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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-04.txt 
Date: Sat, 1 Jul 2006 18:02:18 -0700
To: Jeroen van Bemmel <jbemmel@zonnet.nl>
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On Jun 29, 2006, at 3:28 PM, Jeroen van Bemmel wrote:

> We've had some VERY long and extensive discussions on this  
> (starting with http://www1.ietf.org/mail-archive/web/sip/current/ 
> msg13677.html) but the draft is still not addressing some of the  
> issues that were raised
>
> 1) The UAC needs a positive indication that the selected proxy can  
> indeed handle STUN packets without choking on them. It is not  
> sufficient if the URI it is configured with says 'keepalive=stun',  
> that's a wish but no guarantee.
>
>

Why is this a wish not a guarantee? I'm under the impression that  
devices that follow the RFCs should not have problems with this. I  
understand the argument that we might want DHCP or DNS to be a way to  
discover this configuration information but I think either of theses  
systems, not to mention the config-framework, can be used to get an  
URI. Keep in mind the thing that drove us to this approach - sig- 
comp. We need sig-comp to work too so however we deal with the  
approach for configuration of the URI so it has the sig-comp tag will  
also work for this. These options were discussed several times - at  
one point I was proposing a DNS scheme but I got the input that it  
could and should be separated.

>
> 2) Section 5.3: the Edge Proxy MUST Record-Route dialog forming  
> INVITE requests that it forwards over the flow towards the UA  
> (reverse direction is not strictly necessary, but probably common  
> too). If it does not do this, an ACK for 2xx cannot arrive. For  
> incoming SUBSCRIBE it might work without (the NOTIFY could create a  
> direct flow towards the target or some other proxy, using the local  
> socket as a Contact), but then there would be an issue with keeping  
> that flow alive (the STUN question). Probably safer to say that (in  
> addition to MUST for INVITE) "the Edge Proxy SHOULD Record-Route  
> all dialog-forming requests it forwards to the UA"

This could be one way to solve the problem -  others are possible  
too. I don't think this is in scope for outbound. I don't think  
outbound does anything that makes it so this solution (or some of the  
other proposed ones) can't be used. A long time ago, outbound got  
stuck in a huge rat hole of trying to define how every system should  
form it's high reliability and high availability solutions. That was  
too big a task to take on so I am happy to leave it out of scope as  
long as people can build good system.

>
> 3) The Proxy-Require mechanism we have discussed could be useful to  
> UACs that want to ensure that a selected edge proxy supports  
> outbound, aborting if not.
>

I'll let Rohan reply to this but I don't think I see what problem it  
solves.

> 4) (minor remark) Perhaps some words could be added that the UAC  
> SHOULD perform keepalives when configured to do so, even if it  
> detects that there is no NAT. There could be a transparent firewall  
> device that needs to be kept open.
>

I agree this is the behavior we want  - I thought it already more or  
less said that. It  does not suggest detecting if their is a NAT  
there at all - if the URI has the tag, you do keep alive.


> Regards,
>
> Jeroen

I'm sorry my time is limited on this right now but I'm reading a ton  
of documents and posting comments so forgive me as I suffer from a  
response implosion problem.

Cullen <my crazy comments as an individual only>

>
>
>
>
>
>
> ----- Original Message ----- From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Cc: <sip@ietf.org>
> Sent: Wednesday, June 28, 2006 12:50 AM
> Subject: [Sip] I-D ACTION:draft-ietf-sip-outbound-04.txt
>
>
>> A New Internet-Draft is available from the on-line Internet-Drafts  
>> directories.
>> This draft is a work item of the Session Initiation Protocol  
>> Working Group of the IETF.
>>
>> Title : Managing Client Initiated Connections in the
>>                          Session Initiation Protocol (SIP)
>> Author(s) : C. Jennings, R. Mahy
>> Filename : draft-ietf-sip-outbound-04.txt
>> Pages : 34
>> Date : 2006-6-27
>>
>> The Session Initiation Protocol (SIP) allows proxy servers to
>>   initiate TCP connections and send asynchronous UDP datagrams to  
>> User
>>   Agents in order to deliver requests.  However, many practical
>>   considerations, such as the existence of firewalls and Network
>>   Address Translators (NATs), prevent servers from connecting to User
>>   Agents in this way.  This specification defines behaviors for User
>>   Agents, registrars and proxy servers that allow requests to be
>>   delivered on existing connections established by the User  
>> Agent.  It
>>   also defines keep alive behaviors needed to keep NAT bindings open
>>   and specifies the usage of multiple connections.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-04.txt
>>
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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Comments on rosenberg-sip-identity-coexistence
Date: Sun, 2 Jul 2006 10:02:27 -0700
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On Jun 30, 2006, at 12:57 PM, Hadriel Kaplan wrote:

>
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>>
>> I think one of the major reason for creating PAI was to provide
>> malicious call trace for anonymous calls. I don't think it was for
>> dealing with spoofing the From header. If that was the goal, we could
>> have solved it just by having device that ends up inserting the PAI
>> just reject any request that has spoofed From headers.
>
> I agree with you, but it is used for detecting suspicious From  
> headers even
> for non-anonymous calls.  Obviously if the proxy inserting the PAI  
> _knows_ a
>> From is spoofed, it could/should reject it.  But if the proxy does  
>> _not_
> know whether the From is spoofed or not, it just doesn't insert the  
> PAI.
> For example, ingress proxy receives a request from another domain -  
> if it
> trusts the domain it passes the PAI (or inserts one).  If it does  
> not trust
> the domain, it removes PAI, but doesn't want to reject the request  
> (else it
> wouldn't accept requests from such untrusted domains to begin with).

I see your point, I can see how a proxy that is deciding not to  
remove a PAI might not know a From was spoofed. I don't see how a  
proxy that was going to *insert* a PAI would know what to put in the  
PAI and not be able to know if a PAI was spoofed. 

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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Comments on rosenberg-sip-identity-coexistence
Date: Sun, 2 Jul 2006 10:17:47 -0700
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On Jun 30, 2006, at 2:59 PM, Hadriel Kaplan wrote:

> More inline...
>
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>>
>> case 2) ingress B2B on terminating domain - this is fairly easy too.
>> This B2BUA resigns the incoming request. To do this it uses a cert
>> that has a wildcard for all domain - the UAS need to be configured to
>> have this cert in it root trust list. What the UAS is doing here is
>> delegating that it trust any name that the
>>   B2BUA trusts. This is the sort of trust delegation one wants to be
>> able to make.
>
> I don't know how practical that is to do on the UASs.  It seems to  
> me like
> most UASs simply trust whatever their proxy tells them, and that  
> should be
> ok if we can guarantee what their proxy tells them is true. (a big IF)

For UA that don't do Identity, I think what I proposed works with no  
changes to them. For UAs that do do Identity, they need to have a  
list or root certs they trust. It seems like when you are determining  
what the UA trusts you are setting up this list. Again, in this case  
I don't see what does not work about my proposal.

>
> If a b2bua before that proxy but in the same trust domain verified the
> identity, and can somehow tell the proxy it verified the identity,  
> and even
> tell the UAS it did as well, why do more?  That's what the PAI  
> does.  It may
> not be what it was supposed to do, but it's what it seems to now.
>

So in the case of an anonymous call where the real caller identity is  
in the PAI for malicious call trace reasons, what are you proposing  
to do. Does the proxy strip the PAI before sending it to the UAS?   
How would the proxy know? Do you deliver the PAI to the UA - this  
would violate privacy laws in Europe. I don't see how this would work.

>
>> case 3) a B2BUA in a transit provider between the originating and
>> terminating domain. There are two very different cases here - one is
>> where the transit provider is an intermediate in providing PSTN
>> termination services and using E.164 addresses. For example some
>> company, contracts its PSTN termination from say MCI who for numbers
>> in Australia perhaps uses Telstra. The company wants to see the phone
>> number asserted by MCI, not Telstra  - it is MCI they trust and can
>> make an authorization decision based on. They have no idea who
>> Telstra is so whatever telstra wants to claim is useless for the
>> company. The point is, whatever things Telstra might assert to MCI,
>> the company wants MCI to make an assertion that the company can use
>> for it authorization decision. In these cases the transit B2BUA needs
>> to resign things.
>
> And that's the rub.  The real use/value of Identity, IMHO, is for the
> transit case - period.  Within a domain or between just 2, and  
> there are
> other ways to trust, except for the really paranoid.  But the  
> longer the
> trust chain of transits is, the greater the probability the chain  
> is broken.
>
>
> When I looked at the original identity draft I thought ok - the  
> b2bua/proxy
> egress can be the signer and the ingress b2bua/proxy can be the  
> verifier.
> Of course then I immediately thought if the 2 domains are connected  
> by just
> those 2 elements, why bother doing the signing/verifying for a  
> single hop
> that is going to be TLS/IPSEC anyway??  Use the PAI you already have.
>
> But for transit cases: it requires the transit domain(s) to either  
> (1) not
> muck with contact, call-id, and bodies, or (2) support identity,  
> and to
> somehow re-sign with something that the verifier in the next domain 
> (s) will
> accept, which goes back to hop-by-hop trust kinda.  But changing  
> (1) or (2)
> is out of the local (and final) domain's control to change - they  
> need the
> transit domain(s) to change, and that's a much bigger hurdle.
>
>

Ok, I am done't think I am manging to explaim myself. Say A has a  
contract with B and B has a crontract with C. Now a call arrives at  
C, it seem that C wants a assertion to be made by B. Anything A says  
to C is clsoe to useless because C has no clue who A even is. I guese  
I feel like we need to sort out the requirements for what we want in  
a transit case and then we might be able to figure out a solution.


>> As a last note, I do think there are workable solutions for the B2BUA
>> case. Having UA accept message with a bad Identity header but that
>> have a PAI seriously breaks Identify. One of the key things that
>> Identity provides is a close to end to end integrity for the bodies.
>
> Yes, it signed the body, and contact, but I consider that its fatal  
> flaw.
> It should have stuck with the From (and Date and whatever else it  
> thinks is
> needed for a reasonable nonce).  The body I know is a hard one to  
> not sign
> (it's so tempting!).  But sign the contact?  What does that have to  
> do with
> identity?
>
>
>> This breaks this and we have no other good way to provide this. What
>> I am suggesting above provides a way to keep this integrity property
>> so that no device can tamper with the integrity other than devices
>> that the UAS or UAC explicitly trusts to tamper with the integrity. I
>> think this is a fairly important property and achievable.
>
> What about just splitting it into 2 Identity headers/signatures?   
> One for
> the From and nonce-ish stuff, the other for the body and same nonce- 
> ish
> stuff.  If a middle-man wants to modify the body, he verifies and  
> re-signs
> that second signature, or if the middle-man is unaware of identity  
> then the
> next one can decide what to do with a bad signature for a body but  
> good
> signature for From.
>

The From itself has no intrinsic value - what you are trying to sign  
is that some body came from the person in the From header. If the  
 From header and it's signature can be cut and paste from one message  
to another, Identity would fail to solve any security problems.

> So when the final UAS gets it (or its domain verifier) she says  
> "hmmm... the
>> From was definitely signed by the URI domain company.com, but this  
>> body is
> signed by provider.com who I trust... ok, I can trust that", vs. "I  
> see
> company.com signed the From but the body was broken... this is  
> suspicious".
> This way a b2bua can change the body but doesn't have to have a  
> certificate
> matching the From-uri domain to do so, and the UAS can decide if  
> it's kosher
> for the b2bua-domain to change the body, while also verifying the  
> From is of
> the right domain.  I haven't thought it through, but it sounds ok.

You are proposing almost exactly the same thing I am. The way you are  
thinking of handling the body, just use that handling for both.

>
> -hadriel

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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-04.txt 
Date: Sun, 2 Jul 2006 19:21:56 +0200
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Cullen,

see inline

Best regards,

Jeroen

----- Original Message ----- 
From: "Cullen Jennings" <fluffy@cisco.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Cc: <sip@ietf.org>
Sent: Sunday, July 02, 2006 3:02 AM
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-04.txt


>
> On Jun 29, 2006, at 3:28 PM, Jeroen van Bemmel wrote:
>
>> We've had some VERY long and extensive discussions on this  (starting 
>> with http://www1.ietf.org/mail-archive/web/sip/current/ msg13677.html) 
>> but the draft is still not addressing some of the  issues that were 
>> raised
>>
>> 1) The UAC needs a positive indication that the selected proxy can 
>> indeed handle STUN packets without choking on them. It is not  sufficient 
>> if the URI it is configured with says 'keepalive=stun',  that's a wish 
>> but no guarantee.
>>
>>
>
> Why is this a wish not a guarantee?

Because the configuration may be wrong, ie the entity that provisioned the 
URI thinks the element supports outbound, but in fact it does not. There can 
be various reasons for this, like invalid manual provisioning, some edge 
proxy is replaced with one that does not support outbound and someone 
forgets to update DNS, etc

I expect you will see a lot of what I'd call "opportunistic outbound": 
implementations that add 'keepalive=stun' to the URI for registration, in 
the hope that it is supported. There is nothing in the current spec to 
prevent this, but I believe it should be discouraged (that is, the sending 
of STUN packets multiplexed on SIP ports to hosts for which support is 
unknown).

> I'm under the impression that  devices that follow the RFCs should not 
> have problems with this. I  understand the argument that we might want 
> DHCP or DNS to be a way to  discover this configuration information but I 
> think either of theses  systems, not to mention the config-framework, can 
> be used to get an  URI. Keep in mind the thing that drove us to this 
> approach - sig- comp. We need sig-comp to work too so however we deal with 
> the  approach for configuration of the URI so it has the sig-comp tag will 
> also work for this. These options were discussed several times - at  one 
> point I was proposing a DNS scheme but I got the input that it  could and 
> should be separated.
>

I suppose a similar argument applies to sigcomp: it should have been 
prevented that implementations try it towards hosts that are not known to 
support it. A difference may be that sigcomp is also used in the first 
request, while STUN can be postponed

A second reason for providing such a 'keepalive confirmation' is that it 
also allows cases where the registrar / edge proxy is able to determine that 
keep-alives are in fact not needed (eg because it has authorative knowledge 
that there is no NAT or firewall in between). Not including 'keepalive=stun' 
would avoid unnecessary traffic. (note that perhaps a different mechanism 
would be more appropriate, eg "keepalive=not-needed" instead, to distinguish 
between not supported and not needed)

>>
>> 2) Section 5.3: the Edge Proxy MUST Record-Route dialog forming  INVITE 
>> requests that it forwards over the flow towards the UA  (reverse 
>> direction is not strictly necessary, but probably common  too). If it 
>> does not do this, an ACK for 2xx cannot arrive. For  incoming SUBSCRIBE 
>> it might work without (the NOTIFY could create a  direct flow towards the 
>> target or some other proxy, using the local  socket as a Contact), but 
>> then there would be an issue with keeping  that flow alive (the STUN 
>> question). Probably safer to say that (in  addition to MUST for INVITE) 
>> "the Edge Proxy SHOULD Record-Route  all dialog-forming requests it 
>> forwards to the UA"
>
> This could be one way to solve the problem -  others are possible  too. I 
> don't think this is in scope for outbound. I don't think  outbound does 
> anything that makes it so this solution (or some of the  other proposed 
> ones) can't be used. A long time ago, outbound got  stuck in a huge rat 
> hole of trying to define how every system should  form it's high 
> reliability and high availability solutions. That was  too big a task to 
> take on so I am happy to leave it out of scope as  long as people can 
> build good system.
>

I agree that the way in which high reliability/availability is done should 
be left to implementations. However, I don't see how ACK for 2xx could 
arrive without Record-Route (the contents of the R-R URI, and wheteher it is 
used for flow mapping is an implementation issue. However, I think its use 
should be mandated).

>>
>> 3) The Proxy-Require mechanism we have discussed could be useful to  UACs 
>> that want to ensure that a selected edge proxy supports  outbound, 
>> aborting if not.
>>
>
> I'll let Rohan reply to this but I don't think I see what problem it 
> solves.
>

It ensures early abortion of the registration process if outbound is not 
supported. As it is currently defined, a UAC has no way of determining 
whether an edge-proxy supports outbound (the flag in the provisioned URI 
makes it very likely, but not guaranteed)

A second usage of a Proxy-Require mechanism is perhaps out of scope for 
outbound, but related to keep-alive (which, as others have pointed out, 
should perhaps be separated). Consider an UAC behind NAT that sends an 
INVITE. Note that it could manage perfectly well without outbound support 
(it could simply not REGISTER), but it would need to keep the flow towards 
its peer open. If it is somehow able to determine that the peer does not 
support STUN, and it cannot use TCP keepalive, what should it do? One way 
could be to ask a proxy that supports keepalives to stay in the signaling 
path. It need not be the same proxy used for registration (but it might well 
be). For this scenario, the UAC would need a way to ask the proxy to provide 
keep-alive services. "Proxy-Require: keep-alive" would be one way; semantics 
would be that the proxy MUST Record-Route such a request (as discussed in a 
parallel thread, another reason is that the ACK might contain the same 
Proxy-Require)


>> 4) (minor remark) Perhaps some words could be added that the UAC  SHOULD 
>> perform keepalives when configured to do so, even if it  detects that 
>> there is no NAT. There could be a transparent firewall  device that needs 
>> to be kept open.
>>
>
> I agree this is the behavior we want  - I thought it already more or  less 
> said that. It  does not suggest detecting if their is a NAT  there at 
> all - if the URI has the tag, you do keep alive.
>

and if there is no such tag, it MUST NOT do keep alive (or at least not 
STUN, TCP keepalives would be harmless) ?

>
>> Regards,
>>
>> Jeroen
>
> I'm sorry my time is limited on this right now but I'm reading a ton  of 
> documents and posting comments so forgive me as I suffer from a  response 
> implosion problem.
>
> Cullen <my crazy comments as an individual only>
>
>>
>>
>>
>>
>>
>>
>> ----- Original Message ----- From: <Internet-Drafts@ietf.org>
>> To: <i-d-announce@ietf.org>
>> Cc: <sip@ietf.org>
>> Sent: Wednesday, June 28, 2006 12:50 AM
>> Subject: [Sip] I-D ACTION:draft-ietf-sip-outbound-04.txt
>>
>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>> directories.
>>> This draft is a work item of the Session Initiation Protocol  Working 
>>> Group of the IETF.
>>>
>>> Title : Managing Client Initiated Connections in the
>>>                          Session Initiation Protocol (SIP)
>>> Author(s) : C. Jennings, R. Mahy
>>> Filename : draft-ietf-sip-outbound-04.txt
>>> Pages : 34
>>> Date : 2006-6-27
>>>
>>> The Session Initiation Protocol (SIP) allows proxy servers to
>>>   initiate TCP connections and send asynchronous UDP datagrams to  User
>>>   Agents in order to deliver requests.  However, many practical
>>>   considerations, such as the existence of firewalls and Network
>>>   Address Translators (NATs), prevent servers from connecting to User
>>>   Agents in this way.  This specification defines behaviors for User
>>>   Agents, registrars and proxy servers that allow requests to be
>>>   delivered on existing connections established by the User  Agent.  It
>>>   also defines keep alive behaviors needed to keep NAT bindings open
>>>   and specifies the usage of multiple connections.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-04.txt
>>>
>>> To remove yourself from the I-D Announcement list, send a message to
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>>>
>>> A list of Internet-Drafts directories can be found in
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>>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
>>>
>>> Internet-Drafts can also be obtained by e-mail.
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>>
>>
>> ---------------------------------------------------------------------- 
>> ----------
>>
>>
>>> _______________________________________________
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>>
>>
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-04.txt 
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On Jul 2, 2006, at 10:21 AM, Jeroen van Bemmel wrote:

>
>>> 4) (minor remark) Perhaps some words could be added that the UAC   
>>> SHOULD perform keepalives when configured to do so, even if it   
>>> detects that there is no NAT. There could be a transparent  
>>> firewall  device that needs to be kept open.
>>>
>>
>> I agree this is the behavior we want  - I thought it already more  
>> or  less said that. It  does not suggest detecting if their is a  
>> NAT  there at all - if the URI has the tag, you do keep alive.
>>
>
> and if there is no such tag, it MUST NOT do keep alive (or at least  
> not STUN, TCP keepalives would be harmless) ?

I'm sympathetic to what you are saying as I had originally proposed  
that it be allowed to do this - I pointed out the a SIP server would  
reject the message. Lots of people felt strongly that we should not  
send STUN messages to SIP servers that did not support it. The issue  
came up in some WG meeting - there was strong consensus that we  
should not send keep alive to things that did not support them. 

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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-04.txt 
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On Jul 2, 2006, at 10:21 AM, Jeroen van Bemmel wrote:

>>> 1) The UAC needs a positive indication that the selected proxy  
>>> can indeed handle STUN packets without choking on them. It is  
>>> not  sufficient if the URI it is configured with says  
>>> 'keepalive=stun',  that's a wish but no guarantee.
>>>
>>>
>>
>> Why is this a wish not a guarantee?
>
> Because the configuration may be wrong, ie the entity that  
> provisioned the URI thinks the element supports outbound, but in  
> fact it does not. There can be various reasons for this, like  
> invalid manual provisioning, some edge proxy is replaced with one  
> that does not support outbound and someone forgets to update DNS, etc

Well totally ignoring anything to do with outbound - I don't really  
see how this is different from when someone configure the proxy as  
foo.com when it is supposed to be bar.com

I do think it is important to get sip so that a user might configure  
an AOR name and password and everything else just happens but that is  
the point of the config framework. I agree configuration is a  
nightmare today and we need to make it easier.

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>> Because the configuration may be wrong, ie the entity that  provisioned 
>> the URI thinks the element supports outbound, but in  fact it does not. 
>> There can be various reasons for this, like  invalid manual provisioning, 
>> some edge proxy is replaced with one  that does not support outbound and 
>> someone forgets to update DNS, etc
>
> Well totally ignoring anything to do with outbound - I don't really  see 
> how this is different from when someone configure the proxy as  foo.com 
> when it is supposed to be bar.com
>

Conceptually not so different, but there would be a good chance that foo.com 
would either accept nothing or SIP on common SIP ports. The problem is that 
STUN is sent to the same port. Were it a different port, then the risk would 
be much lower. The combination of non-SIP (even binary) and same port makes 
it troublesome.

> I do think it is important to get sip so that a user might configure  an 
> AOR name and password and everything else just happens but that is  the 
> point of the config framework. I agree configuration is a  nightmare today 
> and we need to make it easier.

Easier and more foolproof, and more easy to debug / trace when/where things 
go wrong. For all of this, explicit confirmation of outbound support helps. 


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This looks good - a few comments - all very trivial

Does this update 3261? If so, nice if header on first page had that

Section 3 - Long lived definition. I'm really not sure what it means  
to say that a GRUU has a long life than any of the registrations or  
that it is viable for every. I'm not sure it changes behavior of  
anything. Is the important property that if the same AOR/Instance  
registers two different times it gets the same AOR? Anyways, I can  
live with the text exactly as is but I was not getting why it was  
important.

Figure 1: On the diagonal arrow between GRUU and contact, I think the  
end near contact should have "0..1" and the end near the GRUU should  
have "0..n" - right now this is swapped.

Section 6, page 12, 4th para or so ... you are talking about mid- 
dialog requests. Might be nice to mention how a transaction stateful  
proxy knows if something is a middialog request or not.

Section 6, last para before section 7. I'm not sure I get how  
anything behaves differently from an unbound GRUU that exists vs the  
GRUU does not exist. Again, i don't have some huge problem with  
current text but when I read it, it makes me wonder if there is some  
difference which I should understand.

Section 7.1.1.1 - There has been some problems with the way 3261  
allowed a user to install forwarding rules using a REGISTER request.  
Part of the problem is that it allows an amplification attack on  
someone else that did not provide any consent. It makes me wonder if  
now that we have the consent framework moving forward, do we really  
want to continue down this path. I realize this is pretty orthogonal  
problem to GRUU.

Section 8.1.1 - 2nd para. You have "However, that AOR will be  
rewritten ..." - I found this hard to follow. Is it the request URI?  
I'm not sure how to fix but you might want to have a read of the end  
of that para and see if it is all clear to you.




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Something more about Proxy-Require for outbound:

>>
>> 3) The Proxy-Require mechanism we have discussed could be useful to  UACs 
>> that want to ensure that a selected edge proxy supports  outbound, 
>> aborting if not.
>>
>
> I'll let Rohan reply to this but I don't think I see what problem it 
> solves.
>

There's another, more fundamental reason for this. The draft defines an 
option-tag which can occur in Supported, Require, Proxy-Require, and 
Unsupported headers. For each of these cases the semantics and expected 
behavior should be defined, either in this draft or implicitly (eg based on 
RFC3261)

Without stating otherwise, the default behavior for Proxy-Require would be: 
return 420 Bad Extension if not understood, else process as defined for that 
feature. In particular, for the current draft a proxy would not remove the 
Proxy-Require header from the request it forwards. So if an UAC adds 
Proxy-Require, it means that every proxy on the path must support it.

This is not required nor desired, hence we'd better state explicitly that a 
proxy needs to remove any Proxy-Require: outbound when forwarding

Regards,

Jeroen 


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Samir Srivastava wrote:
> Hi Mike,
>  
>      My answers below. IMHO, it's an important piece of work. Now we
> have much cleaner solution with Proxy-Require instead of
> mess of SIPS etc. 
>  
> I remember in wacky thread  Paul Kyzivat mentioning that if SIPS etc so
> bad, why not we define a new clean
> solution. I request his comments on this approach. Hoping that Dave Oran
> who raised the question and Paul listening.

I don't consider myself competent as a security guy.

Seems like there are limits on what you can do with Via header checking. 
Mike covered some of that, but I guess you still have to trust all the 
proxies on the path. But presumably that is assumed.

Having the UAC enumerate the acceptable ciphersuites seems problematic 
to me. (But I am way out of my depth here.) It could be that some part 
of the path has proxies that don't support anything listed, but support 
something newer and stronger that isn't listed. I wonder if it would be 
better to somehow rank ciphersuites by their strength, and have the UAC 
simply specify the minimum strength it will accept.

	Paul

> Thx
> Samir
> 
> ________________________________
> 
> From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com] 
> Sent: Wednesday, June 28, 2006 8:05 AM
> To: sip@ietf.org
> Subject: [Sip] Comments on draft-audet-sip-sips-guidelines
> 
> 
> 
> Samir,
> 
> Read through the draft and had a few nits and comments.
> 
> Nits: 
> 
> Sec 3, 2) "need to use" -> MUST use? 
> 
> page 4, top: "renegotiate"
> 
> Will need UAC, Proxy, UAS sections in future draft.
> 
> Comments:
> 
> Option tag: e2e-ciphersuite: Include description of semantic for
> Supported header as well. When text is expanded these could be separate
> sub-sections.
> 
> Suggestion: q-value should order the suites both by strength of
> algorithm/key size (first) and then by preference (second). Reason is to
> see degree of downgrade. 
> 
> SS>> Will fix in the next version. 
> 
> Via header should describe action that a proxy node MUST verify that the
> Via header added by the previous hop is valid, i.e. that it did in fact
> use the cipher named in the Via for the incoming link. May want to send
> a report directly back to the UAC Contact notifying of the compromise. 
> 
> SS>> Will fix in the next version. 
> 
> Not sure error code sent back would help.
> 
> Not sure how this would work with IPSEC. The applications may not have
> as much control of the host or communication with it (cross-layer
> indicators). I could see the IPSEC cipher listed as a substitute. But,
> then the application needs to have a means to know if it is equivalent,
> unless it names those up front in the new header. For fairly well known
> systems (e.g. follows the ANSI Generic Security standard), this might be
> a managable list. 
> 
> SS>> It's still an OPEN ISSUE. I need to consult IPSEC experts, on how
> difficult it is for application to get the info/control ? 
> 
> Does IANA already have a registry of the exact string names used for
> each ciphersuite? 
> 
> SS>> Will have more details on this, in the next version.
> 
> Last half of second paragraph above Sec 4 should be in Sec 5 security
> considerations. Should also point out that this may not prevent a proxy
> from dropping out Security-Level header. Thought, should the cipher also
> be included in the Record-Route header? The UAC should approve the route
> set for continuing the call? 
> 
> SS>> Good Catch. Will have discussions on this in the next version.
> 
> The draft is a good start. Thanks for taking the ball and running with
> it.
> 
> Mike
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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<html><body>
<p><tt>Hi Joe,<br>
</tt><br>
<tt>My understanding is that since response has to follow the transport=
 published in via header, the UAS does not have choice in choosing tran=
sport for response.<br>
</tt><br>
<br>
<tt>regards<br>
</tt><tt>Rayees<br>
</tt><br>
<br>
<br>
<tt>-----Joe &lt;joe.uelk@gmail.com&gt; wrote: -----<br>
</tt><br>
<tt>To: sip@ietf.org<br>
</tt><tt>From: Joe &lt;joe.uelk@gmail.com&gt;<br>
</tt><tt>Date: 07/01/2006 01:22AM<br>
</tt><tt>Subject: [Sip] SIP Transport Selection Question<br>
</tt><br>
<tt>Under the Clients section 3261 states:<br>
</tt><br>
<tt>&quot;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 b=
e sent using an RFC 2914 congestion controlled transport protocol, such=
 as TCP.&quot;<br>
</tt><br>
<tt>Is this for &quot;requests&quot; (INVITE, etc.) only? What if an IN=
VITE were sent as UDP but a response (200 OK) were to come back with ex=
tra headers which breached the MTU? Must the node sending the OK switch=
 the transport to TCP? Is this optional?<br>
</tt><br>
<tt>Regards,<br>
</tt><tt>Joe<br>
</tt><tt>_______________________________________________<br>
</tt><tt>Sip mailing list =A0https://www1.ietf.org/mailman/listinfo/sip=
<br>
</tt><tt>This list is for NEW development of the core SIP Protocol<br>
</tt><tt>Use sip-implementors@cs.columbia.edu for questions on current =
sip<br>
</tt><tt>Use sipping@ietf.org for new developments on the application o=
f sip</tt><br>
<tt>*****FSS-Private *****&quot; DISCLAIMER: This message is proprietar=
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From sip-bounces@ietf.org Mon Jul 03 06:39:40 2006
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Subject: RE: [Sip] Update to fork-loop-fix submitted
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Hi Robert,

I had suggested one more alternative

(http://www1.ietf.org/mail-archive/web/sip/current/msg14359.html )

Quoted from my previous mail:

<quote>
I was thinking how to reduce the impact of overhead for calculating the
digest.

If it is suggested as follows, I think it will reduce the overhead on
normal scenarios:

Proxy checks to see if its address is there in the via already.=20
If it is not there, it proceeds normally without calculating the digest
and inserting Via to the outgoing request without the digest.
If it is there, then it is possible that the request is looping. So,
this time around, it calculates a digests and inserts and sends it
along.
If it is looping, the request will definitely come in again, which can
be rejected.
This way the impact of calculating the digest is required (reduced to a
smaller percent of the sessions) only when the proxy is suspicious the
request might be looping.
Of course this results in the scenarios mentioned in the id to be
successful in completing the first loop, but the second loop will be
detected and arrested. I think this can be applicable to general loop
detection, and not just fork loop fixes.
</quote>

Regards,
Ravi.

--
=20
Ravishankar. G. Shiroor
Wipro Technologies, Bangalore.
=20
ravishankar.shiroor@wipro.com
--
=20

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: Sunday, June 25, 2006 8:22 PM
> To: IETF SIP List
> Subject: [Sip] Update to fork-loop-fix submitted
>=20
> Until it appears in the repository, see:
> http://www.nostrum.com/~rjsparks/draft-ietf-sip-fork-loop-fix-02.txt
>=20
> This was a moderately heavy touch - based on list feedback,=20
> I've moved and updated the sections covering the=20
> loopdetection algorithm in from 3261. In the process of=20
> writing the clarifying text, I discovered several places that=20
> need clarification. They are listed as Open Issues in the=20
> draft. Basically there are fields that I can't recall or=20
> reconstruct how they would be useful in the part of the hash=20
> used for loop-detection or spiral differentiation. If anyone=20
> can remember why they need to be included, send a note to the=20
> list and we'll capture that in the draft.
>=20
> Please review the updated loop-detection text very carefully.
>=20
> Here's the changelog from the draft:
>=20
>        Integrated several editorial fixes suggested by=20
> Jonathan Rosenberg
>=20
>        Noted that the reduction of the attack to a single registration
>        against a single URI as documented in previous versions, is, in
>        fact, going to be effective against implementations=20
> conforming to
>        the standards before this repair.
>=20
>        Re-incorporated motivation from the original=20
> maxforwards-problem
>        draft into the security considerations section based=20
> on feedback
>        from Cullen Jennings
>=20
>        Introduced replacement text for the loop detection algorithm
>        description in RFC 3261, fixing the bug 648 (the=20
> topmost Via value
>        must not be included in the second part) and clarifying the
>        algorithm.  Removed several other fields suggested by 3261 and
>        placed open issues around their presence.
>=20
>        Added a Notes to Implementors section capturing the=20
> "common way"
>        text and pointing to the interoperability issues that have been
>        observed with loop detection at previous SIPits
>=20
>=20
> Thanks,
>=20
> RjS
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Mon Jul 03 11:06:29 2006
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Subject: Re: [Sip] Proposals for review process to conclude WGLCs
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I really like the first idea - of using a conference call. I think it 
imposes a specific deadline that puts pressure on folks to get the 
reviews done in a timely way. Otherwise, you get embarassed by missing a 
call that you committed to be on. Its the same logic behind the IETF 
draft deadline.

A few more tweaks to what you propose - I'd suggest that reviewers have 
to show up to the call with specific comments on the document, which 
they can work through and raise.

Also, though the output of the review can be go/no-go, lets remember 
everything needs to come through the mailer ultimately.

Thanks,
Jonathan R.

Dean Willis wrote:

> 
> One of the questions we've struggled with is how to get sufficient  
> review on a document under WGLC to make sure it's really done.
> 
> 
> SIPPING is following a new procedure with designated reviewers, doing  
> review by email, and sending stuff back to the. Apparently this  suffers 
> from the expected issues, with designated reviewers sometimes  not quite 
> meeting their review commitments.
> 
> Cisco made me take a peer-review refresher class the other day, so  this 
> started me thinking about some alternatives.
> 
> In the model taught in this class, each peer review  includes 1) a  
> "moderator" or "reader" who frames the questions and keeps things on  
> track, 2) a "recorder" who records conclusions and issues raised  during 
> the review, 3) the author(s), who neither presents no  interprets the 
> work, but does respond to specific questions about it,  and 4) the 
> reviewers, who hopefully have expertise and have committed  time to the 
> review.
> 
> 
> I'd like to propose a couple of alternative models to what SPPING has  
> been doing and seeing if either resonates with our group.
> 
> 
> 1) Real-time review sessions via conference call: When we get ready  to 
> do a final review, the chairs assemble a review team. Presumably,  this 
> review team is assembled by polling the list for volunteers, and  by 
> pushing extra hard for stakeholders and other people with specific  
> expertise. The moderator builds up a checklist and set of basic  
> questions to consider, and distributes it to the review team. At a  tiem 
> agreed to by enough of the participants, we have a conference  call, not 
> to exceed two hours, and work through the checklist in a  moderated 
> fashion. At the conclusion of that two hours, we either  reach a 
> positive decision, a negative decision (with list of issues  and 
> actions), or decide we need more review.
> 
> 
> 2) Moderated email-review, using the entire list as the "review  team". 
> In this model, the chairs pick a moderator and recorder for  the review. 
> The moderator sets up a checklist and sends it to the  list, along with 
> an announcement of the review. The moderator then  goes through the 
> checklist item-by-item on list, polling for  comments. The recorder 
> records any issues or conclusions. Each item  might take two or three 
> days for primary comments, but could be  revisited during the review 
> process if needed. The author could  answer specific questions about the 
> draft, but not paraphrase or  interpret.  At the conclusion of the 
> review, the moderator and  reviewer would report to the chairs, who 
> would decide whether we have  issues, a conclusion, or need further review.
> 
> 
> What do y'all think?
> 
> -- 
> Dean
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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To: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Proposals for review process to conclude WGLCs
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Dean Willis wrote:
> In the model taught in this class, each peer review  includes 1) a  
> "moderator" or "reader" who frames the questions and keeps things on  
> track, 2) a "recorder" who records conclusions and issues raised  during 
> the review, 3) the author(s), who neither presents no  interprets the 
> work, but does respond to specific questions about it,  and 4) the 
> reviewers, who hopefully have expertise and have committed  time to the 
> review.

Dean: This is probably the CMMI peer review (or a derivation
thereof).  I have been part of such reviews for software processes;
they tend to work well assuming the reviewers indeed know
what is going on with the code.  I have also observed that often
times, the reviews get bogged down on how many spaces of indentation
are permissible.  Hence the need for reviewers that are astute.

In the context of your proposals:

> 1) Real-time review sessions via conference call:
[...]
> 2) Moderated email-review, using the entire list as the "review  team". 
[...]

> What do y'all think?

(1) may be more productive, for the reasons that Jonathan mentioned
as well.

Also, I am mapping this to something else that happens in
the IETF -- the NomCom meeting, which is literally, a *peer's*
review instead of a peer review.  At these teleconference meetings,
it is expected that the voting members read the questionnaires/bios
in order to vote on the candidates.  Given that the voting members
are:

   (a) interested in the process, which is why they volunteered
   (b) prepared (by reading the bio and questionnaire)
   (c) on a real-time review session via conference call

things tend to go smoothly in so far as the review of a
candidate is concerned (whether or not everyone agrees that this
is the right candidate is another matter altogether).  Being on
the phone and preparing to verbalize your thoughts appears to
allow introspection of what is it that you are reviewing.

Substitute (a), (b), and (c) above for reviewing WGLC documents,
and I think you'll have a good working model to conclude WGLCs.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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From sip-bounces@ietf.org Mon Jul 03 12:28:45 2006
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Subject: [Sip] -01 version of domain-certs (Domain Certificates in SIP)
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Folks: Dean has been kind enough to add domain-certs-01 to the
reading list of connect-reuse.  The sole open issue in connect-reuse
revolves around virtual SIP servers and certs, and there are
precious few guidelines we have around these except for SIP
Identity (section 13.4).

Domain-certs contains text to foster discussion on this and
a couple of other issues, server farms for instance, which
has not seen any discussion.  In April we had discussions on
the identities stored in domain certificates.  The summary
of these discussions is included in Section 4 of domain-certs-01.

Please eyeball the draft if you have some cycles and can
comment on it.  It is available at
http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-certs-01.txt

Thank you.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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From sip-bounces@ietf.org Mon Jul 03 13:05:24 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
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Subject: [Sip] Presentation slides for SIP WG sessions
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Dean has posted the agenda at 

http://www.softarmor.com/sipwg/meets/ietf66/agenda.html

(and which will also be posted on the official IETF site as well)

https://datatracker.ietf.org/public/meeting_materials.cgi?meeting_num=66

which indicates various presenters on topics. Could we ask (nay demand) that presenters send their slides to Dean and myself at least 30 minutes before the WG sessions are due to start. We would like time to get these posted before the session starts so that remote participants have the opportunity to access them while the presentation is in progress. 

(If you fail to produce them, then we may make you present without the slides, so that all participants - both in the room and remote - have the opportunity of enjoying the same experience!)

Obviously the earlier you deliver them the better for the chairs and all participants, and we can take them immediately if you so wish.

Remember that presentations are about identifying the issues to the WG to resolve, and not about what you have done to the draft - you should already have presented the history of how you got to the current draft to the SIP list when you submitted it, and ideally it should also be included in a tear off section in the document - and the WG is expected to have read the document. And remember to leave time for the WG to discuss these issues - do not have so many slides that we stand no change of the WG presenting their opinion on issues.

External participants may want to bookmark 

http://videolab.uoregon.edu/events/ietf/

and the presentations will be available at the same locations as the agenda.

as the place where they can access the streaming audio of the various sessions.

regards

Keith

Keith Drage
Lucent Technologies
drage@lucent.com
tel: +44 1793 776249

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From sip-bounces@ietf.org Mon Jul 03 13:11:56 2006
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Currently the SIP diagnostic draft, sip-hop-limit-diagnostics talks
about returning additional diagnostic information in the 483. I am
wondering whether there would be a need to extend this draft to make 
it generic so that caller can request for providing such additional
diagnostic information even in 200 OK. The usecase that I have in
mind is to provide some kind of tool to the service provider who
wants to use this diagnostic information to check the path/domain
the call has traversed and take measures to update/optimize the
routing policies.

If the above requirement is acceptable, then the security aspect of
such a functionality will also have be addressed (may be through authorization
mechanism defined in 3261).

Regards,
Prakash

HTIPL FIN PDT System Engineering Group
Huawei Technologies India Pvt. Ltd
Bangalore, India


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From sip-bounces@ietf.org Mon Jul 03 14:29:34 2006
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Subject: [Sip] Comments on draft-ietf-sip-outbound-04
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Herewith my comments on the above document.

