So as a result, the UAC may receive the provisional response to the 
> INVITE, immediately followed by the final response 2xx to the INVITE. 
> However, the UAC may still send the PRACK for the provisional response 
> here, but the UAS has already terminated the transaction when this 
> late PRACK is received. Consequently the UAS responds to this late 
> PRACK with a 481 Call/Transaction Does Not Exist, which in turn causes 
> the UAC to terminate the already established and ongoing SIP session.
> Can you please clarify how SIP UAs (UAC and UAS and even SIP Proxies) 
> should handle and solve such a race condition? Especially, is the 
> PRACK transaction directly affecting the corresponding SIP dialog, 
> such that a late negative response to the PRACK received on UAC side 
> (after the 2xx final response to the INVITE has been received already 
> before) is justifying to execute the release of the already 
> successfully established session by the UAC?
> What exactly is meant by but[the UAS] MUST be prepared to process 
> PRACK requests for those outstanding responses? Does process refers 
> to respond with a 2xx, or to provide any final response, i.e. even a 
> negative response  compared to just dropping it without any response?
It means that it must keep enough state to not return a 481. (A UAS that 
481s such a PRACK has not followed the requirement). The expectation is 
that the PRACK would get a 200. The "In that case" sentence defends 
against the real problems with the race condition you're looking at. The 
remaining situations where there wouldn't be a 200 to the PRACK are 
places where context requires another response (like a 500 or a failure 
due to use or attempted use of an extension).

> Many thanks in advance!
> Jan Petersen, NOKIA
> email: _jan.1.petersen@nokia.com_ <mailto:jan.1.petersen@nokia.com>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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    <div class="moz-cite-prefix">On 11/21/16 12:38 PM, Petersen, Jan 1.
      (Nokia - DE/Munich) wrote:<br>
    </div>
    <blockquote
cite="mid:HE1PR07MB17223A9DBB88CE3942EBF3DF8DB50@HE1PR07MB1722.eurprd07.prod.outlook.com"
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          <div>Hello SIP Experts and RFC 3262 authors,</div>
          <div><font color="#1F497D"></font></div>
          <div></div>
          <div>I am wondering how the race condition between provisional
            and final responses to the INVITE request that may occur
            based on this rfc 3262 is to be solved.</div>
          <div></div>
          <div>I have checked Errata and subsequent rfcs (such as RFC
            5407), but was not able to find any solution to this
            specific race condition.</div>
          <div></div>
          <div></div>
          <div>The issue occurs if the UAS is responding with an
            (subsequent) provisional response to the INVITE request,
            using 100rel, while the shortly afterwards the UAS is
            actually sending the final response to the INVITE, before
            the PRACK to the provisional response
            has been received by the UAS:</div>
          <div></div>
          <div><font face="Courier New" color="#002060">The UAS MAY send
              a final response to the initial request before</font></div>
          <div><font face="Courier New" color="#002060">having received
              PRACKs for all unacknowledged reliable provisional</font></div>
          <div><font face="Courier New" color="#002060">responses,
              unless the final response is 2xx and any of the</font></div>
          <div><font face="Courier New" color="#002060">unacknowledged
              reliable provisional responses contained a session</font></div>
          <div><font face="Courier New" color="#002060">description. In
              that case, it MUST NOT send a final response until</font></div>
          <div><font face="Courier New" color="#002060">those
              provisional responses are acknowledged. If the UAS does
              send a</font></div>
          <div><font face="Courier New" color="#002060">final response
              when reliable responses are still unacknowledged, it</font></div>
          <div><font face="Courier New" color="#002060">SHOULD NOT
              continue to retransmit the unacknowledged reliable</font></div>
          <div><font face="Courier New" color="#002060">provisional
              responses, but it MUST be prepared to process PRACK</font></div>
          <div><font face="Courier New" color="#002060">requests for
              those outstanding responses.</font></div>
          <div></div>
          <div></div>
          <div>So as a result, the UAC may receive the provisional
            response to the INVITE, immediately followed by the final
            response 2xx to the INVITE. However, the UAC may still send
            the PRACK for the provisional response here, but the UAS has
            already terminated the
            transaction when this late PRACK is received. Consequently
            the UAS responds to this late PRACK with a 481
            Call/Transaction Does Not Exist, which in turn causes the
            UAC to terminate the already established and ongoing SIP
            session.</div>
          <div></div>
          <div>Can you please clarify how SIP UAs (UAC and UAS and even
            SIP Proxies) should handle and solve such a race condition?
            Especially, is the PRACK transaction directly affecting the
            corresponding SIP dialog, such that a late negative response
            to the PRACK received
            on UAC side (after the 2xx final response to the INVITE has
            been received already before) is justifying to execute the
            release of the already successfully established session by
            the UAC?</div>
          <div></div>
          <div>What exactly is meant by <font color="#002060"></font><font
              face="Courier New" color="#C00000"></font><font
              face="Courier New" color="#002060"> </font><font
              face="Courier New" color="#C00000">but[the UAS] MUST be
              prepared to process PRACK requests for
              those outstanding responses</font>? Does <font
              color="#C00000">process</font> refers to respond with a
            2xx, or to provide any final response, i.e. even a negative
            response  compared to just dropping it without any
            response?</div>
        </span></font></blockquote>
    <font size="2"><font face="Calibri">It means that it must keep
        enough state to not return a 481. (A UAS that 481s such a PRACK
        has not followed the requirement). The expectation is that the
        PRACK would get a 200. The "In that case" sentence defends
        against the real problems with the race condition you're looking
        at. The remaining situations where there wouldn't be a 200 to
        the PRACK are places where context requires another response
        (like a 500 or a failure due to use or attempted use of an
        extension).<br>
        <br>
      </font></font>
    <blockquote
cite="mid:HE1PR07MB17223A9DBB88CE3942EBF3DF8DB50@HE1PR07MB1722.eurprd07.prod.outlook.com"
      type="cite"><font face="Calibri" size="2"><span
          style="font-size:11pt;">
          <div></div>
          <div></div>
          <div>Many thanks in advance!</div>
          <div></div>
          <div>Jan Petersen, NOKIA<br>
          </div>
          <div><font face="Nokia Pure Headline" size="2"><span
                style="font-size:9pt;">email: <a moz-do-not-send="true"
                  href="mailto:jan.1.petersen@nokia.com"><font
                    color="#0563C1"><u>jan.1.petersen@nokia.com</u></font></a>
                <br>
                </span></font></div>
          <div style="margin-bottom:5pt;"><font color="#7F7F7F"></font></div>
          <div></div>
          <div></div>
        </span></font>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
sipcore mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.org/mailman/listinfo/sipcore</a>
</pre>
    </blockquote>
    <br>
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This change seems like a good idea to me.=20


