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Date: Thu, 1 May 2014 10:57:00 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, <sipcore@ietf.org>
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Subject: Re: [sipcore] Precondition and status confirmation requested by offerer
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Hi,

thanks for bringing this to my attention, Paul.

When the original offer/answer exchange (which includes the request for
"conf") completes the resources are already reserved. So, the answerer
does not need to send an additional offer. It will only need to send one
if the "threshold is crossed". That is, since the *initial* status has
already all the resources reserved, the threshold is not crossed. If the
resources were to be lost at some point, then an offer would naturally
need to be sent.

Cheers,

Gonzalo

On 30/04/2014 5:37 PM, Paul Kyzivat wrote:
> Ivo,
> 
> I just skimmed this, but before digging into it I think we need to
> settle on the proper place to discuss it. My first thought was that it
> is not really a sip thing and ought to be discussed in mmusic. But
> looking back I see the the drafts leading to 3312 were processed in the
> sip wg. Since sipcore and dispatch are the descendents of the sip wg,
> maybe sipcore isn't wrong. Normally I would say that mmusic would be the
> right place to deal with a SDP issue such as this. But mmusic is fully
> engaged in other things now, and this would likely get lost there. So
> for the time being I guess we might as well discuss it in sipcore.
> 
> Perhaps we can have an opinion on this by Gonzalo, since he was editor
> of 3312 and its update 4032.
> 
>     Thanks,
>     Paul
> 
> On 4/30/14 10:07 AM, Ivo Sedlacek wrote:
>> Hello,
>>
>> can you please help me with a question related to precondition handling
>> in RFC3312?
>>
>> Scenario:
>>
>> Bob receives an SDP offer O1 containing (not all lines are shown):
>>
>>        m=audio <port> <proto> <fmt>
>>
>>        a=curr:qos local sendrecv
>>
>>        a=curr:qos remote none
>>
>>        a=des:qos mandatory local sendrecv
>>
>>        a=des:qos mandatory remote sendrecv
>>
>>        a=conf:qos remote sendrecv
>>
>> Bob has needed resources and generates and sends an SDP answer A1
>> containing (not all lines are shown):
>>
>>        m=audio <port> <proto> <fmt>
>>
>>        a=curr:qos local sendrecv
>>
>>        a=curr:qos remote sendrecv
>>
>>        a=des:qos mandatory local sendrecv
>>
>>        a=des:qos mandatory remote sendrecv
>>
>> Question:
>>
>> Does "a=conf:qos remote sendrecv" included in SDP offer O1 mandate Bob
>> to send an additional SDP offer O2 immediately after sending the SDP
>> answer A1?
>>
>> RFC3312 seems to mandate sending such additional SDP offer since it
>> states:
>>
>> --------------------
>>
>>     The confirm-status attribute MAY be used in both offers and answers.
>>
>>     This attribute represents a threshold for the resource reservation.
>>
>>     >>When this threshold is reached or surpassed, the user agent MUST
>> send
>>
>>     an offer to the peer user agent, reflecting the new current status of
>>
>>     the media line as soon as allowed by the SIP offer/answer
>> rules<<.  If
>>
>>     this threshold is crossed again (e.g., the network stops providing
>>
>>     resources for the media stream), the user agent MUST send a new offer
>>
>>     as well, as soon as allowed by the SIP offer/answer rules.
>>
>>     >>If a peer has requested confirmation on a particular stream, an
>> agent
>>
>>     MUST mark that stream with a flag in its local status table.  When
>>
>>     all the rows with this flag have a "Current" value of "yes", the user
>>
>>     agent MUST send a new offer to the peer.<<  This offer will
>> contain the
>>
>>     current status of resource reservation in the current-status
>>
>>     attributes.  Later, if any of the rows with this flag transition to
>>
>>     "No", a new offer MUST be sent as well.
>>
>> --------------------
>>
>> However:
>>
>> - sending of such additional SDP offer O2 should not be necessary since
>> SDP answer A1 indicates that Bob's resources are reserved.
>>
>> - if order of the SDP answer A1 and the additional SDP offer O2 is
>> swapped on the way to the remote UA, then the remote UA rejects the
>> UPDATE request carrying the additional offer O2 with 491 (Request
>> Pending) response.
>>
>> The situation above occurs with third party call control. I am aware
>> that in regular two party calls, it should not occur.
>>
>> Kind regards
>>
>> Ivo Sedlacek
>>
>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> 


From nobody Tue May  6 04:21:07 2014
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From: Taisto Qvist <taisto.qvist@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Subject: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Hi,=20

I have been figure out how to interpret the MUST rule of rfc3261's chapter =
18.1.1. that says:

   If a request is within 200 bytes of the path MTU, or if it is larger
   than 1300 bytes and the path MTU is unknown, the request MUST be sent
   using an RFC 2914 [43] congestion controlled transport protocol, such
   as TCP.

in relation to DNS NAPTR priorities for selecting transport protocol.

The initial paragraph(just preceding the above rule) of 18.1.1 says that:

   "The user of the transport layer passes the client transport the request=
,=20
   an IP address, port, transport, and possibly TTL for multicast destinati=
ons."

Which to me implies that DNS lookup according to rfc3263 has been performed=
, and that the rule above  will/can OVERRIDE the NAPTR priorities?

Reading rfc3263 sometimes uses the phrase:=20

   However, another transport, such as TCP, MAY be used if the guidelines o=
f=20
   SIP mandate it for this particular request.
  =20
However, only directly in connection with URI's containing IP and/or port, =
or scenarios where NAPTR does not exist.

So, in a scenario where NAPTR *are* configured(for instance with UDP as hig=
hest priority), are you supposed to ignore the MUST rule in 18.1.1 above??

I am definitely in need of clarification, since 18.1.1 doesn't refer at all=
 to the procedures of rfc3263 and how they can impact this rule of protocol=
 selection.

Best Regards
Taisto Qvist


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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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> Which to me implies that DNS lookup according to rfc3263
> has been performed, and that the rule above  will/can
> OVERRIDE the NAPTR priorities?

Correct; the size rule overrides the already determined transport for the
request (unless CANCEL or non-2xx ACK).


> So, in a scenario where NAPTR *are* configured (for
> instance with UDP as highest priority), are you supposed
> to ignore the MUST rule in 18.1.1 above??

No.

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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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> From: Taisto Qvist <taisto.qvist@ericsson.com>
> 
> I have been figure out how to interpret the MUST rule of rfc3261's
> chapter 18.1.1. that says:
> 
>    If a request is within 200 bytes of the path MTU, or if it is larger
>    than 1300 bytes and the path MTU is unknown, the request MUST be sent
>    using an RFC 2914 [43] congestion controlled transport protocol, such
>    as TCP.
> 
> in relation to DNS NAPTR priorities for selecting transport protocol.

My interpretation of the rules is that the DNS lookup process
generates a prioritized list of addresses/ports/transports.  This list
is then filtered depending on the context.  If the rule in 18.1.1 is
triggered, all the elements of the list that have transport=UDP are
removed.  If not, and if a transport parameter was specified, elements
that do not match the transport parameter are removed.  (There is
further filtering if the URI is sips:, but I don't know them.)

Dale


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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "Dale R. Worley" <worley@ariadne.com>, Taisto Qvist <taisto.qvist@ericsson.com>
Thread-Topic: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Agree - it is still possible to take information from the NAPTR record and =
comply with the RFC 3261 requirement, e.g. in selecting TCP over SCTP.

Regards

Keith=20

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of=20
> Dale R. Worley
> Sent: 06 May 2014 14:48
> To: Taisto Qvist
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of=20
> 1300 bytes.
>=20
> > From: Taisto Qvist <taisto.qvist@ericsson.com>
> >=20
> > I have been figure out how to interpret the MUST rule of rfc3261's=20
> > chapter 18.1.1. that says:
> >=20
> >    If a request is within 200 bytes of the path MTU, or if=20
> it is larger
> >    than 1300 bytes and the path MTU is unknown, the request=20
> MUST be sent
> >    using an RFC 2914 [43] congestion controlled transport=20
> protocol, such
> >    as TCP.
> >=20
> > in relation to DNS NAPTR priorities for selecting transport=20
> protocol.
>=20
> My interpretation of the rules is that the DNS lookup process=20
> generates a prioritized list of addresses/ports/transports. =20
> This list is then filtered depending on the context.  If the=20
> rule in 18.1.1 is triggered, all the elements of the list=20
> that have transport=3DUDP are removed.  If not, and if a=20
> transport parameter was specified, elements that do not match=20
> the transport parameter are removed.  (There is further=20
> filtering if the URI is sips:, but I don't know them.)
>=20
> Dale
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =


From nobody Tue May  6 07:40:55 2014
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Date: Tue, 6 May 2014 10:40:46 -0400
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To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, "Dale R. Worley" <worley@ariadne.com>,  Taisto Qvist <taisto.qvist@ericsson.com>, sipcore@ietf.org
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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Hi,

I disagree with both interpretations.  My interpretation is that the RFC
3263 NAPTR/SRV process only selects a single transport type; UDP was
selected within this situation.  The RFC 3261 size rule overrides the
selected transport type (but still uses the address and port associated
with UDP) and provides a fallback to the selected transport type.