Technical
---------

1)	Section 2.1. The word "connection" gets used at various points in the document, including the introduction. I'd appreciate a definition.

2)	I thought that we agreed at the last WG meeting that for TCP, keepalives were perfectly appropriate if they worked. I guess this is primarily a UAC issue rather than a proxy issue. TCP keepalives seem underdocumented, and in some places it seems that STUN is the only mechanism. Should there not be additional information on TCP keepalives at the end of section 3.1. The note in 3.4 seems entirely too negative, implying that the use is deprecated altogether. Section 4.4.1 is also implying that the UAC must only support STUN keepalives - where is the TCP equivalent section. Section 7 applies to all implementations, but ignores the support of TCP keepalives.

3)	Section 3.2:

   The registrar challenges this registration to authenticate Bob. When
   the registrar adds an entry for this contact under the AOR for Bob,
   the registrar also keeps track of the connection over which it
   received this registration.

I am not sure of the relevance of the first sentence. Yes it may happen here, but the way it is written implies that outbound only works if it occurs.

4)	Section 3.4 (last paragraph) indicates:

   Regardless if they are in the same administrative domain, this
   specification requires that Registrars and Edge proxies support the
   Path header mechanism in RFC 3327 [RFC3327].

I found the requirement for UAC sort of in subclause 4.2. I found no requirement for the edge proxy to support the Path header extension in section 5. I did find a clear statement for the registrar in section 6.1.  

5)	Section 4.1:

   The UA SHOULD include a "sip.instance" media feature tag as a UA
   characteristic [RFC3840] in requests and responses.  As described in
   [RFC3840], this media feature tag will be encoded in the Contact
   header field as the "+sip.instance" Contact header field parameter.
   The value of this parameter MUST be a URN [RFC2141].  One case where
   a UA may not want to include the URN in the sip.instance media
   feature tag is when it is making an anonymous request or some other
   privacy concern requires that the UA not reveal its identity.

As currently written this seems to be a "MUST" followed by an instance where the "MUST" doesn't apply, and this is not allowed in RFC 2119. However it appears that this was not the intention, and that some rewording is required.

6)	Section 4.2:

   The UAC MUST indicate that it supports the Path header [RFC3327]
   mechanism, by including the 'path' option-tag in a Supported header
   field value in its REGISTER requests.  Other than optionally
   examining the Path vector in the response, this is all that is
   required of the UAC to support Path.

Either the UAC supports the Path header extension, or not. I am not sure that adding the second sentence helps.

7)	Section 4.2.1: I did not understand this subclause at all. Can you explain what you are trying to communicate?

8)	Section 4.4.2:

   When a flow to a particular URI in the outbound-proxy-set fails, the
   UA needs to form a new flow to replace the old flow and replace any
   registrations that were previously sent over this flow.  Each new
   registration MUST have the same reg-id as the registration it
   replaces.  This is done in much the same way as forming a brand new
   flow as described in Section 4.2; however, if there is a failure in
   forming this flow, the UA needs to wait a certain amount of time
   before retrying to form a flow to this particular next hop.

is almost a dupulication of the requirement in 4.2:

   These ordinary registration requests MUST also add a distinct reg-id
   parameter to the Contact header field.  Each one of these
   registrations will form a new flow from the UA to the proxy.  The
   reg-id sequence does not have to be sequential but MUST be exactly
   the same reg-id sequence each time the device power cycles or reboots
   so that the reg-id values will collide with the previously used
   reg-id values.  This is so the proxy can realize that the older
   registrations are probably not useful.

Can we make clearer that one relates to choosing a new reg-id on an initial registration, and the other relates to reusing the previous reg-id on a reboot and that these are therefore two distinct cases.

9)	Section 4.4.2. Should the default times vary for TCP? 4.4 specifies different times between keepalives for TCP and UDP.

10)	Section 6.2: Under what circumstances does the SHOULD not apply? And if the SHOULD is not followed, what does the registrar do instead?

11)	Not sure that reference RFC 3968 and reference RFC 3969 qualify as normative references. They are only explanation to the IANA on how the text is added to the registry.

12)	GRUU currently defined as an informative reference. Given that it defines the instance id which is needed for this protocol, I would have thought it needed to be normative.

NITS
----

13)	Section 3.3. This text starts out by using the term "administrative domain" and then starts shortening it to just "the domain". Given that "domain" is such an overused word I suggest we use "administrative domain" in full throughout, or find a completely new term.

14)	Section 3.4, 2nd paragraph has a "SHOULD". This is a general introduction, and I am surprised that RFC 2119 terminology appears at all. Suggest requirement is moved to a later clause (unless it is already there).

15)	I counted three different ways of representing notes throughout the text. I like the style of indentation, with a clear first word saying note, which you have used as one of these options. Can we align the remainder of the instances where the word "note" appears with this.

16)	Section 4.2 should probably do with some substructuring with inclusion of some header level 3 titles. The contents of 4.2 and 4.2.1 seem unrelated.

17)	Section 4.2, 3rd paragraph:

   These ordinary registration requests MUST also add a distinct reg-id
   parameter to the Contact header field.  Each one of these
   registrations will form a new flow from the UA to the proxy.  The
   reg-id sequence does not have to be sequential but MUST be exactly
   the same reg-id sequence each time the device power cycles or reboots
   so that the reg-id values will collide with the previously used
   reg-id values.  This is so the proxy can realize that the older
   registrations are probably not useful.

Not sure "collide" is the right word. Perhaps "match the previously..."

18)	Text seems to alternate between using "UAC" and "UA" and "User Agent". Where the UA is a UAC, can we use the correct term?

19)	Where response codes are identified can we include them in full RFC 3261 format in all cases, e.g. correct the following:

         For example, if a proxy forwarded a request to 4 locations, and
         received 503, 407, 501, and 404 responses, it may choose to
         forward the 407 (Proxy Authentication Required) response.

20)	Appendix A is followed by a Section 15.

regards

Keith


Keith Drage
drage@lucent.com
tel: +44 1793 776249

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From: "Drage, Keith (Keith)" <drage@lucent.com>
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Technical:
----------

1) I need to understand whether this updates RFC 3263 or just extends it. Part of this is what appears on the cover page and in the RFC index as and when it is finally approved, but the other part also relates to compatibility, see later comment.

2) I think you have already acknowledged this, but the document requires revision to align with the terminology in the latest version (-04) of draft-ietf-sip-outbound.

3) In section 5, as far as I understand, the NAPTR request is identical for a UA that supports RFC 3263 and for one that supports this extension. Therefore the difference is in the response. Therefore the paragraph of UA requirements:

   As per RFC 2915 [4], the client discards any NAPTR records whose
   services fields are not applicable.  A client resolving a SIPS URI
   MUST discard any services that do not contain "SIPS-O" as the
   protocol in the service field.  A client resolving a SIP URI MUST
   retain records with "SIP-O" and the transport protocols the client
   supports and SHOULD retain records with "SIPS-O", if the client
   supports TLS.

must work with a NAPTR response that has no knowledge of outbound. Is the above paragraph compatible with this? 

Additionally note that the words "if the client supports TLS" are superfluous. As RFC 3261 requires support of TLS, we are not documenting capabilities where TLS is not supported.

regards

Keith

Keith Drage
drage@lucent.com
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From sip-bounces@ietf.org Mon Jul 03 15:42:34 2006
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Subject: Re: [Sip] draft-ietf-sip-hop-limit-diagnostics-03
From: Scott Lawrence <slawrence@pingtel.com>
To: Cullen Jennings <fluffy@cisco.com>
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On Mon, 2006-06-26 at 09:00 -0700, Cullen Jennings wrote:
> On Jun 16, 2006, at 2:59 PM, Scott Lawrence wrote:
> 
> >
> > There are, I think, two issues remaining for discussion:
> 
> I disagree - I don't think the draft of the discussion has addressed  
> the issue I brought up.

I'm going to guess here - do you mean the issue of how large the
response should be?

There is text in the draft on _how_ to reduce the size of the response.

3261 does not provide guidance clear guidance on response size limits.
It _does_ say that all implementations MUST be able to handle messages
up to the maximum UDP packet size, which can probably handle even 70 hop
messages.  If you think that SIP needs better definition of _when_ to
limit response sizes and _what_ size they should be limited to, then I
think that's an interesting discussion, but it's a bigger question than
this error response and a problem that already exists whether this draft
is accepted or not.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Mon Jul 03 16:39:01 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
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Subject: [Sip] Comments on draft-ietf-sip-connect-reuse-05
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I have the following comments on the draft:

Technical
---------

1)	I am interested in the question of whether this draft updates RFC 3261 or not, and would appreciate some list discussion on that issue. What, if any, RFC 3261 requirements does it change?

NITS
----

2)	The IANA considerations may need some additional information. You may want to look at the outbound draft to see if there is information there you could use as a template.

3)	You may want to update Rohan's affiliation on the cover page (I assume he has it right on outbound).

regards

Keith

Keith Drage
Lucent Technologies
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From sip-bounces@ietf.org Mon Jul 03 17:16:48 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: Rayees Khan <rayees.khan@flextronicssoftware.com>, Joe <joe.uelk@gmail.com>
Subject: RE: [Sip] SIP Transport Selection Question
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There is an implicit assumption in the MTU size that may or may not prevent fragmentation.

The request is specified to be MTU minus 200 bytes such that it is unlikely that the addition of headers and parameters by proxies along the path will still keep it in the MTU size, and a further assumption exists that responses will be no larger than requests.

Neither of these assumptions is true in all circumstances.

However a response to a request sent over UDP will be returned over UDP, and the response will in most cases get through even if the MTU size is exceeded. The problem is that the response is not delivered reliably and there may be unnecessary retransmissions of the response in order for it to be delivered.

As over the years the size of SIP messages have exploded, there have been calls for the UDP option to be removed entirely.

regards

Keith


-----Original Message-----
From: Rayees Khan [mailto:rayees.khan@flextronicssoftware.com]
Sent: 03 July 2006 09:39
To: Joe
Cc: sip@ietf.org
Subject: Re: [Sip] SIP Transport Selection Question


Hi Joe,

My understanding is that since response has to follow the transport published in via header, the UAS does not have choice in choosing transport for response.


regards
Rayees



-----Joe <joe.uelk@gmail.com> wrote: -----

To: sip@ietf.org
From: Joe <joe.uelk@gmail.com>
Date: 07/01/2006 01:22AM
Subject: [Sip] SIP Transport Selection Question

Under the Clients section 3261 states:

"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 congestion controlled transport protocol, such as TCP."

Is this for "requests" (INVITE, etc.) only? What if an INVITE were sent as UDP but a response (200 OK) were to come back with extra headers which breached the MTU? Must the node sending the OK switch the transport to TCP? Is this optional?

Regards,
Joe
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As over the years the size of SIP messages have exploded, there have
been calls for the UDP option to be removed entirely.

SS>> Refer followwing thread from MMUSIC, for how big SDP can be with
ICE etc extensions (~10 K)
http://www1.ietf.org/mail-archive/web/mmusic/current/msg04204.html

Should we take action sooner ?=20

Thx
Samir

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From sip-bounces@ietf.org Mon Jul 03 18:53:40 2006
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I have the following comments on the above draft:

Structural
----------

1)	Abstract: From reading the document, it is clear that presence systems are not covered at all in this document for various reasons. Therefore PUBLISH, SUBSCRIBE and NOTIFY are essentially out of scope. It seems to me that this should be made much more explicit in the abstract.

2)	The SIP extension guidelines (RFC 4485) need to be applied to this document, which requires some substantial restructuring. The absence of application of these rules mean that the requirements of the document are extremely unclear. For example RFC 4485 states (section 4.5):

   Developers of protocols often get caught up in syntax issues, without
   spending enough time on semantics.  The semantics of a protocol are
   far more important.  SIP extensions MUST clearly define the semantics
   of the extensions.  Specifically, the extension MUST specify the
   behaviors expected of a UAC, UAS, and proxy in processing the
   extension.  This is often best described by having separate sections
   for each of these three elements.  Each section SHOULD step through
   the processing rules in temporal order of the most common messaging
   scenario.

Absence of this application means that in places in this document it is impossible to determine whether a requirement and its required support occurs as the sender of the location information, or the receiver of the location information.

3)	Section 1 (and possibly section 2) should be redrafted taking into account RFC 4485 section 4.7:

   Too often, extension documents dive into detailed syntax and
   semantics without giving a general overview of operation.  This makes
   understanding of the extension harder.  It is RECOMMENDED that
   extensions have a protocol overview section that discusses the basic
   operation of the extension.  Basic operation usually consists of the
   message flow, in temporal order, for the most common case covered by
   the extension.  The most important processing rules for the elements
   in the call flow SHOULD be mentioned.  Usage of the RFC 2119 [1]
   terminology in the overview section is NOT RECOMMENDED, and the
   specification should explicitly state that the overview is tutorial
   in nature only.  This section SHOULD expand all acronyms, even those
   common in SIP systems, and SHOULD be understandable to readers who
   are not SIP experts. [27] provides additional guidance on writing
   good overview sections.

I found this text much too wordy, and considered that it should be possibly to convey the necessary information to the reader in about half the length.

4)	Section 3 need to have all the RFC 2119 language removed (see history-info and outbound as examples). I remain open as to whether it appears at the start of the document, as here and in history-info, or at the end, as in outbound, but given the length, we may want to consider moving it to the end.

5)	Section 6 should probably follow immediately on from the procedures in the bulk of section 4, and probably be split as in comment 2) above.

6)	Section 7 should be integrated into the main procedures text.

Technical
---------

7)	Section 1, of the three examples given in this introduction, item 2) is out of scope. I had to read an awful lot of text to discover that fact. 

8)	Section 2, 1st paragraph. Probably not appropriate to use the term "trust model". That term is defined in RFC 3324 and RFC 3325, and I am not convinced the usage here is one and the same.

9)	Section 2, 3rd para. 

   No routing of the request based on the location contents is required
   in this case, therefore no SIP Proxies between these two UAs need to
   view the location information contained in the SIP message(s).  The 
   UAC should know certain types of messages will be routed based on 
   the UA's location when creating a message.  

Did not understand what this paragraph is meant to be telling me - can you explain it.

10)	Section 3.1 UAC-4: This is not a generic requirement as it is already identified that certain methods are outside the scope of this document.

11)	Section 3.1 UAC-9, UAC-10: I assume this is specified this way so that misconfiguration releases the location rather than hides it. I believe this contravenes the regulatory requirements of some countries. I believe in Japan privacy requirements are more important that the transmission of the information for emergency calls. If someone from Japan was writing these requirements they would therefore indicate exactly the opposite.

12)	Section 3.2 UAS-6 and section 3.3 Proxy-4. Firstly this is halfway into implementation. Secondly if it is required to address location errors in requests, why is the same consideration not given to location errors in responses. 

13)	Section 3.3 Proxy-1: This merely repeats RFC 3261, rather than developing a new requirement.

14)	Section 3.3 Proxy-3: This seems to be rewriting the requirements to correspond to the solution. I understand the problem is that the solution cannot distinguish between multiple locations, nor understand the reliability, and therefore a receiver may get confused. Firstly this does not seem to me to be a SIP issue, and secondly in general the ability to transport multiple locations could be regarded as useful - they just need some form of assertion as to who provided which location.

15)	Section 4 and elsewhere. I don't find location in responses well described at all. It seems to be an afterthought, rather than something that is decided based on a full set of requirements. I have already made a comment with regard to requirements on error handling. Should we support location in response, or force the UAS to generate a request of its own in order to send a location?

16)	Section 4. Should the instances of "URI" here be "URN".

17)	Section 4 specifies:

   Because a person's location is generally considered to be sensitive 
   in nature, certain security measures need to be taken into account 
   when transmitting such information.  Section 26 of [RFC3261] defines
   the security functionality SIPS for transporting SIP messages with 
   either TLS or IPSec, and S/MIME for encrypting message bodies from 
   SIP intermediaries that would otherwise have access to reading the 
   clear-text bodies.  SIP endpoints SHOULD implement S/MIME to encrypt
   the PIDF-LO message body (part) end-to-end.  The SIPS-URI from 
   [RFC3261] MUST be implemented for message protection (message 
   integrity and confidentiality) and SHOULD be used when S/MIME is not
   used.  

   The entities sending and receiving location MUST obey the privacy 
   and security rules in the PIDF-LO, regarding retransmission and 
   retention, to be compliant with this specification.

   Self-signed certificates SHOULD NOT be used for protecting PIDF-LO, 
   as the sender does not have a secure identity of the recipient.

However RFC 4119 is already normatively referenced which contains the following text:

   For the purposes of this document, only the securing of a PIDF
   document as a whole, rather than element-by-element security, is
   considered.  None of the requirements [10] suggest that only part of
   the information in a location object might need to be protected while
   other parts are unprotected; virtually any such configuration would
   introduce potentials for privacy leakage.  Consequently, the use of
   MIME-level security is appropriate.

   S/MIME [5] allows security properties (including confidentiality,
   integrity, and authentication properties) to be applied to the
   contents of a MIME body.  Therefore, all PIDF implementations that
   support the XML Schema extensions for location information described
   in this document MUST support S/MIME; in particular, they MUST
   support the CMS [6] EnvelopedData and SignedData content types, which
   are used for encryption and digital signatures, respectively.  It is
   believed that this mechanism meets REQs 2.10, 13, 14.1, 14.2, 14.3,
   and 14.4.

   Additionally, all compliant applications MUST implement the AES
   encryption algorithm for S/MIME, as specified in [8] (and per REQ
   15.1).  Of course, implementations MUST also support the baseline
   encryption and digital signature algorithms described in the S/MIME
   specification.

   S/MIME generally entails the use of X.509 [9] certificates.  In order
   to encrypt a request for a particular destination end-to-end (i.e.,
   to a Location Recipient), the Location Generator must possess
   credentials (typically an X.509 certificate) that have been issued to
   the Location Recipient.  Implementations of this specification SHOULD
   support X.509 certificates for S/MIME, and MUST support password-
   based CMS encryption (see [6]).  Any symmetric keying systems SHOULD
   derive high-entropy content encoding keys (CEKs).  When X.509
   certificates are used to sign PIDF Location Objects, the
   subjectAltName of the certificate SHOULD use the "pres" URI scheme.

Which set of text takes precedence?

18)	Section 4:

   More than one location representation or format MAY be included in 
   the same message body part, but all MUST point at the same position 
   on the earth (altitude not withstanding), as this would confuse the 
   recipient by pointing at more than one position within the same 
   message body part.  There MAY be a case in which part or parts of 
   one location format and part or parts of another format exist in the
   same message body part.  These complementary pieces of information 
   MUST point at the same position on the earth, yet are incomplete 
   within their own format. For example, there maybe be a latitude and 
   longitude in coordinate format and a civic altitude value to 
   complete a 3-dimensional position of a thing (i.e. which floor of a 
   building the UA is on in a building at a particular lat/long 
   coordinates pair).

Not sure what "altitude not withstanding" means, particular as this is a RFC 2119 MUST sentence.

Additionally, the paragraph repeats the essence of the "same position on the earth" in two separate MUST statements.

19)	Section 4:

   There MAY be more than one PIDF-LO in the same SIP message, but each
   in separate message body parts. Each location body part MAY point at
   different positions on the earth (altitude not withstanding).  If 
   the message length exceeds the maximum message length of a single 
   packet (1300 bytes), TCP MUST to be used for proper message 
   fragmentation and reassembly.  

Why is the final sentence stronger than RFC 3261. I assume your thinking is emergency calls, but that is only one of the uses of this document. And surely if emergency calls is the problem, then the requirement has to be made as an update to RFC 3261 for all emergency calls, not just those that happen to carry location.

20)	Section 4:

   Several push-based SIP Request Methods are capable (and applicable) 
   of carrying location, including:

      INVITE, 
      REGISTER, 
      UPDATE, and
      MESSAGE, 

   While the authors do not yet see a reason to have location conveyed 
   in the ACK, PRACK, BYE, REFER and CANCEL Methods, we do not see a 
   reason to prevent carrying a PIDF-LO within these Method Requests as
   long as the SIP message meets the requirements stated within this 
   document.  Discussing Location in the PUBLISH Request Method will be
   for another document.  

The draft seems to go to a great deal of effort to ensure the information is carried end to end, but then suggests that the information might be entrusted to non end-to-end messages. I would suggest that the header and body are declared out of scope for ACK and CANCEL.

21)	Section 4:

   A 200 OK to a SIP Request MAY carry the UAS's PIDF-LO back to the 
   UAC that provided its location in the original request, but this is 
   not something that can be required due to the timing of the request 
   to 200 OK messages, with potential local/user policy requiring the 
   called user to get involved in determining if the caller is someone 
   they wish to give their location to (and at what precision).

I don't understand what this is trying to convey. Please explain?

22)	Section 4.1:

   It is envisioned that HTTP, through the http_URL in [RFC216], and 
   HTTPS [RFC2818] MAY be used to dereference a location-by-reference 
   PIDF-LO.

Do we place any requirement to support HTTP at any location receiver, so that location by reference when received actually works?

23)	Is it not sufficient to say that RFC 2119 applies, and therefore the security considerations of RFC 2119 applies.

NITS
----

24)	Section 3.1 UAC-8: As by-reference doesn't carry the PIDF-LO within SIP, but within HTTP, the text is wrong.

25)	Can we consistently use "request" or "response" rather than "message" following RFC 3261 terminology?

26)	Section 4:

   Several push-based SIP Request Methods are capable (and applicable) 
   of carrying location, including:

      INVITE, 
      REGISTER, 
      UPDATE, and
      MESSAGE, 

   While the authors do not yet see a reason to have location conveyed 
   in the ACK, PRACK, BYE, REFER and CANCEL Methods, we do not see a 
   reason to prevent carrying a PIDF-LO within these Method Requests as
   long as the SIP message meets the requirements stated within this 
   document.  Discussing Location in the PUBLISH Request Method will be
   for another document.  

This document is under WG control, the contents have nothing to do with the authors vision!!

27)	Section 4.1: 1st para: delete "and IANA registers" as superfluous and change "is to be used" to "can be used"

28)	Suggest using lower case throughout, including initial letter, for the name of the option-tag. 

29)	Section 4.1, table of headers needs a table number. As it follows the conventions of RFC 3261 (as indicated by reference), it should have an empty space under the "where" column. However see other technical comments about appearance in responses.

30)	Section 4.2, delete the words "a new 4XX level error is created here", as this is totally obvious from the rest of the words.

31)	Section 9. Delete "4xx error" before "response code"


regards

Keith

Keith Drage
drage@lucent.com
tel: +44 1793 776249

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From sip-bounces@ietf.org Mon Jul 03 20:11:49 2006
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Ready for WGLC on Outbound?
Date: Mon, 3 Jul 2006 17:09:54 -0700
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Hi Erkki,

The only time a UAC sends a STUN request as directed by outbound -04 is 
when the URI in the outbound-proxy-set includes the keepalive=stun 
parameter.  This is 100% backwards safe.  We don't need any text to 
prevent existing implementations from getting a bunch of STUN requests.

That DHCP does not provide URI is a limitation of the SIP DHCP 
configuration mechanism, not outbound.  The same problem exists for the 
"comp" parameter for example.  If you want to write a draft on how to 
"shift-up" to using keepalives or using compression from some SIP or 
external stimulus, you can still do this.  No text in outbound prevents 
that from happening.

thanks,
-rohan


On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
>
> Hi Rohan,
>
>>> Also, the issue is that what may be configured (no matter how you
>>> configure it) may not be a URI. It may simply be an IP address/FQDN,
>>> which is then used to build a SIP-URI. I think we shall
>>> allow that, and make it more clear in the spec. [/CHH]
>>
>> the outbound spec defines a piece of configurable data (the
>> 'outbound-proxy-set'), which is a list of URIs.  This list of URIs may
>> never have a one-to-one correspondence to a user or administrator
>> manually configurable field.  That piece of configurable data may be
>> derived from several fields, manually configurable, discovered via
>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
>> download, or via a random character generator seeded by the phase of
>> the moon.
>>
>> The outbound-proxy-set was defined as a list of URIs because the UA
>> acts on each item in the list, because each URI is expected to be
>> places in a Route header, and because each URI is expected to
>> typically have a keepalive URI parameter in it.
>
> Like mentioned in other recent emails the problem is that if the UA
> derives the proxy URI(s) from DHCP SIP server option or e.g. the AOR
> of the user, in those cases the derived URI will NOT contain the
> keepalive URI parameter in it. DHCP just returns IP addresses or
> domain names instead of URIs with parameters.
>
> Thus in those cases it would be critical for the UA to be able
> to ensure in REGISTER - 200 OK sequence that the contacted proxy
> supports Outbound and can handle STUN requests sent to its SIP
> port (not to cause any damage to an uncompliant proxy). I fully
> agree with this proposal from Jeroen:
>
> --
> The text should then say "the UAC MUST wait for a 200 OK to REGISTER
> before
> it starts sending STUN packets. If the last Path header of the 200 OK
> does
> not contain keepalive=stun, the UAC MUST NOT use STUN towards this URI.
> If
> there are no Path headers and no "Supported: outbound", the UAC also
> MUST
> NOT use STUN towards the registrar.". It could still use TCP keepalive
> if it
> supports that
> --
>
> Please excuse me if I am just repeating myself and what e.g. Jeroen
> has said but I believe the importance of this issue justifies some
> redundancy :-)
>
> Regards,
>
> Erkki


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I'm thinking it would be a Good Thing to define a Content-Disposition
value for the message/sipfrag to unambiguously tag its purpose.

Dale

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At 11:51 PM 7/3/2006 +0100, Drage, Keith (Keith) wrote:
Keith

in-line

>I have the following comments on the above draft:
>
>Structural
>----------
>
>1)      Abstract: From reading the document, it is clear that presence 
>systems are not covered at all in this document for various reasons. 
>Therefore PUBLISH, SUBSCRIBE and NOTIFY are essentially out of scope. It 
>seems to me that this should be made much more explicit in the abstract.

The first line of the abstact states:

"This document defines how the Session Initiation Protocol (SIP) conveys, 
or pushes, geographic location information from one SIP entity to another 
SIP entity."

Since SUB/NOT is not a "push" form of transaction, I believe this satisfies 
what you are looking for.  I admit I do not call out each of the 14 Methods 
that this extension does affect, but I rarely see this done in any SIP 
document, so I didn't know it was necessary here. For example RFC 4538 
doesn't say in the abstract it is only used in INVITE, SUB and NOT, and not 
all the others.

Section 4 clearly states that this extension can be used in SUB and NOT, 
but this is out of scope for this document because they are pull based 
transactions.  Section 4 also states that PUB is out of scope.


>2)      The SIP extension guidelines (RFC 4485) need to be applied to this 
>document, which requires some substantial restructuring.

I believe this will only result in section 4 and 5 being modified, and not 
any other section (structurally - unless the reqs are moved to an appendix)

>The absence of application of these rules mean that the requirements of 
>the document are extremely unclear. For example RFC 4485 states (section 4.5):
>
>    Developers of protocols often get caught up in syntax issues, without
>    spending enough time on semantics.  The semantics of a protocol are
>    far more important.  SIP extensions MUST clearly define the semantics
>    of the extensions.  Specifically, the extension MUST specify the
>    behaviors expected of a UAC, UAS, and proxy in processing the
>    extension.  This is often best described by having separate sections
>    for each of these three elements.  Each section SHOULD step through
>    the processing rules in temporal order of the most common messaging
>    scenario.
>
>Absence of this application means that in places in this document it is 
>impossible to determine whether a requirement and its required support 
>occurs as the sender of the location information, or the receiver of the 
>location information.

This document is written almost exclusively as a UAC pov extension, since 
it, by definition, is what is "conveying" location.  Thus, little about 
this document is about a UAS, save a line that discusses that element can 
response with location.

A little bit more of this doc discusses what a proxy does with this 
extension, with most of the original material from past version removed, to 
be put into ECRIT documents.

I will make element operation wrt this extension much more explicit in the 
next rev.


>3)      Section 1 (and possibly section 2) should be redrafted taking into 
>account RFC 4485 section 4.7:
>
>    Too often, extension documents dive into detailed syntax and
>    semantics without giving a general overview of operation.  This makes
>    understanding of the extension harder.  It is RECOMMENDED that
>    extensions have a protocol overview section that discusses the basic
>    operation of the extension.  Basic operation usually consists of the
>    message flow, in temporal order, for the most common case covered by
>    the extension.

This document, per a conversation between you, me and Dean, in Dallas, is 
to include the requirements for this extension in the front of the 
document, and not deleted or put in an appendix.  This is due to this 
document origin as a SIPPING Requirements document.  This means I have to 
get to the requirements section, which is section 3.  This means I cannot 
get into the classic "Basic Operation" until the section after that - 
Section 4.  Section 4 states it is a Basic Operation section at the top of 
the third paragraph, followed by the two basic message flows (Figure 1 and 
2).  Perhaps I should have called the section "Basic Operation of SIP 
Location Conveyance" instead of "Location Conveyance Using SIP". I will fix 
this in the next rev.  But section 4 is what you are looking for per RFC 
4485 section 4.7.

>The most important processing rules for the elements
>    in the call flow SHOULD be mentioned.

they are

>Usage of the RFC 2119 [1]
>    terminology in the overview section is NOT RECOMMENDED, and the
>    specification should explicitly state that the overview is tutorial
>    in nature only.

I do need to work on this.  This will be addressed in the next rev.  There 
is a good segway in section 4 right now, I just didn't divide that point 
into a new section - but I will now.

>This section SHOULD expand all acronyms, even those
>    common in SIP systems, and SHOULD be understandable to readers who
>    are not SIP experts.

I believe it does

>[27] provides additional guidance on writing
>    good overview sections.
>
>I found this text much too wordy, and considered that it should be 
>possibly to convey the necessary information to the reader in about half 
>the length.

"this text" meaning exactly what text? Section 1, Section 2 or both?  I 
think section 1 is discussing scenarios that offer unique ideas about 
location conveyance, and what is going into which section.  Section 2 has 
more words than it probably should.


>4)      Section 3 need to have all the RFC 2119 language removed (see 
>history-info and outbound as examples).

Section 3 is only requirements, and you said in Dallas they stay where they 
are because this ID stated as a Reqs only doc in SIPPING, so I'm confused 
as to which you want.

>I remain open as to whether it appears at the start of the document, as 
>here and in history-info, or at the end, as in outbound, but given the 
>length, we may want to consider moving it to the end.

Again, this is where you said to keep them (in Dallas), so I did.  Are you 
changing your mind?  If so, given that you hadn't told me prior to after 
the document deadline, I'm not surprised I didn't do it another way.  That 
said, I can move the reqs to an appendix, and mention this is where they 
are in section 1.


>5)      Section 6 should probably follow immediately on from the 
>procedures in the bulk of section 4, and probably be split as in comment 
>2) above.

Based on what you said above, I agree that section 4 and 5 will be modified.


>6)      Section 7 should be integrated into the main procedures text.

hmm... I think calling out how this SIP extension complies with RFC 3693 in 
its own section is appropriate (readers can see this glaring at them in the 
TOC), but it doesn't make sense until the extension is explained, which 
would be hard to undertand before section 5.


>Technical
>---------
>
>7)      Section 1, of the three examples given in this introduction, item 
>2) is out of scope. I had to read an awful lot of text to discover that fact.

I disagree. Think of the REGISTER request, in which a Registrar responds 
with different information based on where the UAC is.

I have a couple of IDs in another WG that take this usage and run with it:
http://www.ietf.org/internet-drafts/draft-polk-ecrit-lost-server-uri-00.txt
http://www.ietf.org/internet-drafts/draft-polk-ecrit-emergency-dialstring-00.txt
http://www.ietf.org/internet-drafts/draft-polk-ecrit-mapping-during-registration-00.txt


>8)      Section 2, 1st paragraph. Probably not appropriate to use the term 
>"trust model". That term is defined in RFC 3324 and RFC 3325, and I am not 
>convinced the usage here is one and the same.

Should I use "e2e trust" vs. non-e2e trust"?

3325 talks about the UAS trusting the Proxy to assert the PAI is accurate 
about the UAC.

Here, this is from the UAC's pov, and making a decision based on where it 
believes the information will be used (only at the destination UAS, or by a 
proxy to use for message routing based on where the UAC is).  This seems 
like a model; either e2e or used by an intermediary to get the message to 
the right endpoint.

Any suggestions?


>9)      Section 2, 3rd para.
>
>    No routing of the request based on the location contents is required
>    in this case, therefore no SIP Proxies between these two UAs need to
>    view the location information contained in the SIP message(s).  The
>    UAC should know certain types of messages will be routed based on
>    the UA's location when creating a message.
>
>Did not understand what this paragraph is meant to be telling me - can you 
>explain it.

yes

In the previous paragraphs it was concluded by the UAC no help was needed 
in the network, from a location conveyance pov, to find the destination 
UAS, therefore merely sending location in the message body was enough.  The 
forth paragraph continues this train of thought stating RFC 3261 does not 
allow message bodies to be modified or deleted, and if a UAC wants location 
to be conveyed confidentially, it needs to employ an encryption mechanism 
(S/MIME).  Paragraphs 2, 3 and 4 could be a single, larger paragraph - but 
I broke it up.  Perhaps that was a mistake.


>10)     Section 3.1 UAC-4: This is not a generic requirement as it is 
>already identified that certain methods are outside the scope of this document.

This allows Location to be included in other SIP Requests (than those 
listed above this requirement).  Without this requirement, table 2 and 3 
would be pretty empty (with only INVITE dialog related requests with an 
'o', and everything else having a '-', meaning not allowed, and the doc 
states that 'some other Request MAY include location, but the authors do 
not see a reason for this at this time'.


>11)     Section 3.1 UAC-9, UAC-10: I assume this is specified this way so 
>that misconfiguration releases the location rather than hides it.

that's what UAC-9 is for, also if the first INVITE hid location, but a 
subsequent message (after recieving a 424) doesn't want to hide it to make 
sure this isn't the reason the message didn't get through.

UAC-10 allows for a user to "not" include location during an emergency 
call.  We already know of places that have this law on the books.

>I believe this contravenes the regulatory requirements of some countries.

Yes, and in those jurisdictions, that's an configuration issue, not a 
protocol issue.  My mobile phone needs to be able to roam between a 
jurisdiction that has such a law, and one that doesn't.

>I believe in Japan privacy requirements are more important that the 
>transmission of the information for emergency calls. If someone from Japan 
>was writing these requirements they would therefore indicate exactly the 
>opposite.

Again, this is an configuration issue.  There are places where the user has 
no choice in the matter, other places the user has a choice.


>12)     Section 3.2 UAS-6 and section 3.3 Proxy-4. Firstly this is halfway 
>into implementation. Secondly if it is required to address location errors 
>in requests, why is the same consideration not given to location errors in 
>responses.

Are you asking if there is an error in a SIP response?  How do you address 
that anytime?

Basically, I'm not understanding your question/comment 12. Can you rephrase?


>13)     Section 3.3 Proxy-1: This merely repeats RFC 3261, rather than 
>developing a new requirement.

Fair - it does. I guess this was to remind readers of this.  It should be 
removed.


>14)     Section 3.3 Proxy-3: This seems to be rewriting the requirements 
>to correspond to the solution.

I disagree.  We have a problem in that there is no indicator of who 
generated a PIDF-LO (by-value or by-reference).  This is very real.  This 
requirement is stating that if there already is location in a message, not 
to add to the confusion factor if you can help it.

>I understand the problem is that the solution cannot distinguish between 
>multiple locations, nor understand the reliability, and therefore a 
>receiver may get confused. Firstly this does not seem to me to be a SIP 
>issue, and secondly in general the ability to transport multiple locations 
>could be regarded as useful - they just need some form of assertion as to 
>who provided which location.

but there is no form of assertion now... so what do we do, ignore the 
problem as it is today?


>15)     Section 4 and elsewhere. I don't find location in responses well 
>described at all.

I state that " Bob MAY choose to include his location in a response back to 
Alice."  Otherwise, I don't get into this other than it must obey the 
PIDF-LO format and privacy mechanisms of a request.

>It seems to be an afterthought, rather than something that is decided 
>based on a full set of requirements. I have already made a comment with 
>regard to requirements on error handling. Should we support location in 
>response, or force the UAS to generate a request of its own in order to 
>send a location?

fair question - I'm guess I'm ok with saying Location Conveyance is only in 
requests, but I didn't see the harm in Location being in a 200 OK to a 
request that had location in it.  Guidance/opinions welcome


>16)     Section 4. Should the instances of "URI" here be "URN".

I take URNs to be services (like urn:services:sos), and URIs to be records 
on a remote node to be accessed.  Based on that, I think locations are 
URIs, of which, a URL is a subset.


>17)     Section 4 specifies:
>
>    Because a person's location is generally considered to be sensitive
>    in nature, certain security measures need to be taken into account
>    when transmitting such information.  Section 26 of [RFC3261] defines
>    the security functionality SIPS for transporting SIP messages with
>    either TLS or IPSec, and S/MIME for encrypting message bodies from
>    SIP intermediaries that would otherwise have access to reading the
>    clear-text bodies.  SIP endpoints SHOULD implement S/MIME to encrypt
>    the PIDF-LO message body (part) end-to-end.  The SIPS-URI from
>    [RFC3261] MUST be implemented for message protection (message
>    integrity and confidentiality) and SHOULD be used when S/MIME is not
>    used.
>
>    The entities sending and receiving location MUST obey the privacy
>    and security rules in the PIDF-LO, regarding retransmission and
>    retention, to be compliant with this specification.
>
>    Self-signed certificates SHOULD NOT be used for protecting PIDF-LO,
>    as the sender does not have a secure identity of the recipient.
>
>However RFC 4119 is already normatively referenced which contains the 
>following text:
>
>    For the purposes of this document, only the securing of a PIDF
>    document as a whole, rather than element-by-element security, is
>    considered.  None of the requirements [10] suggest that only part of
>    the information in a location object might need to be protected while
>    other parts are unprotected; virtually any such configuration would
>    introduce potentials for privacy leakage.  Consequently, the use of
>    MIME-level security is appropriate.
>
>    S/MIME [5] allows security properties (including confidentiality,
>    integrity, and authentication properties) to be applied to the
>    contents of a MIME body.  Therefore, all PIDF implementations that
>    support the XML Schema extensions for location information described
>    in this document MUST support S/MIME; in particular, they MUST
>    support the CMS [6] EnvelopedData and SignedData content types, which
>    are used for encryption and digital signatures, respectively.  It is
>    believed that this mechanism meets REQs 2.10, 13, 14.1, 14.2, 14.3,
>    and 14.4.
>
>    Additionally, all compliant applications MUST implement the AES
>    encryption algorithm for S/MIME, as specified in [8] (and per REQ
>    15.1).  Of course, implementations MUST also support the baseline
>    encryption and digital signature algorithms described in the S/MIME
>    specification.
>
>    S/MIME generally entails the use of X.509 [9] certificates.  In order
>    to encrypt a request for a particular destination end-to-end (i.e.,
>    to a Location Recipient), the Location Generator must possess
>    credentials (typically an X.509 certificate) that have been issued to
>    the Location Recipient.  Implementations of this specification SHOULD
>    support X.509 certificates for S/MIME, and MUST support password-
>    based CMS encryption (see [6]).  Any symmetric keying systems SHOULD
>    derive high-entropy content encoding keys (CEKs).  When X.509
>    certificates are used to sign PIDF Location Objects, the
>    subjectAltName of the certificate SHOULD use the "pres" URI scheme.
>
>Which set of text takes precedence?

I don't see the conflict.  Can you point any such conflict out?

Is it the SHOULD vs. MUST implement S/MIME?  If so, I'll change this ID 
back to MUST implement.


>18)     Section 4:
>
>    More than one location representation or format MAY be included in
>    the same message body part, but all MUST point at the same position
>    on the earth (altitude not withstanding), as this would confuse the
>    recipient by pointing at more than one position within the same
>    message body part.  There MAY be a case in which part or parts of
>    one location format and part or parts of another format exist in the
>    same message body part.  These complementary pieces of information
>    MUST point at the same position on the earth, yet are incomplete
>    within their own format. For example, there maybe be a latitude and
>    longitude in coordinate format and a civic altitude value to
>    complete a 3-dimensional position of a thing (i.e. which floor of a
>    building the UA is on in a building at a particular lat/long
>    coordinates pair).
>
>Not sure what "altitude not withstanding" means, particular as this is a 
>RFC 2119 MUST sentence.

It means with or without, it makes no difference.  There are 4000 datums. 
Some in 2D, and others in 3D.  WGS84 has both.  There is a movement in 
Geopriv to have coordinate format location only provide 2D and a civic 
format provide altitude within the same <location-info> element of the 
PIDF-LO.  This statement is regarding during times where civic is not 
supported, or a altitude is not available, consider the 2D location to be 
enough information to present location by.