> On Nov 4, 2016, at 2:28 PM, Adam Roach <adam@nostrum.com> wrote:
>=20
> [as SIPCORE chair]
>=20
> Back when we transitioned from the SIPPING/SIP working group structure =
to SIPCORE, we put relatively tight limits on the SIPCORE charter. This =
was a recognition of the fact that the volume of SIP-related work at the =
time was too large for any one working group to reasonably handle. =
Instead, the intention was that the DISPATCH model of creating small, =
short-lived working groups for specific topics would be a better way to =
manage the relatively heavy workload.
>=20
> Fast forward seven years to today. SIP is a much more mature protocol, =
and the inflow of new work is significantly smaller than it was back =
then. At the same time, we have had several small, self-contained work =
items show up in DISPATCH that were deemed too small to create a new =
working group for, but precluded from being dispatched to SIPCORE by its =
charter. This has led to unnecessary delays as we determined the best =
disposition for these items.
> We, the SIPCORE chairs and area director, seek to fix that. We would =
like to amend the SIPCORE charter to allow it to take on small, =
self-contained protocol extensions in addition to changes to the core =
SIP protocol. This is a relatively minor update to the existing charter, =
which expands the scope as described above, and adds back in two of the =
architectural principles from the SIP charter which are applicable only =
to protocol extensions.
>=20
> Please provide feedback on the proposed charter by sending email to =
the SIPCORE mailing list. We will also discuss this briefly during the =
DISPATCH session in Seoul.
>=20
> The proposed charter text follows.
>=20
>=20
>=20
>> The Session Initiation Protocol Core (SIPCore) working group is
>> chartered to maintain and continue the development of the SIP =
protocol,
>> currently defined as proposed standard RFCs 3261, 3262, 3263, 3264, =
and
>> 6665.
>>=20
>> The SIPCore working group will concentrate on specifications that =
update
>> or replace the core SIP specifications named above as well as
>> specifications pertaining to small, self-contained SIP protocol
>> extensions.  The process and requirements for new SIP extensions are
>> documented in RFC5727, "Change Process for the Session Initiation
>> Protocol".
>>=20
>> Throughout its work, the group will strive to maintain the basic =
model
>> and architecture defined by SIP. In particular:
>>=20
>> 1. Services and features are provided end-to-end whenever possible.
>>=20
>> 2. Reuse of existing Internet protocols and architectures and
>>    integrating with other Internet applications is crucial.
>>=20
>> 3. Standards-track extensions and new features must be generally
>>    applicable, and not applicable only to a specific set of session
>>    types.
>>=20
>> 4. Simpler solutions that solve a given problem should be favored
>>     over more complex solutions.
>>=20
>> The primary source of change requirements to the core SIP =
specifications
>> (enumerated above) will be a) interoperability problems that stem =
from
>> ambiguous, or under-defined specification, and b) requirements from
>> other working groups in the ART Area. The primary source of new =
protocol
>> extensions is the DISPATCH working group, which will generally make =
the
>> determination regarding whether new SIP-related work warrants a new
>> working group or belongs in an existing one.
>=20
>=20
> For ease of reference, the existing SIPCORE charter is =
athttps://datatracker.ietf.org/doc/charter-ietf-sipcore/ =
<https://datatracker.ietf.org/doc/charter-ietf-sipcore/>
> Thanks!
> /a
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div class=""><br class=""></div>This change seems like a good idea to me.&nbsp;<div class=""><br class=""></div><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Nov 4, 2016, at 2:28 PM, Adam Roach &lt;<a href="mailto:adam@nostrum.com" class="">adam@nostrum.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><div class="">
  

    <meta http-equiv="content-type" content="text/html; charset=utf-8" class="">
  
  <div bgcolor="#FFFFFF" text="#000000" class=""><p class="">[as SIPCORE chair]</p><p class="">Back when we transitioned from the SIPPING/SIP working group
      structure to SIPCORE, we put relatively tight limits on the
      SIPCORE charter. This was a recognition of the fact that the
      volume of SIP-related work at the time was too large for any one
      working group to reasonably handle. Instead, the intention was
      that the DISPATCH model of creating small, short-lived working
      groups for specific topics would be a better way to manage the
      relatively heavy workload.</p><p class="">Fast forward seven years to today. SIP is a much more mature
      protocol, and the inflow of new work is significantly smaller than
      it was back then. At the same time, we have had several small,
      self-contained work items show up in DISPATCH that were deemed too
      small to create a new working group for, but precluded from being
      dispatched to SIPCORE by its charter. This has led to unnecessary
      delays as we determined the best disposition for these items.<br class="">
    </p><p class="">We, the SIPCORE chairs and area director, seek to fix that. We
      would like to amend the SIPCORE charter to allow it to take on
      small, self-contained protocol extensions in addition to changes
      to the core SIP protocol. This is a relatively minor update to the
      existing charter, which expands the scope as described above, and
      adds back in two of the architectural principles from the SIP
      charter which are applicable only to protocol extensions.</p><p class="">Please provide feedback on the proposed charter by sending email
      to the SIPCORE mailing list. We will also discuss this briefly
      during the DISPATCH session in Seoul.</p><p class="">The proposed charter text follows.</p><p class=""><br class="">
    </p><div class="">
      <br class="webkit-block-placeholder"></div><blockquote type="cite" class="">
        <meta http-equiv="content-type" content="text/html;
          charset=utf-8" class="">
        <div id="magicdomid2" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The
            Session Initiation Protocol Core (SIPCore) working group is</span></div>
        <div id="magicdomid3" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">chartered
            to maintain and continue the development of the SIP
            protocol,</span></div>
        <div id="magicdomid4" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">currently
            defined as proposed standard RFCs 3261, 3262, 3263, 3264,
            and</span></div>
        <div id="magicdomid5" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">6665.</span></div>
        <div id="magicdomid6" class=""><br class="">
        </div>
        <div id="magicdomid7" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The
            SIPCore working group will concentrate on specifications
            that update</span></div>
        <div id="magicdomid8" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">or
            replace the core SIP specifications named above as well as</span></div>
        <div id="magicdomid9" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">specifications
            pertaining to small, self-contained SIP protocol</span></div>
        <div id="magicdomid10" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">extensions.&nbsp;
            The process and requirements for new SIP extensions are</span></div>
        <div id="magicdomid11" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">documented
            in RFC5727, "Change Process for the Session Initiation</span></div>
        <div id="magicdomid12" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Protocol".</span></div>
        <div id="magicdomid13" class=""><br class="">
        </div>
        <div id="magicdomid14" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">Throughout
            its work, the group will strive to maintain the basic model</span></div>
        <div id="magicdomid15" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">and
            architecture defined by SIP. In particular:</span></div>
        <div id="magicdomid16" class=""><br class="">
        </div>
        <div id="magicdomid17" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">1.
            Services and features are provided end-to-end whenever
            possible.</span></div>
        <div id="magicdomid18" class=""><br class="">
        </div>
        <div id="magicdomid19" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">2.
            Reuse of existing Internet protocols and architectures and</span></div>
        <div id="magicdomid20" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">&nbsp;&nbsp;
            integrating with other Internet applications is crucial.</span></div>
        <div id="magicdomid21" class=""><br class="">
        </div>
        <div id="magicdomid22" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">3.
            Standards-track extensions and new features must be
            generally</span></div>
        <div id="magicdomid23" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">&nbsp;&nbsp;
            applicable, and not applicable only to a specific set of
            session</span></div>
        <div id="magicdomid24" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">&nbsp;&nbsp;
            types.</span></div>
        <div id="magicdomid25" class=""><br class="">
        </div>
        <div id="magicdomid26" class=""><span class="author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">4.
            Simpler solutions that solve a given problem should be
            favored</span></div>
        <div id="magicdomid27" class=""><span class="author-a-lz76zz72zz68zz76zfnz80zicdfplz82zz84z">&nbsp;&nbsp;&nbsp;
            over more complex solutions.</span></div>
        <div id="magicdomid28" class=""><br class="">
        </div>
        <div id="magicdomid29" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">The
            primary source of change requirements to the core SIP
            specifications</span></div>
        <div id="magicdomid30" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">(enumerated
            above) will be a) interoperability problems that stem from</span></div>
        <div id="magicdomid31" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">ambiguous,
            or under-defined specification, and b) requirements from</span></div>
        <div id="magicdomid32" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">other
            working groups in the ART Area. The primary source of new
            protocol</span></div>
        <div id="magicdomid33" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">extensions
            is the DISPATCH working group, which will generally make the</span></div>
        <div id="magicdomid34" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">determination
            regarding whether new SIP-related work warrants a new</span></div>
        <div id="magicdomid35" class=""><span class="author-a-lz70zz73ztz73zz74zedkismz67z9z85zz122z">working
            group or belongs in an existing one.</span></div>
      </blockquote><div class=""><br class="webkit-block-placeholder"></div><p class=""><br class="">
    </p><p class="">For ease of reference, the existing SIPCORE charter is at
      <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/charter-ietf-sipcore/">https://datatracker.ietf.org/doc/charter-ietf-sipcore/</a></p><p class="">Thanks!<br class="">
    </p><p class="">/a<br class="">
    </p>
  </div>