This is the reason for the following RFC 3261 section 18.2.1 snippet:

"For any port and interface
 that a server listens on for UDP, it MUST listen on that same port
 and interface for TCP.  This is because a message may need to be sent
 using TCP, rather than UDP, if it is too large."


> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DRAGE,
> Keith (Keith)
> Sent: Tuesday, May 06, 2014 10:08 AM
> To: Dale R. Worley; Taisto Qvist
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
>
> Agree - it is still possible to take information from the NAPTR record
> and comply with the RFC 3261 requirement, e.g. in selecting TCP over
> SCTP.
>
> Regards
>
> Keith
>
> > -----Original Message-----
> > From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of
> > Dale R. Worley
> > Sent: 06 May 2014 14:48
> > To: Taisto Qvist
> > Cc: sipcore@ietf.org
> > Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of
> > 1300 bytes.
> >
> > > From: Taisto Qvist <taisto.qvist@ericsson.com>
> > >
> > > I have been figure out how to interpret the MUST rule of rfc3261's
> > > chapter 18.1.1. that says:
> > >
> > >    If a request is within 200 bytes of the path MTU, or if
> > it is larger
> > >    than 1300 bytes and the path MTU is unknown, the request
> > MUST be sent
> > >    using an RFC 2914 [43] congestion controlled transport
> > protocol, such
> > >    as TCP.
> > >
> > > in relation to DNS NAPTR priorities for selecting transport
> > protocol.
> >
> > My interpretation of the rules is that the DNS lookup process
> > generates a prioritized list of addresses/ports/transports.
> > This list is then filtered depending on the context.  If the
> > rule in 18.1.1 is triggered, all the elements of the list
> > that have transport=UDP are removed.  If not, and if a
> > transport parameter was specified, elements that do not match
> > the transport parameter are removed.  (There is further
> > filtering if the URI is sips:, but I don't know them.)
> >
> > Dale
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

-- 

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From nobody Tue May  6 08:27:52 2014
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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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On 5/6/14 9:47 AM, Dale R. Worley wrote:
>> From: Taisto Qvist <taisto.qvist@ericsson.com>
>>
>> I have been figure out how to interpret the MUST rule of rfc3261's
>> chapter 18.1.1. that says:
>>
>>     If a request is within 200 bytes of the path MTU, or if it is larger
>>     than 1300 bytes and the path MTU is unknown, the request MUST be sent
>>     using an RFC 2914 [43] congestion controlled transport protocol, such
>>     as TCP.
>>
>> in relation to DNS NAPTR priorities for selecting transport protocol.
>
> My interpretation of the rules is that the DNS lookup process
> generates a prioritized list of addresses/ports/transports.  This list
> is then filtered depending on the context.  If the rule in 18.1.1 is
> triggered, all the elements of the list that have transport=UDP are
> removed.  If not, and if a transport parameter was specified, elements
> that do not match the transport parameter are removed.  (There is
> further filtering if the URI is sips:, but I don't know them.)

This is consistent with my feeling of how it *ought* to work.

Whether that is what was *intended* is a slightly different question, 
and one I don't know the answer to.

	Thanks,
	Paul


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From: Taisto Qvist <taisto.qvist@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Thanks Keith, but I was more thinking about the situation that NAPTR config=
uration prioritizes UDP, and then the message is larger than 1300 bytes.

In this case it I don't know which rule is more important? When NAPTR confi=
g "disagrees" with rfc3261, which takes precedence?

Regards
Taisto

-----Original Message-----
From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]=20
Sent: den 6 maj 2014 16:08
To: Dale R. Worley; Taisto Qvist
Cc: sipcore@ietf.org
Subject: RE: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.

Agree - it is still possible to take information from the NAPTR record and =
comply with the RFC 3261 requirement, e.g. in selecting TCP over SCTP.

Regards

Keith=20

> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Dale R.=20
> Worley
> Sent: 06 May 2014 14:48
> To: Taisto Qvist
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of
> 1300 bytes.
>=20
> > From: Taisto Qvist <taisto.qvist@ericsson.com>
> >=20
> > I have been figure out how to interpret the MUST rule of rfc3261's=20
> > chapter 18.1.1. that says:
> >=20
> >    If a request is within 200 bytes of the path MTU, or if
> it is larger
> >    than 1300 bytes and the path MTU is unknown, the request
> MUST be sent
> >    using an RFC 2914 [43] congestion controlled transport
> protocol, such
> >    as TCP.
> >=20
> > in relation to DNS NAPTR priorities for selecting transport
> protocol.
>=20
> My interpretation of the rules is that the DNS lookup process=20
> generates a prioritized list of addresses/ports/transports.
> This list is then filtered depending on the context.  If the rule in=20
> 18.1.1 is triggered, all the elements of the list that have=20
> transport=3DUDP are removed.  If not, and if a transport parameter was=20
> specified, elements that do not match the transport parameter are=20
> removed.  (There is further filtering if the URI is sips:, but I don't=20
> know them.)
>=20
> Dale
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20


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From: Taisto Qvist <taisto.qvist@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Hi,=20

This is somewhat similar to what my interpretation has been before as well.=
  But after some discussions with Brett on sip-implementors, I've realized =
that I've missed the rules of 2915/NAPTR Order values, which quite(?) clear=
ly says that once you've selected a NAPTR RR with a specific Order value, t=
hat *it*.=20

      Low numbers are processed before high numbers, and once a
      NAPTR is found whose rule "matches" the target, the client MUST
      NOT consider any NAPTRs with a higher value for order....

So at least in the case of different Order values for the different protoco=
ls, you should only use the first match. If the Order values are the same t=
hough.....?

But still...it feels like the two algorithms/rules collide a bit, or are at=
 least in need of clarification. Especially that rfc 3263, chapter 4.1 only=
 refers to the size-rule as a *MAY* in scenarios where we don't have or use=
 NAPTR.=20

A MAY requirement, referring to a MUST requirement.

Regards
Taisto Qvist

-----Original Message-----
From: Dale R. Worley [mailto:worley@ariadne.com]=20
Sent: den 6 maj 2014 15:48
To: Taisto Qvist
Cc: sipcore@ietf.org
Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.

> From: Taisto Qvist <taisto.qvist@ericsson.com>
>=20
> I have been figure out how to interpret the MUST rule of rfc3261's=20
> chapter 18.1.1. that says:
>=20
>    If a request is within 200 bytes of the path MTU, or if it is larger
>    than 1300 bytes and the path MTU is unknown, the request MUST be sent
>    using an RFC 2914 [43] congestion controlled transport protocol, such
>    as TCP.
>=20
> in relation to DNS NAPTR priorities for selecting transport protocol.

My interpretation of the rules is that the DNS lookup process generates a p=
rioritized list of addresses/ports/transports.  This list is then filtered =
depending on the context.  If the rule in 18.1.1 is triggered, all the elem=
ents of the list that have transport=3DUDP are removed.  If not, and if a t=
ransport parameter was specified, elements that do not match the transport =
parameter are removed.  (There is further filtering if the URI is sips:, bu=
t I don't know them.)

Dale


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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Taisto Qvist <taisto.qvist@ericsson.com>, "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300 bytes.
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Exactly as others have indicated. If the length exceeds the limit then the =
absolute RFC 3261 requirement (MUST) applies, i.e. not use UDP.=20

I am aware of implementations that can be set to override this, but they ar=
e non-conformant to the RFC.

Regards

Keith

> -----Original Message-----
> From: Taisto Qvist [mailto:taisto.qvist@ericsson.com]=20
> Sent: 07 May 2014 07:52
> To: DRAGE, Keith (Keith); Dale R. Worley
> Cc: sipcore@ietf.org
> Subject: RE: [sipcore] NAPTR priorities vs rfc3261s rule of=20
> 1300 bytes.
>=20
> Thanks Keith, but I was more thinking about the situation=20
> that NAPTR configuration prioritizes UDP, and then the=20
> message is larger than 1300 bytes.
>=20
> In this case it I don't know which rule is more important?=20
> When NAPTR config "disagrees" with rfc3261, which takes precedence?
>=20
> Regards
> Taisto
>=20
> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
> Sent: den 6 maj 2014 16:08
> To: Dale R. Worley; Taisto Qvist
> Cc: sipcore@ietf.org
> Subject: RE: [sipcore] NAPTR priorities vs rfc3261s rule of=20
> 1300 bytes.
>=20
> Agree - it is still possible to take information from the=20
> NAPTR record and comply with the RFC 3261 requirement, e.g.=20
> in selecting TCP over SCTP.
>=20
> Regards
>=20
> Keith=20
>=20
> > -----Original Message-----
> > From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf=20
> Of Dale R.=20
> > Worley
> > Sent: 06 May 2014 14:48
> > To: Taisto Qvist
> > Cc: sipcore@ietf.org
> > Subject: Re: [sipcore] NAPTR priorities vs rfc3261s rule of 1300=20
> > bytes.
> >=20
> > > From: Taisto Qvist <taisto.qvist@ericsson.com>
> > >=20
> > > I have been figure out how to interpret the MUST rule of=20
> rfc3261's=20
> > > chapter 18.1.1. that says:
> > >=20
> > >    If a request is within 200 bytes of the path MTU, or if
> > it is larger
> > >    than 1300 bytes and the path MTU is unknown, the request
> > MUST be sent
> > >    using an RFC 2914 [43] congestion controlled transport
> > protocol, such
> > >    as TCP.
> > >=20
> > > in relation to DNS NAPTR priorities for selecting transport
> > protocol.
> >=20
> > My interpretation of the rules is that the DNS lookup process=20
> > generates a prioritized list of addresses/ports/transports.
> > This list is then filtered depending on the context.  If the rule in
> > 18.1.1 is triggered, all the elements of the list that have=20
> > transport=3DUDP are removed.  If not, and if a transport=20
> parameter was=20
> > specified, elements that do not match the transport parameter are=20
> > removed.  (There is further filtering if the URI is sips:,=20
> but I don't=20
> > know them.)
> >=20
> > Dale
> >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> =


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 15 May 2014 11:35:26 +0200
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Subject: [sipcore] Trickle ICE is a MUST for SIP
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Hi,

IMHO we need Trickle ICE for SIP ASAP. Otherwise any custom protocol
designed by people with less RTC knowledge than SIP folks will be
better than SIP when ICE is in use (of course I mean WebRTC).