>Additionally, the paragraph repeats the essence of the "same position on 
>the earth" in two separate MUST statements.

I'll work on that, I was attempting to cover a separate angle of the same 
meaning.


>19)     Section 4:
>
>    There MAY be more than one PIDF-LO in the same SIP message, but each
>    in separate message body parts. Each location body part MAY point at
>    different positions on the earth (altitude not withstanding).  If
>    the message length exceeds the maximum message length of a single
>    packet (1300 bytes), TCP MUST to be used for proper message
>    fragmentation and reassembly.
>
>Why is the final sentence stronger than RFC 3261. I assume your thinking 
>is emergency calls, but that is only one of the uses of this document. And 
>surely if emergency calls is the problem, then the requirement has to be 
>made as an update to RFC 3261 for all emergency calls, not just those that 
>happen to carry location.

hmmm... so this is a problem.  I hadn't made that distinction wrt 3261 
language.  If I change this to a SHOULD, will that be acceptable?


>20)     Section 4:
>
>    Several push-based SIP Request Methods are capable (and applicable)
>    of carrying location, including:
>
>       INVITE,
>       REGISTER,
>       UPDATE, and
>       MESSAGE,
>
>    While the authors do not yet see a reason to have location conveyed
>    in the ACK, PRACK, BYE, REFER and CANCEL Methods, we do not see a
>    reason to prevent carrying a PIDF-LO within these Method Requests as
>    long as the SIP message meets the requirements stated within this
>    document.  Discussing Location in the PUBLISH Request Method will be
>    for another document.
>
>The draft seems to go to a great deal of effort to ensure the information 
>is carried end to end, but then suggests that the information might be 
>entrusted to non end-to-end messages.

Maybe I'm missing where a message other than 100 Trying is not 
end2end.  Certainly all requests are intended to be e2e.

>I would suggest that the header and body are declared out of scope for ACK 
>and CANCEL.

I can modify the above to state 'we do not see the point/reason of 
including location in either ACK or CANCEL'.


>21)     Section 4:
>
>    A 200 OK to a SIP Request MAY carry the UAS's PIDF-LO back to the
>    UAC that provided its location in the original request, but this is
>    not something that can be required due to the timing of the request
>    to 200 OK messages, with potential local/user policy requiring the
>    called user to get involved in determining if the caller is someone
>    they wish to give their location to (and at what precision).
>
>I don't understand what this is trying to convey. Please explain?

A 200 OK can carry Location back to the UAC; however, this cannot be an 
expected operation due to the user of the UAS possibly needing to get 
included in whether or not to include their location to the user of the 
UAC.  This delay may be based on who the UAC user is.  This could greatly 
delay call set-up, and delay sending of any response message to the UAC.

Or, the UAS user may have a policy to let certain offerers know the UAS's 
location if they match certain rules within the request (i.e. are on a 
buddy list).

I'm finding this answer to be awkward, but I feel something needs to be 
said here that a UAS may or may not response with location, based on some 
rule match in the request.

suggestions are welcome.


>22)     Section 4.1:
>
>    It is envisioned that HTTP, through the http_URL in [RFC216], and
>    HTTPS [RFC2818] MAY be used to dereference a location-by-reference
>    PIDF-LO.
>
>Do we place any requirement to support HTTP at any location receiver, so 
>that location by reference when received actually works?

JDR got on me about lacking a means to dereference a 
location-by-reference.  Since I don't believe we need another lame usage 
for SIP INFO, I thought we'd suggest that HTTP would be a good fetching 
protocol for a PIDF-LO not locally stored.

What should I do here, change the BNF

         locationURI = absoluteURI / cidURI
to
         locationURI = http_url / https_url / absoluteURI / cidURI
  or
add a requirement in section 3 that receivers of a location by reference 
URI be able to use HTTP/HTTPS to dereference the PIDF-LO?


>23)     Is it not sufficient to say that RFC 2119 applies, and therefore 
>the security considerations of RFC 2119 applies.

I do not understand this comment.  Can you be more verbose?


>NITS
>----
>
>24)     Section 3.1 UAC-8: As by-reference doesn't carry the PIDF-LO 
>within SIP, but within HTTP, the text is wrong.

I think within a by-value message body, it is within SIP.  If I understand 
your point above (about 2 inches), you think this requirement needs to 
change where it states location by-reference within HTTP/HTTPS, correct?


>25)     Can we consistently use "request" or "response" rather than 
>"message" following RFC 3261 terminology?

fair point.  I will change all transactions to requests and responses.


>26)     Section 4:
>
>    Several push-based SIP Request Methods are capable (and applicable)
>    of carrying location, including:
>
>       INVITE,
>       REGISTER,
>       UPDATE, and
>       MESSAGE,
>
>    While the authors do not yet see a reason to have location conveyed
>    in the ACK, PRACK, BYE, REFER and CANCEL Methods, we do not see a
>    reason to prevent carrying a PIDF-LO within these Method Requests as
>    long as the SIP message meets the requirements stated within this
>    document.  Discussing Location in the PUBLISH Request Method will be
>    for another document.
>
>This document is under WG control, the contents have nothing to do with 
>the authors vision!!

OK, and this usage of pov is quite prevelent in documents.  Nonetheless, I 
will change it to omit author's visions.


>27)     Section 4.1: 1st para: delete "and IANA registers"

this comes from an old correction made by an AD (who said to always say 
this if a doc registers anything), but I will remove it

>as superfluous and change "is to be used" to "can be used"

ack


>28)     Suggest using lower case throughout, including initial letter, for 
>the name of the option-tag.

For the option-tag, I thought I had - but I will make sure of 
this.  Location wrt the header needs to be upper case though (correct still?).


>29)     Section 4.1, table of headers needs a table number.

ack

>As it follows the conventions of RFC 3261 (as indicated by reference), it 
>should have an empty space under the "where" column.

my mistake

>However see other technical comments about appearance in responses.
>
>30)     Section 4.2, delete the words "a new 4XX level error is created 
>here", as this is totally obvious from the rest of the words.

ack


>31)     Section 9. Delete "4xx error" before "response code"

ack



>regards
>
>Keith
>
>Keith Drage
>drage@lucent.com
>tel: +44 1793 776249

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BACKGROUND
SIP allows "retargeting" in which the endpoints (on either end) of
a call can change. So, for instance, Alice might be calling Bob
but actually get forwarded to Charlie. This draft describes a
mechanism for the new endpoint to notify their peer that their
identity is not the same as the peer expected.


BASIC MESSAGE FLOW
The basic message flow without this draft looks something like this:

Alice                         Proxy                       Carol
-----                         -----                       -----
INVITE (To=Bob) -> 
                                INVITE (To=Bob) ->     
                                                 <- OK (To=Bob)
                   <- OK (To=Bob)                
ACK (To=Bob) ->                
                                ACK (To=Bob) ->


This draft adds a new set of messages for Carol to tell Alice about
the retargeting. Like this:


Alice                         Proxy                       Carol
-----                         -----                       -----
INVITE (To=Bob) -> 
                                INVITE (To=Bob) ->     
                                                 <- OK (To=Bob)
                   <- OK (To=Bob)                
ACK (To=Bob) ->                
                                ACK (To=Bob) ->

                                         <- UPDATE (From=Carol)
            <- UPDATE (From=Carol)
ACK (From=Carol) ->                
                                ACK (From=Carol) ->




SECURITY ISSUES
This draft relies on the ordinary SIP authentication mechanisms
(primarily Identity, I would imagine) in order to validate the update
message. Unfortunately, this just tells the other side that they're
now talking to someone new securely. They have no way of determining
whether the retargeting was authorized or not. This is explicitly
acknowledged in the Security Considerations section. Clearly 
something along these lines is needed. Perhaps it could
go in a different draft, but this does seem like the obvious one.


EDITORIAL
1. I found this document fairly hard to read until I figured out what was
going on, perhaps due to my lack of familiarity with SIP. ISTM that a
diagram like the one above would be appropriate here--some diagram
definitely seems advisable.

2. It would be helpful to have unique and distinctive terms for the
UA that isn't being retargeted (Alice above) and the one that is
(Bob/Carol). I found myself trying to figure that out. These
terms should be called out explicitly early on.


S 3. 
You should explicitly state that To doesn't change during retargeting.
That is an unobvious behavior of SIP.

I think some diagrams of these scenarios would help.

The following paragraph is a good example of what I mean above about
distinctive terms:

   However, call re-arrangements are not always carried out using a new
   INVITE request.  Sometimes a B2BUA performs call re-arrangements
   using third party call control (3PCC) techniques.  With such
   techniques the UA involved in the original call and still involved in
   the re-arranged call receives only a re-INVITE or UPDATE request in
   the context of the original dialog between that UA and the B2BUA.
   This forces the UA to re-negotiate the session with the new remote
   party, but introduces a need to convey the identity of the new remote
   party to the UA.  Because there is no new INVITE request (outside the
   context of the existing dialog), techniques applicable to new calls
   do not apply.

"the UA involved in the original call and still involved in
the re-arranged call" is too long to process as one chunk.


S 4.
It should be made clear that this is background material that
will be used later. It's not an alternative to the current
approach, which I kind of got the impression from "Existing".

It would probably halp to name these mechanisms in some easy
to remember way. I'm sure they already have names....

S 6.2.
Just for clarity, you don't need to send an UPDATE/re-INVITEif you
don't support id-change. So, the peers already need to be able to deal
with a non-UPDATE-sending situation, right? So, I'm not sure
I understandw hy you need to send the UPDATE/re-INVITE if nothing
has changed...

7.2.
There is a very bad break in this table. The headers need to be on
the same page as the columns.



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From sip-bounces@ietf.org Tue Jul 04 02:56:44 2006
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Hi Keith, Erkki:

To answer your first question, I believe this draft updates RFC 3263.

And with respect your third question, I agree that the draft does not 
consider the case when Outbound is not deployed, so the text has to be 
updated to reflect the case.

I have also a couple of other non-nit comments.

1) I think the draft has to be a bit more strict on where this draft 
ends and when Outbound begins. At the moment I found this line quite 
fuzzy. Assuming this draft is about discovery, but not execution of the 
outbound procedures, I believe requirement REQ-OUTB-DISC-6 has not much 
to do with discovery, but with outbound. The same applies to several 
paragraphs in Section 5, when the draft describes actions taken upon a 
reception of a 200 response  for a REGISTER request.

2) The first paragraph in Section 2.4.4 suggests that a non-standardized 
autodiscovery mechanism will require the UA to insert a "Proxy-Require: 
outbound". Care must be taken to avoid that this becomes the de-facto 
behavior. It must noted that such Proxy-Require will require each of the 
proxies en route to the registrar to support outbound, which might not 
be the desired behavior. I would suggest to add some more text 
describing this case and highlighting the problem.

3) IANA: something is not clear to me. Is this draft creating a new 
registry (as it suggest Sections 4 and 7), named "Registry for the NAPTR 
Resource Record Services Fiedl", or is it updating the existing registry 
at http://www.iana.org/assignments/sip-table

I thought it would be the latter, but the fact that the existing 
registry refers to SRV records rather than NAPTR makes me doubt. Can 
someone clarify?

BR,

      Miguel

Drage, Keith (Keith) wrote:
> Technical:
> ----------
> 
> 1) I need to understand whether this updates RFC 3263 or just extends it. Part of this is what appears on the cover page and in the RFC index as and when it is finally approved, but the other part also relates to compatibility, see later comment.
> 
> 2) I think you have already acknowledged this, but the document requires revision to align with the terminology in the latest version (-04) of draft-ietf-sip-outbound.
> 
> 3) In section 5, as far as I understand, the NAPTR request is identical for a UA that supports RFC 3263 and for one that supports this extension. Therefore the difference is in the response. Therefore the paragraph of UA requirements:
> 
>    As per RFC 2915 [4], the client discards any NAPTR records whose
>    services fields are not applicable.  A client resolving a SIPS URI
>    MUST discard any services that do not contain "SIPS-O" as the
>    protocol in the service field.  A client resolving a SIP URI MUST
>    retain records with "SIP-O" and the transport protocols the client
>    supports and SHOULD retain records with "SIPS-O", if the client
>    supports TLS.
> 
> must work with a NAPTR response that has no knowledge of outbound. Is the above paragraph compatible with this? 
> 
> Additionally note that the words "if the client supports TLS" are superfluous. As RFC 3261 requires support of TLS, we are not documenting capabilities where TLS is not supported.
> 
> regards
> 
> Keith
> 
> Keith Drage
> drage@lucent.com
> tel: +44 1793 776249
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
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> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Subject: RE: [Sip] Ready for WGLC on Outbound?
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Hi Rohan,

Sorry to be persistent with this issue, but what you say
seems to conflict with what was discussed in March.

> The only time a UAC sends a STUN request as directed by=20
> outbound -04 is when the URI in the outbound-proxy-set=20
> includes the keepalive=3Dstun parameter. That DHCP does not=20
> provide URI is a limitation of the SIP DHCP configuration=20
> mechanism, not outbound. If you want to write a draft on=20
> how to "shift-up" to using keepalives from some SIP or=20
> external stimulus, you can still do this.

What you say seems to suggest that you agree there is a problem
when using DHCP SIP server option (as it is currently defined)
for finding out if the discovered proxy supports outbound and
STUN keepalives, but the solution is not in scope of Outbound
draft itself.=20

However I came to a different understanding based on the emails
seen in the list earlier. For instance this email from Dean
Willis (SIP WG chair) seems to suggest that the problem should
be in scope of the outbound draft:
http://www1.ietf.org/mail-archive/web/sip/current/msg13667.html

"Ok, I think we've boiled this down to a new requirement:

for the "outbound" technique to work, we MUST have a means of=20
discovering whether a proxy supports the extensions of outbound.

While we could do that by doing a REGISTER exchange that includes=20
an appropriate option-tag in Require:, it would be more efficient=20
if the proxy discovery mechanism informed us of this detail. So,=20
we posit that the Outbound draft SHOULD select and discuss such a=20
mechanism."

And in yet another email
http://www1.ietf.org/mail-archive/web/sip/current/msg13692.html
Dean tells this:

"There are two ways we can establish compatibility with an extensions:

1) By foreknowledge
2) By experimentation -- try it and see if it works.

Both have limitations:

In the case of #1, we might have incorrect knowledge. So we'll probably
end up testing anyhow . . .

In the case of #2, the experiment might cause a meltdown. So the "probe
for discovery" must be made completely safe. Also, we can eventually end
up with so many wierd hooks and idiosyncratic implementations that we
end up doing a BUNCH of probing before we do any real communicating. And
it's possible for the conditions to deteriorate so much that probes for
two different possible conditions are mutually exclusive -- if one
condition applies, the probe for another will cause a meltdown.

So what happens if we start STUN-probing a proxy on its SIP port and it
doesn't support STUN? My guess is, it's random.=20

...

One approach to "forward compatibility" design is to make sure that we
can use both foreknowledge and experimentation -- the "trust but verify"
model.

...

While it's easy to decide whether a configured registrar URI has a
keepalive parameter, what happens to proxies we select based on
something besides direct configuration?

Specifically,

1) Can the DHCP option for default proxy present the keepalive
parameter?"
----

During these three months after seeing those comments in the list
I have been expecting the outbound draft to provide answer to
the question how UA could (within SIP signalling exchange)
safely to check whether or not a proxy supports STUN keepalive.
Two mechanims have been suggested how the UA could do this in
a simple way:

1. The UA to check that the Path header in 200 OK returned for
   REGISTER would contain keepalive=3Dstun parameter for the proxy URI

2. The UA to add Proxy-Reequire: stun-keepalive parameter into
   the REGISTER request and the proxy to remove this tag from
   the request before forwarding it towards the registrar

For me, any of those mechanisms would sound feasible and for
safety reasons I would hope the draft to deny the UA for sending
STUN keepalives to the proxy which did not pass such a test.

> the same problem exists for the "comp" parameter for example.=20

Yes but please bear in mind that for comp=3Dsigcomp URI parameter
RFC 3486 Compressing SIP gives several solutions how the UA
would be able to check if the proxy supports SigComp while
outbound does not define any other than relying on configuration.=20
A cite from RFC 3486:

4.1   Obtaining a SIP or SIPS URI with comp=3Dsigcomp

   For requests within a dialog, a next-hop URI with the comp=3Dsigcomp
   parameter is obtained from a Record-Route header field when the
   dialog is established.  A client sending a request outside a dialog
   can also obtain SIP URIs with comp=3Dsigcomp in a Contact header =
field
   in a 3xx or 485 response to the request.

   However, clients establishing a session will not typically be willing
   to wait until the dialog is established in order to begin compressing
   messages.  One of the biggest gains that SigComp can bring to SIP is
   the ability to compress the initial INVITE of a dialog, when the user
   is waiting for the session to be established.  Therefore, clients
   need a means to obtain a comp=3Dsigcomp URI from their outbound proxy
   before the user decides to establish a session.

   One solution to this problem is manual configuration.  However,
   sometimes it is necessary to have clients configured in an automatic
   fashion.  Unfortunately, current mechanisms for SIP client
   configuration (e.g., using DHCP [6]) do not allow to provide the
   client with URI parameters.  In this case, the client SHOULD send an
   uncompressed OPTIONS request to its outbound proxy.  The outbound
   proxy can provide an alternative SIP URI with the comp=3Dsigcomp
   parameter in a Contact header field in a 200 OK response to the
   OPTIONS.  The client can use this URI for subsequent requests that
   are sent through the same outbound proxy using compression.

   RFC 3261 [1] does not define how a proxy should respond to an OPTIONS
   request addressed to itself.  It only describes how servers respond
   to OPTIONS addressed to a particular user.  Section 11.2 of RFC 3261
   says:

      Contact header fields MAY be present in a 200 (OK) response and
      have the same semantics as in a 3xx response.  That is, they may
      list a set of alternative names and methods of reaching the user.

   We extend this behavior to proxy servers responding to OPTIONS
   addressed to them.  They MAY list a set of alternative URIs to
   contact the proxy.

What has been called for is that outbound draft would also=20
specify one solution how the UA could check the STUN keepalive
support of a proxy. In my opinion good proposals seem
to exist how to do it in a reasonable way.

Regards,

Erkki

>-----Original Message-----
>From: ext Rohan Mahy [mailto:rohan@ekabal.com]=20
>Sent: 04.July.2006 03:10
>To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>Cc: dean.willis@softarmor.com; fluffy@cisco.com; Rohan Mahy;=20
>sip@ietf.org; jbemmel@zonnet.nl
>Subject: Re: [Sip] Ready for WGLC on Outbound?
>
>Hi Erkki,
>
>The only time a UAC sends a STUN request as directed by=20
>outbound -04 is=20
>when the URI in the outbound-proxy-set includes the keepalive=3Dstun=20
>parameter.  This is 100% backwards safe.  We don't need any text to=20
>prevent existing implementations from getting a bunch of STUN requests.
>
>That DHCP does not provide URI is a limitation of the SIP DHCP=20
>configuration mechanism, not outbound.  The same problem=20
>exists for the=20
>"comp" parameter for example.  If you want to write a draft on how to=20
>"shift-up" to using keepalives or using compression from some SIP or=20
>external stimulus, you can still do this.  No text in outbound=20
>prevents=20
>that from happening.
>
>thanks,
>-rohan
>
>
>On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
>>
>> Hi Rohan,
>>
>>>> Also, the issue is that what may be configured (no matter how you
>>>> configure it) may not be a URI. It may simply be an IP=20
>address/FQDN,
>>>> which is then used to build a SIP-URI. I think we shall
>>>> allow that, and make it more clear in the spec. [/CHH]
>>>
>>> the outbound spec defines a piece of configurable data (the
>>> 'outbound-proxy-set'), which is a list of URIs.  This list=20
>of URIs may
>>> never have a one-to-one correspondence to a user or administrator
>>> manually configurable field.  That piece of configurable data may be
>>> derived from several fields, manually configurable, discovered via
>>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
>>> download, or via a random character generator seeded by the phase of
>>> the moon.
>>>
>>> The outbound-proxy-set was defined as a list of URIs because the UA
>>> acts on each item in the list, because each URI is expected to be
>>> places in a Route header, and because each URI is expected to
>>> typically have a keepalive URI parameter in it.
>>
>> Like mentioned in other recent emails the problem is that if the UA
>> derives the proxy URI(s) from DHCP SIP server option or e.g. the AOR
>> of the user, in those cases the derived URI will NOT contain the
>> keepalive URI parameter in it. DHCP just returns IP addresses or
>> domain names instead of URIs with parameters.
>>
>> Thus in those cases it would be critical for the UA to be able
>> to ensure in REGISTER - 200 OK sequence that the contacted proxy
>> supports Outbound and can handle STUN requests sent to its SIP
>> port (not to cause any damage to an uncompliant proxy). I fully
>> agree with this proposal from Jeroen:
>>
>> --
>> The text should then say "the UAC MUST wait for a 200 OK to REGISTER
>> before
>> it starts sending STUN packets. If the last Path header of the 200 OK
>> does
>> not contain keepalive=3Dstun, the UAC MUST NOT use STUN=20
>towards this URI.
>> If
>> there are no Path headers and no "Supported: outbound", the UAC also
>> MUST
>> NOT use STUN towards the registrar.". It could still use TCP=20
>keepalive
>> if it
>> supports that
>> --
>>
>> Please excuse me if I am just repeating myself and what e.g. Jeroen
>> has said but I believe the importance of this issue justifies some
>> redundancy :-)
>>
>> Regards,
>>
>> Erkki
>
>

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Because TLS is used in SIP in hop-by-hop mode, the cipher suite that
you negotiate with your first hop proxy is no guarantee of what will
be negotiated with future proxies. Consider the classic trapezoid:

        Atlanta ---------------------- Biloxi
         /                                \
        /                                  \
      /	                                    \ 
alice@atlanta                             bob@biloxi

Alice can negotiate AES-256 with Atlanta but then have Atlanta
negotiate RC4-40 with Biloxi. This draft suggests a series of
extensions to SIP to carry the initial client's preferences and that
proxies further down the chain are expected to enforce.

It seems to me that there is a simpler mechanism: any proxy X
can only offer the intersection of the cipher suites that was offered
to it and the ciphersuites that it is willing to support. This has
the result that no downstream proxy can negotiate a cipher suite
that Alice wasn't willing to negotiate with Atlanta. 

There are two obvious disadvantages with respect to the proposed
draft:

1. Alice doesn't get any signal of what was eventually negotiated.
   I would solve this problem by extending Via (I'm actually surprised
   this isn't done). MTAs typically put it in Received lines, FWIW...
2. It's less expressive. In particular, the downstream cipher suites
   always narrow and never grow. So, if there were a third proxy in
   between Atlanta and Biloxi that only spoke DES, then Biloxi and Bob
   would only get DES even if they both spoke AES. Based on my experience
   with what real implementations support and the MUST for AES, I'm not
   much concerned about this. It's generally impossible to pickle
   the full client policy and have it sent to a proxy. It's mostly
   a matter of where you draw the line, and drawing it where the
   protocol already draws it seems natural.

The big advantage of the approach I propose is that it doesn't
require any new mechanisms and can be incrementally deployed in
an interoperable way.







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Oh, one more note. This draft claims:

   An obvious inference of this statement is that SIP allows use of
   different cipher suites at each hop along the path.  At the worst,
   SIP doesn't forbid the use of TLS_NULL_WITH_NULL_NULL ciphersuite for
   the SIPS URI scheme also.

This is explicitly forbidden by TLS.

   TLS_NULL_WITH_NULL_NULL is specified and is the initial state of a
   TLS connection during the first handshake on that channel, but must
   not be negotiated, as it provides no more protection than an
   unsecured connection.

-Ekr

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From sip-bounces@ietf.org Tue Jul 04 12:33:29 2006
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Subject: [Sip] Comments on draft-gurbani-sip-domain-certs-01
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It seems to me that this document addresses two issues:

1. What name(s) should appear in certificates.
2. Where they should appear and how they should be encoded.

I'd like to separate these two, on the theory that (1)
is difficult and (2) is bit-mangling.

Let's take a step back and examine first principles: The basic
assumption behind RFC 2818 (and most other uses of TLS) is that you
start with some URI X that you want to dereference. Through some
series of (untrusted) manipulations you end up with a connection to
some endpoint which presents you a certificate C. You then need to
determine whether X matches C. In order for this process to work
properly, X needs to either contain C or there needs to be some
trusted transformation between C and X.

In the Web context, what this means is that if you have a server
farm with A.example.com, B.example.com, C.example.com, etc.
then they all have certificates containing "www.example.com".
This can either be done via key-sharing or can be done by 
issuing them separate certificates that all contain the 
same domain name ("www.example.com").

ISTM that the same situation needs to apply with SIP. If
you're dereferencing <sips:alice@atlanta.com> then the
certificate presented to you needs to either be for a
known (and trusted!) proxy or contain "atlanta.com". Anything
else requires trusting the DNS resolution process, which
isn't generally considered acceptable.

So, this leaves us with the question of whether there should
be multiple names. As I understand S 10, the claimed disadvantages
of a single name are:

   1.  It does not correspond to common TLS usage, which is to use a
       certificate identifying the host.  If proxy.biloxi.com is allowed
       to require that the subject of the certificate contains a DNS
       entry corresponding to the IP address of proxy1.atlanta.com, then
       this in turn requires every authoritative proxy for atlanta.com
       to be registered with the DNS entry atlanta.com.  Hence it is
       impossible for a domain to use different SIP proxies as incoming
       and outgoing proxies.
   2.  It makes forensics in the case of compromise more complex.  If a
       malicious principal is discovered making use of a certificate for
       a private key issued to proxy1.atlanta.com, there is little to
       identify which of atlanta.com's SIP proxies has been compromised.
       If the certificate contains proxy1.atlanta.com's name as well as
       atlanta.com's name, then the forensics are simpler.

I'm not sure what "common TLS usage" is being referred to. As I 
noted above, in my experience HTTPS behaves exactly this way 
(putting the generic domain name and not host name in the
certificate). I agree that it does make forensics more difficult,
but this problem can be solved iwth a separate certificate 
(with a separate serial #) for each host, even if they contain
the same name. There's no need for a second name here.



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From sip-bounces@ietf.org Tue Jul 04 14:45:12 2006
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Subject: RE: [Sip] Comments on draft-koivusalo-sip-outbound-discovery-02
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With regard to 

> Additionally note that the words "if the client supports TLS" 
> are superfluous. As RFC 3261 requires support of TLS, we are 
> not documenting capabilities where TLS is not supported.
> 

It has been pointed out that UAC support is only SHOULD strength in RFC 3261 therefore the extra words are appropriate.

regards

Keith

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: 03 July 2006 20:12
> To: sip@ietf.org
> Cc: erkki.koivusalo@nokia.com
> Subject: [Sip] Comments on draft-koivusalo-sip-outbound-discovery-02
> 
> 
> Technical:
> ----------
> 
> 1) I need to understand whether this updates RFC 3263 or just 
> extends it. Part of this is what appears on the cover page 
> and in the RFC index as and when it is finally approved, but 
> the other part also relates to compatibility, see later comment.
> 
> 2) I think you have already acknowledged this, but the 
> document requires revision to align with the terminology in 
> the latest version (-04) of draft-ietf-sip-outbound.
> 
> 3) In section 5, as far as I understand, the NAPTR request is 
> identical for a UA that supports RFC 3263 and for one that 
> supports this extension. Therefore the difference is in the 
> response. Therefore the paragraph of UA requirements:
> 
>    As per RFC 2915 [4], the client discards any NAPTR records whose
>    services fields are not applicable.  A client resolving a SIPS URI
>    MUST discard any services that do not contain "SIPS-O" as the
>    protocol in the service field.  A client resolving a SIP URI MUST
>    retain records with "SIP-O" and the transport protocols the client
>    supports and SHOULD retain records with "SIPS-O", if the client
>    supports TLS.
> 
> must work with a NAPTR response that has no knowledge of 
> outbound. Is the above paragraph compatible with this? 
> 
> Additionally note that the words "if the client supports TLS" 
> are superfluous. As RFC 3261 requires support of TLS, we are 
> not documenting capabilities where TLS is not supported.
> 
> regards
> 
> Keith
> 
> Keith Drage
> drage@lucent.com
> tel: +44 1793 776249
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Subject: Re: [Sip] Ready for WGLC on Outbound?
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I think people have stopped posting on this because they can't say  
anything that they have not already said. Don't take that as they agree.

We got very strong feed back on one point - don't send STUN messages  
to a proxy unless you know it supports them. People felt strongly  
about that - I don't know why - but there was lots of people that  
agreed with that.

I imagine people have sent me around 10 options on how to get the  
stun supported information and the set of outbound proxy URIs -  
certainly DNS and DHCP are some of them. We can write some drafts on  
various ways to do it. Right now my concern is making sure that no  
matter which of these many ways the WG decided to move forward with,  
outbound will work. So far every one of the possibilities I have  
heard would be possible with the current outbound. Given that, I'm  
not sure what we need to change in outbound. I'm not even sure what  
you are proposing that we change in outbound - I think what you are  
proposing is we need to hold up outbound until we choose which one of  
the 10 we will use and furthermore you feel very strongly on which  
one of the 10 we should use. I'm not keen on this - I think we have  
shown they are separable problems. The WG decided on this separation  
of problems awhile back - I think it was in Vancouver - and many  
people supported it. I think it was a good idea  - we have shown we  
can move outbound forward before we have determined all the ways that  
this configuration information is delivered to the UA. I would be  
very hesitant to reverse this decision now.

I don't think Dean's posts were sent as decries of the requirements  
from the chair but purely as his opinion in trying to help bring the  
discussion to consensus. But I do think the draft meets the  
requirements and ideas he was outlining - I think the current draft  
does meet the requirements he outlines. It does have a way for the  
"discovering whether a proxy supports the extensions of outbound" and  
on the topic of "there are two ways we can establish compatibility  
with an extensions", the WG rejected option 2 and we have gone with 1.

Cullen (with my outbound author hat on)


On Jul 4, 2006, at 3:50 AM, <Erkki.Koivusalo@nokia.com>  
<Erkki.Koivusalo@nokia.com> wrote:

>
> Hi Rohan,
>
> Sorry to be persistent with this issue, but what you say
> seems to conflict with what was discussed in March.
>
>> The only time a UAC sends a STUN request as directed by
>> outbound -04 is when the URI in the outbound-proxy-set
>> includes the keepalive=stun parameter. That DHCP does not
>> provide URI is a limitation of the SIP DHCP configuration
>> mechanism, not outbound. If you want to write a draft on
>> how to "shift-up" to using keepalives from some SIP or
>> external stimulus, you can still do this.
>
> What you say seems to suggest that you agree there is a problem
> when using DHCP SIP server option (as it is currently defined)
> for finding out if the discovered proxy supports outbound and
> STUN keepalives, but the solution is not in scope of Outbound
> draft itself.
>
> However I came to a different understanding based on the emails
> seen in the list earlier. For instance this email from Dean
> Willis (SIP WG chair) seems to suggest that the problem should
> be in scope of the outbound draft:
> http://www1.ietf.org/mail-archive/web/sip/current/msg13667.html
>
> "Ok, I think we've boiled this down to a new requirement:
>
> for the "outbound" technique to work, we MUST have a means of
> discovering whether a proxy supports the extensions of outbound.
>
> While we could do that by doing a REGISTER exchange that includes
> an appropriate option-tag in Require:, it would be more efficient
> if the proxy discovery mechanism informed us of this detail. So,
> we posit that the Outbound draft SHOULD select and discuss such a
> mechanism."
>
> And in yet another email
> http://www1.ietf.org/mail-archive/web/sip/current/msg13692.html
> Dean tells this:
>
> "There are two ways we can establish compatibility with an extensions:
>
> 1) By foreknowledge
> 2) By experimentation -- try it and see if it works.
>
> Both have limitations:
>
> In the case of #1, we might have incorrect knowledge. So we'll  
> probably
> end up testing anyhow . . .
>
> In the case of #2, the experiment might cause a meltdown. So the  
> "probe
> for discovery" must be made completely safe. Also, we can  
> eventually end
> up with so many wierd hooks and idiosyncratic implementations that we
> end up doing a BUNCH of probing before we do any real  
> communicating. And
> it's possible for the conditions to deteriorate so much that probes  
> for
> two different possible conditions are mutually exclusive -- if one
> condition applies, the probe for another will cause a meltdown.
>
> So what happens if we start STUN-probing a proxy on its SIP port  
> and it
> doesn't support STUN? My guess is, it's random.
>
> ...
>
> One approach to "forward compatibility" design is to make sure that we
> can use both foreknowledge and experimentation -- the "trust but  
> verify"
> model.
>
> ...
>
> While it's easy to decide whether a configured registrar URI has a
> keepalive parameter, what happens to proxies we select based on
> something besides direct configuration?
>
> Specifically,
>
> 1) Can the DHCP option for default proxy present the keepalive
> parameter?"
> ----
>
> During these three months after seeing those comments in the list
> I have been expecting the outbound draft to provide answer to
> the question how UA could (within SIP signalling exchange)
> safely to check whether or not a proxy supports STUN keepalive.
> Two mechanims have been suggested how the UA could do this in
> a simple way:
>
> 1. The UA to check that the Path header in 200 OK returned for
>    REGISTER would contain keepalive=stun parameter for the proxy URI
>
> 2. The UA to add Proxy-Reequire: stun-keepalive parameter into
>    the REGISTER request and the proxy to remove this tag from
>    the request before forwarding it towards the registrar
>
> For me, any of those mechanisms would sound feasible and for
> safety reasons I would hope the draft to deny the UA for sending
> STUN keepalives to the proxy which did not pass such a test.
>
>> the same problem exists for the "comp" parameter for example.
>
> Yes but please bear in mind that for comp=sigcomp URI parameter
> RFC 3486 Compressing SIP gives several solutions how the UA
> would be able to check if the proxy supports SigComp while
> outbound does not define any other than relying on configuration.
> A cite from RFC 3486:
>
> 4.1   Obtaining a SIP or SIPS URI with comp=sigcomp
>
>    For requests within a dialog, a next-hop URI with the comp=sigcomp
>    parameter is obtained from a Record-Route header field when the
>    dialog is established.  A client sending a request outside a dialog
>    can also obtain SIP URIs with comp=sigcomp in a Contact header  
> field
>    in a 3xx or 485 response to the request.
>
>    However, clients establishing a session will not typically be  
> willing
>    to wait until the dialog is established in order to begin  
> compressing
>    messages.  One of the biggest gains that SigComp can bring to  
> SIP is
>    the ability to compress the initial INVITE of a dialog, when the  
> user
>    is waiting for the session to be established.  Therefore, clients
>    need a means to obtain a comp=sigcomp URI from their outbound proxy
>    before the user decides to establish a session.
>
>    One solution to this problem is manual configuration.  However,
>    sometimes it is necessary to have clients configured in an  
> automatic
>    fashion.  Unfortunately, current mechanisms for SIP client
>    configuration (e.g., using DHCP [6]) do not allow to provide the
>    client with URI parameters.  In this case, the client SHOULD  
> send an
>    uncompressed OPTIONS request to its outbound proxy.  The outbound
>    proxy can provide an alternative SIP URI with the comp=sigcomp
>    parameter in a Contact header field in a 200 OK response to the
>    OPTIONS.  The client can use this URI for subsequent requests that
>    are sent through the same outbound proxy using compression.
>
>    RFC 3261 [1] does not define how a proxy should respond to an  
> OPTIONS
>    request addressed to itself.  It only describes how servers respond
>    to OPTIONS addressed to a particular user.  Section 11.2 of RFC  
> 3261
>    says:
>
>       Contact header fields MAY be present in a 200 (OK) response and
>       have the same semantics as in a 3xx response.  That is, they may
>       list a set of alternative names and methods of reaching the  
> user.
>
>    We extend this behavior to proxy servers responding to OPTIONS
>    addressed to them.  They MAY list a set of alternative URIs to
>    contact the proxy.
>
> What has been called for is that outbound draft would also
> specify one solution how the UA could check the STUN keepalive
> support of a proxy. In my opinion good proposals seem
> to exist how to do it in a reasonable way.
>
> Regards,
>
> Erkki
>
>> -----Original Message-----
>> From: ext Rohan Mahy [mailto:rohan@ekabal.com]
>> Sent: 04.July.2006 03:10
>> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>> Cc: dean.willis@softarmor.com; fluffy@cisco.com; Rohan Mahy;
>> sip@ietf.org; jbemmel@zonnet.nl
>> Subject: Re: [Sip] Ready for WGLC on Outbound?
>>
>> Hi Erkki,
>>
>> The only time a UAC sends a STUN request as directed by
>> outbound -04 is
>> when the URI in the outbound-proxy-set includes the keepalive=stun
>> parameter.  This is 100% backwards safe.  We don't need any text to
>> prevent existing implementations from getting a bunch of STUN  
>> requests.
>>
>> That DHCP does not provide URI is a limitation of the SIP DHCP
>> configuration mechanism, not outbound.  The same problem
>> exists for the
>> "comp" parameter for example.  If you want to write a draft on how to
>> "shift-up" to using keepalives or using compression from some SIP or
>> external stimulus, you can still do this.  No text in outbound
>> prevents
>> that from happening.
>>
>> thanks,
>> -rohan
>>
>>
>> On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
>>>
>>> Hi Rohan,
>>>
>>>>> Also, the issue is that what may be configured (no matter how you
>>>>> configure it) may not be a URI. It may simply be an IP
>> address/FQDN,
>>>>> which is then used to build a SIP-URI. I think we shall
>>>>> allow that, and make it more clear in the spec. [/CHH]
>>>>
>>>> the outbound spec defines a piece of configurable data (the
>>>> 'outbound-proxy-set'), which is a list of URIs.  This list
>> of URIs may
>>>> never have a one-to-one correspondence to a user or administrator
>>>> manually configurable field.  That piece of configurable data  
>>>> may be
>>>> derived from several fields, manually configurable, discovered via
>>>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
>>>> download, or via a random character generator seeded by the  
>>>> phase of
>>>> the moon.
>>>>
>>>> The outbound-proxy-set was defined as a list of URIs because the UA
>>>> acts on each item in the list, because each URI is expected to be
>>>> places in a Route header, and because each URI is expected to
>>>> typically have a keepalive URI parameter in it.
>>>
>>> Like mentioned in other recent emails the problem is that if the UA
>>> derives the proxy URI(s) from DHCP SIP server option or e.g. the AOR
>>> of the user, in those cases the derived URI will NOT contain the
>>> keepalive URI parameter in it. DHCP just returns IP addresses or
>>> domain names instead of URIs with parameters.
>>>
>>> Thus in those cases it would be critical for the UA to be able
>>> to ensure in REGISTER - 200 OK sequence that the contacted proxy
>>> supports Outbound and can handle STUN requests sent to its SIP
>>> port (not to cause any damage to an uncompliant proxy). I fully
>>> agree with this proposal from Jeroen:
>>>
>>> --
>>> The text should then say "the UAC MUST wait for a 200 OK to REGISTER
>>> before
>>> it starts sending STUN packets. If the last Path header of the  
>>> 200 OK
>>> does
>>> not contain keepalive=stun, the UAC MUST NOT use STUN
>> towards this URI.
>>> If
>>> there are no Path headers and no "Supported: outbound", the UAC also
>>> MUST
>>> NOT use STUN towards the registrar.". It could still use TCP
>> keepalive
>>> if it
>>> supports that
>>> --
>>>
>>> Please excuse me if I am just repeating myself and what e.g. Jeroen
>>> has said but I believe the importance of this issue justifies some
>>> redundancy :-)
>>>
>>> Regards,
>>>
>>> Erkki
>>
>>

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From sip-bounces@ietf.org Tue Jul 04 17:36:46 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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	<Erkki.Koivusalo@nokia.com>
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Subject: Re: [Sip] Ready for WGLC on Outbound?
Date: Tue, 4 Jul 2006 23:34:48 +0200
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Cullen,

The point is that these are really 2 separate issues:
1. Discovering the set of outbound proxy URIs (with keepalive=stun 
parameters in them)
2. Confirming that a given proxy in the set actually supports outbound (ie 
validating the keepalive=stun information)

Of these 2 options, (1) is indeed a separable problem. (2) - to me - is not, 
for me outbound is not acceptable without something to address (2). It 
really should say "the UAC MUST NOT use STUN until { mechanism for (2) says 
its supported }"

There have been several options mentioned to do this, a simple way is to 
have the edge proxy insert a flag. Question is: why not?

Regards,

Jeroen

----- Original Message ----- 
From: "Cullen Jennings" <fluffy@cisco.com>
To: <Erkki.Koivusalo@nokia.com>
Cc: <rohan@ekabal.com>; <dean.willis@softarmor.com>; <sip@ietf.org>; 
<jbemmel@zonnet.nl>
Sent: Tuesday, July 04, 2006 10:37 PM
Subject: Re: [Sip] Ready for WGLC on Outbound?