_______________________________________________<br class="">dispatch mailing list<br class=""><a href="mailto:dispatch@ietf.org" class="">dispatch@ietf.org</a><br class="">https://www.ietf.org/mailman/listinfo/dispatch<br class=""></div></blockquote></div><br class=""></div></body></html>
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From: Adam Roach <adam@nostrum.com>
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Subject: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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[as chair]

SIPCORE:

The chairs would like to call for adoption of 
draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft, 
provisional on approval of the currently proposed SIPCORE charter 
revisions (which we believe have a high probability of approval). If you 
have any opinion on such adoption, please comment on the SIPCORE mailing 
list within the next two weeks.

While we do not wish to rush discussion unnecessarily, we have been 
informally told by 3GPP that the registration of a SIP status code 
meaning "reject this call and any subsequent calls from the same caller" 
is both important and time-sensitive. To that end, we plan to run this 
call for adoption over the next two weeks (ending Monday, December 12). 
Assuming adoption and approval of the new charter, we will then run a 
three-week working group last call (ending shortly after the end of the 
year). Given the relatively simple mechanism and limited scope of the 
current document, we believe this is a reasonable timeframe. If you 
disagree, please speak up.

Working group participants are strongly encouraged to review and comment 
on the document at this time.

Thanks!

/a


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From: "DOLLY, MARTIN C" <md3135@att.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
Thread-Index: AQHSSd92AVGSlf+ZMU25j7USfLZWiaDvMwNm
Date: Tue, 29 Nov 2016 01:53:28 +0000
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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I support

Martin C. Dolly
Lead Member of Technical Staff
Core & Government/Regulatory Standards
AT&T
Cell: +1.609.903.3360<tel:+1.609.903.3360>
Email: md3135@att.com<mailto:md3135@att.com>


On Nov 28, 2016, at 8:25 PM, Adam Roach <adam@nostrum.com<mailto:adam@nostr=
um.com>> wrote:

[as chair]

SIPCORE:

The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
atus-unwanted as a SIPCORE draft, provisional on approval of the currently =
proposed SIPCORE charter revisions (which we believe have a high probabilit=
y of approval). If you have any opinion on such adoption, please comment on=
 the SIPCORE mailing list within the next two weeks.

While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning "reject=
 this call and any subsequent calls from the same caller" is both important=
 and time-sensitive. To that end, we plan to run this call for adoption ove=
r the next two weeks (ending Monday, December 12). Assuming adoption and ap=
proval of the new charter, we will then run a three-week working group last=
 call (ending shortly after the end of the year). Given the relatively simp=
le mechanism and limited scope of the current document, we believe this is =
a reasonable timeframe. If you disagree, please speak up.

Working group participants are strongly encouraged to review and comment on=
 the document at this time.

Thanks!