I actively try to argue that SIP is suitable for browsers and WebRTC,
but this is no longer true if Trickle ICE is not feasible with SIP.
Having a VPN or a virtual interface (VMware, VirtualBox, etc) in the
computer makes SIP a bad choice because the ICE triggering process
takes too long (timeout due to virtual interfaces).

Is there interest in draft-ivov-mmusic-trickle-ice-sip [*] ? Really, I
don't think we can wait one year for a standard. There are lot of
people using SIP for WebRTC. Trickle ICE is a MUST.

Best regards.



[*] http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-01


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


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Hi,

Will join this topic in the MMUSIC WG where the draft is being proposed.

Regards.

2014-05-15 11:35 GMT+02:00 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> Hi,
>
> IMHO we need Trickle ICE for SIP ASAP. Otherwise any custom protocol
> designed by people with less RTC knowledge than SIP folks will be
> better than SIP when ICE is in use (of course I mean WebRTC).
>
> I actively try to argue that SIP is suitable for browsers and WebRTC,
> but this is no longer true if Trickle ICE is not feasible with SIP.
> Having a VPN or a virtual interface (VMware, VirtualBox, etc) in the
> computer makes SIP a bad choice because the ICE triggering process
> takes too long (timeout due to virtual interfaces).
>
> Is there interest in draft-ivov-mmusic-trickle-ice-sip [*] ? Really, I
> don't think we can wait one year for a standard. There are lot of
> people using SIP for WebRTC. Trickle ICE is a MUST.
>
> Best regards.
>
>
>
> [*] http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-01
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>



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


From nobody Thu May 15 08:47:33 2014
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Iñaki,

O/A was carefully partitioned so that the basic O/A mechanism is 
independent of sip, so RFC 3264 could be handled by mmusic, while the 
means of conveying O/A in sip is defined in 3261 that was handled by the 
sip wg.

We have similar turf issues for trickle ICE. The basic mechanism for 
trickle ICE is independent of SIP, but the mechanism for conveying it in 
sip is a different matter. So the conveyance of trickle ICE in sip via 
INFO or most likely should be handled in DISPATCH or SIPCORE, but it 
*may* be most appropriate to handle the actual message content 
describing the trickling in MMUSIC.

It is not necessary to process info packages in sipcore. In principle 
one could be done in mmusic, if people are comfortable with that.

I know this is all bureaucratic BS, but better to sort it out up front 
than later on. OTOH mmusic is very busy, so doing the work elsewhere 
might be quicker.

	Thanks,
	Paul

On 5/15/14 7:36 AM, Iñaki Baz Castillo wrote:
> Hi,
>
> Will join this topic in the MMUSIC WG where the draft is being proposed.
>
> Regards.
>
> 2014-05-15 11:35 GMT+02:00 Iñaki Baz Castillo <ibc@aliax.net>:
>> Hi,
>>
>> IMHO we need Trickle ICE for SIP ASAP. Otherwise any custom protocol
>> designed by people with less RTC knowledge than SIP folks will be
>> better than SIP when ICE is in use (of course I mean WebRTC).
>>
>> I actively try to argue that SIP is suitable for browsers and WebRTC,
>> but this is no longer true if Trickle ICE is not feasible with SIP.
>> Having a VPN or a virtual interface (VMware, VirtualBox, etc) in the
>> computer makes SIP a bad choice because the ICE triggering process
>> takes too long (timeout due to virtual interfaces).
>>
>> Is there interest in draft-ivov-mmusic-trickle-ice-sip [*] ? Really, I
>> don't think we can wait one year for a standard. There are lot of
>> people using SIP for WebRTC. Trickle ICE is a MUST.
>>
>> Best regards.
>>
>>
>>
>> [*] http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-01
>>
>>
>> --
>> Iñaki Baz Castillo
>> <ibc@aliax.net>
>
>
>


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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
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To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Trickle ICE is a MUST for SIP
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Subject: Re: [sipcore] Trickle ICE is a MUST for SIP
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From: "Olle E. Johansson" <oej@edvina.net>
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Cc: sipcore@ietf.org, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] Trickle ICE is a MUST for SIP
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On 15 May 2014, at 17:47, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> I=F1aki,
>=20
> O/A was carefully partitioned so that the basic O/A mechanism is =
independent of sip, so RFC 3264 could be handled by mmusic, while the =
means of conveying O/A in sip is defined in 3261 that was handled by the =
sip wg.
>=20
> We have similar turf issues for trickle ICE. The basic mechanism for =
trickle ICE is independent of SIP, but the mechanism for conveying it in =
sip is a different matter. So the conveyance of trickle ICE in sip via =
INFO or most likely should be handled in DISPATCH or SIPCORE, but it =
*may* be most appropriate to handle the actual message content =
describing the trickling in MMUSIC.
>=20
> It is not necessary to process info packages in sipcore. In principle =
one could be done in mmusic, if people are comfortable with that.
>=20
> I know this is all bureaucratic BS, but better to sort it out up front =
than later on. OTOH mmusic is very busy, so doing the work elsewhere =
might be quicker.

I propose we get it dispatched to SIPCORE and handle it quickly.
I=F1aki - please resend your message to dispatch and we'll handle it =
from there.

/O
>=20
> 	Thanks,
> 	Paul
>=20
> On 5/15/14 7:36 AM, I=F1aki Baz Castillo wrote:
>> Hi,
>>=20
>> Will join this topic in the MMUSIC WG where the draft is being =
proposed.
>>=20
>> Regards.
>>=20
>> 2014-05-15 11:35 GMT+02:00 I=F1aki Baz Castillo <ibc@aliax.net>:
>>> Hi,
>>>=20
>>> IMHO we need Trickle ICE for SIP ASAP. Otherwise any custom protocol
>>> designed by people with less RTC knowledge than SIP folks will be
>>> better than SIP when ICE is in use (of course I mean WebRTC).
>>>=20
>>> I actively try to argue that SIP is suitable for browsers and =
WebRTC,
>>> but this is no longer true if Trickle ICE is not feasible with SIP.
>>> Having a VPN or a virtual interface (VMware, VirtualBox, etc) in the
>>> computer makes SIP a bad choice because the ICE triggering process
>>> takes too long (timeout due to virtual interfaces).
>>>=20
>>> Is there interest in draft-ivov-mmusic-trickle-ice-sip [*] ? Really, =
I
>>> don't think we can wait one year for a standard. There are lot of
>>> people using SIP for WebRTC. Trickle ICE is a MUST.
>>>=20
>>> Best regards.
>>>=20
>>>=20
>>>=20
>>> [*] http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-01
>>>=20
>>>=20
>>> --
>>> I=F1aki Baz Castillo
>>> <ibc@aliax.net>
>>=20
>>=20
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>, <sipcore@ietf.org>
References: <CALiegfmqevE3t7otsyzNa5b+VUrA3xUmt76GuSGQ1y4ASpnj8w@mail.gmail.com> <CALiegf=uV3Ruc+5Sp035=M__OOWaQWLqKmQg9BRUd+-7VQ_OdA@mail.gmail.com> <5374E17F.8090806@alum.mit.edu> <7594FB04B1934943A5C02806D1A2204B1D2FDE35@ESESSMB209.ericsson.se> <7594FB04B1934943A5C02806D1A2204B1D2FDEBC@ESESSMB209.ericsson.se>
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Date: Mon, 19 May 2014 04:34:41 +0530
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Subject: Re: [sipcore] Trickle ICE is a MUST for SIP
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Hi all,

I agree that Trickle ICE is a special case but it should not be designed =
in a way to break the existing O/A model. I have explained the usecases =
in MMUSIC WG wherein INFO creates race condition within O/A =
(http://www.ietf.org/mail-archive/web/mmusic/current/msg13287.html). =
Irrespective of WG where this usecase is developed, this issue has to be =
resolved.