>
> I think people have stopped posting on this because they can't say 
> anything that they have not already said. Don't take that as they agree.
>
> We got very strong feed back on one point - don't send STUN messages  to a 
> proxy unless you know it supports them. People felt strongly  about that - 
> I don't know why - but there was lots of people that  agreed with that.
>
> I imagine people have sent me around 10 options on how to get the  stun 
> supported information and the set of outbound proxy URIs -  certainly DNS 
> and DHCP are some of them. We can write some drafts on  various ways to do 
> it. Right now my concern is making sure that no  matter which of these 
> many ways the WG decided to move forward with,  outbound will work. So far 
> every one of the possibilities I have  heard would be possible with the 
> current outbound. Given that, I'm  not sure what we need to change in 
> outbound. I'm not even sure what  you are proposing that we change in 
> outbound - I think what you are  proposing is we need to hold up outbound 
> until we choose which one of  the 10 we will use and furthermore you feel 
> very strongly on which  one of the 10 we should use. I'm not keen on 
> this - I think we have  shown they are separable problems. The WG decided 
> on this separation  of problems awhile back - I think it was in 
> Vancouver - and many  people supported it. I think it was a good idea  - 
> we have shown we  can move outbound forward before we have determined all 
> the ways that  this configuration information is delivered to the UA. I 
> would be  very hesitant to reverse this decision now.
>
> I don't think Dean's posts were sent as decries of the requirements  from 
> the chair but purely as his opinion in trying to help bring the 
> discussion to consensus. But I do think the draft meets the  requirements 
> and ideas he was outlining - I think the current draft  does meet the 
> requirements he outlines. It does have a way for the  "discovering whether 
> a proxy supports the extensions of outbound" and  on the topic of "there 
> are two ways we can establish compatibility  with an extensions", the WG 
> rejected option 2 and we have gone with 1.
>
> Cullen (with my outbound author hat on)
>
>
> On Jul 4, 2006, at 3:50 AM, <Erkki.Koivusalo@nokia.com> 
> <Erkki.Koivusalo@nokia.com> wrote:
>
>>
>> Hi Rohan,
>>
>> Sorry to be persistent with this issue, but what you say
>> seems to conflict with what was discussed in March.
>>
>>> The only time a UAC sends a STUN request as directed by
>>> outbound -04 is when the URI in the outbound-proxy-set
>>> includes the keepalive=stun parameter. That DHCP does not
>>> provide URI is a limitation of the SIP DHCP configuration
>>> mechanism, not outbound. If you want to write a draft on
>>> how to "shift-up" to using keepalives from some SIP or
>>> external stimulus, you can still do this.
>>
>> What you say seems to suggest that you agree there is a problem
>> when using DHCP SIP server option (as it is currently defined)
>> for finding out if the discovered proxy supports outbound and
>> STUN keepalives, but the solution is not in scope of Outbound
>> draft itself.
>>
>> However I came to a different understanding based on the emails
>> seen in the list earlier. For instance this email from Dean
>> Willis (SIP WG chair) seems to suggest that the problem should
>> be in scope of the outbound draft:
>> http://www1.ietf.org/mail-archive/web/sip/current/msg13667.html
>>
>> "Ok, I think we've boiled this down to a new requirement:
>>
>> for the "outbound" technique to work, we MUST have a means of
>> discovering whether a proxy supports the extensions of outbound.
>>
>> While we could do that by doing a REGISTER exchange that includes
>> an appropriate option-tag in Require:, it would be more efficient
>> if the proxy discovery mechanism informed us of this detail. So,
>> we posit that the Outbound draft SHOULD select and discuss such a
>> mechanism."
>>
>> And in yet another email
>> http://www1.ietf.org/mail-archive/web/sip/current/msg13692.html
>> Dean tells this:
>>
>> "There are two ways we can establish compatibility with an extensions:
>>
>> 1) By foreknowledge
>> 2) By experimentation -- try it and see if it works.
>>
>> Both have limitations:
>>
>> In the case of #1, we might have incorrect knowledge. So we'll  probably
>> end up testing anyhow . . .
>>
>> In the case of #2, the experiment might cause a meltdown. So the  "probe
>> for discovery" must be made completely safe. Also, we can  eventually end
>> up with so many wierd hooks and idiosyncratic implementations that we
>> end up doing a BUNCH of probing before we do any real  communicating. And
>> it's possible for the conditions to deteriorate so much that probes  for
>> two different possible conditions are mutually exclusive -- if one
>> condition applies, the probe for another will cause a meltdown.
>>
>> So what happens if we start STUN-probing a proxy on its SIP port  and it
>> doesn't support STUN? My guess is, it's random.
>>
>> ...
>>
>> One approach to "forward compatibility" design is to make sure that we
>> can use both foreknowledge and experimentation -- the "trust but  verify"
>> model.
>>
>> ...
>>
>> While it's easy to decide whether a configured registrar URI has a
>> keepalive parameter, what happens to proxies we select based on
>> something besides direct configuration?
>>
>> Specifically,
>>
>> 1) Can the DHCP option for default proxy present the keepalive
>> parameter?"
>> ----
>>
>> During these three months after seeing those comments in the list
>> I have been expecting the outbound draft to provide answer to
>> the question how UA could (within SIP signalling exchange)
>> safely to check whether or not a proxy supports STUN keepalive.
>> Two mechanims have been suggested how the UA could do this in
>> a simple way:
>>
>> 1. The UA to check that the Path header in 200 OK returned for
>>    REGISTER would contain keepalive=stun parameter for the proxy URI
>>
>> 2. The UA to add Proxy-Reequire: stun-keepalive parameter into
>>    the REGISTER request and the proxy to remove this tag from
>>    the request before forwarding it towards the registrar
>>
>> For me, any of those mechanisms would sound feasible and for
>> safety reasons I would hope the draft to deny the UA for sending
>> STUN keepalives to the proxy which did not pass such a test.
>>
>>> the same problem exists for the "comp" parameter for example.
>>
>> Yes but please bear in mind that for comp=sigcomp URI parameter
>> RFC 3486 Compressing SIP gives several solutions how the UA
>> would be able to check if the proxy supports SigComp while
>> outbound does not define any other than relying on configuration.
>> A cite from RFC 3486:
>>
>> 4.1   Obtaining a SIP or SIPS URI with comp=sigcomp
>>
>>    For requests within a dialog, a next-hop URI with the comp=sigcomp
>>    parameter is obtained from a Record-Route header field when the
>>    dialog is established.  A client sending a request outside a dialog
>>    can also obtain SIP URIs with comp=sigcomp in a Contact header  field
>>    in a 3xx or 485 response to the request.
>>
>>    However, clients establishing a session will not typically be  willing
>>    to wait until the dialog is established in order to begin  compressing
>>    messages.  One of the biggest gains that SigComp can bring to  SIP is
>>    the ability to compress the initial INVITE of a dialog, when the  user
>>    is waiting for the session to be established.  Therefore, clients
>>    need a means to obtain a comp=sigcomp URI from their outbound proxy
>>    before the user decides to establish a session.
>>
>>    One solution to this problem is manual configuration.  However,
>>    sometimes it is necessary to have clients configured in an  automatic
>>    fashion.  Unfortunately, current mechanisms for SIP client
>>    configuration (e.g., using DHCP [6]) do not allow to provide the
>>    client with URI parameters.  In this case, the client SHOULD  send an
>>    uncompressed OPTIONS request to its outbound proxy.  The outbound
>>    proxy can provide an alternative SIP URI with the comp=sigcomp
>>    parameter in a Contact header field in a 200 OK response to the
>>    OPTIONS.  The client can use this URI for subsequent requests that
>>    are sent through the same outbound proxy using compression.
>>
>>    RFC 3261 [1] does not define how a proxy should respond to an  OPTIONS
>>    request addressed to itself.  It only describes how servers respond
>>    to OPTIONS addressed to a particular user.  Section 11.2 of RFC  3261
>>    says:
>>
>>       Contact header fields MAY be present in a 200 (OK) response and
>>       have the same semantics as in a 3xx response.  That is, they may
>>       list a set of alternative names and methods of reaching the  user.
>>
>>    We extend this behavior to proxy servers responding to OPTIONS
>>    addressed to them.  They MAY list a set of alternative URIs to
>>    contact the proxy.
>>
>> What has been called for is that outbound draft would also
>> specify one solution how the UA could check the STUN keepalive
>> support of a proxy. In my opinion good proposals seem
>> to exist how to do it in a reasonable way.
>>
>> Regards,
>>
>> Erkki
>>
>>> -----Original Message-----
>>> From: ext Rohan Mahy [mailto:rohan@ekabal.com]
>>> Sent: 04.July.2006 03:10
>>> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>>> Cc: dean.willis@softarmor.com; fluffy@cisco.com; Rohan Mahy;
>>> sip@ietf.org; jbemmel@zonnet.nl
>>> Subject: Re: [Sip] Ready for WGLC on Outbound?
>>>
>>> Hi Erkki,
>>>
>>> The only time a UAC sends a STUN request as directed by
>>> outbound -04 is
>>> when the URI in the outbound-proxy-set includes the keepalive=stun
>>> parameter.  This is 100% backwards safe.  We don't need any text to
>>> prevent existing implementations from getting a bunch of STUN  requests.
>>>
>>> That DHCP does not provide URI is a limitation of the SIP DHCP
>>> configuration mechanism, not outbound.  The same problem
>>> exists for the
>>> "comp" parameter for example.  If you want to write a draft on how to
>>> "shift-up" to using keepalives or using compression from some SIP or
>>> external stimulus, you can still do this.  No text in outbound
>>> prevents
>>> that from happening.
>>>
>>> thanks,
>>> -rohan
>>>
>>>
>>> On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
>>>>
>>>> Hi Rohan,
>>>>
>>>>>> Also, the issue is that what may be configured (no matter how you
>>>>>> configure it) may not be a URI. It may simply be an IP
>>> address/FQDN,
>>>>>> which is then used to build a SIP-URI. I think we shall
>>>>>> allow that, and make it more clear in the spec. [/CHH]
>>>>>
>>>>> the outbound spec defines a piece of configurable data (the
>>>>> 'outbound-proxy-set'), which is a list of URIs.  This list
>>> of URIs may
>>>>> never have a one-to-one correspondence to a user or administrator
>>>>> manually configurable field.  That piece of configurable data  may be
>>>>> derived from several fields, manually configurable, discovered via
>>>>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
>>>>> download, or via a random character generator seeded by the  phase of
>>>>> the moon.
>>>>>
>>>>> The outbound-proxy-set was defined as a list of URIs because the UA
>>>>> acts on each item in the list, because each URI is expected to be
>>>>> places in a Route header, and because each URI is expected to
>>>>> typically have a keepalive URI parameter in it.
>>>>
>>>> Like mentioned in other recent emails the problem is that if the UA
>>>> derives the proxy URI(s) from DHCP SIP server option or e.g. the AOR
>>>> of the user, in those cases the derived URI will NOT contain the
>>>> keepalive URI parameter in it. DHCP just returns IP addresses or
>>>> domain names instead of URIs with parameters.
>>>>
>>>> Thus in those cases it would be critical for the UA to be able
>>>> to ensure in REGISTER - 200 OK sequence that the contacted proxy
>>>> supports Outbound and can handle STUN requests sent to its SIP
>>>> port (not to cause any damage to an uncompliant proxy). I fully
>>>> agree with this proposal from Jeroen:
>>>>
>>>> --
>>>> The text should then say "the UAC MUST wait for a 200 OK to REGISTER
>>>> before
>>>> it starts sending STUN packets. If the last Path header of the  200 OK
>>>> does
>>>> not contain keepalive=stun, the UAC MUST NOT use STUN
>>> towards this URI.
>>>> If
>>>> there are no Path headers and no "Supported: outbound", the UAC also
>>>> MUST
>>>> NOT use STUN towards the registrar.". It could still use TCP
>>> keepalive
>>>> if it
>>>> supports that
>>>> --
>>>>
>>>> Please excuse me if I am just repeating myself and what e.g. Jeroen
>>>> has said but I believe the importance of this issue justifies some
>>>> redundancy :-)
>>>>
>>>> Regards,
>>>>
>>>> Erkki
>>>
>>> 


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From sip-bounces@ietf.org Tue Jul 04 18:32:53 2006
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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
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Hi Eric,

   Thx for your review and detailed comments. Pl find my replies in line
below.

Thx
Samir



Because TLS is used in SIP in hop-by-hop mode, the cipher suite that you
negotiate with your first hop proxy is no guarantee of what will be
negotiated with future proxies. Consider the classic trapezoid:

        Atlanta ---------------------- Biloxi
         /                                \
        /                                  \
      /	                                    \=20
alice@atlanta                             bob@biloxi

Alice can negotiate AES-256 with Atlanta but then have Atlanta negotiate
RC4-40 with Biloxi. This draft suggests a series of extensions to SIP to
carry the initial client's preferences and that proxies further down the
chain are expected to enforce.

It seems to me that there is a simpler mechanism: any proxy X can only
offer the intersection of the cipher suites that was offered to it and
the ciphersuites that it is willing to support. This has the result that
no downstream proxy can negotiate a cipher suite that Alice wasn't
willing to negotiate with Atlanta.=20

SS>> Here it is assumed that cipher-suite X which is between
alice@atlanta and Proxy for atlanta is supported between proxies for
biloxi and atlanta. This may not be true. Suppose Proxy for atlanta is
upgraded with the latest TLS version, extension, and cipher suites. But
proxy for biloxi is not upgraded. In this case we cannot make a secure
call. But with the proposal, we can make a call though it will be less
secure. The level of security (qValue) is indicated back in the
response.


There are two obvious disadvantages with respect to the proposed
draft:

1. Alice doesn't get any signal of what was eventually negotiated.
   I would solve this problem by extending Via (I'm actually surprised
   this isn't done). MTAs typically put it in Received lines, FWIW...

SS>> Proposal extends the via to include all TLS security information.
Here we agree.

2. It's less expressive. In particular, the downstream cipher suites
   always narrow and never grow. So, if there were a third proxy in
   between Atlanta and Biloxi that only spoke DES, then Biloxi and Bob
   would only get DES even if they both spoke AES. Based on my
experience
   with what real implementations support and the MUST for AES, I'm not
   much concerned about this. It's generally impossible to pickle
   the full client policy and have it sent to a proxy. It's mostly
   a matter of where you draw the line, and drawing it where the
   protocol already draws it seems natural.

SS>> Here if proxy for atlanta has good control over it's network, and
it is using less secure cipher-suite (or no-security) in that hop, then
as per your proposal it is going to use same less secure cipher-suite
all along the path even in open internet where it is crossing provider's
boundary, even if edge proxies are supporting the most secure
cipher-suites etc. Here it should be noted that upgrading edge proxies
is easy, as compared to upgrading every UA within provider's domain
with the latest cipher-suite, protocol etc. In your example, If proxies
are crossing the provider's boundaries over the open internet, then I
would like to have AES in that hop. This cannot be done in your
proposal, even if it is supported.=20


The big advantage of the approach I propose is that it doesn't require
any new mechanisms and can be incrementally deployed in an interoperable
way.

SS>> My problem is that current mechanism are not future proof. Suppose
down the line current cipher-suites, protocols etc are broken, and we
have new cipher-suites etc . Then we will have different cipher-suites
in the network, we cannot assume that every SIP entity in the network is
upgraded quickly. Proposal caters that with clear indication of
security.=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol Use
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From sip-bounces@ietf.org Tue Jul 04 18:37:15 2006
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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
Date: Tue, 4 Jul 2006 17:35:25 -0500
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Hi Eric,

Thx for pointing this out. My mistake. Will be fixed in the next
version.

Thx
Samir=20

-----Original Message-----
From: Eric Rescorla [mailto:ekr@networkresonance.com]=20
Sent: Tuesday, July 04, 2006 9:22 AM
To: sip@ietf.org
Subject: Re: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00

Oh, one more note. This draft claims:

   An obvious inference of this statement is that SIP allows use of
   different cipher suites at each hop along the path.  At the worst,
   SIP doesn't forbid the use of TLS_NULL_WITH_NULL_NULL ciphersuite for
   the SIPS URI scheme also.

This is explicitly forbidden by TLS.

   TLS_NULL_WITH_NULL_NULL is specified and is the initial state of a
   TLS connection during the first handshake on that channel, but must
   not be negotiated, as it provides no more protection than an
   unsecured connection.

-Ekr

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From sip-bounces@ietf.org Tue Jul 04 18:53:27 2006
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"Samir Srivastava" <samirsr@nortel.com> writes:
> SS>> Here it is assumed that cipher-suite X which is between
> alice@atlanta and Proxy for atlanta is supported between proxies for
> biloxi and atlanta. This may not be true. Suppose Proxy for atlanta is
> upgraded with the latest TLS version, extension, and cipher suites. But
> proxy for biloxi is not upgraded. In this case we cannot make a secure
> call.

No, this is incorrect. I didn't say the *selected* cipher suite,
I said the intersection of the offered and acceptable. Consider
the following case for what various agents support:

Alice        Atlanta         Biloxi
-----        -------         ------
AES-SHA      AES-SHA         3DES-SHA
AES-SHA256   AES-SHA256      AES-SHA

In this case, Alice and Atlanta will (most likely) negotiate
AES-SHA256.  Atlanta will offer AES-SHA and AES-SHA256. Biloxi will
select AES-SHA. The only way in which the second hop could fail
would be if Atlanta didn't speak AES-SHA, but remember that 
AES-SHA is a MUST implement.


> 2. It's less expressive. In particular, the downstream cipher suites
>    always narrow and never grow. So, if there were a third proxy in
>    between Atlanta and Biloxi that only spoke DES, then Biloxi and Bob
>    would only get DES even if they both spoke AES. Based on my
> experience
>    with what real implementations support and the MUST for AES, I'm not
>    much concerned about this. It's generally impossible to pickle
>    the full client policy and have it sent to a proxy. It's mostly
>    a matter of where you draw the line, and drawing it where the
>    protocol already draws it seems natural.
>
> SS>> Here if proxy for atlanta has good control over it's network, and
> it is using less secure cipher-suite (or no-security) in that hop, then
> as per your proposal it is going to use same less secure cipher-suite
> all along the path even in open internet where it is crossing provider's
> boundary, even if edge proxies are supporting the most secure
> cipher-suites etc.

I don't see your point here. You seem to be suggesting that we should
deliberately be fostering a mechanism that allows providers to use
weak mechanisms inside their internal networks. That doesn't seem
like a particularly worthy goal, so I don't see why that merits 
the amount of extra protocol complexity you want to introduce.


> Here it should be noted that upgrading edge proxies
> is easy, as compared to upgrading every UA within provider's domain
> with the latest cipher-suite, protocol etc. In your example, If proxies
> are crossing the provider's boundaries over the open internet, then I
> would like to have AES in that hop. This cannot be done in your
> proposal, even if it is supported. 

It would seem easier to simply relax the narrowing constraint I
indicated in cases where both ends are under the control of
the same operator. This seems a lot simpler than lugging a
bunch of policy information around inside your internal network.


> The big advantage of the approach I propose is that it doesn't require
> any new mechanisms and can be incrementally deployed in an interoperable
> way.
>
> SS>> My problem is that current mechanism are not future proof. Suppose
> down the line current cipher-suites, protocols etc are broken, and we
> have new cipher-suites etc . Then we will have different cipher-suites
> in the network, we cannot assume that every SIP entity in the network is
> upgraded quickly. Proposal caters that with clear indication of
> security. 

Can you name an instance in which this has actually happened?

But suppose it's true? You just relax the constraint that you need to
offer only incoming cipher suites, upgrade your edge proxies,
and the problem goes away. The reason not to do that ordinarily
is that we don't have an unambigous ordering of cipher suite
strengths so we'd like to let the original UA have an opinion
but if you *know* that the UA's offered algorithm is broken,
I think it's safe to override it.

-Ekr

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Another update, including more feedback from the list.

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20
> Sent: Tuesday, June 27, 2006 12:50
> To: i-d-announce@ietf.org
> Subject: I-D ACTION:draft-audet-sip-sips-guidelines-02.txt
>=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>=20
>=20
> 	Title		: Guidelines for the use of the SIPS=20
> URI Scheme in the Session Initiation Protocol (SIP)
> 	Author(s)	: F. Audet
> 	Filename	: draft-audet-sip-sips-guidelines-02.txt
> 	Pages		: 24
> 	Date		: 2006-6-27
> =09
> This document provides clarifications and guidelines concerning the
>    use of SIPS URI Scheme in the Session Initiation Protocol (SIP).
>=20
> A URL for this Internet-Draft is:=20
> http://www.ietf.org/internet-drafts/draft-audet-sip-sips-guide
lines-02.txt

To remove yourself from the I-D Announcement list, send a message to=20
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Internet-Drafts are also available by anonymous FTP. Login with the
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A list of Internet-Drafts directories can be found in
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Internet-Drafts can also be obtained by e-mail.

Send a message to:
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=09
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From sip-bounces@ietf.org Tue Jul 04 21:44:57 2006
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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
Date: Tue, 4 Jul 2006 20:42:57 -0500
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Hi Eric,

   Refer the  http://www.cryptosystem.net/aes/ . Some points from there
are quoted below w.r.t algebraic attacks on it...


"Should AES be used ? Most people will answer yes, for now, due to the
lack of a better standard. AES remains still millions of times better
than many snake oil ciphers you can buy here or there. Moreover, even if
AES will be broken in the coming years by a fast algebraic attack, it
will probably still remain more secure than DES, and can still safely be
used for applications that do not require long term security.=20

However, though AES seems overdesigned with respect to differential or
linear cryptanalysis, from the point of view of algebraic attacks, it is
an extremely bad cipher. It is rather difficult to imagine a worse
cipher, among ciphers that resist to other known attacks... (worse with
respect to algebraic attacks). Therefore, even if it turns out that the
XSL is NOT faster than the exhaustive search of a 128-bit key, NIST
should sooner or later revoke AES. One should not take chances with
these things. Another open call for a cryptographic standard will
probably be done in the coming years (no need to hurry). "

I am not expert on the algebraic attacks, but I sensed the need for
openess in SIP from this prespective, as breaking of older crypto
algorithm forced to create the new algorithms (AES). IMHO one cannot
deny this trend in future. I will appreciate your views on this.

More replies inline below.

Thx
Samir


"Samir Srivastava" <samirsr@nortel.com> writes:
> SS>> Here it is assumed that cipher-suite X which is between
> alice@atlanta and Proxy for atlanta is supported between proxies for=20
> biloxi and atlanta. This may not be true. Suppose Proxy for atlanta is

> upgraded with the latest TLS version, extension, and cipher suites.=20
> But proxy for biloxi is not upgraded. In this case we cannot make a=20
> secure call.

No, this is incorrect. I didn't say the *selected* cipher suite, I said
the intersection of the offered and acceptable. Consider the following
case for what various agents support:

Alice        Atlanta         Biloxi
-----        -------         ------
AES-SHA      AES-SHA         3DES-SHA
AES-SHA256   AES-SHA256      AES-SHA

In this case, Alice and Atlanta will (most likely) negotiate AES-SHA256.
Atlanta will offer AES-SHA and AES-SHA256. Biloxi will select AES-SHA.
The only way in which the second hop could fail would be if Atlanta
didn't speak AES-SHA, but remember that AES-SHA is a MUST implement.

SS>> Fine. It will work in the current state of affairs. Breaking of
AES-SHA in future, will need another cipher-suite. Then we fix TLS with
that (newer version etc). But SIP doesn't have any knowledge of how much
secure hop it is.=20

It is an open issue, how the UAC or proxy preferences of cipher-suites
etc will be overidden. My first goal is to have an agreement on the need
of this in SIP. UAC needs to have the knowledge with what level of
security, it is able to reach UAS. IMHO, Level of Security cannot be
compared with other features of underlying protocols. Without being
having a guranteed level of security, I cannot share senstive info.
There is no such indication in SIP. This level of security can be
upgraded (At UAC and UAS) when SRTP key negotiation etc is done and
media is guranteed to secure also.


> 2. It's less expressive. In particular, the downstream cipher suites
>    always narrow and never grow. So, if there were a third proxy in
>    between Atlanta and Biloxi that only spoke DES, then Biloxi and Bob
>    would only get DES even if they both spoke AES. Based on my
> experience=20
>    with what real implementations support and the MUST for AES, I'm
not
>    much concerned about this. It's generally impossible to pickle
>    the full client policy and have it sent to a proxy. It's mostly
>    a matter of where you draw the line, and drawing it where the
>    protocol already draws it seems natural.
>
> SS>> Here if proxy for atlanta has good control over it's network, and
> it is using less secure cipher-suite (or no-security) in that hop,
then
> as per your proposal it is going to use same less secure cipher-suite
> all along the path even in open internet where it is crossing
provider's
> boundary, even if edge proxies are supporting the most secure
> cipher-suites etc.

I don't see your point here. You seem to be suggesting that we should
deliberately be fostering a mechanism that allows providers to use
weak mechanisms inside their internal networks. That doesn't seem
like a particularly worthy goal, so I don't see why that merits=20
the amount of extra protocol complexity you want to introduce.

SS>> This is being inherited from SIPS. Where it doesn't even have TLS
in the last hop. I don't like  this mechanism, if I am interested in
complete ironclad e2e AES (as per the current state of encryption) . But
closed walled garden supporters like it, and I don't want to start
debate on this.

> Here it should be noted that upgrading edge proxies
> is easy, as compared to upgrading every UA within provider's domain
> with the latest cipher-suite, protocol etc. In your example, If
proxies
> are crossing the provider's boundaries over the open internet, then I
> would like to have AES in that hop. This cannot be done in your
> proposal, even if it is supported.=20

It would seem easier to simply relax the narrowing constraint I
indicated in cases where both ends are under the control of
the same operator. This seems a lot simpler than lugging a
bunch of policy information around inside your internal network.

SS>> Being at UAC, I don't know in advance how I will be able to reach
the UAS. Take the case of retargeting from the another domain's proxy to
the UAC domain proxy. The proposal is cleaner w.r.t. all these aspects.=20

> The big advantage of the approach I propose is that it doesn't require
> any new mechanisms and can be incrementally deployed in an
interoperable
> way.
>
> SS>> My problem is that current mechanism are not future proof.
Suppose
> down the line current cipher-suites, protocols etc are broken, and we
> have new cipher-suites etc . Then we will have different cipher-suites
> in the network, we cannot assume that every SIP entity in the network
is
> upgraded quickly. Proposal caters that with clear indication of
> security.=20

Can you name an instance in which this has actually happened?

But suppose it's true? You just relax the constraint that you need to
offer only incoming cipher suites, upgrade your edge proxies,
and the problem goes away. The reason not to do that ordinarily
is that we don't have an unambigous ordering of cipher suite
strengths so we'd like to let the original UA have an opinion
but if you *know* that the UA's offered algorithm is broken,
I think it's safe to override it.

SS>> Refer the earlier text in the reply. Ordering of strength of
cipher-suites etc and overriding is not considered currently. If we are
in the agreement that we need a security-level kind of header in SIP,
then we can hash it out.


-Ekr

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From sip-bounces@ietf.org Tue Jul 04 23:31:10 2006
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"Samir Srivastava" <samirsr@nortel.com> writes:
> In this case, Alice and Atlanta will (most likely) negotiate AES-SHA256.
> Atlanta will offer AES-SHA and AES-SHA256. Biloxi will select AES-SHA.
> The only way in which the second hop could fail would be if Atlanta
> didn't speak AES-SHA, but remember that AES-SHA is a MUST implement.
>
> SS>> Fine. It will work in the current state of affairs. Breaking of
> AES-SHA in future, will need another cipher-suite. Then we fix TLS with
> that (newer version etc). But SIP doesn't have any knowledge of how much
> secure hop it is. 

Again, that's not true with the scheme I described. Only 
cipher suites that the first-hop client is willing to negotiate
will be negotiated on any hop. Consider the simple case where
there is currently only one algorithm X. When a new algorithm
Y is introduced (whatever the status of X), if the original
UAC only supports X, then there are two possibilities:

1. Every proxy along the path supports X as well as Y.
2. Some proxy along the path only supports Y.
 
Given the algorithm I described, in case 1 you get X (which by
definition is OK with the UAC). In case 2, the connection fails
somewhere along the line. In neither case can you get a cipher
suite that X does not consider acceptable.


> SS>> Being at UAC, I don't know in advance how I will be able to reach
> the UAS. Take the case of retargeting from the another domain's proxy to
> the UAC domain proxy. The proposal is cleaner w.r.t. all these aspects. 

I don't understand the point you're making here. I'm not talking
about the UAC, but about the various proxies inside a single
trust domain.


> Can you name an instance in which this has actually happened?
>
> But suppose it's true? You just relax the constraint that you need to
> offer only incoming cipher suites, upgrade your edge proxies,
> and the problem goes away. The reason not to do that ordinarily
> is that we don't have an unambigous ordering of cipher suite
> strengths so we'd like to let the original UA have an opinion
> but if you *know* that the UA's offered algorithm is broken,
> I think it's safe to override it.
>
> SS>> Refer the earlier text in the reply. Ordering of strength of
> cipher-suites etc and overriding is not considered currently.

I don't understand this point at all. What I'm saying is that
the only reason why you need to consult the UAC's cipher
suite preferences at all is that we don't have a single
agreed upon ordering of algorithm strengths. If we did,
then it would be a simple matter of saying "on outgoing
you can negotiate any algorithm as strong or stronger
as the one on incoming." However, in the case where the
proxy *knows* that the incoming algorithm is broken, then
it's safe to negotiate a stronger algorithm even if the
client did not offer it.


> If we are
> in the agreement that we need a security-level kind of header in SIP,
> then we can hash it out.

I don't agree that we need such a header.

-Ekr

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From: Ben Campbell <ben@estacado.net>
Subject: Re: [Sip] Ready for WGLC on Outbound?
Date: Tue, 4 Jul 2006 23:10:28 -0500
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I've been a  bystander, but will jump in here. Comment below:

On Jul 4, 2006, at 3:37 PM, Cullen Jennings wrote:

>
> I think people have stopped posting on this because they can't say  
> anything that they have not already said. Don't take that as they  
> agree.
>
> We got very strong feed back on one point - don't send STUN  
> messages to a proxy unless you know it supports them. People felt  
> strongly about that - I don't know why - but there was lots of  
> people that agreed with that.
>
> I imagine people have sent me around 10 options on how to get the  
> stun supported information and the set of outbound proxy URIs -  
> certainly DNS and DHCP are some of them. We can write some drafts  
> on various ways to do it. Right now my concern is making sure that  
> no matter which of these many ways the WG decided to move forward  
> with, outbound will work. So far every one of the possibilities I  
> have heard would be possible with the current outbound. Given that,  
> I'm not sure what we need to change in outbound. I'm not even sure  
> what you are proposing that we change in outbound - I think what  
> you are proposing is we need to hold up outbound until we choose  
> which one of the 10 we will use and furthermore you feel very  
> strongly on which one of the 10 we should use. I'm not keen on this  
> - I think we have shown they are separable problems. The WG decided  
> on this separation of problems awhile back - I think it was in  
> Vancouver - and many people supported it. I think it was a good  
> idea  - we have shown we can move outbound forward before we have  
> determined all the ways that this configuration information is  
> delivered to the UA. I would be very hesitant to reverse this  
> decision now.

I strongly agree with Cullen. We do not need to block the outbound  
mechanism draft on figuring out the discovery stuff. The separation  
was a good idea. If for no other reason, it lets us get work done.




>
> I don't think Dean's posts were sent as decries of the requirements  
> from the chair but purely as his opinion in trying to help bring  
> the discussion to consensus. But I do think the draft meets the  
> requirements and ideas he was outlining - I think the current draft  
> does meet the requirements he outlines. It does have a way for the  
> "discovering whether a proxy supports the extensions of outbound"  
> and on the topic of "there are two ways we can establish  
> compatibility with an extensions", the WG rejected option 2 and we  
> have gone with 1.
>
> Cullen (with my outbound author hat on)
>
>
> On Jul 4, 2006, at 3:50 AM, <Erkki.Koivusalo@nokia.com>  
> <Erkki.Koivusalo@nokia.com> wrote:
>
>>
>> Hi Rohan,
>>
>> Sorry to be persistent with this issue, but what you say
>> seems to conflict with what was discussed in March.
>>
>>> The only time a UAC sends a STUN request as directed by
>>> outbound -04 is when the URI in the outbound-proxy-set
>>> includes the keepalive=stun parameter. That DHCP does not
>>> provide URI is a limitation of the SIP DHCP configuration
>>> mechanism, not outbound. If you want to write a draft on
>>> how to "shift-up" to using keepalives from some SIP or
>>> external stimulus, you can still do this.
>>
>> What you say seems to suggest that you agree there is a problem
>> when using DHCP SIP server option (as it is currently defined)
>> for finding out if the discovered proxy supports outbound and
>> STUN keepalives, but the solution is not in scope of Outbound
>> draft itself.
>>
>> However I came to a different understanding based on the emails
>> seen in the list earlier. For instance this email from Dean
>> Willis (SIP WG chair) seems to suggest that the problem should
>> be in scope of the outbound draft:
>> http://www1.ietf.org/mail-archive/web/sip/current/msg13667.html
>>
>> "Ok, I think we've boiled this down to a new requirement:
>>
>> for the "outbound" technique to work, we MUST have a means of
>> discovering whether a proxy supports the extensions of outbound.
>>
>> While we could do that by doing a REGISTER exchange that includes
>> an appropriate option-tag in Require:, it would be more efficient
>> if the proxy discovery mechanism informed us of this detail. So,
>> we posit that the Outbound draft SHOULD select and discuss such a
>> mechanism."
>>
>> And in yet another email
>> http://www1.ietf.org/mail-archive/web/sip/current/msg13692.html
>> Dean tells this:
>>
>> "There are two ways we can establish compatibility with an  
>> extensions:
>>
>> 1) By foreknowledge
>> 2) By experimentation -- try it and see if it works.
>>
>> Both have limitations:
>>
>> In the case of #1, we might have incorrect knowledge. So we'll  
>> probably
>> end up testing anyhow . . .
>>
>> In the case of #2, the experiment might cause a meltdown. So the  
>> "probe
>> for discovery" must be made completely safe. Also, we can  
>> eventually end
>> up with so many wierd hooks and idiosyncratic implementations that we
>> end up doing a BUNCH of probing before we do any real  
>> communicating. And
>> it's possible for the conditions to deteriorate so much that  
>> probes for
>> two different possible conditions are mutually exclusive -- if one
>> condition applies, the probe for another will cause a meltdown.
>>
>> So what happens if we start STUN-probing a proxy on its SIP port  
>> and it
>> doesn't support STUN? My guess is, it's random.
>>
>> ...
>>
>> One approach to "forward compatibility" design is to make sure  
>> that we
>> can use both foreknowledge and experimentation -- the "trust but  
>> verify"
>> model.
>>
>> ...
>>
>> While it's easy to decide whether a configured registrar URI has a
>> keepalive parameter, what happens to proxies we select based on
>> something besides direct configuration?
>>
>> Specifically,
>>
>> 1) Can the DHCP option for default proxy present the keepalive
>> parameter?"
>> ----
>>
>> During these three months after seeing those comments in the list
>> I have been expecting the outbound draft to provide answer to
>> the question how UA could (within SIP signalling exchange)
>> safely to check whether or not a proxy supports STUN keepalive.
>> Two mechanims have been suggested how the UA could do this in
>> a simple way:
>>
>> 1. The UA to check that the Path header in 200 OK returned for
>>    REGISTER would contain keepalive=stun parameter for the proxy URI
>>
>> 2. The UA to add Proxy-Reequire: stun-keepalive parameter into
>>    the REGISTER request and the proxy to remove this tag from
>>    the request before forwarding it towards the registrar
>>
>> For me, any of those mechanisms would sound feasible and for
>> safety reasons I would hope the draft to deny the UA for sending
>> STUN keepalives to the proxy which did not pass such a test.
>>
>>> the same problem exists for the "comp" parameter for example.
>>
>> Yes but please bear in mind that for comp=sigcomp URI parameter
>> RFC 3486 Compressing SIP gives several solutions how the UA
>> would be able to check if the proxy supports SigComp while
>> outbound does not define any other than relying on configuration.
>> A cite from RFC 3486:
>>
>> 4.1   Obtaining a SIP or SIPS URI with comp=sigcomp
>>
>>    For requests within a dialog, a next-hop URI with the comp=sigcomp
>>    parameter is obtained from a Record-Route header field when the
>>    dialog is established.  A client sending a request outside a  
>> dialog
>>    can also obtain SIP URIs with comp=sigcomp in a Contact header  
>> field
>>    in a 3xx or 485 response to the request.
>>
>>    However, clients establishing a session will not typically be  
>> willing
>>    to wait until the dialog is established in order to begin  
>> compressing
>>    messages.  One of the biggest gains that SigComp can bring to  
>> SIP is
>>    the ability to compress the initial INVITE of a dialog, when  
>> the user
>>    is waiting for the session to be established.  Therefore, clients
>>    need a means to obtain a comp=sigcomp URI from their outbound  
>> proxy
>>    before the user decides to establish a session.
>>
>>    One solution to this problem is manual configuration.  However,
>>    sometimes it is necessary to have clients configured in an  
>> automatic
>>    fashion.  Unfortunately, current mechanisms for SIP client
>>    configuration (e.g., using DHCP [6]) do not allow to provide the
>>    client with URI parameters.  In this case, the client SHOULD  
>> send an
>>    uncompressed OPTIONS request to its outbound proxy.  The outbound
>>    proxy can provide an alternative SIP URI with the comp=sigcomp
>>    parameter in a Contact header field in a 200 OK response to the
>>    OPTIONS.  The client can use this URI for subsequent requests that
>>    are sent through the same outbound proxy using compression.
>>
>>    RFC 3261 [1] does not define how a proxy should respond to an  
>> OPTIONS
>>    request addressed to itself.  It only describes how servers  
>> respond
>>    to OPTIONS addressed to a particular user.  Section 11.2 of RFC  
>> 3261
>>    says:
>>
>>       Contact header fields MAY be present in a 200 (OK) response and
>>       have the same semantics as in a 3xx response.  That is, they  
>> may
>>       list a set of alternative names and methods of reaching the  
>> user.
>>
>>    We extend this behavior to proxy servers responding to OPTIONS
>>    addressed to them.  They MAY list a set of alternative URIs to
>>    contact the proxy.
>>
>> What has been called for is that outbound draft would also
>> specify one solution how the UA could check the STUN keepalive
>> support of a proxy. In my opinion good proposals seem
>> to exist how to do it in a reasonable way.
>>
>> Regards,
>>
>> Erkki
>>
>>> -----Original Message-----
>>> From: ext Rohan Mahy [mailto:rohan@ekabal.com]
>>> Sent: 04.July.2006 03:10
>>> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>>> Cc: dean.willis@softarmor.com; fluffy@cisco.com; Rohan Mahy;
>>> sip@ietf.org; jbemmel@zonnet.nl
>>> Subject: Re: [Sip] Ready for WGLC on Outbound?
>>>
>>> Hi Erkki,
>>>
>>> The only time a UAC sends a STUN request as directed by
>>> outbound -04 is
>>> when the URI in the outbound-proxy-set includes the keepalive=stun
>>> parameter.  This is 100% backwards safe.  We don't need any text to
>>> prevent existing implementations from getting a bunch of STUN  
>>> requests.
>>>
>>> That DHCP does not provide URI is a limitation of the SIP DHCP
>>> configuration mechanism, not outbound.  The same problem
>>> exists for the
>>> "comp" parameter for example.  If you want to write a draft on  
>>> how to
>>> "shift-up" to using keepalives or using compression from some SIP or
>>> external stimulus, you can still do this.  No text in outbound
>>> prevents
>>> that from happening.
>>>
>>> thanks,
>>> -rohan
>>>
>>>
>>> On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
>>>>
>>>> Hi Rohan,
>>>>
>>>>>> Also, the issue is that what may be configured (no matter how you
>>>>>> configure it) may not be a URI. It may simply be an IP
>>> address/FQDN,
>>>>>> which is then used to build a SIP-URI. I think we shall
>>>>>> allow that, and make it more clear in the spec. [/CHH]
>>>>>
>>>>> the outbound spec defines a piece of configurable data (the
>>>>> 'outbound-proxy-set'), which is a list of URIs.  This list
>>> of URIs may
>>>>> never have a one-to-one correspondence to a user or administrator
>>>>> manually configurable field.  That piece of configurable data  
>>>>> may be
>>>>> derived from several fields, manually configurable, discovered via
>>>>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
>>>>> download, or via a random character generator seeded by the  
>>>>> phase of
>>>>> the moon.
>>>>>
>>>>> The outbound-proxy-set was defined as a list of URIs because  
>>>>> the UA
>>>>> acts on each item in the list, because each URI is expected to be
>>>>> places in a Route header, and because each URI is expected to
>>>>> typically have a keepalive URI parameter in it.
>>>>
>>>> Like mentioned in other recent emails the problem is that if the UA
>>>> derives the proxy URI(s) from DHCP SIP server option or e.g. the  
>>>> AOR
>>>> of the user, in those cases the derived URI will NOT contain the
>>>> keepalive URI parameter in it. DHCP just returns IP addresses or
>>>> domain names instead of URIs with parameters.
>>>>
>>>> Thus in those cases it would be critical for the UA to be able
>>>> to ensure in REGISTER - 200 OK sequence that the contacted proxy
>>>> supports Outbound and can handle STUN requests sent to its SIP
>>>> port (not to cause any damage to an uncompliant proxy). I fully
>>>> agree with this proposal from Jeroen:
>>>>
>>>> --
>>>> The text should then say "the UAC MUST wait for a 200 OK to  
>>>> REGISTER
>>>> before
>>>> it starts sending STUN packets. If the last Path header of the  
>>>> 200 OK
>>>> does
>>>> not contain keepalive=stun, the UAC MUST NOT use STUN
>>> towards this URI.
>>>> If
>>>> there are no Path headers and no "Supported: outbound", the UAC  
>>>> also
>>>> MUST
>>>> NOT use STUN towards the registrar.". It could still use TCP
>>> keepalive
>>>> if it
>>>> supports that
>>>> --
>>>>
>>>> Please excuse me if I am just repeating myself and what e.g. Jeroen
>>>> has said but I believe the importance of this issue justifies some
>>>> redundancy :-)
>>>>
>>>> Regards,
>>>>
>>>> Erkki
>>>
>>>
>
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From sip-bounces@ietf.org Wed Jul 05 01:52:58 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: [Sip] Relationship between GRUU and callee capabilities
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Errors-To: sip-bounces@ietf.org

This is a multi-part message in MIME format.