/a

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

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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>I support<br>
<br>
<p class=3D"MsoNormal" style=3D"margin: 0in 0in 0.0001pt;"><span style=3D"b=
ackground-color: rgba(255, 255, 255, 0);"><b>Martin C. Dolly</b><o:p></o:p>=
</span></p>
<p class=3D"MsoNormal" style=3D"margin: 0in 0in 0.0001pt;"><span style=3D"b=
ackground-color: rgba(255, 255, 255, 0);">Lead Member of Technical Staff<o:=
p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin: 0in 0in 0.0001pt;"><span style=3D"b=
ackground-color: rgba(255, 255, 255, 0);">Core &amp; Government/Regulatory =
Standards<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin: 0in 0in 0.0001pt;"><span style=3D"b=
ackground-color: rgba(255, 255, 255, 0);">AT&amp;T<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin: 0in 0in 0.0001pt;"><span style=3D"b=
ackground-color: rgba(255, 255, 255, 0);">Cell:&nbsp;<a dir=3D"ltr" href=3D=
"tel:&#43;1.609.903.3360" x-apple-data-detectors=3D"true" x-apple-data-dete=
ctors-type=3D"telephone" x-apple-data-detectors-result=3D"2/0">&#43;1.609.9=
03.3360</a><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin: 0in 0in 0.0001pt;"><span style=3D"b=
ackground-color: rgba(255, 255, 255, 0);">Email:&nbsp;<u><a href=3D"mailto:=
md3135@att.com">md3135@att.com</a></u><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin: 0in 0in 0.0001pt;"><o:p style=3D"ba=
ckground-color: rgba(255, 255, 255, 0);">&nbsp;</o:p></p>
</div>
<div><br>
On Nov 28, 2016, at 8:25 PM, Adam Roach &lt;<a href=3D"mailto:adam@nostrum.=
com">adam@nostrum.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div><span>[as chair]</span><br>
<span></span><br>
<span>SIPCORE:</span><br>
<span></span><br>
<span>The chairs would like to call for adoption of draft-schulzrinne-dispa=
tch-status-unwanted as a SIPCORE draft, provisional on approval of the curr=
ently proposed SIPCORE charter revisions (which we believe have a high prob=
ability of approval). If you have
 any opinion on such adoption, please comment on the SIPCORE mailing list w=
ithin the next two weeks.</span><br>
<span></span><br>
<span>While we do not wish to rush discussion unnecessarily, we have been i=
nformally told by 3GPP that the registration of a SIP status code meaning &=
quot;reject this call and any subsequent calls from the same caller&quot; i=
s both important and time-sensitive. To that
 end, we plan to run this call for adoption over the next two weeks (ending=
 Monday, December 12). Assuming adoption and approval of the new charter, w=
e will then run a three-week working group last call (ending shortly after =
the end of the year). Given the
 relatively simple mechanism and limited scope of the current document, we =
believe this is a reasonable timeframe. If you disagree, please speak up.</=
span><br>
<span></span><br>
<span>Working group participants are strongly encouraged to review and comm=
ent on the document at this time.</span><br>
<span></span><br>
<span>Thanks!</span><br>
<span></span><br>
<span>/a</span><br>
<span></span><br>
<span>_______________________________________________</span><br>
<span>sipcore mailing list</span><br>
<span><a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www=
.ietf.org/mailman/listinfo/sipcore</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_7C7531C3212C43448CE19A1F54A25F51attcom_--


From nobody Mon Nov 28 19:13:22 2016
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To: sipcore@ietf.org
References: <c291b1bb-72e4-c19c-ba23-35a1b23b4644@nostrum.com>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Mon, 28 Nov 2016 22:13:16 -0500
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/3jOsjiHnc1SDhbkbhMl_NeT8NjM>
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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I support adoption.

	Thanks,
	Paul

On 11/28/16 8:25 PM, Adam Roach wrote:
> [as chair]
>
> SIPCORE:
>
> The chairs would like to call for adoption of
> draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft,
> provisional on approval of the currently proposed SIPCORE charter
> revisions (which we believe have a high probability of approval). If you
> have any opinion on such adoption, please comment on the SIPCORE mailing
> list within the next two weeks.
>
> While we do not wish to rush discussion unnecessarily, we have been
> informally told by 3GPP that the registration of a SIP status code
> meaning "reject this call and any subsequent calls from the same caller"
> is both important and time-sensitive. To that end, we plan to run this
> call for adoption over the next two weeks (ending Monday, December 12).
> Assuming adoption and approval of the new charter, we will then run a
> three-week working group last call (ending shortly after the end of the
> year). Given the relatively simple mechanism and limited scope of the
> current document, we believe this is a reasonable timeframe. If you
> disagree, please speak up.
>
> Working group participants are strongly encouraged to review and comment
> on the document at this time.
>
> Thanks!
>
> /a
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From nobody Mon Nov 28 19:42:15 2016
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: SIPCORE <sipcore@ietf.org>
Thread-Topic: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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I support adoption too.

Cheers,

Gonzalo

> On Nov 28, 2016, at 10:13 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
> I support adoption.
>=20
> 	Thanks,
> 	Paul
>=20
> On 11/28/16 8:25 PM, Adam Roach wrote:
>> [as chair]
>>=20
>> SIPCORE:
>>=20
>> The chairs would like to call for adoption of
>> draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft,
>> provisional on approval of the currently proposed SIPCORE charter
>> revisions (which we believe have a high probability of approval). If you
>> have any opinion on such adoption, please comment on the SIPCORE mailing
>> list within the next two weeks.
>>=20
>> While we do not wish to rush discussion unnecessarily, we have been
>> informally told by 3GPP that the registration of a SIP status code
>> meaning "reject this call and any subsequent calls from the same caller"
>> is both important and time-sensitive. To that end, we plan to run this
>> call for adoption over the next two weeks (ending Monday, December 12).
>> Assuming adoption and approval of the new charter, we will then run a
>> three-week working group last call (ending shortly after the end of the
>> year). Given the relatively simple mechanism and limited scope of the
>> current document, we believe this is a reasonable timeframe. If you
>> disagree, please speak up.
>>=20
>> Working group participants are strongly encouraged to review and comment
>> on the document at this time.
>>=20
>> Thanks!
>>=20
>> /a
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: SIPCORE <sipcore@ietf.org>
Thread-Topic: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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Date: Tue, 29 Nov 2016 04:42:45 +0000
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References: <c291b1bb-72e4-c19c-ba23-35a1b23b4644@nostrum.com> <1f290ece-96d7-b9a5-c41c-bafc0454ab9d@alum.mit.edu> <B5C4B3A5-5BA4-4DF9-8BB0-5EEC01C6D031@cisco.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/NgUPhsgSLxlmsxpS25_ZJtbbcRk>
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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Me too.

Thanks,
Tolga

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Gonzalo
> Salgueiro (gsalguei)
> Sent: Monday, November 28, 2016 10:42 PM
> To: SIPCORE <sipcore@ietf.org>
> Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne=
-
> dispatch-status-unwanted
>=20
> I support adoption too.
>=20
> Cheers,
>=20
> Gonzalo
>=20
> > On Nov 28, 2016, at 10:13 PM, Paul Kyzivat <pkyzivat@alum.mit.edu>
> wrote:
> >
> > I support adoption.
> >
> > 	Thanks,
> > 	Paul
> >
> > On 11/28/16 8:25 PM, Adam Roach wrote:
> >> [as chair]
> >>
> >> SIPCORE:
> >>
> >> The chairs would like to call for adoption of
> >> draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft,
> >> provisional on approval of the currently proposed SIPCORE charter
> >> revisions (which we believe have a high probability of approval). If
> >> you have any opinion on such adoption, please comment on the SIPCORE
> >> mailing list within the next two weeks.
> >>
> >> While we do not wish to rush discussion unnecessarily, we have been
> >> informally told by 3GPP that the registration of a SIP status code
> >> meaning "reject this call and any subsequent calls from the same calle=
r"
> >> is both important and time-sensitive. To that end, we plan to run
> >> this call for adoption over the next two weeks (ending Monday,
> December 12).
> >> Assuming adoption and approval of the new charter, we will then run a
> >> three-week working group last call (ending shortly after the end of
> >> the year). Given the relatively simple mechanism and limited scope of
> >> the current document, we believe this is a reasonable timeframe. If
> >> you disagree, please speak up.
> >>
> >> Working group participants are strongly encouraged to review and
> >> comment on the document at this time.
> >>
> >> Thanks!
> >>
> >> /a
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> >>
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Mon Nov 28 21:32:39 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, 'SIPCORE' <sipcore@ietf.org>
Thread-Topic: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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Date: Tue, 29 Nov 2016 05:32:32 +0000
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Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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I support adoption.