Thanks
Partha


> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer
> Holmberg
> Sent: Thursday, May 15, 2014 11:05 PM
> To: Christer Holmberg; Paul Kyzivat; sipcore@ietf.org
> Subject: Re: [sipcore] Trickle ICE is a MUST for SIP
>=20
> In addition, an Info Package doesn't even have to be specified in =
IETF.
>=20
> Trickle ICE is obviously a special case, though, and not just some
> application data sent between two entities :)
>=20
> Regards,
>=20
> Christer
>=20
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer
> Holmberg
> Sent: 15 May 2014 20:28
> To: Paul Kyzivat; sipcore@ietf.org
> Subject: Re: [sipcore] Trickle ICE is a MUST for SIP
>=20
> It is true that MMUSIC is very busy during the meetings, but between
> the meetings there is lots of space available on the mailing list...
>=20
> Regards,
>=20
> Christer
>=20
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul
> Kyzivat
> Sent: 15 May 2014 18:47
> To: sipcore@ietf.org
> Subject: Re: [sipcore] Trickle ICE is a MUST for SIP
>=20
> I=C3=B1aki,
>=20
> O/A was carefully partitioned so that the basic O/A mechanism is
> independent of sip, so RFC 3264 could be handled by mmusic, while the
> means of conveying O/A in sip is defined in 3261 that was handled by
> the sip wg.
>=20
> We have similar turf issues for trickle ICE. The basic mechanism for
> trickle ICE is independent of SIP, but the mechanism for conveying it
> in sip is a different matter. So the conveyance of trickle ICE in sip
> via INFO or most likely should be handled in DISPATCH or SIPCORE, but
> it
> *may* be most appropriate to handle the actual message content
> describing the trickling in MMUSIC.
>=20
> It is not necessary to process info packages in sipcore. In principle
> one could be done in mmusic, if people are comfortable with that.
>=20
> I know this is all bureaucratic BS, but better to sort it out up front
> than later on. OTOH mmusic is very busy, so doing the work elsewhere
> might be quicker.
>=20
> 	Thanks,
> 	Paul
>=20
> On 5/15/14 7:36 AM, I=C3=B1aki Baz Castillo wrote:
> > Hi,
> >
> > Will join this topic in the MMUSIC WG where the draft is being
> proposed.
> >
> > Regards.
> >
> > 2014-05-15 11:35 GMT+02:00 I=C3=B1aki Baz Castillo <ibc@aliax.net>:
> >> Hi,
> >>
> >> IMHO we need Trickle ICE for SIP ASAP. Otherwise any custom =
protocol
> >> designed by people with less RTC knowledge than SIP folks will be
> >> better than SIP when ICE is in use (of course I mean WebRTC).
> >>
> >> I actively try to argue that SIP is suitable for browsers and
> WebRTC,
> >> but this is no longer true if Trickle ICE is not feasible with SIP.
> >> Having a VPN or a virtual interface (VMware, VirtualBox, etc) in =
the
> >> computer makes SIP a bad choice because the ICE triggering process
> >> takes too long (timeout due to virtual interfaces).
> >>
> >> Is there interest in draft-ivov-mmusic-trickle-ice-sip [*] ? =
Really,
> >> I don't think we can wait one year for a standard. There are lot of
> >> people using SIP for WebRTC. Trickle ICE is a MUST.
> >>
> >> Best regards.
> >>
> >>
> >>
> >> [*] http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-01
> >>
> >>
> >> --
> >> I=C3=B1aki Baz Castillo
> >> <ibc@aliax.net>
> >
> >
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Goren, Ofer (Ofer)" <oferg@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: SIP response is larger than MTU
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Hi

I saw this question raised @ 2006, but I did not see any follow up or a fin=
al decision. There was a draft suggested that became dead  @ 2006 as well.
Anyway, what if I send an empty INVITE, for example, using UDP, and the res=
ulted response with SDP offer, exceeds the MTU? I cannot switch to TCP for =
the response, as it needs to follow the request's VIAs.

Thank you,

Ofer.

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<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I saw this question raised @ 2006, but I did not see=
 any follow up or a final decision. There was a draft suggested that became=
 dead &nbsp;@ 2006 as well.<o:p></o:p></p>
<p class=3D"MsoNormal">Anyway, what if I send an empty INVITE, for example,=
 using UDP, and the resulted response with SDP offer, exceeds the MTU? I ca=
nnot switch to TCP for the response, as it needs to follow the request&#821=
7;s VIAs.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thank you,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Ofer.<o:p></o:p></p>
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From nobody Mon May 19 04:18:47 2014
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From: Brett Tate <brett@broadsoft.com>
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To: "Goren, Ofer (Ofer)" <oferg@avaya.com>, sipcore@ietf.org
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Subject: Re: [sipcore] SIP response is larger than MTU
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> what if I send an empty INVITE, for example, using
> UDP, and the resulted response with SDP offer,
> exceeds the MTU? I cannot switch to TCP for the
> response, as it needs to follow the request's VIAs.

RFC 3261 section 18.1.1 discusses that 1) UDP fragmentation will occur for
SIP messages over (approximately) 1300 bytes, and 2) implementations MUST
be able to handle 65535 byte SIP messages sent over UDP.

The fragmented UDP response can typically be successfully
delivered/handled unless someone has disabled fragmentation support or
there was excessive fragmentation.  However as you noticed, the attempt to
deploy "preferring UDP and switch to TCP only for large messages" does
have limitations.

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--047d7b3a982ee5d65604fa54822f
Content-Type: text/plain; charset=ISO-8859-1

Hi,

I have just submitted a draft that has some initial thoughts for an
authorization framework for SIP that is based on OAuth.
The draft is still missing lots of details, but I think it has enough
information to allow us to start the discussion on the mailing list.

We would appreciate it if people review the document and provide us with
their thoughts.

Regards,
 Rifaat



---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Mon, May 26, 2014 at 5:37 PM
Subject: New Version Notification for draft-yusef-sipcore-sip-oauth-00.txt
To: Victor Pascual <victor.pascual@quobis.com>, Rifaat Shekh-Yusef <
rifaat.ietf@gmail.com>



A new version of I-D, draft-yusef-sipcore-sip-oauth-00.txt
has been successfully submitted by Rifaat Shekh-Yusef and posted to the
IETF repository.

Name:           draft-yusef-sipcore-sip-oauth
Revision:       00
Title:          The Session Initiation Protocol (SIP) OAuth
Document date:  2014-05-26
Group:          Individual Submission
Pages:          18
URL:
http://www.ietf.org/internet-drafts/draft-yusef-sipcore-sip-oauth-00.txt
Status:
https://datatracker.ietf.org/doc/draft-yusef-sipcore-sip-oauth/
Htmlized:       http://tools.ietf.org/html/draft-yusef-sipcore-sip-oauth-00


Abstract:
   The Session Initiation Protocol (SIP) uses a challenge-response
   framework to authenticate the user, but it does not have an
   authorization framework to control the user's access to various
   services in the system.

   This document defines an authorization framework for SIP that is
   based on the OAuth 2.0 framework.





Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat

--047d7b3a982ee5d65604fa54822f
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi,<div><br></div><div>I have just submitted a draft that =
has some initial thoughts for an authorization framework for SIP that is ba=
sed on OAuth.</div><div>The draft is still missing lots of details, but I t=
hink it has enough information to allow us to start the discussion on the m=
ailing list.</div>
<div><br></div><div>We would appreciate it if people review the document an=
d provide us with their thoughts.</div><div><br></div><div>Regards,</div><d=
iv>=A0Rifaat</div><div><br></div><div><br><br><div class=3D"gmail_quote">--=
-------- Forwarded message ----------<br>
From: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>&gt;</span><br>=
Date: Mon, May 26, 2014 at 5:37 PM<br>Subject: New Version Notification for=
 draft-yusef-sipcore-sip-oauth-00.txt<br>
To: Victor Pascual &lt;<a href=3D"mailto:victor.pascual@quobis.com">victor.=
pascual@quobis.com</a>&gt;, Rifaat Shekh-Yusef &lt;<a href=3D"mailto:rifaat=
.ietf@gmail.com">rifaat.ietf@gmail.com</a>&gt;<br><br><br><br>
A new version of I-D, draft-yusef-sipcore-sip-oauth-00.txt<br>
has been successfully submitted by Rifaat Shekh-Yusef and posted to the<br>
IETF repository.<br>
<br>
Name: =A0 =A0 =A0 =A0 =A0 draft-yusef-sipcore-sip-oauth<br>
Revision: =A0 =A0 =A0 00<br>
Title: =A0 =A0 =A0 =A0 =A0The Session Initiation Protocol (SIP) OAuth<br>
Document date: =A02014-05-26<br>
Group: =A0 =A0 =A0 =A0 =A0Individual Submission<br>
Pages: =A0 =A0 =A0 =A0 =A018<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/internet-drafts/=
draft-yusef-sipcore-sip-oauth-00.txt" target=3D"_blank">http://www.ietf.org=
/internet-drafts/draft-yusef-sipcore-sip-oauth-00.txt</a><br>
Status: =A0 =A0 =A0 =A0 <a href=3D"https://datatracker.ietf.org/doc/draft-y=
usef-sipcore-sip-oauth/" target=3D"_blank">https://datatracker.ietf.org/doc=
/draft-yusef-sipcore-sip-oauth/</a><br>
Htmlized: =A0 =A0 =A0 <a href=3D"http://tools.ietf.org/html/draft-yusef-sip=
core-sip-oauth-00" target=3D"_blank">http://tools.ietf.org/html/draft-yusef=
-sipcore-sip-oauth-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0The Session Initiation Protocol (SIP) uses a challenge-response<br>
=A0 =A0framework to authenticate the user, but it does not have an<br>
=A0 =A0authorization framework to control the user&#39;s access to various<=
br>
=A0 =A0services in the system.<br>
<br>
=A0 =A0This document defines an authorization framework for SIP that is<br>
=A0 =A0based on the OAuth 2.0 framework.<br>
<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div></div>

--047d7b3a982ee5d65604fa54822f--


From nobody Wed May 28 04:55:58 2014
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From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Precondition and status confirmation in regular two party calls
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Subject: [sipcore] Precondition and status confirmation in regular two party calls
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Hello,

can you please help me with a one more question related to precondition han=
dling in RFC3312?