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In http://www.jdrosen.net/papers/draft-ietf-sip-gruu-09.txt section 6 =
figure 1, a GRUU is shown to be associated with 0..n Contacts. Does this =
include callee capabilities that were indicated in the registration of =
that contact? In other words, should the GRUU proxy be able to =
reconstruct those capabilities, e.g. for caller preference processing =
purposes?

Thanks,

Jeroen
------=_NextPart_000_0021_01C6A007.BB752670
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.2912" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>In <A=20
href=3D"http://www.jdrosen.net/papers/draft-ietf-sip-gruu-09.txt">http://=
www.jdrosen.net/papers/draft-ietf-sip-gruu-09.txt</A>&nbsp;section=20
6 figure 1, a GRUU is shown to be associated with 0..n Contacts. Does =
this=20
include callee capabilities that were indicated in the registration of =
that=20
contact? In other words, should the GRUU proxy be able to reconstruct =
those=20
capabilities, e.g. for caller preference processing =
purposes?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Jeroen</FONT></DIV></BODY></HTML>

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--===============1405987573==--





From sip-bounces@ietf.org Wed Jul 05 03:13:01 2006
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Subject: Re: [Sip] FW: I-D ACTION:draft-audet-sip-sips-guidelines-02.txt
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Francois,

Perhaps I've mentioned this before, but the registration of multiple 
Contacts (one per transport) to indicate which transports are supported is 
not workable: usage of different q-values does not stop a proxy from doing 
parallel forking. And there are other considerations, for example callee 
capabilities would have to be repeated for each contact, making the request 
bulky

In any case, it seems that transport support/selection is a different issue 
than sip-sips, and should be left out of scope.

For the three cases that you mention:

o  AOR is to be reachable only with secure transport
Contact: <sips:bob@bobspc.example.com>;schemes="sips", RFC3263 section 4.1 
says that TCP SHOULD be chosen as transport, and "sips" implies TLS. Since 
"sip" is not included, a downgrade is not allowed

o  AOR is to be reachable preferably with secure transport
Contact: <sips:bob@bobspc.example.com>;schemes="sips,sip"

o  AOR is to be reachable only without secure transport
Contact: <sip:bob@bobspc.example.com>;schemes="sip"

The definition of "implicit preferences" (RFC3841 section 7.2.2) might have 
to be extended to include scheme considerations for contact selection to 
work properly. Also, it should be defined that when explicit capabilities 
are provided, a proxy must not rewrite to a scheme that is not indicated.

Some nits:
- the URIs used in REGISTER throughout the draft should be 
"{sip/sips}:example.com", without 'transport' parameter and without 
"registrar."
- Contacts in REGISTER responses MUST reflect all Contact header parameters 
(including 'q')
- "If a UAS does not support SIPS, it MUST reject a request over TLS with 
response code 416" (section 6): Though not strictly necessary, since SIPS 
implies TLS one reason for "not support SIPS" here could be that TLS is not 
supported. In such a case, it will be impossible for the UAS to respond to 
an incoming request over TLS. Perhaps the domain proxy would be the element 
responding with 416 here?

Regards,

Jeroen

----- Original Message ----- 
From: "Francois Audet" <audet@nortel.com>
To: "IETF SIP List" <sip@ietf.org>
Sent: Wednesday, July 05, 2006 2:39 AM
Subject: [Sip] FW: I-D ACTION:draft-audet-sip-sips-guidelines-02.txt


Another update, including more feedback from the list.

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Tuesday, June 27, 2006 12:50
> To: i-d-announce@ietf.org
> Subject: I-D ACTION:draft-audet-sip-sips-guidelines-02.txt
>
>
> A New Internet-Draft is available from the on-line
> Internet-Drafts directories.
>
>
> Title : Guidelines for the use of the SIPS
> URI Scheme in the Session Initiation Protocol (SIP)
> Author(s) : F. Audet
> Filename : draft-audet-sip-sips-guidelines-02.txt
> Pages : 24
> Date : 2006-6-27
>
> This document provides clarifications and guidelines concerning the
>    use of SIPS URI Scheme in the Session Initiation Protocol (SIP).
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-audet-sip-sips-guide
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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
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"Samir Srivastava" <samirsr@nortel.com> writes:
> In this case, Alice and Atlanta will (most likely) negotiate
AES-SHA256.
> Atlanta will offer AES-SHA and AES-SHA256. Biloxi will select AES-SHA.
> The only way in which the second hop could fail would be if Atlanta=20
> didn't speak AES-SHA, but remember that AES-SHA is a MUST implement.
>
> SS>> Fine. It will work in the current state of affairs. Breaking of
> AES-SHA in future, will need another cipher-suite. Then we fix TLS=20
> with that (newer version etc). But SIP doesn't have any knowledge of=20
> how much secure hop it is.

Again, that's not true with the scheme I described. Only cipher suites
that the first-hop client is willing to negotiate will be negotiated on
any hop. Consider the simple case where there is currently only one
algorithm X. When a new algorithm Y is introduced (whatever the status
of X), if the original UAC only supports X, then there are two
possibilities:

1. Every proxy along the path supports X as well as Y.
2. Some proxy along the path only supports Y.
=20
Given the algorithm I described, in case 1 you get X (which by
definition is OK with the UAC). In case 2, the connection fails
somewhere along the line. In neither case can you get a cipher suite
that X does not consider acceptable.

SS>> Why you want to fail the call setup in 2. In my proposal, When we
have overriding defined, call will be setup, with the clear indication,
that it is less/more secure ( 0 being no secure, 1 being Max i.e. for
Algo Y in this case and their will be some value for X say 0.8 ). Making
more no of retries will not be acceptable by the users and not desired
by the providers. Your proposal will require more no of resubmission of
INVITEs with changing algorithms etc when there are large no of
algorithms.

Moreover, What about the case when UAC and Proxy at the first hop is
upgraded to Y and other hops for particular session are having still X.
In your scheme, secure call setup will fail and more no of retries etc
will be needed. In my proposal, call will be set, with the indication
that it is=20
less secure with a single INVITE transaction.

=20
> SS>> Being at UAC, I don't know in advance how I will be able to reach
> the UAS. Take the case of retargeting from the another domain's proxy=20
> to the UAC domain proxy. The proposal is cleaner w.r.t. all these
aspects.

I don't understand the point you're making here. I'm not talking about
the UAC, but about the various proxies inside a single trust domain.

SS>> My point is, UAC and it's home proxy doesn't know at prior, what
domains it will be visiting. The proxies visted may or may not be in a
single trust domain. When INVITE is issued, they may be in the single
trust domain, and retargeting makes it to go different trust domain.
Retargetor entity (Proxy) needs to know where it is going. The earlier
proxies in path will not have any idea. In the single trust domain, the
original call might be using algorithm X, but going out in a different
domain might need different algorithm Y. There will be cases when X is
more/less stronger than Y.
So again we are back to similar issues of upgrade/downgrade of SIP and
SIPS. My proposal provides clear guideline what proxy should do. Your
proposal will reject the calls based on relative stengths of X and Y and
what UAC has. And require UAC to reInitiate the calls when it is failed.

> Can you name an instance in which this has actually happened?
>
> But suppose it's true? You just relax the constraint that you need to=20
> offer only incoming cipher suites, upgrade your edge proxies, and the=20
> problem goes away. The reason not to do that ordinarily is that we=20
> don't have an unambigous ordering of cipher suite strengths so we'd=20
> like to let the original UA have an opinion but if you *know* that the

> UA's offered algorithm is broken, I think it's safe to override it.
>
> SS>> Refer the earlier text in the reply. Ordering of strength of
> cipher-suites etc and overriding is not considered currently.

I don't understand this point at all. What I'm saying is that the only
reason why you need to consult the UAC's cipher suite preferences at all
is that we don't have a single agreed upon ordering of algorithm
strengths. If we did, then it would be a simple matter of saying "on
outgoing you can negotiate any algorithm as strong or stronger as the
one on incoming." However, in the case where the proxy *knows* that the
incoming algorithm is broken, then it's safe to negotiate a stronger
algorithm even if the client did not offer it.

SS>> Refer the previous comments. In your proposal calls are failed. And
they need to retried with less security. All together new transactions
etc. are needed. In my proposal call is setup with the available level
of security in the network, if it is acceptable to UAC with a single
INVITE. Then user can accept/reject the calls. In your proposal *more
trust* is given to a single Proxy and UAC doesn't know what is being
done in the path. If a single evil proxy downgrades, it cannot be
indicated back to UAC, as you don't agree on the new header which
carries the information for least strong algorithm being used in path
when RESPONSE is sent back to UAC. My proposal can catch single bad
proxy if it lies. My proposal doesn't cover the case when two adjacent
proxies lie in collaboration.=20




> If we are
> in the agreement that we need a security-level kind of header in SIP,=20
> then we can hash it out.

I don't agree that we need such a header.

SS>> I don't agree with you here. Refer my concerns in above replies.
Let me divide this into two
    1) Header for REQUEST where ciphersuites etc are listed.
    2) Header for RESPONSE where least strong used algorithm in the path
is indicated back=20
       to UAC by UAS by inspecting via trails.

    Let me know your opinions for 1 and 2. Then we can focus on the
discussion for that case.

-Ekr

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From: Juha Heinanen <jh@tutpro.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Subject: Re: [Sip] Ready for WGLC on Outbound?
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Jeroen van Bemmel writes:

 > 2. Confirming that a given proxy in the set actually supports outbound (ie 
 > validating the keepalive=stun information)

i don't understand this one.  if a proxy is listed as an outbound proxy
(for example in the result returned from DNS), why can't we require that it
supports stun keepalives?

-- juha

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From sip-bounces@ietf.org Wed Jul 05 05:49:59 2006
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Subject: Re: [Sip] Ready for WGLC on Outbound?
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Juha,

Do you mean
A) you dont understand what I am trying to say
B) you dont see / agree that this is needed

Let me try to rephrase:

Currently deployed proxies generally do not support STUN, especially not on 
the same ports that they use for SIP. Sending a STUN packet to such a proxy 
could very well be harmless, but it could also cause it to crash.

Adhering to the principle "be strict in what you send", it is the UAC's 
responsibility to make reasonably sure that a proxy it is about to STUN can 
take it. "keepalive=stun" in the URI it received through configuration is 
likely a good indicator, depending on which provisioning mechanism is used. 
But *independent* of the provisioning mechanism, a confirmation from the 
proxy itself is better.

Sure, we can state "any proxy listed as outbound proxy MUST support STUN". 
We might even say "all proxies SHOULD support STUN". But no matter what 
requirements we come up with, they won't be met by currently deployed 
proxies.

Summarizing:
1 - there is an implicit requirement "MUST NOT break current deployments". 
For outbound, this includes "proxies that do not support STUN MUST NOT 
receive STUN packets" (could consider to include something like this as a 
9th requirement in section 12 of 
http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-04.txt)
2 - since we cannot influence all receivers, this translates into a 
requirement on the sender of STUN packets (the UAC): MUST NOT send STUN 
unless supported by target proxy
3 - to meet this requirement, we currently have "MUST check provisioned URI 
has keepalive=stun". It should say explicitly "if not, then MUST NOT send 
STUN to that URI"
4 - I am argueing: "MUST check provisioned keepalive=stun" is not 
sufficient, because the configuration may be incorrect (for various reasons, 
also depending on what you use for provisioning). As an additional check, we 
need "MUST check confirmation from proxy"

We cannot depend only on the configuration mechanism (which is still TBD, 
and in principle open ended)

Regards,

Jeroen

----- Original Message ----- 
From: "Juha Heinanen" <jh@tutpro.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Cc: "Cullen Jennings" <fluffy@cisco.com>; <Erkki.Koivusalo@nokia.com>; 
<sip@ietf.org>; <rohan@ekabal.com>; <dean.willis@softarmor.com>
Sent: Wednesday, July 05, 2006 9:05 AM
Subject: Re: [Sip] Ready for WGLC on Outbound?


> Jeroen van Bemmel writes:
>
> > 2. Confirming that a given proxy in the set actually supports outbound 
> > (ie
> > validating the keepalive=stun information)
>
> i don't understand this one.  if a proxy is listed as an outbound proxy
> (for example in the result returned from DNS), why can't we require that 
> it
> supports stun keepalives?
>
> -- juha 


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From sip-bounces@ietf.org Wed Jul 05 05:53:25 2006
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Subject: Re: [Sip] Ready for WGLC on Outbound?
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Jeroen van Bemmel writes:

 > Currently deployed proxies generally do not support STUN, especially not on 
 > the same ports that they use for SIP. Sending a STUN packet to such a proxy 
 > could very well be harmless, but it could also cause it to crash.

sure, but those proxies are not outbound proxys and should not be listed
as such.  problem solved.

 > Sure, we can state "any proxy listed as outbound proxy MUST support STUN". 
 > We might even say "all proxies SHOULD support STUN". But no matter what 
 > requirements we come up with, they won't be met by currently deployed 
 > proxies.

if your currently deployed proxy is not outbound proxy as specified in
the i-d, you don't list it as such.  it is a regular proxy and can do
whatever is specified in rfc3261.

-- juha

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From sip-bounces@ietf.org Wed Jul 05 06:34:11 2006
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From: "Rajnish Jain" <rjain@precisionscientific.org>
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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
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> It seems to me that there is a simpler mechanism: any proxy X 
> can only offer the intersection of the cipher suites that was 
> offered to it and the ciphersuites that it is willing to 
> support. 

There seems to be a fundamental disconnect in the comment above. The comment
above is basically describing a hop-by-hop mechanism. However, the draft is
fundamentally about solving the e2e problem. The keyword here is e2e. How
will the proxy know that its narrowing down the UAC's preferred ciphersuites
will not be detrimental to the setting up of the call somewhere downstream
in the e2e path? 


> This has the result that no downstream proxy can 
> negotiate a cipher suite that Alice wasn't willing to 
> negotiate with Atlanta. 

But this has also has the undesirable result that no downstream proxy can
negotiate a cipher suite that was agreeable to Alice.


> 1. Alice doesn't get any signal of what was eventually negotiated.
>    I would solve this problem by extending Via (I'm actually surprised
>    this isn't done). MTAs typically put it in Received lines, FWIW...

How will extending 'Via' help Alice, the UAC, learn about the negotiated
ciphersuite? The draft does talk about extending 'Via' but that is to inform
Bob, the UAS. The draft also mentions that the Security-Level header
containing the lowest 'q' value is sent back to Alice. 


> 2. It's less expressive. In particular, the downstream cipher suites
>    always narrow and never grow. So, if there were a third proxy in
>    between Atlanta and Biloxi that only spoke DES, then Biloxi and Bob
>    would only get DES even if they both spoke AES. 

This comment doesn't seem to reflect the scheme described in the draft.
Instead, this seems like a comment reflecting the hop-by-hop mechanism
suggested in your own email. 


Rajnish


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Subject: [Sip] A coment to draft-ietf-sip-outbound-04
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Reading Section 4.2 of Outbound-04, there is a sentence that came to my 
attention.

    For each outbound proxy URI in the set, the UA MUST send a
    REGISTER in the normal way using this URI as the default outbound
    proxy.

I think it is not appropriate to use a MUST strength in this case. I 
suspect that the spirit of the draft is to offer high availability by 
making the UA to establish one flow per URI, which is a good spirit. 
However, one has to evaluate the cost of establishing one flow per URI, 
which might be quite high in terms of battery consumption, for example 
(due to STUN refreshes).

I also believe the MUST is misused here because it is not an absolute 
requirement for the specification. In other words, the SIP outbound 
procedures work OK even if the UA establishes just one flow (even when 
it is provisioned with several outbound URIs). Nothing is broken.

So, we should be looking at adding a paragraph that describes the 
benefit of establishing one flow per outbound URI, but letting up to the 
UA's policy to decide on the absolute number of established flows.

I would suggest to replace the above mentioned sentence with something 
around these lines:

    For each outbound proxy URI in the set, the UA MAY send a
    REGISTER in the normal way using this URI as the default outbound
    proxy. Sending one REGISTER per outbound proxy URI decreases the
    likelihood of interrupted service due to dropped connections at the
    cost of maintaining a living flow which each outbound proxy server.

Comments?

/Miguel

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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From: Juha Heinanen <jh@tutpro.com>
To: Miguel Garcia <Miguel.An.Garcia@nokia.com>
Subject: [Sip] A coment to draft-ietf-sip-outbound-04
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Miguel Garcia writes:

 > I also believe the MUST is misused here because it is not an absolute 
 > requirement for the specification. In other words, the SIP outbound 
 > procedures work OK even if the UA establishes just one flow (even when 
 > it is provisioned with several outbound URIs). Nothing is broken.

reliability may get broken.  wasn't this issue discussed already
earlier?  i remember that a conclusion was reached something along the
lines that the default is to establish at least two flows unless
instructed otherwise by the domain?

-- juha

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Juha Heinanen wrote:
> Miguel Garcia writes:
> 
>  > I also believe the MUST is misused here because it is not an absolute 
>  > requirement for the specification. In other words, the SIP outbound 
>  > procedures work OK even if the UA establishes just one flow (even when 
>  > it is provisioned with several outbound URIs). Nothing is broken.
> 
> reliability may get broken.  wasn't this issue discussed already
> earlier?  i remember that a conclusion was reached something along the
> lines that the default is to establish at least two flows unless
> instructed otherwise by the domain?
> 

That is what I remember as well, but I don't recall any conclusion of 
"unless instructed otherwise by the domain". In otherwise, there is 
still some "magic" for the domain instructions.

But coming back to my point... I don't understand why there is a MUST 
here, because the protocol is not broken if the MUST is not followed.

/Miguel

> -- juha

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Subject: RE: [Sip] Ready for WGLC on Outbound?
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Thread-Topic: [Sip] Ready for WGLC on Outbound?
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=20
Hi Cullen, Rohan and Dean,

>I think people have stopped posting on this because they can't say =20
>anything that they have not already said. Don't take that as=20
>they agree.

This is the last email which I intend to post on this subject
(and hopefully I can keep that intention). The reason to keep on
posting is that I am unable to attend the IETF meeting in Montreal,
so I can only raise issues by email prior to the meeting and ask
those issues to be discussed as open questions in the WG meeting.
If this happens and the WG is able to reach a consensus in the meeting
as recorded to the minutes, that will be time for me to eventually=20
shut up and outbound draft to move on for the WGLC.

>you are proposing that we change in outbound - I think what you are =20
>proposing is we need to hold up outbound until we choose which one of =20
>the 10 we will use and furthermore you feel very strongly on which =20
>one of the 10 we should use.

Please do not understand me incorrectly. My intention is not to block or
delay the approval of outbound draft, I also feel that it is important
to get it approved as soon as possible, to be able to cope with the
NAT/FW traversal problems better than now. Further on I do not try
to push the outbound draft to contain a solution for outbound discovery,
but a safe probe that a UA could do in order to check that sending STUN
packets will do no harm for the proxy. I always believed that this was
a requirement for Outbound-04, based on the comments I saw on the list=20
prior to and right after the Dallas meeting (revolving around the=20
issue whether to use STUN or SIP PING as the keepalive mechanism).=20
Finding that Outbound-04 did not provide a mechanism, I tried to=20
raise the issue as soon as I could, to get it closed asap, believing
that it would be sufficient that the authors to select the mechanism
amongst the options available and add 1-2 related paragraphs to=20
Outbound-05 before the WGLC. I still hope it could be done like
that and just try to be helpful to avoid one pitfall with the draft.

>The WG decided on this separation =20
>of problems awhile back - I think it was in Vancouver - and many =20
>people supported it. I think it was a good idea  - we have shown we =20
>can move outbound forward before we have determined all the ways that =20
>this configuration information is delivered to the UA. I would be =20
>very hesitant to reverse this decision now.

Even if I have been working on SIP for some years, I only joined
the WG (including sip@ietf.org follow-up) in the beginning of this
year, so I have not been aware of what was discussed and decided
earlier. So thanks for being updated. However I would tend to agree
with how Jeroen put it in his email (it is the issue 2 which I also
tried to highlight):

>The point is that these are really 2 separate issues:
>1. Discovering the set of outbound proxy URIs (with keepalive=3Dstun=20
>parameters in them)
>2. Confirming that a given proxy in the set actually supports=20
>outbound (ie validating the keepalive=3Dstun information)
>
>Of these 2 options, (1) is indeed a separable problem.=20
>(2) - to me - is not, for me outbound is not acceptable=20
>without something to address (2).

The specific concerns that I have (and would like the WG to=20
consider in the meeting) are closely related to what you told:

>We got very strong feed back on one point - don't send=20
>STUN messages to a proxy unless you know it supports them.=20

1. This is what Outbound-04 says on the subject:

   User Agents that form flows MUST check if the configured URI they are
   connecting to has a 'keepalive' URI parameter (defined in Section 10)
   with the value of 'stun'.  If the parameter is present, the UA needs
   to periodically perform keepalive checks by sending a STUN [I-D.ietf-
   behave-rfc3489bis] Binding Requests over the flow.

Does the WG consider checking the configured URI as a safe enough
check for the UA to ensure that the SIP proxy support STUN ?

In my opinion it is not. As an example a vendor X could implement
a compliant but bad UA which provides an option for the user to=20
configure the settings via an User Interface e.g. like this:
"Is the phone behind a NAT: Yes/No". When the user selects
"Yes" the UA would add the keepalive=3Dstun flag to the proxy URI
even if the user does not have a clue if the proxy actually
supports outbound or not. There is nothing in outbound draft
to prevent such unsafe implementations to appear.

2. draft-koivusalo-sip-outbound-discovery identified a security
threat like this:

   DNS NAPTR records are used to allow a client to discover that the
   server supports SIP Outbound.  An attacker could potentially modify
   these records, resulting the client to send STUN request to a proxy
   that does not support STUN requests in its SIP port.  This kind of
   attack could cause the proxy to behave in an unpredictable way and
   might cause it to crash.

   This kind of attack is prevented by a mechansim introduced within
   draft-ietf-sip-outbound [8].  When sending REGISTER request, the UA
   has to add an option tag to it, requiring the proxy to reject the
   request if it does not support SIP Outbound.

   [[Author's note: the mechanism to be used within REGISTER request is
   still to be decided.  After the decision is made the text above must
   be rewritten.]]

Without such a mechanism in outbound draft there is nothing to do
to migitate the identified threat.

3. There is a separate draft being written about how the UA could
automatically discover if a proxy supports outbound or not.
Someone could claim that using that draft would make up the "safe
check". However I do not think it would be wise the outbound draft
itself to rely on any such additional draft to be ever approved
or used. Outbound draft should be sufficiently self-contained so
that applying it would not requrire the UA or proxy vendor to=20
implement some related draft (which might be available much
later if ever) to do some mandatory things.


I hope I was able formulate those concerns clearly enough so that
the WG members can draw their opinions on those and make a=20
judgement whether or not this needs to be addressed in the=20
outbound draft before WGLC.

It is in your hands now ;-)

Regards,

Erkki

>-----Original Message-----
>From: ext Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]=20
>Sent: 05.July.2006 00:35
>To: Cullen Jennings; Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>Cc: rohan@ekabal.com; dean.willis@softarmor.com; sip@ietf.org
>Subject: Re: [Sip] Ready for WGLC on Outbound?
>
>Cullen,
>
>The point is that these are really 2 separate issues:
>1. Discovering the set of outbound proxy URIs (with keepalive=3Dstun=20
>parameters in them)
>2. Confirming that a given proxy in the set actually supports=20
>outbound (ie=20
>validating the keepalive=3Dstun information)
>
>Of these 2 options, (1) is indeed a separable problem. (2) -=20
>to me - is not,=20
>for me outbound is not acceptable without something to address (2). It=20
>really should say "the UAC MUST NOT use STUN until { mechanism=20
>for (2) says=20
>its supported }"
>
>There have been several options mentioned to do this, a simple=20
>way is to=20
>have the edge proxy insert a flag. Question is: why not?
>
>Regards,
>
>Jeroen
>
>----- Original Message -----=20
>From: "Cullen Jennings" <fluffy@cisco.com>
>To: <Erkki.Koivusalo@nokia.com>
>Cc: <rohan@ekabal.com>; <dean.willis@softarmor.com>; <sip@ietf.org>;=20
><jbemmel@zonnet.nl>
>Sent: Tuesday, July 04, 2006 10:37 PM
>Subject: Re: [Sip] Ready for WGLC on Outbound?
>
>
>>
>> I think people have stopped posting on this because they can't say=20
>> anything that they have not already said. Don't take that as=20
>they agree.
>>
>> We got very strong feed back on one point - don't send STUN=20
>messages  to a=20
>> proxy unless you know it supports them. People felt strongly=20
> about that -=20
>> I don't know why - but there was lots of people that  agreed=20
>with that.
>>
>> I imagine people have sent me around 10 options on how to=20
>get the  stun=20
>> supported information and the set of outbound proxy URIs - =20
>certainly DNS=20
>> and DHCP are some of them. We can write some drafts on =20
>various ways to do=20
>> it. Right now my concern is making sure that no  matter=20
>which of these=20
>> many ways the WG decided to move forward with,  outbound=20
>will work. So far=20
>> every one of the possibilities I have  heard would be=20
>possible with the=20
>> current outbound. Given that, I'm  not sure what we need to=20
>change in=20
>> outbound. I'm not even sure what  you are proposing that we=20
>change in=20
>> outbound - I think what you are  proposing is we need to=20
>hold up outbound=20
>> until we choose which one of  the 10 we will use and=20
>furthermore you feel=20
>> very strongly on which  one of the 10 we should use. I'm not keen on=20
>> this - I think we have  shown they are separable problems.=20
>The WG decided=20
>> on this separation  of problems awhile back - I think it was in=20
>> Vancouver - and many  people supported it. I think it was a=20
>good idea  -=20
>> we have shown we  can move outbound forward before we have=20
>determined all=20
>> the ways that  this configuration information is delivered=20
>to the UA. I=20
>> would be  very hesitant to reverse this decision now.
>>
>> I don't think Dean's posts were sent as decries of the=20
>requirements  from=20
>> the chair but purely as his opinion in trying to help bring the=20
>> discussion to consensus. But I do think the draft meets the =20
>requirements=20
>> and ideas he was outlining - I think the current draft  does=20
>meet the=20
>> requirements he outlines. It does have a way for the =20
>"discovering whether=20
>> a proxy supports the extensions of outbound" and  on the=20
>topic of "there=20
>> are two ways we can establish compatibility  with an=20
>extensions", the WG=20
>> rejected option 2 and we have gone with 1.
>>
>> Cullen (with my outbound author hat on)
>>
>>
>> On Jul 4, 2006, at 3:50 AM, <Erkki.Koivusalo@nokia.com>=20
>> <Erkki.Koivusalo@nokia.com> wrote:
>>
>>>
>>> Hi Rohan,
>>>
>>> Sorry to be persistent with this issue, but what you say
>>> seems to conflict with what was discussed in March.
>>>
>>>> The only time a UAC sends a STUN request as directed by
>>>> outbound -04 is when the URI in the outbound-proxy-set
>>>> includes the keepalive=3Dstun parameter. That DHCP does not
>>>> provide URI is a limitation of the SIP DHCP configuration
>>>> mechanism, not outbound. If you want to write a draft on
>>>> how to "shift-up" to using keepalives from some SIP or
>>>> external stimulus, you can still do this.
>>>
>>> What you say seems to suggest that you agree there is a problem
>>> when using DHCP SIP server option (as it is currently defined)
>>> for finding out if the discovered proxy supports outbound and
>>> STUN keepalives, but the solution is not in scope of Outbound
>>> draft itself.
>>>
>>> However I came to a different understanding based on the emails
>>> seen in the list earlier. For instance this email from Dean
>>> Willis (SIP WG chair) seems to suggest that the problem should
>>> be in scope of the outbound draft:
>>> http://www1.ietf.org/mail-archive/web/sip/current/msg13667.html
>>>
>>> "Ok, I think we've boiled this down to a new requirement:
>>>
>>> for the "outbound" technique to work, we MUST have a means of
>>> discovering whether a proxy supports the extensions of outbound.
>>>
>>> While we could do that by doing a REGISTER exchange that includes
>>> an appropriate option-tag in Require:, it would be more efficient
>>> if the proxy discovery mechanism informed us of this detail. So,
>>> we posit that the Outbound draft SHOULD select and discuss such a
>>> mechanism."
>>>
>>> And in yet another email
>>> http://www1.ietf.org/mail-archive/web/sip/current/msg13692.html
>>> Dean tells this:
>>>
>>> "There are two ways we can establish compatibility with an=20
>extensions:
>>>
>>> 1) By foreknowledge
>>> 2) By experimentation -- try it and see if it works.
>>>
>>> Both have limitations:
>>>
>>> In the case of #1, we might have incorrect knowledge. So=20
>we'll  probably
>>> end up testing anyhow . . .
>>>
>>> In the case of #2, the experiment might cause a meltdown.=20
>So the  "probe
>>> for discovery" must be made completely safe. Also, we can =20
>eventually end
>>> up with so many wierd hooks and idiosyncratic=20
>implementations that we
>>> end up doing a BUNCH of probing before we do any real =20
>communicating. And
>>> it's possible for the conditions to deteriorate so much=20
>that probes  for
>>> two different possible conditions are mutually exclusive -- if one
>>> condition applies, the probe for another will cause a meltdown.
>>>
>>> So what happens if we start STUN-probing a proxy on its SIP=20
>port  and it
>>> doesn't support STUN? My guess is, it's random.
>>>
>>> ...
>>>
>>> One approach to "forward compatibility" design is to make=20
>sure that we
>>> can use both foreknowledge and experimentation -- the=20
>"trust but  verify"
>>> model.
>>>
>>> ...
>>>
>>> While it's easy to decide whether a configured registrar URI has a
>>> keepalive parameter, what happens to proxies we select based on
>>> something besides direct configuration?
>>>
>>> Specifically,
>>>
>>> 1) Can the DHCP option for default proxy present the keepalive
>>> parameter?"
>>> ----
>>>
>>> During these three months after seeing those comments in the list
>>> I have been expecting the outbound draft to provide answer to
>>> the question how UA could (within SIP signalling exchange)
>>> safely to check whether or not a proxy supports STUN keepalive.
>>> Two mechanims have been suggested how the UA could do this in
>>> a simple way:
>>>
>>> 1. The UA to check that the Path header in 200 OK returned for
>>>    REGISTER would contain keepalive=3Dstun parameter for the proxy =
URI
>>>
>>> 2. The UA to add Proxy-Reequire: stun-keepalive parameter into
>>>    the REGISTER request and the proxy to remove this tag from
>>>    the request before forwarding it towards the registrar
>>>
>>> For me, any of those mechanisms would sound feasible and for
>>> safety reasons I would hope the draft to deny the UA for sending
>>> STUN keepalives to the proxy which did not pass such a test.
>>>
>>>> the same problem exists for the "comp" parameter for example.
>>>
>>> Yes but please bear in mind that for comp=3Dsigcomp URI parameter
>>> RFC 3486 Compressing SIP gives several solutions how the UA
>>> would be able to check if the proxy supports SigComp while
>>> outbound does not define any other than relying on configuration.
>>> A cite from RFC 3486:
>>>
>>> 4.1   Obtaining a SIP or SIPS URI with comp=3Dsigcomp
>>>
>>>    For requests within a dialog, a next-hop URI with the=20
>comp=3Dsigcomp
>>>    parameter is obtained from a Record-Route header field when the
>>>    dialog is established.  A client sending a request=20
>outside a dialog
>>>    can also obtain SIP URIs with comp=3Dsigcomp in a Contact=20
>header  field
>>>    in a 3xx or 485 response to the request.
>>>
>>>    However, clients establishing a session will not=20
>typically be  willing
>>>    to wait until the dialog is established in order to=20
>begin  compressing
>>>    messages.  One of the biggest gains that SigComp can=20
>bring to  SIP is
>>>    the ability to compress the initial INVITE of a dialog,=20
>when the  user
>>>    is waiting for the session to be established.  Therefore, clients
>>>    need a means to obtain a comp=3Dsigcomp URI from their=20
>outbound proxy
>>>    before the user decides to establish a session.
>>>
>>>    One solution to this problem is manual configuration.  However,
>>>    sometimes it is necessary to have clients configured in=20
>an  automatic
>>>    fashion.  Unfortunately, current mechanisms for SIP client
>>>    configuration (e.g., using DHCP [6]) do not allow to provide the
>>>    client with URI parameters.  In this case, the client=20
>SHOULD  send an
>>>    uncompressed OPTIONS request to its outbound proxy.  The outbound
>>>    proxy can provide an alternative SIP URI with the comp=3Dsigcomp
>>>    parameter in a Contact header field in a 200 OK response to the
>>>    OPTIONS.  The client can use this URI for subsequent=20
>requests that
>>>    are sent through the same outbound proxy using compression.
>>>
>>>    RFC 3261 [1] does not define how a proxy should respond=20
>to an  OPTIONS
>>>    request addressed to itself.  It only describes how=20
>servers respond
>>>    to OPTIONS addressed to a particular user.  Section 11.2=20
>of RFC  3261
>>>    says:
>>>
>>>       Contact header fields MAY be present in a 200 (OK)=20
>response and
>>>       have the same semantics as in a 3xx response.  That=20
>is, they may
>>>       list a set of alternative names and methods of=20
>reaching the  user.
>>>
>>>    We extend this behavior to proxy servers responding to OPTIONS
>>>    addressed to them.  They MAY list a set of alternative URIs to
>>>    contact the proxy.
>>>
>>> What has been called for is that outbound draft would also
>>> specify one solution how the UA could check the STUN keepalive
>>> support of a proxy. In my opinion good proposals seem
>>> to exist how to do it in a reasonable way.
>>>
>>> Regards,
>>>
>>> Erkki
>>>
>>>> -----Original Message-----
>>>> From: ext Rohan Mahy [mailto:rohan@ekabal.com]
>>>> Sent: 04.July.2006 03:10
>>>> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>>>> Cc: dean.willis@softarmor.com; fluffy@cisco.com; Rohan Mahy;
>>>> sip@ietf.org; jbemmel@zonnet.nl
>>>> Subject: Re: [Sip] Ready for WGLC on Outbound?
>>>>
>>>> Hi Erkki,
>>>>
>>>> The only time a UAC sends a STUN request as directed by
>>>> outbound -04 is
>>>> when the URI in the outbound-proxy-set includes the =
keepalive=3Dstun
>>>> parameter.  This is 100% backwards safe.  We don't need any text to
>>>> prevent existing implementations from getting a bunch of=20
>STUN  requests.
>>>>
>>>> That DHCP does not provide URI is a limitation of the SIP DHCP
>>>> configuration mechanism, not outbound.  The same problem
>>>> exists for the
>>>> "comp" parameter for example.  If you want to write a=20
>draft on how to
>>>> "shift-up" to using keepalives or using compression from=20
>some SIP or
>>>> external stimulus, you can still do this.  No text in outbound
>>>> prevents
>>>> that from happening.
>>>>
>>>> thanks,
>>>> -rohan
>>>>
>>>>
>>>> On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
>>>>>
>>>>> Hi Rohan,
>>>>>
>>>>>>> Also, the issue is that what may be configured (no=20
>matter how you
>>>>>>> configure it) may not be a URI. It may simply be an IP
>>>> address/FQDN,
>>>>>>> which is then used to build a SIP-URI. I think we shall
>>>>>>> allow that, and make it more clear in the spec. [/CHH]
>>>>>>
>>>>>> the outbound spec defines a piece of configurable data (the
>>>>>> 'outbound-proxy-set'), which is a list of URIs.  This list
>>>> of URIs may
>>>>>> never have a one-to-one correspondence to a user or administrator
>>>>>> manually configurable field.  That piece of configurable=20
>data  may be
>>>>>> derived from several fields, manually configurable,=20
>discovered via
>>>>>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
>>>>>> download, or via a random character generator seeded by=20
>the  phase of
>>>>>> the moon.
>>>>>>
>>>>>> The outbound-proxy-set was defined as a list of URIs=20
>because the UA
>>>>>> acts on each item in the list, because each URI is expected to be
>>>>>> places in a Route header, and because each URI is expected to
>>>>>> typically have a keepalive URI parameter in it.
>>>>>
>>>>> Like mentioned in other recent emails the problem is that=20
>if the UA
>>>>> derives the proxy URI(s) from DHCP SIP server option or=20
>e.g. the AOR
>>>>> of the user, in those cases the derived URI will NOT contain the
>>>>> keepalive URI parameter in it. DHCP just returns IP addresses or
>>>>> domain names instead of URIs with parameters.
>>>>>
>>>>> Thus in those cases it would be critical for the UA to be able
>>>>> to ensure in REGISTER - 200 OK sequence that the contacted proxy
>>>>> supports Outbound and can handle STUN requests sent to its SIP
>>>>> port (not to cause any damage to an uncompliant proxy). I fully
>>>>> agree with this proposal from Jeroen:
>>>>>
>>>>> --
>>>>> The text should then say "the UAC MUST wait for a 200 OK=20
>to REGISTER
>>>>> before
>>>>> it starts sending STUN packets. If the last Path header=20
>of the  200 OK
>>>>> does
>>>>> not contain keepalive=3Dstun, the UAC MUST NOT use STUN
>>>> towards this URI.
>>>>> If
>>>>> there are no Path headers and no "Supported: outbound",=20
>the UAC also
>>>>> MUST
>>>>> NOT use STUN towards the registrar.". It could still use TCP
>>>> keepalive
>>>>> if it
>>>>> supports that
>>>>> --
>>>>>
>>>>> Please excuse me if I am just repeating myself and what=20
>e.g. Jeroen
>>>>> has said but I believe the importance of this issue justifies some
>>>>> redundancy :-)
>>>>>
>>>>> Regards,
>>>>>
>>>>> Erkki
>>>>
>>>>=20
>
>

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From sip-bounces@ietf.org Wed Jul 05 08:14:36 2006
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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
Date: Wed, 5 Jul 2006 14:12:42 +0200
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I agree with Miguel here

Outbound can be used for (at least) two things:
1) firewall/NAT traversal (especially for inbound requests)
2) high(er) reliability for registrations, by using multiple

These are separate use cases, it should be possible to do (1), (2) or both 
together (for example: I might be in a situation where there is no NAT, but 
I'd still like to use 2 registrations with different reg-ids). A MUST which 
forces an implementation to always do (2) makes it impossible to do only 
(1).