Regards,

Christer

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
Sent: 29 November 2016 03:25
To: 'SIPCORE' <sipcore@ietf.org>
Subject: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispa=
tch-status-unwanted

[as chair]

SIPCORE:

The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
atus-unwanted as a SIPCORE draft, provisional on approval of the currently =
proposed SIPCORE charter revisions (which we believe have a high probabilit=
y of approval). If you have any opinion on such adoption, please comment on=
 the SIPCORE mailing list within the next two weeks.

While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning "reject=
 this call and any subsequent calls from the same caller"=20
is both important and time-sensitive. To that end, we plan to run this call=
 for adoption over the next two weeks (ending Monday, December 12).=20
Assuming adoption and approval of the new charter, we will then run a three=
-week working group last call (ending shortly after the end of the year). G=
iven the relatively simple mechanism and limited scope of the current docum=
ent, we believe this is a reasonable timeframe. If you disagree, please spe=
ak up.

Working group participants are strongly encouraged to review and comment on=
 the document at this time.

Thanks!

/a

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


From nobody Tue Nov 29 05:37:35 2016
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From: Richard Shockey <richard@shockey.us>
To: SIPCORE <sipcore@ietf.org>
Message-ID: <A47D2D6D-E906-427D-8FF3-5ACEB513E6FD@shockey.us>
Thread-Topic: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
References: <c291b1bb-72e4-c19c-ba23-35a1b23b4644@nostrum.com> <7C7531C3-212C-4344-8CE1-9A1F54A25F51@att.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/XYtBtrOimfV1ysdwy8aZLmM6JNo>
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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=20

+1=C2=A0=20

=20

=E2=80=94=20

Richard Shockey

Shockey Consulting LLC

Chairman of the Board SIP Forum

www.shockey.us

www.sipforum.org

richard<at>shockey.us

Skype-Linkedin-Facebook rshockey101

PSTN +1 703-593-2683

=20

=20

From: sipcore <sipcore-bounces@ietf.org> on behalf of "DOLLY, MARTIN C" <md=
3135@att.com>
Date: Monday, November 28, 2016 at 8:53 PM
To: Adam Roach <adam@nostrum.com>
Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-d=
ispatch-status-unwanted

=20

I support

Martin C. Dolly

Lead Member of Technical Staff

Core & Government/Regulatory Standards

AT&T

Cell: +1.609.903.3360

Email: md3135@att.com

=20


On Nov 28, 2016, at 8:25 PM, Adam Roach <adam@nostrum.com> wrote:

[as chair]

SIPCORE:

The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
atus-unwanted as a SIPCORE draft, provisional on approval of the currently p=
roposed SIPCORE charter revisions (which we believe have a high probability =
of approval). If you have any opinion on such adoption, please comment on th=
e SIPCORE mailing list within the next two weeks.

While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning "reject =
this call and any subsequent calls from the same caller" is both important a=
nd time-sensitive. To that end, we plan to run this call for adoption over t=
he next two weeks (ending Monday, December 12). Assuming adoption and approv=
al of the new charter, we will then run a three-week working group last call=
 (ending shortly after the end of the year). Given the relatively simple mec=
hanism and limited scope of the current document, we believe this is a reaso=
nable timeframe. If you disagree, please speak up.

Working group participants are strongly encouraged to review and comment on=
 the document at this time.

Thanks!

/a

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

_______________________________________________ sipcore mailing list sipcor=
e@ietf.org https://www.ietf.org/mailman/listinfo/sipcore=20