RFC3312 describes usage of confirm-status attribute - in summary, if a UA r=
eceives the confirm-status attribute, the UA shall send UPDATE once the UA =
finishes resource reservation.

However, RFC3312 does not explicitly state anything about sending UPDATE if=
 the confirm-status attribute has NOT been received. If the UA does *NOT* r=
eceive confirm-status attribute, and if the UA sent the UPDATE request once=
 the UA finishes resource reservation, it may cause unnecessary race condit=
ions since both UAs can send UPDATEs at about the same time.

Question:
-----------
I assume that if a UA does *NOT* receive confirm-status attribute, then whe=
n the UA finishes the resource reservation, the UA SHOULD NOT send UPDATE r=
equest. Is this correct understanding?
-----------

Thanks for clarification.

Kind regards

Ivo Sedlacek


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Hello,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">can you please help me with a one more qu=
estion related to precondition handling in RFC3312?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">RFC3312 describes usage of confirm-status=
 attribute - in summary, if a UA receives the confirm-status attribute, the=
 UA shall send UPDATE once the UA finishes resource reservation.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">However, RFC3312 does not explicitly stat=
e anything about sending UPDATE if the confirm-status attribute has NOT bee=
n received. If the UA does *NOT* receive confirm-status
 attribute, and if the UA sent the UPDATE request once the UA finishes reso=
urce reservation, it may cause unnecessary race conditions since both UAs c=
an send UPDATEs at about the same time.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Question:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">-----------<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">I assume that if a UA does *NOT* receive =
confirm-status attribute, then when the UA finishes the resource reservatio=
n, the UA SHOULD NOT send UPDATE request. Is this correct
 understanding?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">-----------<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Thanks for clarification.<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Kind regards<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Ivo Sedlacek<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_39B5E4D390E9BD4890E2B31079006101126DDBABESESSMB301erics_--


From nobody Wed May 28 11:32:00 2014
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Date: Wed, 28 May 2014 14:31:52 -0400
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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
In-reply-to: <39B5E4D390E9BD4890E2B31079006101126DDBAB@ESESSMB301.ericsson.se> (ivo.sedlacek@ericsson.com)
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Precondition and status confirmation in regular two party calls
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> From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>

> However, RFC3312 does not explicitly state anything about sending
> UPDATE if the confirm-status attribute has NOT been received. If the
> UA does *NOT* receive confirm-status attribute, and if the UA sent
> the UPDATE request once the UA finishes resource reservation, it may
> cause unnecessary race conditions since both UAs can send UPDATEs at
> about the same time.

The handling of race conditions regarding UPDATE is described in RFC
3331 section 5.2:

   A UAS that receives an UPDATE before it has generated a final
   response to a previous UPDATE on the same dialog MUST return a 500
   response to the new UPDATE, and MUST include a Retry-After header
   field with a randomly chosen value between 0 and 10 seconds.

   If an UPDATE is received that contains an offer, and the UAS has
   generated an offer (in an UPDATE, PRACK or INVITE) to which it has
   not yet received an answer, the UAS MUST reject the UPDATE with a 491
   response.  Similarly, if an UPDATE is received that contains an
   offer, and the UAS has received an offer (in an UPDATE, PRACK, or
   INVITE) to which it has not yet generated an answer, the UAS MUST
   reject the UPDATE with a 500 response, and MUST include a Retry-After
   header field with a randomly chosen value between 0 and 10 seconds.
   [...]

> Question:
> -----------
> I assume that if a UA does *NOT* receive confirm-status attribute,
> then when the UA finishes the resource reservation, the UA SHOULD
> NOT send UPDATE request. Is this correct understanding?
> -----------

In principle, a UA can send an UPDATE at any time when it is not
forbidden to.  RFC 3311 section 5.1 gives some guidelines:

   The UPDATE request is constructed as would any other request within
   an existing dialog, as described in Section 12.2.1 of RFC 3261.  It
   MAY be sent for both early and confirmed dialogs, and MAY be sent by
   either caller or callee.  Although UPDATE can be used on confirmed
   dialogs, it is RECOMMENDED that a re-INVITE be used instead.  This is
   because an UPDATE needs to be answered immediately, ruling out the
   possibility of user approval.  Such approval will frequently be
   needed, and is possible with a re-INVITE.

Dale


From nobody Wed May 28 17:22:46 2014
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From: "MORTON, ALFRED C (AL)" <acmorton@att.com>
To: "bmwg@ietf.org" <bmwg@ietf.org>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 28 May 2014 20:22:28 -0400
Thread-Topic: WGLC: draft-ietf-bmwg-sip-bench-term and -meth
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Subject: [sipcore] WGLC: draft-ietf-bmwg-sip-bench-term and -meth
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BMWG (and SIPCORE):

A WG Last Call period for the Internet-Drafts on SIP Device
Benchmarking:

   http://datatracker.ietf.org/doc/draft-ietf-bmwg-sip-bench-term/
   http://datatracker.ietf.org/doc/draft-ietf-bmwg-sip-bench-meth/

will be open from 28 May 2014 through 11 June 2014.

These drafts are continuing the BMWG Last Call Process. See
http://www1.ietf.org/mail-archive/web/bmwg/current/msg00846.html
The first WGLC was completed on 5 April 2010 with comments.
The second WGLC was completed on 18 May 2012 with comments.
The third WGLC was completed on 10 Dec 2012 with comments.
An IETF Last Call followed, and completed on 30 Jan 2013 with comments.

Please read and express your opinion on whether or not these
Internet-Drafts should be forwarded to the Area Directors for
publication as Informational RFCs.  Send your comments
to this list or acmorton@att.com and sbanks@akamai.com

Al
bmwg co-chair =


From nobody Thu May 29 06:51:08 2014
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From: Michael Procter <michael@voip.co.uk>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
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Hi Rifaat,

On 26 May 2014 22:46, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com> wrote:
> We would appreciate it if people review the document and provide us with
> their thoughts.

Would you be able to provide a couple of motivating use cases?  The
ones I've come up with either aren't fully met by the outlines in this
document, or might be achieved more easily with existing pieces of
protocol.

A couple of typos:
Section 1, last paragraph:
"Single Sing-On" should be "Single Sign-On"

Section 3.4, last paragraph:
"with toke, token refresh, and the master-key"
should be
"with token, refresh_token, and the master-key"

Regards,

Michael


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From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Precondition and status confirmation in regular two party	calls
Thread-Index: AQHPeqMZ2vpklsp7EkukvXxbyikpPZtXnK2Q
Date: Thu, 29 May 2014 14:33:22 +0000
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References: <39B5E4D390E9BD4890E2B31079006101126DDBAB@ESESSMB301.ericsson.se> <201405281831.s4SIVqf4026532@hobgoblin.ariadne.com>
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Subject: Re: [sipcore] Precondition and status confirmation in regular two party	calls
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Hello,

The issue is not the race condition as such.  As you say, UPDATE is allowed=
 to be sent at any time and there are procedures to handle UPDATEs being se=
nt by both UAs at about the same time.

The question was rather: Is it advisable for a UA to send an UPDATE indicat=
ing that its resources are available if confirm-status attribute was *NOT* =
received?

Sending such non-requested UPDATE can generate unnecessary signalling (if b=
oth UAs reserve resources at about the same time, both UAs send UPDATE at a=
bout the same time and 491 would be generated) and call setup delay (since =
at last one of the UPDATEs will need to be repeated).

The whole reason for the confirm-status attribute is to request a confirmat=
ion when resources becomes available. So if the confirm-status attribute is=
 *NOT* present, when resources becomes available, a UA should refrain from =
sending UPDATE (unless there is another reason, not related to resource res=
ervation, for doing so).

Do you agree?

Kind regards

Ivo Sedlacek

This Communication is Confidential. We only send and receive email on the b=
asis of the terms set out at www.ericsson.com/email_disclaimer=20


-----Original Message-----
From: Dale R. Worley [mailto:worley@ariadne.com]=20
Sent: 28. kv=ECtna 2014 20:32
To: Ivo Sedlacek
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Precondition and status confirmation in regular two =
party calls

> From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>

> However, RFC3312 does not explicitly state anything about sending=20
> UPDATE if the confirm-status attribute has NOT been received. If the=20
> UA does *NOT* receive confirm-status attribute, and if the UA sent the=20
> UPDATE request once the UA finishes resource reservation, it may cause=20
> unnecessary race conditions since both UAs can send UPDATEs at about=20
> the same time.

The handling of race conditions regarding UPDATE is described in RFC
3331 section 5.2:

   A UAS that receives an UPDATE before it has generated a final
   response to a previous UPDATE on the same dialog MUST return a 500
   response to the new UPDATE, and MUST include a Retry-After header
   field with a randomly chosen value between 0 and 10 seconds.