The abstract only talks about (1). The introduction talks mainly about (1), 
but mentions (2) in relation to flow failure (basically (2) is needed to 
overcome the window of max. 29 (UDP) or 120 (TCP) seconds it may take to 
detect a failed flow)

The benefit of (2) is highly dependent on the deployment environment and on 
the type of failures that occur. A UAC may set up 2 flows, but if both go 
through the same NAT device and this NAT device reboots, there may only be a 
benefit (in terms of period for which the UAC is not reachable) if the 
keep-alives for both flows are not too synchronized. It does help to 
maintain reachability in case one of the registrars (or edge proxies) 
reboots and looses its state

What I miss in the draft is discussion on the above aspects (i.e. 
synchronization of flow keep-alive timers, types of failures). A minimum of 
2 flows should IMO be RECOMMENDED (with explanation that this protects 
against single registrar / edge proxy failure, and reduces non-reachability 
to at most 15 (UDP) resp. 60 (TCP) seconds in case of NAT reboots, using the 
recommended timer values and provided the keep-alive timers are synchronized 
on half periods )

Note that for deployments where registrar failure is rare, one might decide 
to use only a single flow with half the keep-alive period. So if we go with 
the 'domain suggests number of flows", we should also include "domain 
suggests keep-alive timer periods"

Regards,

Jeroen

----- Original Message ----- 
From: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>
To: "Juha Heinanen" <jh@tutpro.com>
Cc: <sip@ietf.org>; "Cullen Jennings" <fluffy@cisco.com>; "Rohan Mahy" 
<rohan@ekabal.com>
Sent: Wednesday, July 05, 2006 1:29 PM
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04


> Juha Heinanen wrote:
>> Miguel Garcia writes:
>>
>>  > I also believe the MUST is misused here because it is not an absolute 
>>  > requirement for the specification. In other words, the SIP outbound > 
>> procedures work OK even if the UA establishes just one flow (even when > 
>> it is provisioned with several outbound URIs). Nothing is broken.
>>
>> reliability may get broken.  wasn't this issue discussed already
>> earlier?  i remember that a conclusion was reached something along the
>> lines that the default is to establish at least two flows unless
>> instructed otherwise by the domain?
>>
>
> That is what I remember as well, but I don't recall any conclusion of 
> "unless instructed otherwise by the domain". In otherwise, there is still 
> some "magic" for the domain instructions.
>
> But coming back to my point... I don't understand why there is a MUST 
> here, because the protocol is not broken if the MUST is not followed.
>
> /Miguel
>
>> -- juha
>
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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> Use sipping@ietf.org for new developments on the application of sip 


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From sip-bounces@ietf.org Wed Jul 05 09:38:58 2006
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Subject: Re: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
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Editorial note:

Samir, can I ask you to use standard ">" quoting markers?
I'm finding this exchange very hard to read with the quote
markers you'r;e using.
 
"Samir Srivastava" <samirsr@nortel.com> writes:

> "Samir Srivastava" <samirsr@nortel.com> writes:
>> In this case, Alice and Atlanta will (most likely) negotiate
> AES-SHA256.
>> Atlanta will offer AES-SHA and AES-SHA256. Biloxi will select AES-SHA.
>> The only way in which the second hop could fail would be if Atlanta 
>> didn't speak AES-SHA, but remember that AES-SHA is a MUST implement.
>>
>> SS>> Fine. It will work in the current state of affairs. Breaking of
>> AES-SHA in future, will need another cipher-suite. Then we fix TLS 
>> with that (newer version etc). But SIP doesn't have any knowledge of 
>> how much secure hop it is.
>
> Again, that's not true with the scheme I described. Only cipher suites
> that the first-hop client is willing to negotiate will be negotiated on
> any hop. Consider the simple case where there is currently only one
> algorithm X. When a new algorithm Y is introduced (whatever the status
> of X), if the original UAC only supports X, then there are two
> possibilities:
>
> 1. Every proxy along the path supports X as well as Y.
> 2. Some proxy along the path only supports Y.
>  
> Given the algorithm I described, in case 1 you get X (which by
> definition is OK with the UAC). In case 2, the connection fails
> somewhere along the line. In neither case can you get a cipher suite
> that X does not consider acceptable.
>
> SS>> Why you want to fail the call setup in 2.

I don't want it to, particularly. But I'm pointing out the scheme I
described fails safe. You may not have "any knowledge of how secure
it is" but you never end up sending data over a channel you
wouldn't be willing to.

> In my proposal, When we
> have overriding defined, call will be setup, with the clear indication,
> that it is less/more secure ( 0 being no secure, 1 being Max i.e. for
> Algo Y in this case and their will be some value for X say 0.8 ). Making
> more no of retries will not be acceptable by the users and not desired
> by the providers. Your proposal will require more no of resubmission of
> INVITEs with changing algorithms etc when there are large no of
> algorithms.

No, I don't believe it will. Please show the flow diagrams that
you think demonstrate this.


> Moreover, What about the case when UAC and Proxy at the first hop is
> upgraded to Y and other hops for particular session are having still X.
> In your scheme, secure call setup will fail and more no of retries etc
> will be needed. In my proposal, call will be set, with the indication
> that it is 
> less secure with a single INVITE transaction.

No, that's not correct either, as I showed in an earlier diagram,
of exactly this case, repeated here.

Alice        Atlanta         Biloxi
-----        -------         ------
AES-SHA      AES-SHA         3DES-SHA
AES-SHA256   AES-SHA256      AES-SHA

In this case, Alice and Atlanta will (most likely) negotiate
AES-SHA256.  Atlanta will offer AES-SHA and AES-SHA256. Biloxi will
select AES-SHA. 

Now, it's certainly true that if Alice configures her client to
be AES-SHA256 only then she won't be able to get a call through,
but that's a feature because it's a deliberate advertisement
that she's unwilling to do other algorithms. 


>> SS>> Being at UAC, I don't know in advance how I will be able to reach
>> the UAS. Take the case of retargeting from the another domain's proxy 
>> to the UAC domain proxy. The proposal is cleaner w.r.t. all these
> aspects.
>
> I don't understand the point you're making here. I'm not talking about
> the UAC, but about the various proxies inside a single trust domain.
>
> SS>> My point is, UAC and it's home proxy doesn't know at prior, what
> domains it will be visiting. The proxies visted may or may not be in a
> single trust domain. When INVITE is issued, they may be in the single
> trust domain, and retargeting makes it to go different trust domain.
> Retargetor entity (Proxy) needs to know where it is going. The earlier
> proxies in path will not have any idea. In the single trust domain, the
> original call might be using algorithm X, but going out in a different
> domain might need different algorithm Y. There will be cases when X is
> more/less stronger than Y.
> So again we are back to similar issues of upgrade/downgrade of SIP and
> SIPS. My proposal provides clear guideline what proxy should do. Your
> proposal will reject the calls based on relative stengths of X and Y and
> what UAC has. And require UAC to reInitiate the calls when it is failed.

Let's put this issue aside until we're clear on how the system I
described works in the simple case above.


> I don't agree that we need such a header.
>
> SS>> I don't agree with you here. Refer my concerns in above replies.
> Let me divide this into two
>     1) Header for REQUEST where ciphersuites etc are listed.

This is totally unnecessary.


>     2) Header for RESPONSE where least strong used algorithm in the path
> is indicated back 
>        to UAC by UAS by inspecting via trails.

The whole concept of "least strong" is a mistake. What should be
reported back is the entire list of proxies the message went
through and the relevant crypto algorithms.

-Ekr



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From sip-bounces@ietf.org Wed Jul 05 09:41:54 2006
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"Rajnish Jain" <rjain@precisionscientific.org> writes:

>> It seems to me that there is a simpler mechanism: any proxy X 
>> can only offer the intersection of the cipher suites that was 
>> offered to it and the ciphersuites that it is willing to 
>> support. 
>
> There seems to be a fundamental disconnect in the comment above. The comment
> above is basically describing a hop-by-hop mechanism. However, the draft is
> fundamentally about solving the e2e problem. The keyword here is e2e.

This draft *isn't* end-to-end. End-to-end would be like in RFC 2818
where there's a single TLS channel between UAC and UAS. What this
is is hop-by-hop with policy instructions.

> How
> will the proxy know that its narrowing down the UAC's preferred ciphersuites
> will not be detrimental to the setting up of the call somewhere downstream
> in the e2e path? 

It might be, but a lot of experience with real TLS deployments
and the fact that there is a mandatory-to-implement ciphersuite
suggests that this will not be a problem in practice.


>> This has the result that no downstream proxy can 
>> negotiate a cipher suite that Alice wasn't willing to 
>> negotiate with Atlanta. 
>
> But this has also has the undesirable result that no downstream proxy can
> negotiate a cipher suite that was agreeable to Alice.

Huh? No, that's incorrect.


>> 1. Alice doesn't get any signal of what was eventually negotiated.
>>    I would solve this problem by extending Via (I'm actually surprised
>>    this isn't done). MTAs typically put it in Received lines, FWIW...
>
> How will extending 'Via' help Alice, the UAC, learn about the negotiated
> ciphersuite? The draft does talk about extending 'Via' but that is to inform
> Bob, the UAS. The draft also mentions that the Security-Level header
> containing the lowest 'q' value is sent back to Alice. 

Yes, you'll need some reverse channel that carries the same data
as the Via.

>> 2. It's less expressive. In particular, the downstream cipher suites
>>    always narrow and never grow. So, if there were a third proxy in
>>    between Atlanta and Biloxi that only spoke DES, then Biloxi and Bob
>>    would only get DES even if they both spoke AES. 
>
> This comment doesn't seem to reflect the scheme described in the draft.
> Instead, this seems like a comment reflecting the hop-by-hop mechanism
> suggested in your own email. 

Yes, that's why I listed it as a disadvantage of the scheme described
in my e-mail message.

-Ekr


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From sip-bounces@ietf.org Wed Jul 05 09:49:11 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: sip@ietf.org
Subject: RE: [Sip] Ready for WGLC on Outbound?
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Attempt to resend as the previous version seems to have disappeared into the ether.

regards

Keith

> -----Original Message-----
> From: Drage, Keith (Keith) 
> Sent: 05 July 2006 11:25
> To: Cullen Jennings
> Cc: sip@ietf.org; rohan@ekabal.com; dean.willis@softarmor.com
> Subject: RE: [Sip] Ready for WGLC on Outbound?
> 
> 
> (As chair)
> 
> Right now both Dean and myself believe outbound is blocked.
> 
> We asked the question a month or so ago about whether 
> outbound could proceed without a discovery mechanism (either 
> included or normatively referenced), and there was a 
> significant number of people who apparently said no (we do 
> have difficulties on this list who respond to a direct 
> question from the chairs by going away and discussing 
> something completely different).
> 
> It appears that for these people, outbound could proceed if 
> we have a normative reference to a discovery mechanism, but 
> in that case, is that one mechanism or several, and do all or 
> any of these require mandatory support.
> 
> Conversely we have a set of people who appear to be saying 
> "use any mechanism you like" and therefore no normative 
> reference is needed.
> 
> Now obviously, we would like to get this resolved one way or 
> the other. We will quite probably be conducting hums on this 
> issue in the SIP WG session in Montreal, having failed to 
> achieve concensus on the list - unless in the next two or 
> three days we see a breaking of this impasse.
> 
> And remember, if outbound is blocked, then GRUU is dependent 
> on outbound, so at the end of the day that will also be 
> blocked, even if we progress it to the RFC editor's queue.
> 
> We are formulating the questions we need to ask the WG on 
> this issue, so any guidance from those involved would be 
> useful. It would be nice to get the problem stated as clearly 
> as possible so that the WG is able to give an intelligent answer.
> 
> regards
> 
> Keith
> 
> 

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On Jun 29, 2006, at 10:55 PM, <Erkki.Koivusalo@nokia.com>  
<Erkki.Koivusalo@nokia.com> wrote:

>
>> I can think of a number of ways to make this work where
>> the intended semantics are entirely embedded in DNS.
>
> If you have any ideas like that, could you please pass them
> to me since I am working on a related I-D where no such ideas
> have been introduced. I would be happy to receive those to be
> able to check if they work with the rest of the DNS scheme
> currently proposed. If yes, I could take them into my draft.

Well long and short of it is, if you are embedding the config  
information in DNS, just put it in DNS - look at how much information  
a ENUM or SRV record provides. I'm not sure if it is a good idea to  
do this or not but I'm just pointing out ways you might consider.


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From sip-bounces@ietf.org Wed Jul 05 10:32:00 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
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(as chair)

Well I believe we got a grand total of TWO responses on this question.

That means we are going to have to institute a more formal review, probably on the lines of the review process Dean outlined in a previous mail.

We will be discussing the logistics for this process offline in Montreal, but we will be looking for critical reviewers to formally assess this from various sectors of the IETF community.

If you consider yourself or someone else a person who should be a critical reviewer let us (the WG chairs) know. Also if have knowledge of a sector that should be addressing the review (e.g. SIPIT, security, CableLabs...) then please let us know.

regards

Keith

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: 27 June 2006 15:03
> To: 'sip@ietf.org'
> Cc: 'sip-chairs@ietf.org'
> Subject: RE: [Sip] draft-ietf-sip-gruu-09.txt ready for IESG?
> 
> 
> (As WG chair)
> 
> I'm afraid I am going to have to ask you to be much more 
> proactive on this question, as the IESG need something a 
> little more positive from the WG. So...
> 
> i)	If you have thoroughly reviewed this, and as a result 
> believe it is ready, and I know there are at least some of 
> you out there who have torn GRUU to pieces, throughly 
> digested it and put it back together again, then send Dean 
> and myself an email to that effect at sip-chairs@ietf.org.
> 
> ii)	If you are intending to review this but want more time 
> before responding, then also please say.
> 
> Remember this document is still in the WGLC state. We trying 
> to get it into the next state of IESG review. That means WG 
> consensus that it is ready.
> 
> Please try to respond by my original deadline of 30th June 
> (although we may give a couple of extra days grace here).
> 
> regards
> 
> Keith
> 
> > -----Original Message-----
> > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > Sent: 23 June 2006 11:57
> > To: 'sip@ietf.org'
> > Cc: 'sip-chairs@ietf.org'
> > Subject: [Sip] draft-ietf-sip-gruu-09.txt ready for IESG?
> > 
> > 
> > (As WG chair)
> > 
> > The SIP WG chairs and the author all think this document is 
> > ready for sending to the IESG.
> > 
> > We have had WGLC announced 5th July 2004 to complete 17th 
> July 2004. 
> > We had a further last round of WG review requested 26th 
> > October 2005 to complete 6th November 2005.
> > We have had significant discussion on the lists since then of 
> > this draft, and discussion in SIP WG sessions, and all 
> > comments have been resolved.
> > 
> > If you believe this document IS NOT ready to send to IESG, 
> > can you please indicate this to the list and the WG chairs by 
> > 12:00 midday UK time (7:00 am US Eastern Time) on 30th June 
> > 2006, giving technical justification for your objection.
> > 
> > Please limit this thread to answering the question above. If 
> > you have other discussion relating to GRUU generally, then 
> > please use a different thread.
> > 
> > regards
> > 
> > Keith
> > 
> > -----Original Message-----
> > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > Sent: 22 June 2006 20:50
> > To: i-d-announce@ietf.org
> > Cc: sip@ietf.org
> > Subject: [Sip] I-D ACTION:draft-ietf-sip-gruu-09.txt 
> > 
> > 
> > A New Internet-Draft is available from the on-line 
> > Internet-Drafts directories.
> > This draft is a work item of the Session Initiation Protocol 
> > Working Group of the IETF.
> > 
> > 	Title		: Obtaining and Using Globally Routable 
> > User Agent (UA) URIs (GRUU) in the Session Initiation Protocol (SIP)
> > 	Author(s)	: J. Rosenberg
> > 	Filename	: draft-ietf-sip-gruu-09.txt
> > 	Pages		: 38
> > 	Date		: 2006-6-22
> > 	
> > Several applications of the Session Initiation Protocol 
> (SIP) require
> > a user agent (UA) to construct and distribute a URI that can be used
> > by anyone on the Internet to route a call to that specific UA
> > instance.  A URI that routes to a specific UA instance is called a
> > Globally Routable UA URI (GRUU).  This document describes an
> > extension to SIP for obtaining a GRUU from a server and for
> > communicating a GRUU to a peer within a dialog.
> > 
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-sip-gruu-09.txt
> > 
> > To remove yourself from the I-D Announcement list, send a 
> message to 
> > i-d-announce-request@ietf.org with the word unsubscribe in 
> > the body of the message.  
> > You can also visit 
> https://www1.ietf.org/mailman/listinfo/I-D-announce 
> to change your subscription settings.
> 
> 
> Internet-Drafts are also available by anonymous FTP. Login 
> with the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-sip-gruu-09.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
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> _______________________________________________
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> 

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My impression of what the WG wanted was that the service provider  
could provide between 1 and the max number the UA supported of URIs.  
The UA must implement support for at least 2 and should support 4.

So let's say that the domain use two and this domains has to have two  
do to reliability. If the UA has "local policy" and only decided to  
use one - things don't work. I think this is more or less the same as  
if you are doing a single registration and the registry tells you  
your registration expires after 1 hour but the UA decided to refresh  
after two hours to reduce network load, well things are broken.

Now you say that the UA might decide by local policy if it wanted to  
do say 1 or 2 flows - but where will this local policy come from? The  
end user is not going too provide it. I suspect the domain will  
provide it. I'm fine with some times the domain tells the UA to use 1  
uri and other times two. I can imagine the domain changing this  
configuration as the UA moves from different types of access networks.

I expect you are right in that this MUST is too strong here but I  
think there is an argument for SHOULD. The problem is that if the  
domain assume the UA is doing two flows and the UA does one, things  
are broken.



On Jul 5, 2006, at 3:44 AM, Miguel Garcia wrote:

> Reading Section 4.2 of Outbound-04, there is a sentence that came  
> to my attention.
>
>    For each outbound proxy URI in the set, the UA MUST send a
>    REGISTER in the normal way using this URI as the default outbound
>    proxy.
>
> I think it is not appropriate to use a MUST strength in this case.  
> I suspect that the spirit of the draft is to offer high  
> availability by making the UA to establish one flow per URI, which  
> is a good spirit. However, one has to evaluate the cost of  
> establishing one flow per URI, which might be quite high in terms  
> of battery consumption, for example (due to STUN refreshes).
>
> I also believe the MUST is misused here because it is not an  
> absolute requirement for the specification. In other words, the SIP  
> outbound procedures work OK even if the UA establishes just one  
> flow (even when it is provisioned with several outbound URIs).  
> Nothing is broken.
>
> So, we should be looking at adding a paragraph that describes the  
> benefit of establishing one flow per outbound URI, but letting up  
> to the UA's policy to decide on the absolute number of established  
> flows.
>
> I would suggest to replace the above mentioned sentence with  
> something around these lines:
>
>    For each outbound proxy URI in the set, the UA MAY send a
>    REGISTER in the normal way using this URI as the default outbound
>    proxy. Sending one REGISTER per outbound proxy URI decreases the
>    likelihood of interrupted service due to dropped connections at the
>    cost of maintaining a living flow which each outbound proxy server.
>
> Comments?
>
> /Miguel
>
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Copying forward an older thread from SIPPING in case those =20
participating in the fork-loop-fix conversation here hadn't seen it.

RjS


Begin forwarded message:

> From: Thomas Froment <Thomas.Froment@alcatel.fr>
> Date: May 19, 2006 8:56:59 AM CDT
> To: Jeroen van Bemmel <jbemmel@zonnet.nl>
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] comments on loop detection anddraft-campen-=20
> sipping-stack-loop-detect-00
>
> Jeroen van Bemmel wrote:
>
>> Thomas,
>>
>> Very interesting to have some numbers here
>>
>> 260/305 ~ 15% less calls/sec with loop detection. That would be ~ =20
>> 15% CPU, iff it is at 100% for the whole test - is it? (you say =20
>> 13%, so perhaps you already compensated for this - I just want to =20
>> understand how you came to this conclusion)
>
> You are right, when doing a test, CPU is not busy at 100%, it is =20
> approximately between 85 and 90%.
> It is to notice we perform this kind of tests on our implementation =20=

> for a long time (based on sipstone, http://www.sipstone.org/),
> and I am really confident in injector system and in the results and =20=

> in general...
>
>>
>> RFC3261 is wrong in many aspects of loop detection.
>
> Is it documented somewhere? did you find a bug describing these =20
> problems?
>
>> Could you run the same test, but this time only including the =20
>> request URI and the topmost Route header (if any) in the hash?
>
> The current version *already* includes the usage of request URI and =20=

> the topmost Route header (only!) ... I did not even try to
> test the previous algorithm....
>
>> ----- Original Message ----- From: "Thomas Froment" =20
>> <Thomas.Froment@alcatel.fr>
>> To: <sipping@ietf.org>
>> Sent: Friday, May 19, 2006 12:43 PM
>> Subject: [Sipping] comments on loop detection anddraft-campen-=20
>> sipping-stack-loop-detect-00
>>
>>
>> Taking into account draft-ietf-sip-fork-loop-fix-01.txt which =20
>> recommends
>> the support of RFC3261 loop detection algorithm (instead
>> of the Max-Forwards mechanism), I would like to make some comments:
>>
>> 1/ The loop detection mechanism described in RFC 3261 seems =20
>> confusing in
>> some points:
>>
>> - Section 16.6, =A78 (when sending the request, adding the loop =20
>> detection
>> hash on branch parameter) says:
>>
>> " Loop detection is performed by verifying that, when a request =20
>> returns to a proxy,
>> those fields having an impact on the processing of the request =20
>> have not changed.
>> The value placed in this part of the branch parameter SHOULD =20
>> reflect all of
>> those fields (including any Route, Proxy-Require and Proxy-=20
>> Authorization header fields).
>> This is to ensure that if the
>> request is routed back to the proxy and one of those fields
>> changes, it is treated as a spiral and not a loop (see Section
>> 16.3). (...)"
>>
>> First of all the "*fields having an impact on the processing of =20
>> the request have not changed*" is confusing
>> for some people, because you really need to have a deep knowledge =20
>> of SIP protocol to be sure on this point. (And
>> what about SIP extensions like Service Route?)
>>
>> Then "any Route" is, in my opinion confusing, is it really any =20
>> Route or only top most Route?
>> I imagine this is only the top most route, and it would match the =20
>> statement "
>> fields having an impact on the processing of the request", am I =20
>> wrong?
>>
>> Third, the last paragraph is even more confusing:
>> "A common way to create this value is to compute a
>> cryptographic hash of the To tag, =46rom tag, Call-ID header
>> field, the Request-URI of the request received (before
>> translation), the topmost Via header, and the sequence number
>> from the CSeq header field, in addition to any Proxy-Require"
>>
>> - Here, no Route header is mentionned anymore (neither Proxy-=20
>> Authorization), should be clarified...
>> - the "topmost Via header":  this assertion is FALSE, if you take =20
>> the "branch parameter" of Via Header, to compute your hash,
>> you will never detect any loop, since the generated branch will =20
>> always be different !
>>
>> I first tried to find a bug in database, and only found one =20
>> problem related to loop detection:
>> http://bugs.sipit.net/show_bug.cgi?id=3D648
>> "branch hash calculacation needs explicit comment for loop-=20
>> detecting proxies"
>> Which is not pointing out the problem...
>>
>> Correct wording of the second paragraph may look like:
>> "A common way to create this value is to compute a
>> cryptographic hash of the To tag, =46rom tag, Call-ID header
>> field, the Request-URI of the request received (before
>> translation), the topmost Via header *without branch parameter*, =20
>> the sequence number
>> from the CSeq header field, in addition to *the top most Route* =20
>> and any Proxy-Require or Proxy-Authorization header(s)*"
>>
>>
>> 2/ Looking at "draft-campen-sipping-stack-loop-detect-00", we also =20=

>> decided to bench the *existing* RFC 3261 algorithm
>> to be able to evaluate if an optimization is needed.
>> It is to mentionned, this bench was performed on a really =20
>> optimized SIP stack and the loop detection, as well,
>> has been really carefully coded in order to minimize processing =20
>> and memory usage
>> (just to anticipate objections like "is your algorithm really well =20=

>> coded?): as an illustration, loop detection is not performed if
>> a request is recognized as being a retransmission, and an =20
>> optimized MD5 implementation is used as hash algorithm...
>>
>> Doing a simple proxying scenario,
>> using in injector the following "CALL ATTEMPT PER SECOND" (CAPS) =20
>> scenario:
>>
>> injector
>> INVITE --------------> Proxy (under test) ---------------> server =20
>> (UAS test)
>>
>> 100 TRYING  <---------- Proxy (under test)
>> 180 RINGING <---------- Proxy (under test) <-------------- server
>> 200 OK      <---------- Proxy (under test) <-------------- server
>>
>> ACK ------------------> Proxy (under test) ------------->  server
>> BYE ------------------> Proxy (under test) ------------->  server
>>
>> on a basic configuration,
>> WITHOUT loop detection, result is ~ 305 CAPS
>> WITH loop detection, result is ~ 260 CAPS
>> It means, ~13% of CPU is used for this fonction for a basic =20
>> proxying scenario...
>>
>> We will try to provide the figures if we decide to implement draft-=20=

>> campen-sipping-stack-loop-detect-00
>> algorithm but, at first, it seems valuable to think of clarifying =20
>> and/or optimizing current RFC 3261 algorithm...
>>
>> Thomas
>>
>>
>>
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


--Apple-Mail-2--142151341
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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; ">Copying forward an older thread =
from SIPPING in case those participating in the fork-loop-fix =
conversation here hadn't seen it.<DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>RjS</DIV><DIV><BR><DIV><BR><D=
IV>Begin forwarded message:</DIV><BR =
class=3D"Apple-interchange-newline"><BLOCKQUOTE type=3D"cite"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><B>From: =
</B></FONT><FONT face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica">Thomas Froment &lt;<A =
href=3D"mailto:Thomas.Froment@alcatel.fr">Thomas.Froment@alcatel.fr</A>&gt=
;</FONT></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><B>Date: </B></FONT><FONT face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">May 19, 2006 8:56:59 AM =
CDT</FONT></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><B>To: </B></FONT><FONT face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">Jeroen van Bemmel &lt;<A =
href=3D"mailto:jbemmel@zonnet.nl">jbemmel@zonnet.nl</A>&gt;</FONT></DIV><D=
IV style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><B>Cc: </B></FONT><FONT =
face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica"><A =
href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><B>Subject: =
</B></FONT><FONT face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><B>Re: [Sipping] comments on loop detection =
anddraft-campen-sipping-stack-loop-detect-00</B></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV> <DIV style=3D"margin-top:=
 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Jeroen =
van Bemmel wrote:</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV> =
<BLOCKQUOTE type=3D"cite"><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; ">Thomas,</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Very =
interesting to have some numbers here</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">260/305 ~ 15% =
less calls/sec with loop detection. That would be ~ 15% CPU, iff it is =
at 100% for the whole test - is it? (you say 13%, so perhaps you already =
compensated for this - I just want to understand how you came to this =
conclusion)</DIV> </BLOCKQUOTE><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">You are right, when doing a =
test, CPU is not busy at 100%, it is approximately between 85 and =
90%.</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">It is to notice we perform this =
kind of tests on our implementation for a long time (based on sipstone, =
<A =
href=3D"http://www.sipstone.org">http://www.sipstone.org</A>/),</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">and I am really confident in injector system and in =
the results and in general...</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV> <BLOCKQUOTE type=3D"cite"><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">RFC3261 is =
wrong in many aspects of loop detection.</DIV> </BLOCKQUOTE><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Is it =
documented somewhere? did you find a bug describing these =
problems?</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV> =
<BLOCKQUOTE type=3D"cite"><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; ">Could you run the same =
test, but this time only including the request URI and the topmost Route =
header (if any) in the hash?</DIV> </BLOCKQUOTE><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">The current =
version *already* includes the usage of request URI and the topmost =
Route header (only!) ... I did not even try to</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">test the previous algorithm....</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV> <BLOCKQUOTE =
type=3D"cite"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">----- Original Message ----- =
From: "Thomas Froment" &lt;<A =
href=3D"mailto:Thomas.Froment@alcatel.fr">Thomas.Froment@alcatel.fr</A>&gt=
;</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; ">To: &lt;<A =
href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A>&gt;</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Sent: Friday, May 19, 2006 12:43 PM</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Subject: [Sipping] comments on loop detection =
anddraft-campen-sipping-stack-loop-detect-00</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Taking into =
account draft-ietf-sip-fork-loop-fix-01.txt which recommends</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">the support of RFC3261 loop detection algorithm =
(instead</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">of the Max-Forwards mechanism), =
I would like to make some comments:</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">1/ The loop detection mechanism =
described in RFC 3261 seems confusing in</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">some =
points:</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">- Section 16.6, =A78 (when sending the request, =
adding the loop detection</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">hash on =
branch parameter) says:</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">" Loop detection is performed by =
verifying that, when a request returns to a proxy,</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">those fields having an impact on the processing of =
the request have not changed.</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">The value =
placed in this part of the branch parameter SHOULD reflect all =
of</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; ">those fields (including any Route, =
Proxy-Require and Proxy-Authorization header fields).</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">This is to ensure that if the</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">request is routed back to the proxy and one of those =
fields</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">changes, it is treated as a =
spiral and not a loop (see Section</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">16.3). =
(...)"</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">First of all the "*fields having an impact on the =
processing of the request have not changed*" is confusing</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">for some people, because you really need to have a =
deep knowledge of SIP protocol to be sure on this point. (And</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">what about SIP extensions like Service =
Route?)</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Then "any Route" is, in my opinion confusing, is it =
really any Route or only top most Route?</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">I =
imagine this is only the top most route, and it would match the =
statement "</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">fields having an impact on the =
processing of the request", am I wrong?</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Third, the =
last paragraph is even more confusing:</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">"A =
common way to create this value is to compute a</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">cryptographic hash of the To tag, =46rom tag, =
Call-ID header</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">field, the Request-URI of the =
request received (before</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">translation), =
the topmost Via header, and the sequence number</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">from the CSeq header field, in addition to any =
Proxy-Require"</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">- Here, no Route header is mentionned anymore =
(neither Proxy-Authorization), should be clarified...</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">- the "topmost Via header":<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN>this assertion is FALSE, if =
you take the "branch parameter" of Via Header, to compute your =
hash,</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">you will never detect any loop, =
since the generated branch will always be different !</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">I first =
tried to find a bug in database, and only found one problem related to =
loop detection:</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://bugs.sipit.net/show_bug.cgi?id=3D648">http://bugs.sipit.net=
/show_bug.cgi?id=3D648</A></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">"branch hash =
calculacation needs explicit comment for loop-detecting =
proxies"</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Which is not pointing out the =
problem...</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Correct wording of the second paragraph may look =
like:</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">"A common way to create this =
value is to compute a</DIV><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; ">cryptographic hash of the =
To tag, =46rom tag, Call-ID header</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">field, the =
Request-URI of the request received (before</DIV><DIV style=3D"margin-top:=
 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">translation), the topmost Via header *without branch parameter*, the =
sequence number</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">from the CSeq header field, in =
addition to *the top most Route* and any Proxy-Require or =
Proxy-Authorization header(s)*"</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">2/ Looking at =
"draft-campen-sipping-stack-loop-detect-00", we also decided to bench =
the *existing* RFC 3261 algorithm</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">to be able to =
evaluate if an optimization is needed.</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">It is to =
mentionned, this bench was performed on a really optimized SIP stack and =
the loop detection, as well,</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">has been =
really carefully coded in order to minimize processing and memory =
usage</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">(just to anticipate objections =
like "is your algorithm really well coded?): as an illustration, loop =
detection is not performed if</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">a request is =
recognized as being a retransmission, and an optimized MD5 =
implementation is used as hash algorithm...</DIV><DIV style=3D"margin-top:=
 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Doing a =
simple proxying scenario,</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">using in =
injector the following "CALL ATTEMPT PER SECOND" (CAPS) =
scenario:</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">injector</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">INVITE =
--------------&gt; Proxy (under test) ---------------&gt; server (UAS =
test)</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">100 TRYING<SPAN class=3D"Apple-converted-space">=A0 =
</SPAN>&lt;---------- Proxy (under test)</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">180 =
RINGING &lt;---------- Proxy (under test) &lt;-------------- =
server</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">200 OK<SPAN =
class=3D"Apple-converted-space">=A0 =A0 =A0 </SPAN>&lt;---------- Proxy =
(under test) &lt;-------------- server</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">ACK =
------------------&gt; Proxy (under test) -------------&gt;<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN>server</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">BYE ------------------&gt; Proxy (under test) =
-------------&gt;<SPAN class=3D"Apple-converted-space">=A0 =
</SPAN>server</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">on a basic configuration,</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">WITHOUT loop detection, result is ~ 305 =
CAPS</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">WITH loop detection, result is ~ =
260 CAPS</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">It means, ~13% of CPU is used =
for this fonction for a basic proxying scenario...</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">We will =
try to provide the figures if we decide to implement =
draft-campen-sipping-stack-loop-detect-00</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">algorithm but, at first, it seems valuable to think of clarifying =
and/or optimizing current RFC 3261 algorithm...</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">Thomas</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; =
">_______________________________________________</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Sipping mailing list<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN><A =
href=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf.=
org/mailman/listinfo/sipping</A></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">This list is =
for NEW development of the application of SIP</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Use <A =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colum=
bia.edu</A> for questions on current sip</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Use <A =
href=3D"mailto:sip@ietf.org">sip@ietf.org</A> for new developments of =
core SIP<SPAN class=3D"Apple-converted-space">=A0</SPAN></DIV> =
</BLOCKQUOTE><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">_______________________________________________</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Sipping mailing list<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN><A =
href=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf.=
org/mailman/listinfo/sipping</A></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">This list is =
for NEW development of the application of SIP</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Use <A =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colum=
bia.edu</A> for questions on current sip</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Use <A =
href=3D"mailto:sip@ietf.org">sip@ietf.org</A> for new developments of =
core SIP</DIV> </BLOCKQUOTE></DIV><BR></DIV></BODY></HTML>=

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From sip-bounces@ietf.org Wed Jul 05 11:25:34 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Cullen Jennings" <fluffy@cisco.com>,
	"Miguel Garcia" <Miguel.An.Garcia@nokia.com>
References: <44AB9820.4050200@nokia.com>
	<4E14323C-8338-4AF5-AF2A-4602ADB7F00D@cisco.com>
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Wed, 5 Jul 2006 17:23:39 +0200
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Cullen,

see inline,

Regards,

jeroen


>
> My impression of what the WG wanted was that the service provider  could 
> provide between 1 and the max number the UA supported of URIs.  The UA 
> must implement support for at least 2 and should support 4.
>

yes, agree. And "the service provider SHOULD provide at least 2" I suppose

> So let's say that the domain use two and this domains has to have two  do 
> to reliability. If the UA has "local policy" and only decided to  use 
> one - things don't work.

This depends on the definition of "work". The resillience against certain 
types of failures is less, but things will work provided no such failures 
occur.

> I think this is more or less the same as  if you are doing a single 
> registration and the registry tells you  your registration expires after 1 
> hour but the UA decided to refresh  after two hours to reduce network 
> load, well things are broken.
>

This is a clear case because *it is the UAs responsibility to stay 
registered*. In this scenario, we would say that specifically the UA is 
broken (RFC3261 section 10.2.4 : "Each UA is responsible for refreshing the 
bindings that it has previously established.")

Things are not so clear cut for outbound. The question is: who is 
responsible for achieving a certain level of reliability? who decides what 
is sufficient?

> Now you say that the UA might decide by local policy if it wanted to  do 
> say 1 or 2 flows - but where will this local policy come from? The  end 
> user is not going too provide it. I suspect the domain will  provide it. 
> I'm fine with some times the domain tells the UA to use 1  uri and other 
> times two. I can imagine the domain changing this  configuration as the UA 
> moves from different types of access networks.
>

This type of policy could very well be implemented in the UA, in fact it may 
be in a better position to estimate the reliability of the access network(s) 
it is connected to (eg by monitoring channel conditions). Therefore, a model 
in which the UA decides how many flows it uses is defendable too.

The decision on the number of flows depends on the desired level of 
reliability, which in turn depends on the failure characteristics of a 
number of things including the access network, any NAT devices, any edge 
proxies, and the home proxy.

What if we adopt a modus operandi in which the UA decides what level of 
reliability it desires and (thus) how many flows it needs, and the domain 
provides information to support this decision (eg MTBF figures for the 
registrar, or derived information such as a *suggestion* for the number of 
flows)

> I expect you are right in that this MUST is too strong here but I  think 
> there is an argument for SHOULD. The problem is that if the  domain assume 
> the UA is doing two flows and the UA does one, things  are broken.
>

Not from the point of view of the UA, it is getting exactly the level of 
reliability that it would expect from one flow. What reason would the domain 
have to assume anything about the number of flows the UA is using?

>
>
> On Jul 5, 2006, at 3:44 AM, Miguel Garcia wrote:
>
>> Reading Section 4.2 of Outbound-04, there is a sentence that came  to my 
>> attention.
>>
>>    For each outbound proxy URI in the set, the UA MUST send a
>>    REGISTER in the normal way using this URI as the default outbound
>>    proxy.
>>
>> I think it is not appropriate to use a MUST strength in this case.  I 
>> suspect that the spirit of the draft is to offer high  availability by 
>> making the UA to establish one flow per URI, which  is a good spirit. 
>> However, one has to evaluate the cost of  establishing one flow per URI, 
>> which might be quite high in terms  of battery consumption, for example 
>> (due to STUN refreshes).
>>
>> I also believe the MUST is misused here because it is not an  absolute 
>> requirement for the specification. In other words, the SIP  outbound 
>> procedures work OK even if the UA establishes just one  flow (even when 
>> it is provisioned with several outbound URIs).  Nothing is broken.
>>
>> So, we should be looking at adding a paragraph that describes the 
>> benefit of establishing one flow per outbound URI, but letting up  to the 
>> UA's policy to decide on the absolute number of established  flows.
>>
>> I would suggest to replace the above mentioned sentence with  something 
>> around these lines:
>>
>>    For each outbound proxy URI in the set, the UA MAY send a
>>    REGISTER in the normal way using this URI as the default outbound
>>    proxy. Sending one REGISTER per outbound proxy URI decreases the
>>    likelihood of interrupted service due to dropped connections at the
>>    cost of maintaining a living flow which each outbound proxy server.
>>
>> Comments?
>>
>> /Miguel
>>
>> -- 
>> Miguel A. Garcia           tel:+358-50-4804586
>> sip:miguel.an.garcia@openlaboratory.net
>> Nokia Research Center      Helsinki, Finland
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
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Date: Wed, 5 Jul 2006 16:22:43 +0100
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From: "Matthew Gardiner" <Matthew.Gardiner@ACULAB.COM>
To: "EKR" <ekr@networkresonance.com>
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Hi Eric,

>In particular, the downstream cipher suites always narrow and never =
grow.

I don't see that this is the case with Samir's scheme. Consider the case =
below where A wants to talk to D, via B and C:

          B ---------------------- C
         /                          \
        /                            \
       /	                          \=20
      A                                D

A, C and D all support cipher suites x,y and z. B only supports x. Just =
because the hops AB and BC must negotiate cipher suite x, it does not =
prevent hop CD negotiating anything from the range x,y,z.

That was my (brief) understanding of Samir's proposal. I can't see where =
the narrowing occurs, though obviously the range of suites will not =
grow.

Matthew Gardiner
Software Enginneer
Aculab

-----Original Message-----
From: EKR [mailto:ekr@networkresonance.com]
Sent: 04 July 2006 16:59
To: sip@ietf.org
Subject: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00


Because TLS is used in SIP in hop-by-hop mode, the cipher suite that
you negotiate with your first hop proxy is no guarantee of what will
be negotiated with future proxies. Consider the classic trapezoid:

        Atlanta ---------------------- Biloxi
         /                                \
        /                                  \
      /	                                    \=20
alice@atlanta                             bob@biloxi

Alice can negotiate AES-256 with Atlanta but then have Atlanta
negotiate RC4-40 with Biloxi. This draft suggests a series of
extensions to SIP to carry the initial client's preferences and that
proxies further down the chain are expected to enforce.

It seems to me that there is a simpler mechanism: any proxy X
can only offer the intersection of the cipher suites that was offered
to it and the ciphersuites that it is willing to support. This has
the result that no downstream proxy can negotiate a cipher suite
that Alice wasn't willing to negotiate with Atlanta.=20

There are two obvious disadvantages with respect to the proposed
draft:

1. Alice doesn't get any signal of what was eventually negotiated.
   I would solve this problem by extending Via (I'm actually surprised
   this isn't done). MTAs typically put it in Received lines, FWIW...
2. It's less expressive. In particular, the downstream cipher suites
   always narrow and never grow. So, if there were a third proxy in
   between Atlanta and Biloxi that only spoke DES, then Biloxi and Bob
   would only get DES even if they both spoke AES. Based on my =
experience
   with what real implementations support and the MUST for AES, I'm not
   much concerned about this. It's generally impossible to pickle
   the full client policy and have it sent to a proxy. It's mostly
   a matter of where you draw the line, and drawing it where the
   protocol already draws it seems natural.

The big advantage of the approach I propose is that it doesn't
require any new mechanisms and can be incrementally deployed in
an interoperable way.