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.0pt;font-family:Calibri;color:black'>Richard Shockey<o:p></o:p></span></p><=
/div><div><p class=3DMsoNormalCxSpMiddle style=3D'mso-margin-top-alt:1.0pt;margi=
n-right:0in;margin-bottom:1.0pt;margin-left:0in;mso-add-space:auto'><span st=
yle=3D'font-size:8.0pt;font-family:Calibri;color:black'>Shockey Consulting LLC=
<o:p></o:p></span></p></div><div><p class=3DMsoNormalCxSpMiddle style=3D'mso-mar=
gin-top-alt:1.0pt;margin-right:0in;margin-bottom:1.0pt;margin-left:0in;mso-a=
dd-space:auto'><span style=3D'font-size:8.0pt;font-family:Calibri;color:black'=
>Chairman of the Board SIP Forum<o:p></o:p></span></p></div><div><p class=3DMs=
oNormalCxSpMiddle style=3D'mso-margin-top-alt:1.0pt;margin-right:0in;margin-bo=
ttom:1.0pt;margin-left:0in;mso-add-space:auto'><span style=3D'font-size:8.0pt;=
font-family:Calibri;color:black'>www.shockey.us<o:p></o:p></span></p></div><=
div><p class=3DMsoNormalCxSpMiddle style=3D'mso-margin-top-alt:1.0pt;margin-righ=
t:0in;margin-bottom:1.0pt;margin-left:0in;mso-add-space:auto'><span style=3D'f=
ont-size:8.0pt;font-family:Calibri;color:black'>www.sipforum.org<o:p></o:p><=
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1.0pt;margin-right:0in;margin-bottom:1.0pt;margin-left:0in;mso-add-space:aut=
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at&gt;shockey.us<o:p></o:p></span></p></div><div><p class=3DMsoNormalCxSpMiddl=
e style=3D'mso-margin-top-alt:1.0pt;margin-right:0in;margin-bottom:1.0pt;margi=
n-left:0in;mso-add-space:auto'><span style=3D'font-size:8.0pt;font-family:Cali=
bri;color:black'>Skype-Linkedin-Facebook rshockey101<o:p></o:p></span></p></=
div><div><p class=3DMsoNormalCxSpMiddle style=3D'mso-margin-top-alt:1.0pt;margin=
-right:0in;margin-bottom:1.0pt;margin-left:0in;mso-add-space:auto'><span sty=
le=3D'font-size:8.0pt;font-family:Calibri;color:black'>PSTN +1 703-593-2683<o:=
p></o:p></span></p></div></div><p class=3DMsoNormal><span style=3D'font-size:11.=
0pt;font-family:Calibri'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><spa=
n style=3D'font-size:11.0pt;font-family:Calibri'><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-family:Calibri;color:black'>Fro=
m: </span></b><span style=3D'font-family:Calibri;color:black'>sipcore &lt;sipc=
ore-bounces@ietf.org&gt; on behalf of &quot;DOLLY, MARTIN C&quot; &lt;md3135=
@att.com&gt;<br><b>Date: </b>Monday, November 28, 2016 at 8:53 PM<br><b>To: =
</b>Adam Roach &lt;adam@nostrum.com&gt;<br><b>Cc: </b>SIPCORE &lt;sipcore@ie=
tf.org&gt;<br><b>Subject: </b>Re: [sipcore] Call for (provisional) Adoption:=
 draft-schulzrinne-dispatch-status-unwanted<o:p></o:p></span></p></div><div>=
<p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal style=3D=
'margin-bottom:12.0pt'>I support<o:p></o:p></p><p class=3DMsoNormal><b>Martin =
C. Dolly</b><o:p></o:p></p><p class=3DMsoNormal>Lead Member of Technical Staff=
<o:p></o:p></p><p class=3DMsoNormal>Core &amp; Government/Regulatory Standards=
<o:p></o:p></p><p class=3DMsoNormal>AT&amp;T<o:p></o:p></p><p class=3DMsoNormal>=
Cell:&nbsp;<a href=3D"tel:+1.609.903.3360">+1.609.903.3360</a><o:p></o:p></p><=
p class=3DMsoNormal>Email:&nbsp;<u><a href=3D"mailto:md3135@att.com">md3135@att.=
com</a></u><o:p></o:p></p><p class=3DMsoNormal>&nbsp;<o:p></o:p></p></div><div=
><p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>On Nov 28, 2016, at 8:2=
5 PM, Adam Roach &lt;<a href=3D"mailto:adam@nostrum.com">adam@nostrum.com</a>&=
gt; wrote:<o:p></o:p></p></div><blockquote style=3D'margin-top:5.0pt;margin-bo=
ttom:5.0pt'><div><p class=3DMsoNormal>[as chair]<br><br>SIPCORE:<br><br>The ch=
airs would like to call for adoption of draft-schulzrinne-dispatch-status-un=
wanted as a SIPCORE draft, provisional on approval of the currently proposed=
 SIPCORE charter revisions (which we believe have a high probability of appr=
oval). If you have any opinion on such adoption, please comment on the SIPCO=
RE mailing list within the next two weeks.<br><br>While we do not wish to ru=
sh discussion unnecessarily, we have been informally told by 3GPP that the r=
egistration of a SIP status code meaning &quot;reject this call and any subs=
equent calls from the same caller&quot; is both important and time-sensitive=
. To that end, we plan to run this call for adoption over the next two weeks=
 (ending Monday, December 12). Assuming adoption and approval of the new cha=
rter, we will then run a three-week working group last call (ending shortly =
after the end of the year). Given the relatively simple mechanism and limite=
d scope of the current document, we believe this is a reasonable timeframe. =
If you disagree, please speak up.<br><br>Working group participants are stro=
ngly encouraged to review and comment on the document at this time.<br><br>T=
hanks!<br><br>/a<br><br>_______________________________________________<br>s=
ipcore 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">https://www.iet=
f.org/mailman/listinfo/sipcore</a><o:p></o:p></p></div></blockquote><p class=
=3DMsoNormal>_______________________________________________ sipcore mailing l=
ist sipcore@ietf.org https://www.ietf.org/mailman/listinfo/sipcore <o:p></o:=
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To: Adam Roach <adam@nostrum.com>, 'SIPCORE' <sipcore@ietf.org>
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Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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Support.

-- 
Simon Perreault
Director of Engineering, Platform | Jive Communications, Inc.
https://jive.com | +1 418 478 0989 ext. 1241 | sperreault@jive.com


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From: "Gunn, Janet P" <Janet.Gunn@csra.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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+1

Janet

From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY, MARTIN =
C
Sent: Monday, November 28, 2016 8:53 PM
To: Adam Roach <adam@nostrum.com>
Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-d=
ispatch-status-unwanted

I support
Martin C. Dolly
Lead Member of Technical Staff
Core & Government/Regulatory Standards
AT&T
Cell: +1.609.903.3360<tel:+1.609.903.3360>
Email: md3135@att.com<mailto:md3135@att.com>


On Nov 28, 2016, at 8:25 PM, Adam Roach <adam@nostrum.com<mailto:adam@nostr=
um.com>> wrote:
[as chair]

SIPCORE:

The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
atus-unwanted as a SIPCORE draft, provisional on approval of the currently =
proposed SIPCORE charter revisions (which we believe have a high probabilit=
y of approval). If you have any opinion on such adoption, please comment on=
 the SIPCORE mailing list within the next two weeks.

While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning "reject=
 this call and any subsequent calls from the same caller" is both important=
 and time-sensitive. To that end, we plan to run this call for adoption ove=
r the next two weeks (ending Monday, December 12). Assuming adoption and ap=
proval of the new charter, we will then run a three-week working group last=
 call (ending shortly after the end of the year). Given the relatively simp=
le mechanism and limited scope of the current document, we believe this is =
a reasonable timeframe. If you disagree, please speak up.

Working group participants are strongly encouraged to review and comment on=
 the document at this time.

Thanks!

/a

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


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Janet<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,sans-serif">From:</span></b><span style=3D"font-size:11.0pt;=
font-family:&quot;Calibri&quot;,sans-serif"> sipcore [mailto:sipcore-bounce=
s@ietf.org]
<b>On Behalf Of </b>DOLLY, MARTIN C<br>
<b>Sent:</b> Monday, November 28, 2016 8:53 PM<br>
<b>To:</b> Adam Roach &lt;adam@nostrum.com&gt;<br>
<b>Cc:</b> SIPCORE &lt;sipcore@ietf.org&gt;<br>
<b>Subject:</b> Re: [sipcore] Call for (provisional) Adoption: draft-schulz=
rinne-dispatch-status-unwanted<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">I support<o:p></o:p><=
/p>
<p class=3D"MsoNormal"><b>Martin C. Dolly</b><o:p></o:p></p>
<p class=3D"MsoNormal">Lead Member of Technical Staff<o:p></o:p></p>
<p class=3D"MsoNormal">Core &amp; Government/Regulatory Standards<o:p></o:p=
></p>
<p class=3D"MsoNormal">AT&amp;T<o:p></o:p></p>
<p class=3D"MsoNormal">Cell:&nbsp;<a href=3D"tel:&#43;1.609.903.3360">&#43;=
1.609.903.3360</a><o:p></o:p></p>
<p class=3D"MsoNormal">Email:&nbsp;<u><a href=3D"mailto:md3135@att.com">md3=
135@att.com</a></u><o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
On Nov 28, 2016, at 8:25 PM, Adam Roach &lt;<a href=3D"mailto:adam@nostrum.=
com">adam@nostrum.com</a>&gt; wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">[as chair]<br>
<br>
SIPCORE:<br>
<br>
The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
atus-unwanted as a SIPCORE draft, provisional on approval of the currently =
proposed SIPCORE charter revisions (which we believe have a high probabilit=
y of approval). If you have any
 opinion on such adoption, please comment on the SIPCORE mailing list withi=
n the next two weeks.<br>
<br>
While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning &quot;r=
eject this call and any subsequent calls from the same caller&quot; is both=
 important and time-sensitive. To that end,
 we plan to run this call for adoption over the next two weeks (ending Mond=
ay, December 12). Assuming adoption and approval of the new charter, we wil=
l then run a three-week working group last call (ending shortly after the e=
nd of the year). Given the relatively
 simple mechanism and limited scope of the current document, we believe thi=
s is a reasonable timeframe. If you disagree, please speak up.<br>
<br>
Working group participants are strongly encouraged to review and comment on=
 the document at this time.<br>
<br>
Thanks!<br>
<br>
/a<br>
<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">https://www.ietf.=
org/mailman/listinfo/sipcore</a><o:p></o:p></p>
</div>
</blockquote>
</div>
<br>
<p>This electronic message transmission contains information from CSRA that=
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ation in this message is intended only for use by the individual(s) to whom=
 it is addressed. If you believe
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 be aware that any use, disclosure, copying or distribution of the contents=
 of this message is strictly prohibited. NOTE: Regardless of content, this =
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From: Ranjit Avasarala <ranjitkav0811@gmail.com>
Date: Tue, 29 Nov 2016 15:29:29 -0600
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Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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I support