   If an UPDATE is received that contains an offer, and the UAS has
   generated an offer (in an UPDATE, PRACK or INVITE) to which it has
   not yet received an answer, the UAS MUST reject the UPDATE with a 491
   response.  Similarly, if an UPDATE is received that contains an
   offer, and the UAS has received an offer (in an UPDATE, PRACK, or
   INVITE) to which it has not yet generated an answer, the UAS MUST
   reject the UPDATE with a 500 response, and MUST include a Retry-After
   header field with a randomly chosen value between 0 and 10 seconds.
   [...]

> Question:
> -----------
> I assume that if a UA does *NOT* receive confirm-status attribute,=20
> then when the UA finishes the resource reservation, the UA SHOULD NOT=20
> send UPDATE request. Is this correct understanding?
> -----------

In principle, a UA can send an UPDATE at any time when it is not forbidden =
to.  RFC 3311 section 5.1 gives some guidelines:

   The UPDATE request is constructed as would any other request within
   an existing dialog, as described in Section 12.2.1 of RFC 3261.  It
   MAY be sent for both early and confirmed dialogs, and MAY be sent by
   either caller or callee.  Although UPDATE can be used on confirmed
   dialogs, it is RECOMMENDED that a re-INVITE be used instead.  This is
   because an UPDATE needs to be answered immediately, ruling out the
   possibility of user approval.  Such approval will frequently be
   needed, and is possible with a re-INVITE.

Dale


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From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
To: Ivo Sedlacek <ivo.sedlacek@ericsson.com>, "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Precondition and status confirmation in regular two party	calls
Thread-Index: AQHPeqMZ2vpklsp7EkukvXxbyikpPZtXnK2QgAAJwgA=
Date: Thu, 29 May 2014 14:58:53 +0000
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Subject: Re: [sipcore] Precondition and status confirmation in regular two party	calls
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(minor reformulation below, sorry about being sloppy).

Hello,

The issue is not the race condition as such.  As you say, UPDATE is allowed=
 to be sent at any time and there are procedures to handle UPDATEs being se=
nt by both UAs at about the same time.

The question was rather: Is it advisable for a UA to send an UPDATE indicat=
ing that its resources are available if confirm-status attribute was *NOT* =
received?

Sending such non-requested UPDATE can generate unnecessary signalling (if b=
oth UAs reserve resources at about the same time, both UAs send UPDATE at a=
bout the same time and 491 would be generated) and call setup delay (since =
at last one of the UPDATEs will need to be repeated).

The whole reason for the confirm-status attribute is to request a confirmat=
ion when resources becomes available. So if UA has *NOT* received the confi=
rm-status attribute, then when resources of the UA becomes available, the U=
A should refrain from sending UPDATE (unless there is another reason, not r=
elated to resource reservation, for doing so).

Do you agree?

Kind regards

Ivo Sedlacek

-----Original Message-----
From: Dale R. Worley [mailto:worley@ariadne.com]
Sent: 28. kv=ECtna 2014 20:32
To: Ivo Sedlacek
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Precondition and status confirmation in regular two =
party calls

> From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>

> However, RFC3312 does not explicitly state anything about sending=20
> UPDATE if the confirm-status attribute has NOT been received. If the=20
> UA does *NOT* receive confirm-status attribute, and if the UA sent the=20
> UPDATE request once the UA finishes resource reservation, it may cause=20
> unnecessary race conditions since both UAs can send UPDATEs at about=20
> the same time.

The handling of race conditions regarding UPDATE is described in RFC
3331 section 5.2:

   A UAS that receives an UPDATE before it has generated a final
   response to a previous UPDATE on the same dialog MUST return a 500
   response to the new UPDATE, and MUST include a Retry-After header
   field with a randomly chosen value between 0 and 10 seconds.

   If an UPDATE is received that contains an offer, and the UAS has
   generated an offer (in an UPDATE, PRACK or INVITE) to which it has
   not yet received an answer, the UAS MUST reject the UPDATE with a 491
   response.  Similarly, if an UPDATE is received that contains an
   offer, and the UAS has received an offer (in an UPDATE, PRACK, or
   INVITE) to which it has not yet generated an answer, the UAS MUST
   reject the UPDATE with a 500 response, and MUST include a Retry-After
   header field with a randomly chosen value between 0 and 10 seconds.
   [...]

> Question:
> -----------
> I assume that if a UA does *NOT* receive confirm-status attribute,=20
> then when the UA finishes the resource reservation, the UA SHOULD NOT=20
> send UPDATE request. Is this correct understanding?
> -----------

In principle, a UA can send an UPDATE at any time when it is not forbidden =
to.  RFC 3311 section 5.1 gives some guidelines:

   The UPDATE request is constructed as would any other request within
   an existing dialog, as described in Section 12.2.1 of RFC 3261.  It
   MAY be sent for both early and confirmed dialogs, and MAY be sent by
   either caller or callee.  Although UPDATE can be used on confirmed
   dialogs, it is RECOMMENDED that a re-INVITE be used instead.  This is
   because an UPDATE needs to be answered immediately, ruling out the
   possibility of user approval.  Such approval will frequently be
   needed, and is possible with a re-INVITE.

Dale

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


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--90e6ba613c3a0cf91404fa8d80a3
Content-Type: text/plain; charset=ISO-8859-1

Hi Michael,

Thanks for the review and feedback.

Sure, I will add a section to talk about uses cases.
Would you be able to share your use cases to see if we can cover them in
this document?

Thanks,
 Rifaat



On Thu, May 29, 2014 at 9:50 AM, Michael Procter <michael@voip.co.uk> wrote:

> Hi Rifaat,
>
> On 26 May 2014 22:46, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com> wrote:
> > We would appreciate it if people review the document and provide us with
> > their thoughts.
>
> Would you be able to provide a couple of motivating use cases?  The
> ones I've come up with either aren't fully met by the outlines in this
> document, or might be achieved more easily with existing pieces of
> protocol.
>
> A couple of typos:
> Section 1, last paragraph:
> "Single Sing-On" should be "Single Sign-On"
>
> Section 3.4, last paragraph:
> "with toke, token refresh, and the master-key"
> should be
> "with token, refresh_token, and the master-key"
>
> Regards,
>
> Michael
>

--90e6ba613c3a0cf91404fa8d80a3
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr">Hi Michael,<div><br></div><div>Thanks for the review and f=
eedback.</div><div><br></div><div>Sure, I will add a section to talk about =
uses cases.</div><div>Would you be able to share your use cases to see if w=
e can cover them in this document?</div>
<div><br></div><div>Thanks,</div><div>=A0Rifaat</div><div><br></div></div><=
div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu, May 29=
, 2014 at 9:50 AM, Michael Procter <span dir=3D"ltr">&lt;<a href=3D"mailto:=
michael@voip.co.uk" target=3D"_blank">michael@voip.co.uk</a>&gt;</span> wro=
te:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Rifaat,<br>
<div class=3D""><br>
On 26 May 2014 22:46, Rifaat Shekh-Yusef &lt;<a href=3D"mailto:rifaat.ietf@=
gmail.com">rifaat.ietf@gmail.com</a>&gt; wrote:<br>
&gt; We would appreciate it if people review the document and provide us wi=
th<br>
&gt; their thoughts.<br>
<br>
</div>Would you be able to provide a couple of motivating use cases? =A0The=
<br>
ones I&#39;ve come up with either aren&#39;t fully met by the outlines in t=
his<br>
document, or might be achieved more easily with existing pieces of<br>
protocol.<br>
<br>
A couple of typos:<br>
Section 1, last paragraph:<br>
&quot;Single Sing-On&quot; should be &quot;Single Sign-On&quot;<br>
<br>
Section 3.4, last paragraph:<br>
&quot;with toke, token refresh, and the master-key&quot;<br>
should be<br>
&quot;with token, refresh_token, and the master-key&quot;<br>
<br>
Regards,<br>
<br>
Michael<br>
</blockquote></div><br></div>

--90e6ba613c3a0cf91404fa8d80a3--


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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: [sipcore] Fwd: [SIPForum-techwg] SIPNOC 2014: Only Two Weeks Away!
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I'm forwarding this as I believe the event is quite relevant for a lot of
folks on this list.  If your schedule allows I think you'll find a lot of
the topics to be of interest.

Regards,
Mary.

---------- Forwarded message ----------
From: Marc Robins <marc.robins@sipforum.org>
Date: Wed, May 28, 2014 at 1:00 PM
Subject: [SIPForum-techwg] SIPNOC 2014: Only Two Weeks Away!
To: techwg@sipforum.org




[image: Introducing SIPNOC - The SIP Network Operators Conference]
<http://www.tmcnet.com/redir?u=3D1011298>

 <http://www.sipforum.org/>

 <http://www.tmcnet.com/redir?u=3D1011299>

 <http://www.sipforum.org/>

*PLATINUM SPONSORS*

[image: Image] <http://www.audiocodes.com/>

*GOLD SPONSORS *

[image: Image] <http://www.edgewaternetworks.com/>

[image: Image]

<http://www.oracle.com/index.html>

[image: Image]

<http://www.sonus.net/>

[image: Image]

<http://www.sorenson.com/>

*POWERSTATION SPONSORS*

[image: Image] <http://www.audiocodes.com/>

Two-Weeks Till SIPNOC 2014!

Don=E2=80=99t Miss the Opportunity to Network with Some of the IP Communica=
tions
Industry=E2=80=99s Best and Brightest!