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"Matthew Gardiner" <Matthew.Gardiner@ACULAB.COM> writes:

> Hi Eric,
>
>>In particular, the downstream cipher suites always narrow and never grow.
>
> I don't see that this is the case with Samir's scheme. Consider the case below where A wants to talk to D, via B and C:
>
>           B ---------------------- C
>          /                          \
>         /                            \
>        /	                          \ 
>       A                                D
>
> A, C and D all support cipher suites x,y and z. B only supports x. Just because the hops AB and BC must negotiate cipher suite x, it does not prevent hop CD negotiating anything from the range x,y,z.
>
> That was my (brief) understanding of Samir's proposal. I can't see where the narrowing occurs, though obviously the range of suites will not grow.

No, that's a disadvantage of the scheme I proposed.

-Ekr

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From sip-bounces@ietf.org Wed Jul 05 11:47:13 2006
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Update to fork-loop-fix submitted
Date: Wed, 5 Jul 2006 09:50:33 -0500
To: "<ravishankar.shiroor@wipro.com>" <ravishankar.shiroor@wipro.com>
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I understand the suggestion. You are trading off letting the request  
through a loop one extra cycle
for some lower amount of processing the first time the request went  
through the proxy. If you believe that
spirals are relatively rare things, this change would result in  
proxies very rarely having to build the loop/spiral
differentiation hash.

But as Jonathan points out, it's not clear that the hash computation  
is expensive enough to warrant this extra
logic and the extra network traffic it would allow. I expect that the  
hash calculation expense is small compared
to the parse/search/compare operations for "Proxy checks to see if  
its address is there in the via already".

RjS

On Jul 3, 2006, at 5:39 AM, <ravishankar.shiroor@wipro.com>  
<ravishankar.shiroor@wipro.com> wrote:

>
> Hi Robert,
>
> I had suggested one more alternative
>
> (http://www1.ietf.org/mail-archive/web/sip/current/msg14359.html )
>
> Quoted from my previous mail:
>
> <quote>
> I was thinking how to reduce the impact of overhead for calculating  
> the
> digest.
>
> If it is suggested as follows, I think it will reduce the overhead on
> normal scenarios:
>
> Proxy checks to see if its address is there in the via already.
> If it is not there, it proceeds normally without calculating the  
> digest
> and inserting Via to the outgoing request without the digest.
> If it is there, then it is possible that the request is looping. So,
> this time around, it calculates a digests and inserts and sends it
> along.
> If it is looping, the request will definitely come in again, which can
> be rejected.
> This way the impact of calculating the digest is required (reduced  
> to a
> smaller percent of the sessions) only when the proxy is suspicious the
> request might be looping.
> Of course this results in the scenarios mentioned in the id to be
> successful in completing the first loop, but the second loop will be
> detected and arrested. I think this can be applicable to general loop
> detection, and not just fork loop fixes.
> </quote>
>
> Regards,
> Ravi.
>
> --
>
> Ravishankar. G. Shiroor
> Wipro Technologies, Bangalore.
>
> ravishankar.shiroor@wipro.com
> --
>
>
>> -----Original Message-----
>> From: Robert Sparks [mailto:rjsparks@nostrum.com]
>> Sent: Sunday, June 25, 2006 8:22 PM
>> To: IETF SIP List
>> Subject: [Sip] Update to fork-loop-fix submitted
>>
>> Until it appears in the repository, see:
>> http://www.nostrum.com/~rjsparks/draft-ietf-sip-fork-loop-fix-02.txt
>>
>> This was a moderately heavy touch - based on list feedback,
>> I've moved and updated the sections covering the
>> loopdetection algorithm in from 3261. In the process of
>> writing the clarifying text, I discovered several places that
>> need clarification. They are listed as Open Issues in the
>> draft. Basically there are fields that I can't recall or
>> reconstruct how they would be useful in the part of the hash
>> used for loop-detection or spiral differentiation. If anyone
>> can remember why they need to be included, send a note to the
>> list and we'll capture that in the draft.
>>
>> Please review the updated loop-detection text very carefully.
>>
>> Here's the changelog from the draft:
>>
>>        Integrated several editorial fixes suggested by
>> Jonathan Rosenberg
>>
>>        Noted that the reduction of the attack to a single  
>> registration
>>        against a single URI as documented in previous versions,  
>> is, in
>>        fact, going to be effective against implementations
>> conforming to
>>        the standards before this repair.
>>
>>        Re-incorporated motivation from the original
>> maxforwards-problem
>>        draft into the security considerations section based
>> on feedback
>>        from Cullen Jennings
>>
>>        Introduced replacement text for the loop detection algorithm
>>        description in RFC 3261, fixing the bug 648 (the
>> topmost Via value
>>        must not be included in the second part) and clarifying the
>>        algorithm.  Removed several other fields suggested by 3261 and
>>        placed open issues around their presence.
>>
>>        Added a Notes to Implementors section capturing the
>> "common way"
>>        text and pointing to the interoperability issues that have  
>> been
>>        observed with loop detection at previous SIPits
>>
>>
>> Thanks,
>>
>> RjS
>>
>> _______________________________________________
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>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
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> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Sunday, July 02, 2006 1:02 PM
> 
> On Jun 30, 2006, at 12:57 PM, Hadriel Kaplan wrote:
> 
> > I agree with you, but it is used for detecting suspicious From
> > headers even
> > for non-anonymous calls.  Obviously if the proxy inserting the PAI
> > _knows_ a
> >> From is spoofed, it could/should reject it.  But if the proxy does
> >> _not_
> > know whether the From is spoofed or not, it just doesn't insert the
> > PAI.
> > For example, ingress proxy receives a request from another domain -
> > if it
> > trusts the domain it passes the PAI (or inserts one).  If it does
> > not trust
> > the domain, it removes PAI, but doesn't want to reject the request
> > (else it
> > wouldn't accept requests from such untrusted domains to begin with).
> 
> I see your point, I can see how a proxy that is deciding not to
> remove a PAI might not know a From was spoofed. I don't see how a
> proxy that was going to *insert* a PAI would know what to put in the
> PAI and not be able to know if a PAI was spoofed.

I'm not sure I follow - maybe you meant the word "From" instead of "PAI" in
the "if a PAI was spoofed" bit?  
I meant in the context of implicitly trusted connections.  For example: an
operator receiving a request from a directly-attached business customer
proxy, where the operator's ingress proxy may insert a PAI of
bob@company.com based on the From, even though company.com's egress proxy
did not insert the PAI itself.  Obviously for an anonymous request the
ingress proxy would not know enough to put that URI in, and it's certainly
the case that it can't be 100% sure that the request really came from bob to
begin with (e.g., the company proxy could be hijacked, etc.).  But that
could be the case with Identity in use as well.

-hadriel



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Jeroen,

I'd like to address this one point.  My recollection from Dallas is 
that we had a very strong consensus that the existence of 
keepalive=stun was sufficient to send STUN messages.

Based on this, I think we need to make it a requirement of the 
configuration mechanism that  you can depend on its result to have 
accurate keepalive information.

thanks,
-rohan

On Jul 5, 2006, at 1:23, Jeroen van Bemmel wrote:
> 4 - I am argueing: "MUST check provisioned keepalive=stun" is not 
> sufficient, because the configuration may be incorrect (for various 
> reasons, also depending on what you use for provisioning). As an 
> additional check, we need "MUST check confirmation from proxy"
>
> We cannot depend only on the configuration mechanism (which is still 
> TBD, and in principle open ended)


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> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Sunday, July 02, 2006 1:18 PM
>
> On Jun 30, 2006, at 2:59 PM, Hadriel Kaplan wrote:
> 
> > If a b2bua before that proxy but in the same trust domain verified the
> > identity, and can somehow tell the proxy it verified the identity,
> > and even
> > tell the UAS it did as well, why do more?  That's what the PAI
> > does.  It may
> > not be what it was supposed to do, but it's what it seems to now.
> 
> So in the case of an anonymous call where the real caller identity is
> in the PAI for malicious call trace reasons, what are you proposing
> to do. Does the proxy strip the PAI before sending it to the UAS?
> How would the proxy know? Do you deliver the PAI to the UA - this
> would violate privacy laws in Europe. I don't see how this would work.

I don't see how that's germane to the Identity issue?  Is there Identity
support for anonymous requests? (I thought not) The UAS clearly cannot
distinguish a trusted identity for such a request, since there is no
anonymous identity.  

If you mean how does the proxy know to do so for only anonymous requests
while not the others?  You tell me - you wrote the spec. :)  Operators seem
to do every permutation possible.  Some just have a rule where all PAI's for
all requests get stripped at the edge proxy right before the UAS, others do
it only if priv-value=id, some for any priv-value, some don't strip it at
all.  Is your point if the trust domain is not the same, and if the edge
proxy strips them all, then the UAS won't know it's a verified identity?  


> > But for transit cases: it requires the transit domain(s) to either
> > (1) not
> > muck with contact, call-id, and bodies, or (2) support identity,
> > and to
> > somehow re-sign with something that the verifier in the next domain
> > (s) will
> > accept, which goes back to hop-by-hop trust kinda.  But changing
> > (1) or (2)
> > is out of the local (and final) domain's control to change - they
> > need the
> > transit domain(s) to change, and that's a much bigger hurdle.
> 
> Ok, I am done't think I am manging to explaim myself. Say A has a
> contract with B and B has a crontract with C. Now a call arrives at
> C, it seem that C wants a assertion to be made by B. Anything A says
> to C is clsoe to useless because C has no clue who A even is. I guese
> I feel like we need to sort out the requirements for what we want in
> a transit case and then we might be able to figure out a solution.

Yup, we need to sort out the reqs.  In my mind your A-B-C scenario above
doesn't need or benefit from Identity, much (or it just moves the same
problem one step away).  


> > What about just splitting it into 2 Identity headers/signatures?
> > One for
> > the From and nonce-ish stuff, the other for the body and same nonce-
> > ish
> > stuff.  If a middle-man wants to modify the body, he verifies and
> > re-signs
> > that second signature, or if the middle-man is unaware of identity
> > then the
> > next one can decide what to do with a bad signature for a body but
> > good
> > signature for From.
> 
> The From itself has no intrinsic value - what you are trying to sign
> is that some body came from the person in the From header. If the
>  From header and it's signature can be cut and paste from one message
> to another, Identity would fail to solve any security problems.

It has value - at least for the publicly stated benefits of Identity being
useful for SPIT prevention.  

It can't be cut/paste anymore easily than the body also being signed.  Its
hash still includes the To-uri, call-id, date, and CSeq bits.  If you can
cut/paste that info for re-use, you can do it even if the body is also
signed. (although at least with a signed body it would make it hard to fully
establish a pasted call with media, or whatever)


> > This way a b2bua can change the body but doesn't have to have a
> > certificate
> > matching the From-uri domain to do so, and the UAS can decide if
> > it's kosher
> > for the b2bua-domain to change the body, while also verifying the
> > From is of
> > the right domain.  I haven't thought it through, but it sounds ok.
> 
> You are proposing almost exactly the same thing I am. The way you are
> thinking of handling the body, just use that handling for both.

Except your's requires the middlemen to support Identity-coexistence or else
everything breaks - if they don't then the UAS won't know if anything in the
received request is correct, or if only the body got modified - even if only
the body got modified. (of course for some the body is the whole enchilada)

-hadriel



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Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
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Jeroen,

If there are two flows available and one fails, then there is less
urgency to doing the registration refresh.  In other words less of a
need to do an avalanche restart.  I would hope that the UA would
appreciate that there is more to reliability than just the one
connection it cares about.  Because the UA does not have the visibility
that the proxy does, it might allow this type of decision to be deferred
to the network.  Network operators may need to provide such guidance to
maintain the network's stability or deny the UA access altogether.  It
would be good if the implementations of UA and Proxy would support the
possibility of such a negotiation.

Mike
=20

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]=20
> Sent: Wednesday, July 05, 2006 11:24 AM
> To: Cullen Jennings (fluffy); Miguel Garcia
> Cc: sip@ietf.org; Rohan Mahy
> Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
>=20
> Cullen,
>=20
> see inline,
>=20
> Regards,
>=20
> jeroen
>=20
>=20
> >
> > My impression of what the WG wanted was that the service provider =20
> > could provide between 1 and the max number the UA supported=20
> of URIs. =20
> > The UA must implement support for at least 2 and should support 4.
> >
>=20
> yes, agree. And "the service provider SHOULD provide at least=20
> 2" I suppose
>=20
> > So let's say that the domain use two and this domains has=20
> to have two =20
> > do to reliability. If the UA has "local policy" and only=20
> decided to =20
> > use one - things don't work.
>=20
> This depends on the definition of "work". The resillience=20
> against certain types of failures is less, but things will=20
> work provided no such failures occur.
>=20
> > I think this is more or less the same as  if you are doing a single=20
> > registration and the registry tells you  your registration expires=20
> > after 1 hour but the UA decided to refresh  after two hours=20
> to reduce=20
> > network load, well things are broken.
> >
>=20
> This is a clear case because *it is the UAs responsibility to=20
> stay registered*. In this scenario, we would say that=20
> specifically the UA is broken (RFC3261 section 10.2.4 : "Each=20
> UA is responsible for refreshing the bindings that it has=20
> previously established.")
>=20
> Things are not so clear cut for outbound. The question is:=20
> who is responsible for achieving a certain level of=20
> reliability? who decides what is sufficient?
>=20
> > Now you say that the UA might decide by local policy if it=20
> wanted to =20
> > do say 1 or 2 flows - but where will this local policy come=20
> from? The =20
> > end user is not going too provide it. I suspect the domain=20
> will  provide it.
> > I'm fine with some times the domain tells the UA to use 1  uri and=20
> > other times two. I can imagine the domain changing this =20
> configuration=20
> > as the UA moves from different types of access networks.
> >
>=20
> This type of policy could very well be implemented in the UA,=20
> in fact it may=20
> be in a better position to estimate the reliability of the=20
> access network(s)=20
> it is connected to (eg by monitoring channel conditions).=20
> Therefore, a model=20
> in which the UA decides how many flows it uses is defendable too.
>=20
> The decision on the number of flows depends on the desired level of=20
> reliability, which in turn depends on the failure=20
> characteristics of a=20
> number of things including the access network, any NAT=20
> devices, any edge=20
> proxies, and the home proxy.
>=20
> What if we adopt a modus operandi in which the UA decides=20
> what level of=20
> reliability it desires and (thus) how many flows it needs,=20
> and the domain=20
> provides information to support this decision (eg MTBF=20
> figures for the=20
> registrar, or derived information such as a *suggestion* for=20
> the number of=20
> flows)
>=20
> > I expect you are right in that this MUST is too strong here=20
> but I  think=20
> > there is an argument for SHOULD. The problem is that if the=20
>  domain assume=20
> > the UA is doing two flows and the UA does one, things  are broken.
> >
>=20
> Not from the point of view of the UA, it is getting exactly=20
> the level of=20
> reliability that it would expect from one flow. What reason=20
> would the domain=20
> have to assume anything about the number of flows the UA is using?
>=20
> >
> >
> > On Jul 5, 2006, at 3:44 AM, Miguel Garcia wrote:
> >
> >> Reading Section 4.2 of Outbound-04, there is a sentence=20
> that came  to my=20
> >> attention.
> >>
> >>    For each outbound proxy URI in the set, the UA MUST send a
> >>    REGISTER in the normal way using this URI as the=20
> default outbound
> >>    proxy.
> >>
> >> I think it is not appropriate to use a MUST strength in=20
> this case.  I=20
> >> suspect that the spirit of the draft is to offer high =20
> availability by=20
> >> making the UA to establish one flow per URI, which  is a=20
> good spirit.=20
> >> However, one has to evaluate the cost of  establishing one=20
> flow per URI,=20
> >> which might be quite high in terms  of battery=20
> consumption, for example=20
> >> (due to STUN refreshes).
> >>
> >> I also believe the MUST is misused here because it is not=20
> an  absolute=20
> >> requirement for the specification. In other words, the SIP=20
>  outbound=20
> >> procedures work OK even if the UA establishes just one =20
> flow (even when=20
> >> it is provisioned with several outbound URIs).  Nothing is broken.
> >>
> >> So, we should be looking at adding a paragraph that describes the=20
> >> benefit of establishing one flow per outbound URI, but=20
> letting up  to the=20
> >> UA's policy to decide on the absolute number of established  flows.
> >>
> >> I would suggest to replace the above mentioned sentence=20
> with  something=20
> >> around these lines:
> >>
> >>    For each outbound proxy URI in the set, the UA MAY send a
> >>    REGISTER in the normal way using this URI as the=20
> default outbound
> >>    proxy. Sending one REGISTER per outbound proxy URI decreases the
> >>    likelihood of interrupted service due to dropped=20
> connections at the
> >>    cost of maintaining a living flow which each outbound=20
> proxy server.
> >>
> >> Comments?
> >>
> >> /Miguel
> >>
> >> --=20
> >> Miguel A. Garcia           tel:+358-50-4804586
> >> sip:miguel.an.garcia@openlaboratory.net
> >> Nokia Research Center      Helsinki, Finland
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From: "Drage, Keith (Keith)" <drage@lucent.com>
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Of the comments that I made, comment 12) does not apply. I misremembered the relationship between these two documents - it changed before IETF #65.

It also means if you want GRUU, you've got to get outbound finished.

regards

Keith

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: 03 July 2006 19:28
> To: sip@ietf.org
> Cc: Cullen Jennings; Rohan Mahy
> Subject: [Sip] Comments on draft-ietf-sip-outbound-04
> 
> 
> Herewith my comments on the above document.
> 
<snipped> 

> 12)	GRUU currently defined as an informative reference. 
> Given that it defines the instance id which is needed for 
> this protocol, I would have thought it needed to be normative.
> 
<snipped>
> 
> regards
> 
> Keith
> 
> 
> Keith Drage
> drage@lucent.com
> tel: +44 1793 776249
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Subject: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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Cullen Jennings wrote:
> 
> 
> I expect you are right in that this MUST is too strong here but I think 
> there is an argument for SHOULD. The problem is that if the domain 
> assume the UA is doing two flows and the UA does one, things are broken.
> 

I would like to understand which part of the outbound mechanism is 
broken if the domain assumes the UE will do two flows, but the UA does 
one flow. Yes, it isn't reliable, but the mechanism works nicely... or not?

Then... why can't a domain indicate just one outbound URI? At the moment 
the draft has an artificial lower limit for all the domains to provide 
two URIs and impose a reliability that might not be required/needed.

I still think that it makes sense to enable the possibility of the 
network to provide a minimum of URI, although it is not the recommended 
behavior.

With respect the MUST strength... I can accept a SHOULD as well.

BR,

       Miguel

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Subject: Re: [Sip] Ready for WGLC on Outbound?
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inline
----- Original Message ----- 
From: "Rohan Mahy" <rohan@ekabal.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Cc: "Cullen Jennings" <fluffy@cisco.com>; "Rohan Mahy" <rohan@ekabal.com>; 
<sip@ietf.org>; "Juha Heinanen" <jh@tutpro.com>; <dean.willis@softarmor.com>
Sent: Wednesday, July 05, 2006 1:52 PM
Subject: Re: [Sip] Ready for WGLC on Outbound?


> Jeroen,
>
> I'd like to address this one point.  My recollection from Dallas is that 
> we had a very strong consensus that the existence of keepalive=stun was 
> sufficient to send STUN messages.
>
> Based on this, I think we need to make it a requirement of the 
> configuration mechanism that  you can depend on its result to have 
> accurate keepalive information.

Does that mean that a UA should not send STUN if the keepalive=stun is not 
present?

If so, will outbound be useful prior to the completion of the various 
configuration mechanisms? And is it acceptable for a mis-configured UA to 
cause the proxy or the registrar to crash or mis-behave?

A mechanism such as the proposed Proxy-Require/Require method similar to 
sec-agree allows the UA to determine that the proxy and registrar support 
STUN before it sends and potentially harmful messages.

cheers,
(-:bob

>
> thanks,
> -rohan
>
> On Jul 5, 2006, at 1:23, Jeroen van Bemmel wrote:
>> 4 - I am argueing: "MUST check provisioned keepalive=stun" is not 
>> sufficient, because the configuration may be incorrect (for various 
>> reasons, also depending on what you use for provisioning). As an 
>> additional check, we need "MUST check confirmation from proxy"
>>
>> We cannot depend only on the configuration mechanism (which is still TBD, 
>> and in principle open ended)
>
>
> _______________________________________________
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> 


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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Miguel Garcia writes:

 > That is what I remember as well, but I don't recall any conclusion of 
 > "unless instructed otherwise by the domain". In otherwise, there is 
 > still some "magic" for the domain instructions.

if i remember correctly, someone proposed that register reply somehow
tells how many flow the domain needs.  anyway, it was alway decision of
the network, not the UA.

 > But coming back to my point... I don't understand why there is a MUST 
 > here, because the protocol is not broken if the MUST is not followed.

i have told at least eight times now on this mailing list that the only
real added value that outbound brings is reliability due to more than
one flow.  outbound is NOT needed for nat traversal.  if you say MAY in
the spec, UA vendors don't even support setting up more than one flow
and can claim compatibility.

-- juha

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Jeroen van Bemmel writes:

 > I'd still like to use 2 registrations with different reg-ids). A MUST which 
 > forces an implementation to always do (2) makes it impossible to do only 
 > (1).

see old postings.  every implementation must be able to establish at
least two flows (default), but this default may be overridden by
configuration.

-- juha


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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
Date: Wed, 5 Jul 2006 15:27:32 -0400
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Eric,

The Record-Route header should contain the path of those proxies that
intend to stay in the signaling path, whereas the Via only provides the
path back for the responses.

Should this make a difference?  Could A to B agree on one suite, while B
to C agree on another, but A to C not have a means for mid-call?  I
think so long as there is one default suite, this would not be a
problem.

There may be some corner cases here where the ciphersuite used on a hop
associated with a node in the path is not the one that will exist
mid-call due to Via/Record-Route differences.

Mike


> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@networkresonance.com]=20
> Sent: Wednesday, July 05, 2006 9:40 AM
> To: Rajnish Jain
> Cc: sip@ietf.org
> Subject: Re: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00
>=20
> "Rajnish Jain" <rjain@precisionscientific.org> writes:
>=20
> >> It seems to me that there is a simpler mechanism: any proxy X can=20
> >> only offer the intersection of the cipher suites that was=20
> offered to=20
> >> it and the ciphersuites that it is willing to support.
> >
> > There seems to be a fundamental disconnect in the comment=20
> above. The=20
> > comment above is basically describing a hop-by-hop=20
> mechanism. However,=20
> > the draft is fundamentally about solving the e2e problem.=20
> The keyword here is e2e.
>=20
> This draft *isn't* end-to-end. End-to-end would be like in=20
> RFC 2818 where there's a single TLS channel between UAC and=20
> UAS. What this is is hop-by-hop with policy instructions.
>=20
> > How
> > will the proxy know that its narrowing down the UAC's preferred=20
> > ciphersuites will not be detrimental to the setting up of the call=20
> > somewhere downstream in the e2e path?
>=20
> It might be, but a lot of experience with real TLS=20
> deployments and the fact that there is a=20
> mandatory-to-implement ciphersuite suggests that this will=20
> not be a problem in practice.
>=20
>=20
> >> This has the result that no downstream proxy can negotiate=20
> a cipher=20
> >> suite that Alice wasn't willing to negotiate with Atlanta.
> >
> > But this has also has the undesirable result that no=20
> downstream proxy=20
> > can negotiate a cipher suite that was agreeable to Alice.
>=20
> Huh? No, that's incorrect.
>=20
>=20
> >> 1. Alice doesn't get any signal of what was eventually negotiated.
> >>    I would solve this problem by extending Via (I'm=20
> actually surprised
> >>    this isn't done). MTAs typically put it in Received=20
> lines, FWIW...
> >
> > How will extending 'Via' help Alice, the UAC, learn about the=20
> > negotiated ciphersuite? The draft does talk about extending=20
> 'Via' but=20
> > that is to inform Bob, the UAS. The draft also mentions that the=20
> > Security-Level header containing the lowest 'q' value is=20
> sent back to Alice.
>=20
> Yes, you'll need some reverse channel that carries the same=20
> data as the Via.
>=20
> >> 2. It's less expressive. In particular, the downstream=20
> cipher suites
> >>    always narrow and never grow. So, if there were a third proxy in
> >>    between Atlanta and Biloxi that only spoke DES, then=20
> Biloxi and Bob
> >>    would only get DES even if they both spoke AES.=20
> >
> > This comment doesn't seem to reflect the scheme described=20
> in the draft.
> > Instead, this seems like a comment reflecting the=20
> hop-by-hop mechanism=20
> > suggested in your own email.
>=20
> Yes, that's why I listed it as a disadvantage of the scheme=20
> described in my e-mail message.
>=20
> -Ekr
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Juha Heinanen" <jh@tutpro.com>,
	"Miguel Garcia" <Miguel.An.Garcia@nokia.com>
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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
Date: Wed, 5 Jul 2006 21:45:18 +0200
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Juha,

> it was alway decision of the network, not the UA

OK, current section 4.2 says "For each outbound proxy URI in the set, the UA 
MUST send a REGISTER". So that implies that the network could control the 
number of flows, by controlling the number of URIs provided through 
configuration (though this would possibly not work well for manual 
configuration)

I think this warrants some more discussion. The network may have reasons to 
limit the *maximum* number of flows (ie resource allocation), but it could 
enforce such a constraint in other ways too. For example, it could refuse 
registrations beyond the first N

But is it really necessary for the network to impose a minimum?

For flows beyond the maximum the network "suffers" (just like the registrar 
determines the expiration time). But by imposing a minimum number of flows, 
the UA potentially "suffers" (spending scarce resources such as battery 
power). And anyway, when the UA sets up less flows the consequences are for 
its user.

4.2 "UA MUST support sets with at least two outbound proxy URIs" should be 
sufficient to ensure vendor compliance. But to me, the UA should have some 
freedom to choose less than all URIs in the outbound proxy set (and you seem 
to agree: "default may be overridden by configuration", unless you consider 
that this configuration again comes from the network)

> if i remember correctly, someone proposed that register reply somehow
> tells how many flow the domain needs

I'm having trouble with "needs" here. Perhaps "allows" ?

Regards,

Jeroen


----- Original Message ----- 
From: "Juha Heinanen" <jh@tutpro.com>
To: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>
Cc: <sip@ietf.org>; "Cullen Jennings" <fluffy@cisco.com>; "Rohan Mahy" 
<rohan@ekabal.com>
Sent: Wednesday, July 05, 2006 8:58 PM
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04


> Miguel Garcia writes:
>
> > That is what I remember as well, but I don't recall any conclusion of
> > "unless instructed otherwise by the domain". In otherwise, there is
> > still some "magic" for the domain instructions.
>
> if i remember correctly, someone proposed that register reply somehow
> tells how many flow the domain needs.  anyway, it was alway decision of
> the network, not the UA.
>
> > But coming back to my point... I don't understand why there is a MUST
> > here, because the protocol is not broken if the MUST is not followed.
>
> i have told at least eight times now on this mailing list that the only
> real added value that outbound brings is reliability due to more than
> one flow.  outbound is NOT needed for nat traversal.  if you say MAY in
> the spec, UA vendors don't even support setting up more than one flow
> and can claim compatibility.
>
> -- juha
>
> _______________________________________________
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From sip-bounces@ietf.org Wed Jul 05 15:50:00 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: Cullen Jennings <fluffy@cisco.com>
Subject: RE: [Sip] Ready for WGLC on Outbound?
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(As chair)

Right now both Dean and myself believe outbound is blocked.

We asked the question a month or so ago about whether outbound could proceed without a discovery mechanism (either included or normatively referenced), and there was a significant number of people who apparently said no (we do have difficulties on this list who respond to a direct question from the chairs by going away and discussing something completely different).

It appears that for these people, outbound could proceed if we have a normative reference to a discovery mechanism, but in that case, is that one mechanism or several, and do all or any of these require mandatory support.

Conversely we have a set of people who appear to be saying "use any mechanism you like" and therefore no normative reference is needed.

Now obviously, we would like to get this resolved one way or the other. We will quite probably be conducting hums on this issue in the SIP WG session in Montreal, having failed to achieve concensus on the list - unless in the next two or three days we see a breaking of this impasse.

And remember, if outbound is blocked, then GRUU is dependent on outbound, so at the end of the day that will also be blocked, even if we progress it to the RFC editor's queue.

We are formulating the questions we need to ask the WG on this issue, so any guidance from those involved would be useful. It would be nice to get the problem stated as clearly as possible so that the WG is able to give an intelligent answer.

regards

Keith


> -----Original Message-----
> From: Ben Campbell [mailto:ben@estacado.net]
> Sent: 05 July 2006 05:10
> To: Cullen Jennings
> Cc: sip@ietf.org; rohan@ekabal.com; dean.willis@softarmor.com
> Subject: Re: [Sip] Ready for WGLC on Outbound?
> 
> 
> I've been a  bystander, but will jump in here. Comment below:
> 
> On Jul 4, 2006, at 3:37 PM, Cullen Jennings wrote:
> 
> >
> > I think people have stopped posting on this because they can't say  
> > anything that they have not already said. Don't take that as they  
> > agree.
> >
> > We got very strong feed back on one point - don't send STUN  
> > messages to a proxy unless you know it supports them. People felt  
> > strongly about that - I don't know why - but there was lots of  
> > people that agreed with that.
> >
> > I imagine people have sent me around 10 options on how to get the  
> > stun supported information and the set of outbound proxy URIs -  
> > certainly DNS and DHCP are some of them. We can write some drafts  
> > on various ways to do it. Right now my concern is making sure that  
> > no matter which of these many ways the WG decided to move forward  
> > with, outbound will work. So far every one of the possibilities I  
> > have heard would be possible with the current outbound. 
> Given that,  
> > I'm not sure what we need to change in outbound. I'm not even sure  
> > what you are proposing that we change in outbound - I think what  
> > you are proposing is we need to hold up outbound until we choose  
> > which one of the 10 we will use and furthermore you feel very  
> > strongly on which one of the 10 we should use. I'm not keen 
> on this  
> > - I think we have shown they are separable problems. The WG 
> decided  
> > on this separation of problems awhile back - I think it was in  
> > Vancouver - and many people supported it. I think it was a good  
> > idea  - we have shown we can move outbound forward before we have  
> > determined all the ways that this configuration information is  
> > delivered to the UA. I would be very hesitant to reverse this  
> > decision now.
> 
> I strongly agree with Cullen. We do not need to block the outbound  
> mechanism draft on figuring out the discovery stuff. The separation  
> was a good idea. If for no other reason, it lets us get work done.
> 
> 
> 
> 
> >
> > I don't think Dean's posts were sent as decries of the 
> requirements  
> > from the chair but purely as his opinion in trying to help bring  
> > the discussion to consensus. But I do think the draft meets the  
> > requirements and ideas he was outlining - I think the 
> current draft  
> > does meet the requirements he outlines. It does have a way for the  
> > "discovering whether a proxy supports the extensions of outbound"  
> > and on the topic of "there are two ways we can establish  
> > compatibility with an extensions", the WG rejected option 2 and we  
> > have gone with 1.
> >
> > Cullen (with my outbound author hat on)
> >
> >
> > On Jul 4, 2006, at 3:50 AM, <Erkki.Koivusalo@nokia.com>  
> > <Erkki.Koivusalo@nokia.com> wrote:
> >
> >>
> >> Hi Rohan,
> >>
> >> Sorry to be persistent with this issue, but what you say
> >> seems to conflict with what was discussed in March.
> >>
> >>> The only time a UAC sends a STUN request as directed by
> >>> outbound -04 is when the URI in the outbound-proxy-set
> >>> includes the keepalive=stun parameter. That DHCP does not
> >>> provide URI is a limitation of the SIP DHCP configuration
> >>> mechanism, not outbound. If you want to write a draft on
> >>> how to "shift-up" to using keepalives from some SIP or
> >>> external stimulus, you can still do this.
> >>
> >> What you say seems to suggest that you agree there is a problem
> >> when using DHCP SIP server option (as it is currently defined)
> >> for finding out if the discovered proxy supports outbound and
> >> STUN keepalives, but the solution is not in scope of Outbound
> >> draft itself.
> >>
> >> However I came to a different understanding based on the emails
> >> seen in the list earlier. For instance this email from Dean
> >> Willis (SIP WG chair) seems to suggest that the problem should
> >> be in scope of the outbound draft:
> >> http://www1.ietf.org/mail-archive/web/sip/current/msg13667.html
> >>
> >> "Ok, I think we've boiled this down to a new requirement:
> >>
> >> for the "outbound" technique to work, we MUST have a means of
> >> discovering whether a proxy supports the extensions of outbound.
> >>
> >> While we could do that by doing a REGISTER exchange that includes
> >> an appropriate option-tag in Require:, it would be more efficient
> >> if the proxy discovery mechanism informed us of this detail. So,
> >> we posit that the Outbound draft SHOULD select and discuss such a
> >> mechanism."
> >>
> >> And in yet another email
> >> http://www1.ietf.org/mail-archive/web/sip/current/msg13692.html
> >> Dean tells this:
> >>
> >> "There are two ways we can establish compatibility with an  
> >> extensions:
> >>
> >> 1) By foreknowledge
> >> 2) By experimentation -- try it and see if it works.
> >>
> >> Both have limitations:
> >>
> >> In the case of #1, we might have incorrect knowledge. So we'll  
> >> probably
> >> end up testing anyhow . . .
> >>
> >> In the case of #2, the experiment might cause a meltdown. So the  
> >> "probe
> >> for discovery" must be made completely safe. Also, we can  
> >> eventually end
> >> up with so many wierd hooks and idiosyncratic 
> implementations that we
> >> end up doing a BUNCH of probing before we do any real  
> >> communicating. And
> >> it's possible for the conditions to deteriorate so much that  
> >> probes for
> >> two different possible conditions are mutually exclusive -- if one
> >> condition applies, the probe for another will cause a meltdown.
> >>
> >> So what happens if we start STUN-probing a proxy on its SIP port  
> >> and it
> >> doesn't support STUN? My guess is, it's random.
> >>
> >> ...
> >>
> >> One approach to "forward compatibility" design is to make sure  
> >> that we
> >> can use both foreknowledge and experimentation -- the "trust but  
> >> verify"
> >> model.
> >>
> >> ...
> >>
> >> While it's easy to decide whether a configured registrar URI has a
> >> keepalive parameter, what happens to proxies we select based on
> >> something besides direct configuration?
> >>
> >> Specifically,
> >>
> >> 1) Can the DHCP option for default proxy present the keepalive
> >> parameter?"
> >> ----
> >>
> >> During these three months after seeing those comments in the list
> >> I have been expecting the outbound draft to provide answer to
> >> the question how UA could (within SIP signalling exchange)
> >> safely to check whether or not a proxy supports STUN keepalive.
> >> Two mechanims have been suggested how the UA could do this in
> >> a simple way:
> >>
> >> 1. The UA to check that the Path header in 200 OK returned for
> >>    REGISTER would contain keepalive=stun parameter for the 
> proxy URI
> >>
> >> 2. The UA to add Proxy-Reequire: stun-keepalive parameter into
> >>    the REGISTER request and the proxy to remove this tag from
> >>    the request before forwarding it towards the registrar
> >>
> >> For me, any of those mechanisms would sound feasible and for
> >> safety reasons I would hope the draft to deny the UA for sending
> >> STUN keepalives to the proxy which did not pass such a test.
> >>
> >>> the same problem exists for the "comp" parameter for example.
> >>
> >> Yes but please bear in mind that for comp=sigcomp URI parameter
> >> RFC 3486 Compressing SIP gives several solutions how the UA
> >> would be able to check if the proxy supports SigComp while
> >> outbound does not define any other than relying on configuration.
> >> A cite from RFC 3486:
> >>
> >> 4.1   Obtaining a SIP or SIPS URI with comp=sigcomp
> >>
> >>    For requests within a dialog, a next-hop URI with the 
> comp=sigcomp
> >>    parameter is obtained from a Record-Route header field when the
> >>    dialog is established.  A client sending a request outside a  
> >> dialog
> >>    can also obtain SIP URIs with comp=sigcomp in a Contact header  
> >> field
> >>    in a 3xx or 485 response to the request.
> >>
> >>    However, clients establishing a session will not typically be  
> >> willing
> >>    to wait until the dialog is established in order to begin  
> >> compressing
> >>    messages.  One of the biggest gains that SigComp can bring to  
> >> SIP is
> >>    the ability to compress the initial INVITE of a dialog, when  
> >> the user
> >>    is waiting for the session to be established.  
> Therefore, clients
> >>    need a means to obtain a comp=sigcomp URI from their outbound  
> >> proxy
> >>    before the user decides to establish a session.
> >>
> >>    One solution to this problem is manual configuration.  However,
> >>    sometimes it is necessary to have clients configured in an  
> >> automatic
> >>    fashion.  Unfortunately, current mechanisms for SIP client
> >>    configuration (e.g., using DHCP [6]) do not allow to provide the
> >>    client with URI parameters.  In this case, the client SHOULD  
> >> send an
> >>    uncompressed OPTIONS request to its outbound proxy.  
> The outbound
> >>    proxy can provide an alternative SIP URI with the comp=sigcomp
> >>    parameter in a Contact header field in a 200 OK response to the
> >>    OPTIONS.  The client can use this URI for subsequent 
> requests that
> >>    are sent through the same outbound proxy using compression.
> >>
> >>    RFC 3261 [1] does not define how a proxy should respond to an  
> >> OPTIONS
> >>    request addressed to itself.  It only describes how servers  
> >> respond
> >>    to OPTIONS addressed to a particular user.  Section 
> 11.2 of RFC  
> >> 3261
> >>    says:
> >>
> >>       Contact header fields MAY be present in a 200 (OK) 
> response and
> >>       have the same semantics as in a 3xx response.  That 
> is, they  
> >> may
> >>       list a set of alternative names and methods of reaching the  
> >> user.
> >>
> >>    We extend this behavior to proxy servers responding to OPTIONS
> >>    addressed to them.  They MAY list a set of alternative URIs to
> >>    contact the proxy.
> >>
> >> What has been called for is that outbound draft would also
> >> specify one solution how the UA could check the STUN keepalive
> >> support of a proxy. In my opinion good proposals seem
> >> to exist how to do it in a reasonable way.
> >>
> >> Regards,
> >>
> >> Erkki
> >>
> >>> -----Original Message-----
> >>> From: ext Rohan Mahy [mailto:rohan@ekabal.com]
> >>> Sent: 04.July.2006 03:10
> >>> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
> >>> Cc: dean.willis@softarmor.com; fluffy@cisco.com; Rohan Mahy;
> >>> sip@ietf.org; jbemmel@zonnet.nl
> >>> Subject: Re: [Sip] Ready for WGLC on Outbound?
> >>>
> >>> Hi Erkki,
> >>>
> >>> The only time a UAC sends a STUN request as directed by
> >>> outbound -04 is
> >>> when the URI in the outbound-proxy-set includes the keepalive=stun
> >>> parameter.  This is 100% backwards safe.  We don't need 
> any text to
> >>> prevent existing implementations from getting a bunch of STUN  
> >>> requests.
> >>>
> >>> That DHCP does not provide URI is a limitation of the SIP DHCP
> >>> configuration mechanism, not outbound.  The same problem
> >>> exists for the
> >>> "comp" parameter for example.  If you want to write a draft on  
> >>> how to
> >>> "shift-up" to using keepalives or using compression from 
> some SIP or
> >>> external stimulus, you can still do this.  No text in outbound
> >>> prevents
> >>> that from happening.
> >>>
> >>> thanks,
> >>> -rohan
> >>>
> >>>
> >>> On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
> >>>>
> >>>> Hi Rohan,
> >>>>
> >>>>>> Also, the issue is that what may be configured (no 
> matter how you
> >>>>>> configure it) may not be a URI. It may simply be an IP
> >>> address/FQDN,
> >>>>>> which is then used to build a SIP-URI. I think we shall
> >>>>>> allow that, and make it more clear in the spec. [/CHH]
> >>>>>
> >>>>> the outbound spec defines a piece of configurable data (the
> >>>>> 'outbound-proxy-set'), which is a list of URIs.  This list
> >>> of URIs may
> >>>>> never have a one-to-one correspondence to a user or 
> administrator
> >>>>> manually configurable field.  That piece of configurable data  
> >>>>> may be
> >>>>> derived from several fields, manually configurable, 
> discovered via
> >>>>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
> >>>>> download, or via a random character generator seeded by the  
> >>>>> phase of
> >>>>> the moon.
> >>>>>
> >>>>> The outbound-proxy-set was defined as a list of URIs because  
> >>>>> the UA
> >>>>> acts on each item in the list, because each URI is 
> expected to be
> >>>>> places in a Route header, and because each URI is expected to
> >>>>> typically have a keepalive URI parameter in it.
> >>>>
> >>>> Like mentioned in other recent emails the problem is 
> that if the UA
> >>>> derives the proxy URI(s) from DHCP SIP server option or 
> e.g. the  
> >>>> AOR
> >>>> of the user, in those cases the derived URI will NOT contain the
> >>>> keepalive URI parameter in it. DHCP just returns IP addresses or
> >>>> domain names instead of URIs with parameters.
> >>>>
> >>>> Thus in those cases it would be critical for the UA to be able
> >>>> to ensure in REGISTER - 200 OK sequence that the contacted proxy
> >>>> supports Outbound and can handle STUN requests sent to its SIP
> >>>> port (not to cause any damage to an uncompliant proxy). I fully
> >>>> agree with this proposal from Jeroen:
> >>>>
> >>>> --
> >>>> The text should then say "the UAC MUST wait for a 200 OK to  
> >>>> REGISTER
> >>>> before
> >>>> it starts sending STUN packets. If the last Path header of the  
> >>>> 200 OK
> >>>> does
> >>>> not contain keepalive=stun, the UAC MUST NOT use STUN
> >>> towards this URI.
> >>>> If
> >>>> there are no Path headers and no "Supported: outbound", the UAC  
> >>>> also
> >>>> MUST
> >>>> NOT use STUN towards the registrar.". It could still use TCP
> >>> keepalive
> >>>> if it
> >>>> supports that
> >>>> --
> >>>>
> >>>> Please excuse me if I am just repeating myself and what 
> e.g. Jeroen
> >>>> has said but I believe the importance of this issue 
> justifies some
> >>>> redundancy :-)
> >>>>
> >>>> Regards,
> >>>>
> >>>> Erkki
> >>>
> >>>
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From: "Vijay K. Gurbani" <vkg@lucent.com>
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EKR wrote:
> It seems to me that this document addresses two issues:
> 
> 1. What name(s) should appear in certificates.
> 2. Where they should appear and how they should be encoded.