Regards
Ranjit

On Mon, Nov 28, 2016 at 7:25 PM, Adam Roach <adam@nostrum.com> wrote:

> [as chair]
>
> SIPCORE:
>
> The chairs would like to call for adoption of
> draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft,
> provisional on approval of the currently proposed SIPCORE charter revisions
> (which we believe have a high probability of approval). If you have any
> opinion on such adoption, please comment on the SIPCORE mailing list within
> the next two weeks.
>
> While we do not wish to rush discussion unnecessarily, we have been
> informally told by 3GPP that the registration of a SIP status code meaning
> "reject this call and any subsequent calls from the same caller" is both
> important and time-sensitive. To that end, we plan to run this call for
> adoption over the next two weeks (ending Monday, December 12). Assuming
> adoption and approval of the new charter, we will then run a three-week
> working group last call (ending shortly after the end of the year). Given
> the relatively simple mechanism and limited scope of the current document,
> we believe this is a reasonable timeframe. If you disagree, please speak up.
>
> Working group participants are strongly encouraged to review and comment
> on the document at this time.
>
> Thanks!
>
> /a
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr"><div><div>I support<br><br></div>Regards<br></div>Ranjit<b=
r></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, N=
ov 28, 2016 at 7:25 PM, Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:=
adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">[as chair]<br>
<br>
SIPCORE:<br>
<br>
The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
a<wbr>tus-unwanted as a SIPCORE draft, provisional on approval of the curre=
ntly proposed SIPCORE charter revisions (which we believe have a high proba=
bility of approval). If you have any opinion on such adoption, please comme=
nt on the SIPCORE mailing list within the next two weeks.<br>
<br>
While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning &quot;r=
eject this call and any subsequent calls from the same caller&quot; is both=
 important and time-sensitive. To that end, we plan to run this call for ad=
option over the next two weeks (ending Monday, December 12). Assuming adopt=
ion and approval of the new charter, we will then run a three-week working =
group last call (ending shortly after the end of the year). Given the relat=
ively simple mechanism and limited scope of the current document, we believ=
e this is a reasonable timeframe. If you disagree, please speak up.<br>
<br>
Working group participants are strongly encouraged to review and comment on=
 the document at this time.<br>
<br>
Thanks!<br>
<br>
/a<br>
<br>
______________________________<wbr>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/sipcore</a><=
br>
</blockquote></div><br></div>

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From: Andy Hutton <andyhutton.ietf@gmail.com>
Date: Wed, 30 Nov 2016 16:15:57 +0000
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To: Adam Roach <adam@nostrum.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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I support.

Andy

On Tue, Nov 29, 2016 at 1:25 AM, Adam Roach <adam@nostrum.com> wrote:

> [as chair]
>
> SIPCORE:
>
> The chairs would like to call for adoption of
> draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft,
> provisional on approval of the currently proposed SIPCORE charter revisions
> (which we believe have a high probability of approval). If you have any
> opinion on such adoption, please comment on the SIPCORE mailing list within
> the next two weeks.
>
> While we do not wish to rush discussion unnecessarily, we have been
> informally told by 3GPP that the registration of a SIP status code meaning
> "reject this call and any subsequent calls from the same caller" is both
> important and time-sensitive. To that end, we plan to run this call for
> adoption over the next two weeks (ending Monday, December 12). Assuming
> adoption and approval of the new charter, we will then run a three-week
> working group last call (ending shortly after the end of the year). Given
> the relatively simple mechanism and limited scope of the current document,
> we believe this is a reasonable timeframe. If you disagree, please speak up.
>
> Working group participants are strongly encouraged to review and comment
> on the document at this time.
>
> Thanks!
>
> /a
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr">I support.<div><br></div><div>Andy</div></div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Nov 29, 2016 at 1:2=
5 AM, Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" =
target=3D"_blank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">[as chair]<br>
<br>
SIPCORE:<br>
<br>
The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
a<wbr>tus-unwanted as a SIPCORE draft, provisional on approval of the curre=
ntly proposed SIPCORE charter revisions (which we believe have a high proba=
bility of approval). If you have any opinion on such adoption, please comme=
nt on the SIPCORE mailing list within the next two weeks.<br>
<br>
While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning &quot;r=
eject this call and any subsequent calls from the same caller&quot; is both=
 important and time-sensitive. To that end, we plan to run this call for ad=
option over the next two weeks (ending Monday, December 12). Assuming adopt=
ion and approval of the new charter, we will then run a three-week working =
group last call (ending shortly after the end of the year). Given the relat=
ively simple mechanism and limited scope of the current document, we believ=
e this is a reasonable timeframe. If you disagree, please speak up.<br>
<br>
Working group participants are strongly encouraged to review and comment on=
 the document at this time.<br>
<br>
Thanks!<br>
<br>
/a<br>
<br>
______________________________<wbr>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/sipcore</a><=
br>
</blockquote></div><br></div>

--94eb2c0b0c262acf390542870561--


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From: Eric Burger <ewburger@gmail.com>
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Date: Wed, 30 Nov 2016 12:01:58 -0500
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Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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Opinion: Yes, please.