Now in its fourth year, *SIPNOC 2014* is a unique educational conference
for international telecom stakeholders responsible for the deployment,
management and ongoing operation of SIP-based services in service provider
networks.  Come meet the engineering talent from service providers of all
sizes, leading equipment vendors, and the brightest minds doing academic
research.

*Symposium Atmosphere, Absent of Marketing Spin*

This is not a marketing meeting full of vendor sales pitches. It's about
real-world problems and how the technical community is solving them. *Learn
how to* *avoid problems before you even encounter them.*

Attending service providers have included AT&T, Armstrong Utilities,
babyTEL, Bandwidth.com, Bell Canada, Bright House Networks, Broadvox,
Cablevision, Cbeyond, Cinchcast, Cincinnati Bell, Comcast Cable, Cologix,
Consolidated Telecom, Cox Communications, Deutche Telecom, France Telecom
Orange, Global Crossing, Grandstream Networks, iBasis, Level3, Liveops,
nexVortex, NTT, One Source Networks, Optimum Lightpath, Purple
Communications, Rogers Business Solutions, Shoretel, Socket Telecom,
Sorenson Communications, Sprint, Swisscom, TDS Telecom, Time Warner Cable,
Telefonica International, Telmex, UNINETT, Verizon, Vocalocity, Vonage,
Voxbone, WDT (World Discount Telecommunications), Wholesale Carrier
Services, XO Communications, and Ziggo.

In addition to carrier participants, SIPNOC regularly attracts a group of
SIP community stakeholders from vendors, governments and research
organizations such as AudioCodes, Avaya, Broadsoft, CableLabs, Cequint,
Cisco Systems, Commetrex, Current Analysis, Dialogic, ECG Inc, Edgewater
Networks, Ericsson, Faxcore, Federal Communications Commission, GENBAND,
Georgetown University, Huawei Technologies, iconectiv, Integrated Research,
Internet Society (ISOC), Metaswitch Networks, NENA, NetNumber, NTCA,
Oracle, Patton Electronics, Polycom, Public Knowledge, RecordMyCalls,
Sangoma Technologies, Sansay, SecureLogix, ScanSource, SNOM Technology,
Sonus Networks, UNH-IOL,  Unify, and Yealink Network Technology.

You can review the *SIPNOC 2014 conference schedule
<http://www.tmcnet.com/redir?u=3D1011300>* and list of presenters now.
Speakers will explore critical =E2=80=9Creal-world=E2=80=9D service provide=
r challenges
related to SIP applications and infrastructure, including: network
security, SIP trunking, testing and application development, Fax over IP,
call routing and peering, IPv6, troubleshooting and monitoring, emergency
services and more.

*SIPNOC 2014 Registration Information*
For a limited time, you can take advantage of a special discounted rate of
$895, a savings of $100 off the regular rate. To obtain the discount, enter
the following code on the registration page: =E2=80=9Csipnoctmc100=E2=80=9D=
. Your
registration includes access to all conference sessions, keynotes, meals
and breaks, plus networking receptions each evening.

*To register for SIPNOC 2014, please visit
http://www.regonline.com/sipnoc2014 <http://www.regonline.com/sipnoc2014>.*

*Don't Forget!*
Individuals from SIP Forum Full Member companies save even more! Please
contact Marc Robins, SIP Forum President and Managing Director, at
marc.robins@sipforum.org to obtain the exclusive Full Member discount code.

*Hotel Reservations*
SIPNOC 2014 is located at the four-star Hyatt Dulles Hotel. To reserve your
stay at the Hyatt Dulles, please visit:
https://resweb.passkey.com/go/SIPNOC2014


*For more information, please visit:www.sipnoc.org <http://www.sipnoc.org>
or email sipnocinfo@sipforum.org <sipnocinfo@sipforum.org>*


_______________________________________________
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Send mail to: techwg@sipforum.org
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g

--047d7b87371e2d155a04fa9edb53
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I&#39;m forwarding this as I believe the event is quite re=
levant for a lot of folks on this list. =C2=A0If your schedule allows I thi=
nk you&#39;ll find a lot of the topics to be of interest.<div><br></div><di=
v>Regards,</div>
<div>Mary.=C2=A0<br><br><div class=3D"gmail_quote">---------- Forwarded mes=
sage ----------<br>From: <b class=3D"gmail_sendername">Marc Robins</b> <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:marc.robins@sipforum.org">marc.robins@s=
ipforum.org</a>&gt;</span><br>
Date: Wed, May 28, 2014 at 1:00 PM<br>Subject: [SIPForum-techwg] SIPNOC 201=
4: Only Two Weeks Away!<br>To: <a href=3D"mailto:techwg@sipforum.org">techw=
g@sipforum.org</a><br><br><br><div bgcolor=3D"#CFCFCF" lang=3D"EN-US" link=
=3D"blue" vlink=3D"purple">
<div><p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><div align=3D"center"><=
table border=3D"0" cellspacing=3D"0" cellpadding=3D"0" width=3D"650" style=
=3D"width:487.5pt;background:white"><tbody><tr><td colspan=3D"3" style=3D"b=
ackground:#cfcfcf;padding:0in 0in 0in 0in">
</td></tr><tr><td colspan=3D"3" style=3D"padding:0in 0in 0in 0in"><p class=
=3D"MsoNormal"><a href=3D"http://www.tmcnet.com/redir?u=3D1011298" title=3D=
"Introducing SIPNOC - The SIP Network Operators Conference" target=3D"_blan=
k"><span style=3D"text-decoration:none"><img border=3D"0" src=3D"http://ima=
ges.tmcnet.com/mkt/blast/sipnoc/SIPNOC-Header-2014.jpg" alt=3D"Introducing =
SIPNOC - The SIP Network Operators Conference"></span></a><u></u><u></u></p=
>
</td></tr><tr><td colspan=3D"3" style=3D"background:#cfcfcf;padding:0in 0in=
 0in 0in"><p class=3D"MsoNormal"><img border=3D"0" width=3D"2" height=3D"2"=
 src=3D"http://images.tmcnet.com/mkt/blast/microsoft/spacer.gif"><u></u><u>=
</u></p></td>
</tr><tr><td colspan=3D"3" style=3D"padding:0in 0in 0in 0in"><p class=3D"Ms=
oNormal"><img border=3D"0" width=3D"10" height=3D"10" src=3D"http://images.=
tmcnet.com/mkt/blast/microsoft/spacer.gif"><u></u><u></u></p></td></tr><tr>=
<td width=3D"27" style=3D"width:20.25pt;padding:0in 0in 0in 0in">
<p class=3D"MsoNormal"><img border=3D"0" width=3D"10" height=3D"10" src=3D"=
http://images.tmcnet.com/mkt/blast/microsoft/spacer.gif"><u></u><u></u></p>=
</td><td width=3D"650" style=3D"width:487.5pt;padding:0in 0in 0in 0in"><tab=
le border=3D"0" cellspacing=3D"0" cellpadding=3D"0" align=3D"right">
<tbody><tr><td style=3D"padding:0in 0in 0in 0in"><p class=3D"MsoNormal"><u>=
</u><a href=3D"http://www.sipforum.org/" target=3D"_blank"><img border=3D"0=
" width=3D"250" height=3D"166" src=3D"http://images.tmcnet.com/mkt/blast/si=
pnoc/sipnoc_gensession1-small.jpg" align=3D"left"></a><u></u><u></u><u></u>=
</p>
</td></tr><tr><td style=3D"padding:0in 0in 0in 0in"></td></tr><tr><td style=
=3D"padding:0in 0in 0in 0in"><p class=3D"MsoNormal"><u></u><a href=3D"http:=
//www.tmcnet.com/redir?u=3D1011299" target=3D"_blank"><img border=3D"0" wid=
th=3D"250" height=3D"43" src=3D"http://images.tmcnet.com/mkt/blast/sipnoc/v=
iew-agenda-button.jpg" align=3D"left"></a><u></u><u></u><u></u></p>
</td></tr><tr><td style=3D"padding:0in 0in 0in 0in"><p class=3D"MsoNormal">=
<u></u><a href=3D"http://www.sipforum.org/" target=3D"_blank"><img border=
=3D"0" width=3D"200" height=3D"100" src=3D"http://images.tmcnet.com/mkt/bla=
st/sipnoc/sipforum.png" align=3D"left"></a><u></u><u></u><u></u></p>
</td></tr><tr><td style=3D"padding:0in 0in 0in 0in"><table border=3D"1" cel=
lspacing=3D"3" cellpadding=3D"0" align=3D"right" width=3D"271" style=3D"wid=
th:203.25pt;border:solid #efefef 1.0pt"><tbody><tr><td width=3D"257" style=
=3D"width:192.75pt;border:none;padding:2.25pt 2.25pt 2.25pt 2.25pt">
<p><span><b>PLATINUM SPONSORS</b></span><u></u><u></u></p></td></tr><tr><td=
 style=3D"border:none;padding:2.25pt 2.25pt 2.25pt 2.25pt"><p class=3D"MsoN=
ormal"><a href=3D"http://www.audiocodes.com/" target=3D"_blank"><span style=
=3D"text-decoration:none"><img border=3D"0" src=3D"http://images.tmcnet.com=
/mkt/blast/sipnoc/AudioCodes-ScanSource-V.jpg" alt=3D"Image"></span></a><u>=
</u><u></u></p>
</td></tr><tr><td width=3D"257" style=3D"width:192.75pt;border:none;padding=
:2.25pt 2.25pt 2.25pt 2.25pt"><p><span><b>GOLD SPONSORS </b></span><u></u><=
u></u></p></td></tr><tr><td style=3D"border:none;padding:2.25pt 2.25pt 2.25=
pt 2.25pt">
<p class=3D"MsoNormal"><a href=3D"http://www.edgewaternetworks.com/" target=
=3D"_blank"><span style=3D"text-decoration:none"><img border=3D"0" src=3D"h=
ttp://images.tmcnet.com/mkt/blast/sipnoc/edgewaternetworkslogo.jpg" alt=3D"=
Image"></span></a><u></u><u></u></p>
</td></tr><tr><td style=3D"border:none;padding:2.25pt 2.25pt 2.25pt 2.25pt"=
><p class=3D"MsoNormal"><a href=3D"http://www.oracle.com/index.html" target=
=3D"_blank"><span style=3D"text-decoration:none"><img border=3D"0" width=3D=
"120" src=3D"http://images.tmcnet.com/mkt/blast/sipnoc/oracle.gif" alt=3D"I=
mage"></span><br>
<br></a><u></u><u></u></p></td></tr><tr><td style=3D"border:none;padding:2.=
25pt 2.25pt 2.25pt 2.25pt"><p class=3D"MsoNormal"><a href=3D"http://www.son=
us.net/" target=3D"_blank"><span style=3D"text-decoration:none"><img border=
=3D"0" src=3D"http://images.tmcnet.com/mkt/blast/sipnoc/sonus.jpg" alt=3D"I=
mage"></span><br>
<br></a><u></u><u></u></p></td></tr><tr><td style=3D"border:none;padding:2.=
25pt 2.25pt 2.25pt 2.25pt"><p class=3D"MsoNormal"><a href=3D"http://www.sor=
enson.com/" target=3D"_blank"><span style=3D"text-decoration:none"><img bor=
der=3D"0" width=3D"200" src=3D"http://images.tmcnet.com/mkt/blast/sipnoc/sc=
-logo.jpg" alt=3D"Image"></span><br>
<br></a><u></u><u></u></p></td></tr><tr><td width=3D"257" style=3D"width:19=
2.75pt;border:none;padding:2.25pt 2.25pt 2.25pt 2.25pt"><p><span><b>POWERST=
ATION SPONSORS</b></span><u></u><u></u></p></td></tr><tr><td style=3D"borde=
r:none;padding:2.25pt 2.25pt 2.25pt 2.25pt">
<p class=3D"MsoNormal"><a href=3D"http://www.audiocodes.com/" target=3D"_bl=
ank"><span style=3D"text-decoration:none"><img border=3D"0" src=3D"http://i=
mages.tmcnet.com/mkt/blast/sipnoc/AudioCodes-ScanSource-V.jpg" alt=3D"Image=
"></span></a><u></u><u></u></p>
</td></tr></tbody></table></td></tr></tbody></table><p class=3D"MsoNormal">=
<span><span style=3D"font-size:16.0pt;color:#1f497d">Two-Weeks Till SIPNOC =
2014!<u></u><u></u></span></span></p><p class=3D"MsoNormal"><span><span sty=
le=3D"font-size:16.0pt;color:#1f497d">Don=E2=80=99t Miss the Opportunity to=
 Network with Some of the IP Communications Industry=E2=80=99s Best and Bri=
ghtest!</span></span><u></u><u></u></p>
<p>Now in its fourth year, <strong><span style=3D"font-family:&quot;Calibri=
&quot;,&quot;sans-serif&quot;;font-weight:normal">SIPNOC 2014</span></stron=
g> is a unique educational conference for international telecom stakeholder=
s responsible for the deployment, management and ongoing operation of SIP-b=
ased services in service provider networks. =C2=A0Come meet the engineering=
 talent from service providers of all sizes, <span style=3D"color:windowtex=
t">leading equipment vendors, and</span> the brightest minds doing academic=
 research.<br>
<br><strong><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;">Symposium Atmosphere, Absent of Marketing Spin</span></strong> <br>=
<br>This is not a marketing meeting full of vendor sales=C2=A0pitches. It&#=
39;s about real-world problems and how the technical community is solving t=
hem.=C2=A0<em><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-ser=
if&quot;">Learn how to</span></em>=C2=A0<em><span style=3D"font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;">avoid problems=C2=A0before you even=
 encounter them.</span></em><br>
<br>Attending service providers have included AT&amp;T, Armstrong Utilities=
, babyTEL, Bandwidth.com, Bell Canada, Bright House Networks, Broadvox, Cab=
levision, Cbeyond, Cinchcast, Cincinnati Bell, Comcast Cable, Cologix, Cons=
olidated Telecom, Cox Communications, Deutche Telecom, France Telecom Orang=
e, Global Crossing, Grandstream Networks, iBasis, Level3, Liveops,=C2=A0 ne=
xVortex, NTT, One Source Networks, Optimum Lightpath, Purple Communications=
, Rogers Business Solutions, Shoretel, Socket Telecom, Sorenson Communicati=
ons, Sprint, Swisscom, TDS Telecom, Time Warner Cable, Telefonica Internati=
onal, Telmex, UNINETT, Verizon, Vocalocity, Vonage, Voxbone, WDT (World Dis=
count Telecommunications), Wholesale Carrier Services, XO Communications, a=
nd Ziggo.<br>
<br>In addition to carrier participants, SIPNOC regularly attracts a group =
of SIP community stakeholders from vendors, governments and research organi=
zations such as AudioCodes, Avaya, Broadsoft, CableLabs, Cequint, Cisco Sys=
tems, Commetrex, Current Analysis, Dialogic, ECG Inc, Edgewater Networks, E=
ricsson, Faxcore, Federal Communications Commission, GENBAND, Georgetown Un=
iversity, Huawei Technologies, iconectiv, Integrated Research, Internet Soc=
iety (ISOC), Metaswitch Networks, NENA, NetNumber, NTCA, Oracle, Patton Ele=
ctronics, Polycom, Public Knowledge, RecordMyCalls, Sangoma Technologies, S=
ansay, SecureLogix, ScanSource, SNOM Technology, Sonus Networks, UNH-IOL,=
=C2=A0 Unify, and Yealink Network Technology.<u></u><u></u></p>
<p style=3D"margin-bottom:12.0pt">You can review the <u><a href=3D"http://w=
ww.tmcnet.com/redir?u=3D1011300" target=3D"_blank"><span style=3D"color:#00=
6600">SIPNOC 2014 conference schedule</span></a></u> and list of presenters=
 now. Speakers will explore critical =E2=80=9Creal-world=E2=80=9D service p=
rovider challenges related to SIP applications and infrastructure, includin=
g: network security, SIP trunking, testing and application development, Fax=
 over IP, call routing and peering, IPv6, troubleshooting and monitoring, e=
mergency services and more.<br>
<br><strong><u><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-se=
rif&quot;">SIPNOC 2014 Registration Information</span></u></strong><br>For =
a limited time, you can take advantage of a special discounted rate of $895=
, a savings of $100 off the regular rate. To obtain the discount, enter the=
 following code on the registration page: =E2=80=9Csipnoctmc100=E2=80=9D. Y=
our registration includes access to all conference sessions, keynotes, meal=
s and breaks, plus networking receptions each evening. <br>
<br><strong><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;">To register for SIPNOC 2014, please visit <a href=3D"http://www.reg=
online.com/sipnoc2014" target=3D"_blank"><span style=3D"color:#006600">http=
://www.regonline.com/sipnoc2014</span></a>.</span></strong><br>
<br><strong><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;;color:#df3723">Don&#39;t Forget!</span></strong> <br>Individuals fro=
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From: worley@ariadne.com (Dale R. Worley)
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Archived-At: http://mailarchive.ietf.org/arch/msg/sipcore/yjkMSQ_rnuZ7ZQz7QTOMJPyMJak
Subject: Re: [sipcore] Precondition and status confirmation in regular two party	calls
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> From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>

> The whole reason for the confirm-status attribute is to request a
> confirmation when resources becomes available. So if UA has *NOT*
> received the confirm-status attribute, then when resources of the UA
> becomes available, the UA should refrain from sending UPDATE (unless
> there is another reason, not related to resource reservation, for
> doing so).
> 
> Do you agree?

Yes.  It seems to be a case of "Do not send UPDATE unless there is a
reason to."

Dale


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From: "DOLLY, MARTIN C" <md3135@att.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Precondition and status confirmation in regular two party	calls
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Subject: Re: [sipcore] Precondition and status confirmation in regular two party	calls
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I Agree as well

Martin Dolly
Lead Member of Technical Staff
Core Network & Gov't/Regulatory Standards =20
AT&T Standards and Industry Alliances
+1-609-903-3360
md3135@att.com

On May 30, 2014, at 3:14 PM, "Dale R. Worley" <worley@ariadne.com> wrote:

>> From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
>=20
>> The whole reason for the confirm-status attribute is to request a
>> confirmation when resources becomes available. So if UA has *NOT*
>> received the confirm-status attribute, then when resources of the UA
>> becomes available, the UA should refrain from sending UPDATE (unless
>> there is another reason, not related to resource reservation, for
>> doing so).
>>=20
>> Do you agree?
>=20
> Yes.  It seems to be a case of "Do not send UPDATE unless there is a
> reason to."
>=20
> Dale
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