> On Nov 28, 2016, at 8:25 PM, Adam Roach <adam@nostrum.com> wrote:
>=20
> [as chair]
>=20
> SIPCORE:
>=20
> The chairs would like to call for adoption of =
draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft, =
provisional on approval of the currently proposed SIPCORE charter =
revisions (which we believe have a high probability of approval). If you =
have any opinion on such adoption, please comment on the SIPCORE mailing =
list within the next two weeks.
>=20
> While we do not wish to rush discussion unnecessarily, we have been =
informally told by 3GPP that the registration of a SIP status code =
meaning "reject this call and any subsequent calls from the same caller" =
is both important and time-sensitive. To that end, we plan to run this =
call for adoption over the next two weeks (ending Monday, December 12). =
Assuming adoption and approval of the new charter, we will then run a =
three-week working group last call (ending shortly after the end of the =
year). Given the relatively simple mechanism and limited scope of the =
current document, we believe this is a reasonable timeframe. If you =
disagree, please speak up.
>=20
> Working group participants are strongly encouraged to review and =
comment on the document at this time.
>=20
> Thanks!
>=20
> /a
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Wed Nov 30 09:22:21 2016
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From: "Vijay K.Gurbani" <vijay.gurbani@nokia-bell-labs.com>
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Subject: Re: [sipcore] Call for (provisional) Adoption:draft-schulzrinne-dispatch-status-unwanted
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On 11/28/2016 07:25 PM, Adam Roach wrote:
> The chairs would like to call for adoption of
> draft-schulzrinne-dispatch-status-unwanted as a SIPCORE draft,
> provisional on approval of the currently proposed SIPCORE charter
> revisions (which we believe have a high probability of approval). If you
> have any opinion on such adoption, please comment on the SIPCORE mailing
> list within the next two weeks. [...]

Yes for adoption.

Couple minor nits from a quick review:

- S1: s/was unwanted/is unwanted/
   (Reason: to match the present tense verb "express")
- S6: s/is used/be used/

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Nokia Networks
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
Email: vkg@bell-labs.com / vijay.gurbani@nokia-bell-labs.com
Web: http://ect.bell-labs.com/who/vkg/  | Calendar: http://goo.gl/x3Ogq


From nobody Wed Nov 30 09:56:42 2016
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From: Andrew Allen <aallen@blackberry.com>
To: Andy Hutton <andyhutton.ietf@gmail.com>, Adam Roach <adam@nostrum.com>
Thread-Topic: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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Date: Wed, 30 Nov 2016 17:56:31 +0000
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Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-dispatch-status-unwanted
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+1

Sent from my BlackBerry - the most secure mobile device
From: andyhutton.ietf@gmail.com
Sent: November 30, 2016 11:42 AM
To: adam@nostrum.com
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Call for (provisional) Adoption: draft-schulzrinne-d=
ispatch-status-unwanted


I support.

Andy

On Tue, Nov 29, 2016 at 1:25 AM, Adam Roach <adam@nostrum.com<mailto:adam@n=
ostrum.com>> wrote:
[as chair]

SIPCORE:

The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
atus-unwanted as a SIPCORE draft, provisional on approval of the currently =
proposed SIPCORE charter revisions (which we believe have a high probabilit=
y of approval). If you have any opinion on such adoption, please comment on=
 the SIPCORE mailing list within the next two weeks.

While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning "reject=
 this call and any subsequent calls from the same caller" is both important=
 and time-sensitive. To that end, we plan to run this call for adoption ove=
r the next two weeks (ending Monday, December 12). Assuming adoption and ap=
proval of the new charter, we will then run a three-week working group last=
 call (ending shortly after the end of the year). Given the relatively simp=
le mechanism and limited scope of the current document, we believe this is =
a reasonable timeframe. If you disagree, please speak up.

Working group participants are strongly encouraged to review and comment on=
 the document at this time.

Thanks!

/a

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


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<tbody>
<tr>
<td colspan=3D"2" style=3D"padding:initial; font-size:initial; text-align:i=
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&quot;BB Alpha Sans&quot;,&quot;Slate Pro&quot;; font-size:10pt">
<div id=3D"from"><b>From:</b> andyhutton.ietf@gmail.com</div>
<div id=3D"sent"><b>Sent:</b> November 30, 2016 11:42 AM</div>
<div id=3D"to"><b>To:</b> adam@nostrum.com</div>
<div id=3D"cc"><b>Cc:</b> sipcore@ietf.org</div>
<div id=3D"subject"><b>Subject:</b> Re: [sipcore] Call for (provisional) Ad=
option: draft-schulzrinne-dispatch-status-unwanted</div>
</div>
</td>
</tr>
</tbody>
</table>
<div style=3D"border-right:none; border-bottom:none; border-left:none; bord=
er-top:1pt solid rgb(186,188,209); display:block; padding:initial; font-siz=
e:initial; text-align:initial; background-color:rgb(255,255,255)">
</div>
<br>
</div>
<div>
<div dir=3D"ltr">I support.
<div><br>
</div>
<div>Andy</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Tue, Nov 29, 2016 at 1:25 AM, Adam Roach <spa=
n dir=3D"ltr">
&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com<=
/a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
[as chair]<br>
<br>
SIPCORE:<br>
<br>
The chairs would like to call for adoption of draft-schulzrinne-dispatch-st=
a<wbr>tus-unwanted as a SIPCORE draft, provisional on approval of the curre=
ntly proposed SIPCORE charter revisions (which we believe have a high proba=
bility of approval). If you have
 any opinion on such adoption, please comment on the SIPCORE mailing list w=
ithin the next two weeks.<br>
<br>
While we do not wish to rush discussion unnecessarily, we have been informa=
lly told by 3GPP that the registration of a SIP status code meaning &quot;r=
eject this call and any subsequent calls from the same caller&quot; is both=
 important and time-sensitive. To that end,
 we plan to run this call for adoption over the next two weeks (ending Mond=
ay, December 12). Assuming adoption and approval of the new charter, we wil=
l then run a three-week working group last call (ending shortly after the e=
nd of the year). Given the relatively
 simple mechanism and limited scope of the current document, we believe thi=
s is a reasonable timeframe. If you disagree, please speak up.<br>
<br>
Working group participants are strongly encouraged to review and comment on=
 the document at this time.<br>
<br>
Thanks!<br>
<br>
/a<br>
<br>
______________________________<wbr>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/sipcore</a><=
br>
</blockquote>
</div>
<br>
</div>
</div>
</body>
</html>

--_000_BBF5DDFE515C3946BC18D733B20DAD233A84671BXMB122CNCrimnet_--


From nobody Wed Nov 30 14:59:16 2016
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From: "Ben Campbell" <ben@nostrum.com>
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Subject: [sipcore] Ben Campbell's Yes on charter-ietf-sipcore-03-00: (with COMMENT)
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Ben Campbell has entered the following ballot position for
charter-ietf-sipcore-03-00: Yes

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The document, along with other ballot positions, can be found here:
https://datatracker.ietf.org/doc/charter-ietf-sipcore/



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Do people think this needs external review?