Eric: Thanks much for your analysis.  More inline.

> ISTM that the same situation needs to apply with SIP. If you're 
> dereferencing <sips:alice@atlanta.com> then the certificate presented
> to you needs to either be for a known (and trusted!) proxy or
> contain "atlanta.com". Anything else requires trusting the DNS
> resolution process, which isn't generally considered acceptable.

Yes, the discussion on issue (1) on the mailing list converged
on the same outcome you mention above; i.e., it suffices
that when contacting "sips:alice@atlanta.com", the certificate
presented contains "atlanta.com."

However, it is the specific places where SIP differs from HTTP
in its routing of subsequent requests that is of concern.  For
example: consider a dialog that has been established across a
set of proxies that Record-Routed.  The UAC is now ready to send a
new request:

    UPDATE sips:uac@host1.atlanta.com SIP/2.0
    Route: <sips:proxy.atlanta.com;lr>
    ...

The request will go to "sips:proxy.atlanta.com", the Route URI.
What typically appears in a Route URI is the fqdn of a
certain proxy, as opposed to a URI that we can run rfc3263-
resolution on to derive a candidate host.  In other words,
when sending a request to "sips:alice@atlanta.com", it
suffices to see "atlanta.com" in the certificate.  But what
about the case where the UAC is sending a request to
"sips:proxy.atlanta.com"?

Is it sufficient that when the server presents a certificate
that contains "atlanta.com", this certificate is good for
"sips:proxy.atlanta.com" and "sips:proxy3.atlanta.com" and
for subordinate names like "sips:east.proxy.atlanta.com"?
It very well might be the case that this is sufficient (SIP
Identity alludes to the name subordination problem in Section
13.4).  However, if so, then we should state it as such so that
implementations can built the required logic into their code.

So, what are the guidelines to implementors here for comparison?
As an implementor, should I just do a substring comparison
of the contents of a certificate ("atlanta.com") against the
Route URI ("proxy.atlanta.com")?  Today, I am not sure how
implementors are dealing with such cases.  Any thoughts on
this from you or the WG at large will be much appreciated.

> So, this leaves us with the question of whether there should be 
> multiple names. As I understand S 10, the claimed disadvantages of a 
> single name are:
[...]
> I'm not sure what "common TLS usage" is being referred to. As I noted
> above, in my experience HTTPS behaves exactly this way (putting the 
> generic domain name and not host name in the certificate).

So what, then, does it mean to do "mutual authentication" in
SIP?  RFC3261 states that mutual authentication should occur
across inter-domain requests (Section 26.3.2.2).  It's relatively
easy to authenticate the server, but how should the client be
authenticated?  Consider:

     INVITE sips:alice@atlanta.com SIP/2.0
     Via: SIP/2.0/TLS proxy.biloxi.com;branch=...
     To: <sips:alice@atlanta.com>
     From: <sips:bob@biloxi.com>;tag=98jukss-1
     ...

The client authenticates the server using the established norm
of checking for "atlanta.com" in the certificate.  But the
server has only one piece of authoritative information about
the client: the sent-by value in the topmost Via, which contains
the specific host that sent the request.  This could be any host
in the biloxi.com domain, including a host in a subordinate
domain.  Thus, this case now degenerates to the same scenario we
were discussing above.

RFC3261 exhorts the server to match the certificate presented
from the client to the domainname in the From header.  This
will not work if retargetting is used (please see Appendix A and
B in the draft for more information on retargetting).

Now, it could well be that a combination of deprecating
retargetting outside your domain and SIP Identity is the
preferred means to do mutual authentication in SIP.  And if so,
then we ought to deprecate the portions of rfc3261 that talks
about mutual authentication.

I realize that you will be busy with the impending IETF, but
if you have some time for thoughts on virtual server
certificates (section 8) and server farms (section 7), I'd
much appreciate them as well.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Jeroen van Bemmel writes:

 > But is it really necessary for the network to impose a minimum?

in case network's reliability solution is based on more than one
simultaneous flow, then network must impose a minimum.

 > For flows beyond the maximum the network "suffers" (just like the registrar 
 > determines the expiration time). But by imposing a minimum number of flows, 
 > the UA potentially "suffers" (spending scarce resources such as battery 
 > power). And anyway, when the UA sets up less flows the consequences are for 
 > its user.

in the above situation, assuming that network has two proxies, if UA
decides to establish only one flow instead of two, then the likely hood
is 50% that incoming call to the UA cannot be establishes.  it thus
makes no sense for the UA to violate network policy on number of flows.

-- juha

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Drage, Keith (Keith) wrote:
> I have the following comments on the draft:
> 
> Technical ---------
> 
> 1)	I am interested in the question of whether this draft updates RFC
> 3261 or not, and would appreciate some list discussion on that issue.
> What, if any, RFC 3261 requirements does it change?

I beleive the draft updates rfc3261's normative language in
Section 18 on connection oriented transports, especially the
text before Section 18.1.

> NITS ----
> 
> 2)	The IANA considerations may need some additional information. You
> may want to look at the outbound draft to see if there is information
> there you could use as a template.

Sure.  Thanks.

> 3)	You may want to update Rohan's affiliation on the cover page (I
> assume he has it right on outbound).

Sure.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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Drage, Keith (Keith) writes:

 > We asked the question a month or so ago about whether outbound could
 > proceed without a discovery mechanism (either included or normatively
 > referenced), and there was a significant number of people who
 > apparently said no (we do have difficulties on this list who respond
 > to a direct question from the chairs by going away and discussing
 > something completely different). 

yes, i was one of those who said that outbound MUST reference DNS based
discovery in the same way as RFC3261 does.

 > It appears that for these people, outbound could proceed if we have a
 > normative reference to a discovery mechanism, but in that case, is
 > that one mechanism or several, and do all or any of these require
 > mandatory support.

there must be one that every UA MUST implement.  otherwise there is no
interoperability and outbound turn out to be useless in real world.

 > Conversely we have a set of people who appear to be saying "use any
 > mechanism you like" and therefore no normative reference is needed.

this would be a disaster to proxy operator.

 > And remember, if outbound is blocked, then GRUU is dependent on
 > outbound, so at the end of the day that will also be blocked, even if
 > we progress it to the RFC editor's queue.

i don't mind the delay if the result is a deployable package.  it makes
no sense to ietf to release outbound rfc without discovery reference and
pretend that it has accomplished something.  there is already now FAR
TOO MANY sip related rfcs that very few or nobody have implemented.

-- juha

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From sip-bounces@ietf.org Wed Jul 05 21:57:56 2006
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I fear this is an issue that was disposed of before and I don't
realize it:

I think the I-D needs to make clearer that when sending a request with
a GRUU that the request must be sent via an outbound proxy for the
GRUU's domain.  This is needed to make the machinery work, but isn't
stated explicitly as a UA requirement, it's implied in a section of
the I-D that appears to be a proxy requirement.

There's also the question of how the UA discovers the correct outbound
proxy address.  Strictly speaking, without a standardized mechanism,
different vendors could have incompatible implementations.  I admit
that this isn't much worse than the UA configuration problem as it now
stands.  Perhaps a good dodge is to allow (absent other configuration)
the UA to use the proxy for GRUU's domain as an outgoing proxy for
requests using that GRUU as a Contact?

Dale

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Dale,

Thanks for taking a stab at this. The 'hidden semantic' behind REFER is 
a looming potential problem that I don't think we've fully captured. 
Your draft takes a good step towards uncovering it, but I don't think it 
quite gets there.

In particular, the Summary section of the draft is where you actually 
show the differences between the various usages. None of the 'Actions' 
in the table demonstrate the UA having to do something that is 
usage-specific and not directly inferred from the request itself. That, 
to me, is the sign of trouble - that a UA has to manipulate some kind of 
other state, or take some kind of other action, as a side effect of 
REFER, and that other action or state change depends on the usage. Thus, 
at the moment, your draft doesn't seem to point out that there are any 
problems with the REFER mechanism.

One possible place this side-effect issue might show up is in the UI of 
a phone device. Without a doubt, its a better experience for a user to 
get asked, "Joe wants to transfer you to Bob, proceed?" than "You have 
received a REFER with Refer-To of sip:bob@example.com, proceed?", and 
this would seem to require the device to know that the reason for the 
REFER is a transfer. You might want to look at these kinds of UI issues 
across the different usages.

You call out a transfer #1 use case in which the transferee has to send 
a BYE after the transfer. However, this is not the case AFAIK. In none 
of the transfer flows does the transferee need to send a BYE; its the 
entity doing the transfer (which obviously knows its transferring) that 
needs to do a BYE after the REFER for the INVITE succeeds.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Thu Jul 06 00:45:32 2006
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Subject: RE: [Sip] Update to fork-loop-fix submitted
Date: Thu, 6 Jul 2006 10:13:27 +0530
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Thread-Topic: [Sip] Update to fork-loop-fix submitted
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From: <ravishankar.shiroor@wipro.com>
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It is always better to reduce the computation in the path of normal
success case and introduce it in the path of a rarer scenario. I think
it will be a fair assumption to make that Spirals will be rarer than the
non-spiraling scenarios.

In terms of increasing logic, the increase in logic would be very small
if it all any.
It would be a trade-off that should be left to the implementer to take
if he considers that hash computation is expensive on his platform. If
this id recommends the computation of hash at the MUST strength in the
first pass of the SIP request, an implementer would not have that
freedom to make that trade-off..=20

Regards,
Ravi.

--
=20
Ravishankar. G. Shiroor
Wipro Technologies, Bangalore.
=20
ravishankar.shiroor@wipro.com
--
=20

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: Wednesday, July 05, 2006 8:21 PM
> To: RaviShankar Shiroor (WT01 - Product Engineering Solutions)
> Cc: sip@ietf.org
> Subject: Re: [Sip] Update to fork-loop-fix submitted
>=20
> I understand the suggestion. You are trading off letting the=20
> request through a loop one extra cycle for some lower amount=20
> of processing the first time the request went through the=20
> proxy. If you believe that spirals are relatively rare=20
> things, this change would result in proxies very rarely=20
> having to build the loop/spiral differentiation hash.
>=20
> But as Jonathan points out, it's not clear that the hash=20
> computation is expensive enough to warrant this extra logic=20
> and the extra network traffic it would allow. I expect that=20
> the hash calculation expense is small compared to the=20
> parse/search/compare operations for "Proxy checks to see if=20
> its address is there in the via already".
>=20
> RjS
>=20
> On Jul 3, 2006, at 5:39 AM, <ravishankar.shiroor@wipro.com>=20
> <ravishankar.shiroor@wipro.com> wrote:
>=20
> >
> > Hi Robert,
> >
> > I had suggested one more alternative
> >
> > (http://www1.ietf.org/mail-archive/web/sip/current/msg14359.html )
> >
> > Quoted from my previous mail:
> >
> > <quote>
> > I was thinking how to reduce the impact of overhead for=20
> calculating =20
> > the
> > digest.
> >
> > If it is suggested as follows, I think it will reduce the=20
> overhead on
> > normal scenarios:
> >
> > Proxy checks to see if its address is there in the via already.
> > If it is not there, it proceeds normally without calculating the =20
> > digest
> > and inserting Via to the outgoing request without the digest.
> > If it is there, then it is possible that the request is looping. So,
> > this time around, it calculates a digests and inserts and sends it
> > along.
> > If it is looping, the request will definitely come in=20
> again, which can
> > be rejected.
> > This way the impact of calculating the digest is required (reduced =20
> > to a
> > smaller percent of the sessions) only when the proxy is=20
> suspicious the
> > request might be looping.
> > Of course this results in the scenarios mentioned in the id to be
> > successful in completing the first loop, but the second loop will be
> > detected and arrested. I think this can be applicable to=20
> general loop
> > detection, and not just fork loop fixes.
> > </quote>
> >
> > Regards,
> > Ravi.
> >
> > --
> >
> > Ravishankar. G. Shiroor
> > Wipro Technologies, Bangalore.
> >
> > ravishankar.shiroor@wipro.com
> > --
> >
> >
> >> -----Original Message-----
> >> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> >> Sent: Sunday, June 25, 2006 8:22 PM
> >> To: IETF SIP List
> >> Subject: [Sip] Update to fork-loop-fix submitted
> >>
> >> Until it appears in the repository, see:
> >>=20
> http://www.nostrum.com/~rjsparks/draft-ietf-sip-fork-loop-fix-02.txt
> >>
> >> This was a moderately heavy touch - based on list feedback,
> >> I've moved and updated the sections covering the
> >> loopdetection algorithm in from 3261. In the process of
> >> writing the clarifying text, I discovered several places that
> >> need clarification. They are listed as Open Issues in the
> >> draft. Basically there are fields that I can't recall or
> >> reconstruct how they would be useful in the part of the hash
> >> used for loop-detection or spiral differentiation. If anyone
> >> can remember why they need to be included, send a note to the
> >> list and we'll capture that in the draft.
> >>
> >> Please review the updated loop-detection text very carefully.
> >>
> >> Here's the changelog from the draft:
> >>
> >>        Integrated several editorial fixes suggested by
> >> Jonathan Rosenberg
> >>
> >>        Noted that the reduction of the attack to a single =20
> >> registration
> >>        against a single URI as documented in previous versions, =20
> >> is, in
> >>        fact, going to be effective against implementations
> >> conforming to
> >>        the standards before this repair.
> >>
> >>        Re-incorporated motivation from the original
> >> maxforwards-problem
> >>        draft into the security considerations section based
> >> on feedback
> >>        from Cullen Jennings
> >>
> >>        Introduced replacement text for the loop detection algorithm
> >>        description in RFC 3261, fixing the bug 648 (the
> >> topmost Via value
> >>        must not be included in the second part) and clarifying the
> >>        algorithm.  Removed several other fields suggested=20
> by 3261 and
> >>        placed open issues around their presence.
> >>
> >>        Added a Notes to Implementors section capturing the
> >> "common way"
> >>        text and pointing to the interoperability issues that have =20
> >> been
> >>        observed with loop detection at previous SIPits
> >>
> >>
> >> Thanks,
> >>
> >> RjS
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use
> >> sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >>
> >
> >
> > The information contained in this electronic message and any =20
> > attachments to this message are intended for the exclusive use of =20
> > the addressee(s) and may contain proprietary, confidential or =20
> > privileged information. If you are not the intended recipient, you =20
> > should not disseminate, distribute or copy this e-mail. Please =20
> > notify the sender immediately and destroy all copies of this =20
> > message and any attachments.
> >
> > WARNING: Computer viruses can be transmitted via email. The =20
> > recipient should check this email and any attachments for the =20
> > presence of viruses. The company accepts no liability for any =20
> > damage caused by any virus transmitted by this email.
> >
> > www.wipro.com
>=20
>=20

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From sip-bounces@ietf.org Thu Jul 06 00:59:56 2006
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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Juha,

> > For flows beyond the maximum the network "suffers" (just like the 
> > registrar
> > determines the expiration time). But by imposing a minimum number of 
> > flows,
> > the UA potentially "suffers" (spending scarce resources such as battery
> > power). And anyway, when the UA sets up less flows the consequences are 
> > for
> > its user.
>
> in the above situation, assuming that network has two proxies, if UA
> decides to establish only one flow instead of two, then the likely hood
> is 50% that incoming call to the UA cannot be establishes.  it thus
> makes no sense for the UA to violate network policy on number of flows.
>

This hints at a very different setup than I had in my mind. I always assumed 
that any flows beyond the first would be redundant, to be used when the 
first ("primary") flow would fail. Your text here suggests something more 
resembling a load balancing situation, with very different call completion 
probabilities

Indeed, in the case above the UA would need to know that it MUST use two 
flows (or miss 50% of its calls). An alternative setup would have the 
proxies (say P1 and P2) exchange requests when there is no direct connection 
to the UA. If P1 and P2 are setup as a DNS proxyfarm, the routing logic that 
each would follow could be something like "if (UA flow) then forward to UA; 
else if (not from other proxy) forward to other proxy; else send 480"

This looks like an issue to be discussed in the draft (ie what assumptions 
are made regarding how those N flows are used)

Regards,

Jeroen






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From sip-bounces@ietf.org Thu Jul 06 01:15:32 2006
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Subject: [Sip] Outbound Comment: binding key
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 From section 6.1:

   When a reg-id header parameter is present in a Contact header field
   value in a REGISTER request, the corresponding binding is between an
   AOR and the combination of instance-id and reg-id.

I don't think this is quite correct. This statement appears to be
asserting that {instance-id,reg-id} is a unique key to a binding
associated with an AOR. This would, in turn, imply that any successful
registration attempt with an {instance-id,reg-id} that matches an
existing binding would overwrite the existing record.

I don't think that is what's intended, since section 6.2 contains the
following text:

   o  If two bindings have the same instance-id and reg-id, the proxy
      SHOULD prefer the contact that was most recently updated.

If the binding is only to {instance-id,reg-id}, then that text is
nonsensical: you can't have two identical bindings to the same AOR.

I suspect that the text in 6.1 should actually read something more like:

   When a reg-id header parameter is present in a Contact header field
   value in a REGISTER request, the corresponding binding is between an
   AOR and the combination of instance-id, reg-id, and network flow.

/a


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From sip-bounces@ietf.org Thu Jul 06 01:19:33 2006
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The current document doesn't mention q-value handling, which is probably worth some treatment.

In internal discussions, we came up with realistic scenarios whereby a client may reasonably end up maintaining two connections with very different characteristics (UDP versus TLS would be an extreme example), and be willing to use both of them, but *strongly* prefer one over the other.

To handle this situation, it might make sense to put stuff in 6.2 that basically says that, for contacts with matching instance-ids, the proxy MUST NOT try a contact until all contacts at a higher q-value have been tried first. (I could go along with a weaker strength, but I really do think a prescriptive approach is the only thing that allows the clients to handle the situation I describe above in a manner that is predictable to the users).

/a


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From sip-bounces@ietf.org Thu Jul 06 01:53:31 2006
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This one, I might just be confused about, but I combed the draft and 
can't figure out how it is supposed to work.

Section 5 of the outbound document places a number of requirements on 
edge proxies that must be implemented for outbound to work. However, 
negotiation of the outbound mechanism is between the client and the 
registrar. An edge-proxy that doesn't understand outbound will blithely 
shuttle messages back and forth between the client while they signal 
support for the outbound mechanism.

During registration, all a non-outbound edge-proxy will see and care 
about is "Supported: path," which makes it fat and happy. It won't care 
about the funky bits on the end of the Contact. It won't care about the 
"Supported: outbound" header field in the REGISTER response. And it 
won't do anything suspicious that might give the client or the registrar 
some idea that things are about to go quite wrong. The client and 
registrar have done the secret handshake, and are absolutely certain 
that outbound handling is alive and well.

I think this can be fixed with a little more behavior around what 
edge-proxies that support this draft are required to do. Specifically: 
since section 5.1 already has some pretty strong normative language that 
requires edge-proxies to insert Path header fields with certain 
properties, it's a rather small leap to dictate that the form of the 
URIs in these Path header fields must include something that the 
registrar can use to verify outbound support.

For example:

   When an Edge Proxy receives a registration request with a reg-id
   header parameter in the Contact header field, it MUST form a flow
   identifier token that is unique to this network flow.  The Edge Proxy
   MUST insert this token into a URI referring to this proxy and place
   this URI into a Path header field as described in RFC 3327 [RFC3327].
   The token MAY be placed in the userpart of the URI. The URI MUST
   include an ";outbound" parameter.

With that in place, the registrar can now look at an inbound REGISTER 
request and (in addition to checking the Contact header field) look 
through the Path header fields to ensure that all of them contain URIs 
with ";outbound" parameters. If they don't, it can take evasive action 
(perhaps not returning the "Supported: outbound" header field in the 
response?) that prevents things from going wrong.

/a

----

P.S. There is one corner case that this doesn't seem to cover: the 
situation in which the client has a locally provisioned default outbound 
proxy. This locally provisioned default outbound proxy won't use Path at 
all, but it will remain in the signaling for all requests from the 
client. I'm not certain how often this will become a problem, and 
working around it would require something similar to the Path indication 
I describe above, but on Via header fields. That feels like a really 
ugly hack, and I'm not sure it's worth the inelegance to handle cases 
that will be rare, if they exist at all.

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To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Jeroen van Bemmel writes:

 > An alternative setup would have the 
 > proxies (say P1 and P2) exchange requests when there is no direct
 > connection to the UA. 

no thank you.  i don't want to consume resources of both proxies for the
request and, more importantly, i don't want to loose reliability.

conclusion: in some proxy configurations UA MUST set up more than one
flow or miss service.

-- juha

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From sip-bounces@ietf.org Thu Jul 06 02:29:28 2006
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Subject: [Sip] Outbound Comment: q-value
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Adam Roach writes:

 > To handle this situation, it might make sense to put stuff in 6.2
 > that basically says that, for contacts with matching instance-ids,
 > the proxy MUST NOT try a contact until all contacts at a higher
 > q-value have been tried first. (I could go along with a weaker
 > strength, but I really do think a prescriptive approach is the only
 > thing that allows the clients to handle the situation I describe
 > above in a manner that is predictable to the users).

i you suggesting that proxies consult each other about the q-value UA
used when registering with each proxy before forwarding a request to the
UA?  if so, your suggesting makes no sense.

-- juha

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Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04
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=20
Hi Jeroen,

>This hints at a very different setup than I had in my mind. I=20
>always assumed that any flows beyond the first would be=20
>redundant, to be used when the first ("primary") flow would fail.

The concept you had in your mind seems to be a possible implementation
but by no means mandated by -04 version. In that version the network=20
treats all the flows towards an UA in an equal way. It appears already
in the definitions:

  outbound-proxy-set A set of SIP URIs (Uniform Resource Identifiers)
      that represents each of the outbound proxies (often Edge Proxies)
      with which the UA will attempt to maintain a direct flow.  The
      first URI in the set is often refereed to as the primary outbound
      proxy and the second as the secondary outbound proxy.  There is no
      difference between any of the URIs in this set, nor does the
      primary/secondary terminology imply that one is preferred over the
      other.

The normative text leaves it open how the location service will select
a specific contact and related flow for an incoming request. In my
understanding the implementation may either always select one specific
flow as long as it works but it may also balance the load and select
a different contact for different requests. In case the location service
is distributed and colocated with different outbound proxies, it might
prefer using always the local flow towards the UA, as long as it works.

This is why I had to send this proposal for sip-outbound-discovery
draft: http://www1.ietf.org/mail-archive/web/sip/current/msg14985.html
However I have not received any comments to that and will highlight it
as an open issue in the WG meeting.

Regards,

Erkki

>-----Original Message-----
>From: ext Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]=20
>Sent: 06.July.2006 07:58
>To: Juha Heinanen
>Cc: Garcia Miguel.An (Nokia-NRC/Helsinki); sip@ietf.org;=20
>Cullen Jennings; Rohan Mahy
>Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
>
>Juha,
>
>> > For flows beyond the maximum the network "suffers" (just like the=20
>> > registrar
>> > determines the expiration time). But by imposing a minimum=20
>number of=20
>> > flows,
>> > the UA potentially "suffers" (spending scarce resources=20
>such as battery
>> > power). And anyway, when the UA sets up less flows the=20
>consequences are=20
>> > for
>> > its user.
>>
>> in the above situation, assuming that network has two proxies, if UA
>> decides to establish only one flow instead of two, then the=20
>likely hood
>> is 50% that incoming call to the UA cannot be establishes.  it thus
>> makes no sense for the UA to violate network policy on=20
>number of flows.
>>
>
>This hints at a very different setup than I had in my mind. I=20
>always assumed=20
>that any flows beyond the first would be redundant, to be used=20
>when the=20
>first ("primary") flow would fail. Your text here suggests=20
>something more=20
>resembling a load balancing situation, with very different=20
>call completion=20
>probabilities
>
>Indeed, in the case above the UA would need to know that it=20
>MUST use two=20
>flows (or miss 50% of its calls). An alternative setup would have the=20
>proxies (say P1 and P2) exchange requests when there is no=20
>direct connection=20
>to the UA. If P1 and P2 are setup as a DNS proxyfarm, the=20
>routing logic that=20
>each would follow could be something like "if (UA flow) then=20
>forward to UA;=20
>else if (not from other proxy) forward to other proxy; else send 480"
>
>This looks like an issue to be discussed in the draft (ie what=20
>assumptions=20
>are made regarding how those N flows are used)
>
>Regards,
>
>Jeroen
>
>
>
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

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From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Juha:

While I understand that you want to impose reliability to your users, by 
requesting them to establish 2 or more flows, you also need to 
understand that this might not be always the case, because the cost of 
such reliability might be that my battery is drained in half of the 
time, due to the STUN refreshes.

So, what I am advocating is for a way to decoupling reliability from NAT 
traversal. There must be a way for the network to indicate more than one 
outbound URI, used for backup purposes, but still instruct or let the UA 
to establish a single flow. The current text in the draft seems to 
prevent this use case.

/Miguel

Juha Heinanen wrote:
> Jeroen van Bemmel writes:
> 
>  > An alternative setup would have the 
>  > proxies (say P1 and P2) exchange requests when there is no direct
>  > connection to the UA. 
> 
> no thank you.  i don't want to consume resources of both proxies for the
> request and, more importantly, i don't want to loose reliability.
> 
> conclusion: in some proxy configurations UA MUST set up more than one
> flow or miss service.
> 
> -- juha

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

_______________________________________________
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From sip-bounces@ietf.org Thu Jul 06 03:19:52 2006
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Subject: RE: [Sip] Outbound: Validating if the proxy supports outbound or not
Date: Thu, 6 Jul 2006 10:10:24 +0300
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Hi,

The comments sent on the list during last few days seem to
indicate reasonably large concern that it is not sufficient
for the UA to check that the configured proxy URI contains
keepalive=3Dstun parameter, in order for the UA to make sure
that the proxy actually supports outbound. Thus I will=20
expect that the next version of outbound draft will address
this issue. Note that the authors of the draft have promised
to raise this as an open issue in the WG meeting, so please
be prepared to contribute that discussion.

The comments I am referring to are this (mine included):

Bob Penfield:
>Does that mean that a UA should not send STUN if the=20
>keepalive=3Dstun is not present?
>
>If so, will outbound be useful prior to the completion of the various=20
>configuration mechanisms? And is it acceptable for a=20
>mis-configured UA to cause the proxy or the registrar to crash or
mis-behave?
>
>A mechanism such as the proposed Proxy-Require/Require method=20
>similar to sec-agree allows the UA to determine that the proxy and=20
>registrar support STUN before it sends and potentially harmful
messages.

Adam Roach:
>I think this can be fixed with a little more behavior around what=20
>edge-proxies that support this draft are required to do. Specifically:=20
>since section 5.1 already has some pretty strong normative=20
>language that=20
>requires edge-proxies to insert Path header fields with certain=20
>properties, it's a rather small leap to dictate that the form of the=20
>URIs in these Path header fields must include something that the=20
>registrar can use to verify outbound support.
>
>With that in place, the registrar can now look at an inbound REGISTER=20
>reFrom sip-bounces@ietf.org Thu Jul 06 03:19:52 2006
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Subject: RE: [Sip] Outbound: Validating if the proxy supports outbound or not
Date: Thu, 6 Jul 2006 10:10:24 +0300
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From: <Erkki.Koivusalo@nokia.com>
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Hi,

The comments sent on the list during last few days seem to
indicate reasonably large concern that it is not sufficient
for the UA to check that the configured proxy URI contains
keepalive=3Dstun parameter, in order for the UA to make sure
that the proxy actually supports outbound. Thus I will=20
expect that the next version of outbound draft will address
this issue. Note that the authors of the draft have promised
to raise this as an open issue in the WG meeting, so please
be prepared to contribute that discussion.

The comments I am referring to are this (mine included):

Bob Penfield:
>Does that mean that a UA should not send STUN if the=20
>keepalive=3Dstun is not present?
>
>If so, will outbound be useful prior to the completion of the various=20
>configuration mechanisms? And is it acceptable for a=20
>mis-configured UA to cause the proxy or the registrar to crash or
mis-behave?
>
>A mechanism such as the proposed Proxy-Require/Require method=20
>similar to sec-agree allows the UA to determine that the proxy and=20
>registrar support STUN before it sends and potentially harmful
messages.

Adam Roach:
>I think this can be fixed with a little more behavior around what=20
>edge-proxies that support this draft are required to do. Specifically:=20
>since section 5.1 already has some pretty strong normative=20
>language that=20
>requires edge-proxies to insert Path header fields with certain=20
>properties, it's a rather small leap to dictate that the form of the=20
>URIs in these Path header fields must include something that the=20
>registrar can use to verify outbound support.
>
>With that in place, the registrar can now look at an inbound REGISTER=20
>request and (in addition to checking the Contact header field) look=20
>through the Path header fields to ensure that all of them contain URIs=20
>with ";outbound" parameters. If they don't, it can take evasive action=20
>(perhaps not returning the "Supported: outbound" header field in the=20
>response?) that prevents things from going wrong.
=20
Erkki Koivusalo:
>The specific concerns that I have (and would like the WG to=20
>consider in the meeting) are closely related to what you told:
>
>>We got very strong feed back on one point - don't send=20
>>STUN messages to a proxy unless you know it supports them.=20
>
>Does the WG consider checking the configured URI as a safe enough
>check for the UA to ensure that the SIP proxy support STUN ?
>
>In my opinion it is not. As an example a vendor X could implement
>a compliant but bad UA which provides an option for the user to=20
>configure the settings via an User Interface e.g. like this:
>"Is the phone behind a NAT: Yes/No". When the user selects
>"Yes" the UA would add the keepalive=3Dstun flag to the proxy URI
>even if the user does not have a clue if the proxy actually
>supports outbound or not. There is nothing in outbound draft
>to prevent such unsafe implementations to appear.

Jeroen van Bemmel:
>The point is that these are really 2 separate issues:
>1. Discovering the set of outbound proxy URIs (with keepalive=3Dstun=20
>parameters in them)
>2. Confirming that a given proxy in the set actually supports=20
>outbound (ie validating the keepalive=3Dstun information)
>
>Of these 2 options, (1) is indeed a separable problem.=20
>(2) - to me - is not, for me outbound is not acceptable=20
>without something to address (2).

Regards,

Erkki

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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Date: Thu, 6 Jul 2006 10:09:06 +0300
From: Juha Heinanen <jh@tutpro.com>
To: Miguel Garcia <Miguel.An.Garcia@nokia.com>
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
In-Reply-To: <44ACB4EB.4060305@nokia.com>
References: <44AB9820.4050200@nokia.com>
	<17579.41018.861426.943160@rautu.tutpro.com>
	<44ABA2AE.3010001@nokia.com>
	<17580.3033.661988.839303@rautu.tutpro.com>
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Cc: sip@ietf.org, Cullen Jenniquest and (in addition to checking the Contact header field) look=20
>through the Path header fields to ensure that all of them contain URIs=20
>with ";outbound" parameters. If they don't, it can take evasive action=20
>(perhaps not returning the "Supported: outbound" header field in the=20
>response?) that prevents things from going wrong.
=20
Erkki Koivusalo:
>The specific concerns that I have (and would like the WG to=20
>consider in the meeting) are closely related to what you told:
>
>>We got very strong feed back on one point - don't send=20
>>STUN messages to a proxy unless you know it supports them.=20
>
>Does the WG consider checking the configured URI as a safe enough
>check for the UA to ensure that the SIP proxy support STUN ?
>
>In my opinion it is not. As an example a vendor X could implement
>a compliant but bad UA which provides an option for the user to=20
>configure the settings via an User Interface e.g. like this:
>"Is the phone behind a NAT: Yes/No". When the user selects
>"Yes" the UA would add the keepalive=3Dstun flag to the proxy URI
>even if the user does not have a clue if the proxy actually
>supports outbound or not. There is nothing in outbound draft
>to prevent such unsafe implementations to appear.

Jeroen van Bemmel:
>The point is that these are really 2 separate issues:
>1. Discovering the set of outbound proxy URIs (with keepalive=3Dstun=20
>parameters in them)
>2. Confirming that a given proxy in the set actually supports=20
>outbound (ie validating the keepalive=3Dstun information)
>
>Of these 2 options, (1) is indeed a separable problem.=20
>(2) - to me - is not, for me outbound is not acceptable=20
>without something to address (2).

Regards,

Erkki

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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To: Miguel Garcia <Miguel.An.Garcia@nokia.com>
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Miguel Garcia writes:

 > So, what I am advocating is for a way to decoupling reliability from NAT 
 > traversal. There must be a way for the network to indicate more than one 
 > outbound URI, used for backup purposes, but still instruct or let the UA 
 > to establish a single flow. The current text in the draft seems to 
 > prevent this use case.

then suggest some specific mechanism to achieve that you want.  for
example, like someone proposed, introduce some parameter to register 200
OK that tells than one flow is enough.  you cannot simply say that "a UA
may establish only one flow if it doesn't feel like setting up more".

-- juha

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Miguel Garcia writes:

 > So, what I am advocating is for a way to decoupling reliability from NAT 
 > traversal. There must be a way for the network to indicate more than one 
 > outbound URI, used for backup purposes, but still instruct or let the UA 
 > to establish a single flow. The current text in the draft seems to 
 > prevent this use case.

then suggest some specific mechanism to achieve that you want.  for
example, like someone proposed, introduce some parameter to register 200
OK that tells than one flow is enough.  you cannot simply say that "a UA
may establish only one flow if it doesn't feel like setting up more".

-- juha

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Juha Heinanen wrote:
> Miguel Garcia writes:
> 
>  > So, what I am advocating is for a way to decoupling reliability from NAT 
>  > traversal. There must be a way for the network to indicate more than one 
>  > outbound URI, used for backup purposes, but still instruct or let the UA 
>  > to establish a single flow. The current text in the draft seems to 
>  > prevent this use case.
> 
> then suggest some specific mechanism to achieve that you want.  for
> example, like someone proposed, introduce some parameter to register 200
> OK that tells than one flow is enough.  you cannot simply say that "a UA
> may establish only one flow if it doesn't feel like setting up more".

That solution would work for me, but I think it was discussed in the 
past and, I dont' know really why, it was somehow rejected.

The other alternative solution it comes to my mind is to play with the 
Priority of the SRV record. So we could say that the UA establishes a 
flow to each server defined with the lower Priority. Servers defined 
with higher priority will be used in cases of failure.

Within the same priority, the UA establishes one flow per server listed 
there. So, if the network requests several flows, it configures several 
SRV records with the lower priority (say, 0). If the network requests 
one flow, it configures one SRV record with the lower priority. This 
will cover all the use cases we have been discussing.

/Miguel

> 
> -- juha

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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From sip-bounces@ietf.org Thu Jul 06 03:42:38 2006
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Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04
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Hi,

>That solution would work for me, but I think it was discussed in the=20
>past and, I dont' know really why, it was somehow rejected.

Rohan rejected the proposal like this:

>As a scope issue, the chairs (previous and current) and the ADs (at
>time still Allison and Jon) directed the editors to make outbound=20
>separate from gruu, initial proxy discovery/configuration, and=20
>specification of record-routing behavior for high reliability and=20
>per-call state cookies.  We then had WG consensus on this approach. I=20
>think that was a wise choice, and I fear that adding new=20
>parameters now will only unwind that consensus.

So nothing specific against the proposed mechanism, just a
general concern of loading outbound draft with too many
different mechanisms.

Rohan also suggested using DNS for the purpose, however
without mentioning yet any specific way how to use it:

>I can think of a number of ways to make this work where the intended=20
>semantics are entirely embedded in DNS.  Alternatively you can=20
>discover=20
>*candidate* outbound-proxy-URIs via DNS and use a max-flows value=20
>configured/discovered elsewhere to decide how to derive/self-configure=20
>the actually outbound-proxy-set.

What comes to your suggestion on using DNS...

>The other alternative solution it comes to my mind is to play with the=20
>Priority of the SRV record. So we could say that the UA establishes a=20
>flow to each server defined with the lower Priority. Servers defined=20
>with higher priority will be used in cases of failure.

... it seems that there is a drawback. If SRV records would
be used like that, then a service provider could not any more
use the SRV RRs with same level of priority for balancing the
load for the UA population, wishing each UA to maintain lower
number of flows than the number of servers in that priority level.
E.g. balancing the load to 4 servers would require every UA to
maintain 4 flows, which would be impractical and cause extra
load (for keepalives and registration refreshes) for all those
proxies.

Regards,

Erkki

>-----Original Message-----
>From: ext Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
>Sent: 06.July.2006 10:22
>To: Juha Heinanen
>Cc: sip@ietf.org; Cullen Jennings; Rohan Mahy
>Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
>
>Juha Heinanen wrote:
>> Miguel Garcia writes:
>>=20
>>  > So, what I am advocating is for a way to decoupling=20
>reliability from NAT=20
>>  > traversal. There must be a way for the network to=20
>indicate more than one=20
>>  > outbound URI, used for backup purposes, but still=20
>instruct or let the UA=20
>>  > to establish a single flow. The current text in the draft=20
>seems to=20
>>  > prevent this use case.
>>=20
>> then suggest some specific mechanism to achieve that you want.  for
>> example, like someone proposed, introduce some parameter to=20
>register 200
>> OK that tells than one flow is enough.  you cannot simply=20
>say that "a UA
>> may establish only one flow if it doesn't feel like setting up more".
>
>That solution would work for me, but I think it was discussed in the=20
>past and, I dont' know really why, it was somehow rejected.
>
>The other alternative solution it comes to my mind is to play with the=20
>Priority of the SRV record. So we could say that the UA establishes a=20
>flow to each server defined with the lower Priority. Servers defined=20
>with higher priority will be used in cases of failure.
>
>Within the same priority, the UA establishes one flow per=20
>server listed=20
>there. So, if the network requests several flows, it=20
>configures several=20
>SRV records with the lower priority (say, 0). If the network requests=20
>one flow, it configures one SRV record with the lower priority. This=20
>will cover all the use cases we have been discussing.
>
>/Miguel
>
>>=20
>> -- juha
>
>--=20
>Miguel A. Garcia           tel:+358-50-4804586
>sip:miguel.an.garcia@openlaboratory.net
>Nokia Research Center      Helsinki, Finland
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

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Koivusalo Erkki (Nokia-TP-MSW/Helsinki) wrote:

> What comes to your suggestion on using DNS...
> 
>> The other alternative solution it comes to my mind is to play with the 
>> Priority of the SRV record. So we could say that the UA establishes a 
>> flow to each server defined with the lower Priority. Servers defined 
>> with higher priority will be used in cases of failure.
> 
> ... it seems that there is a drawback. If SRV records would
> be used like that, then a service provider could not any more
> use the SRV RRs with same level of priority for balancing the
> load for the UA population, wishing each UA to maintain lower
> number of flows than the number of servers in that priority level.
> E.g. balancing the load to 4 servers would require every UA to
> maintain 4 flows, which would be impractical and cause extra
> load (for keepalives and registration refreshes) for all those
> proxies.
> 

So why would you publish 4 servers with the same Priority in SRV, but 
don't want 4 flows, but still want load balancing among what? Strange...

/Miguel

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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=20
Hi,
=20
This again brings up the question: should the whole usage of STUN for =
keep-alive be described in a separate draft? The outbound spec would =
then just refer to that.
=20
Especially IF we can agree on the fact that the STUN keep-alive is a =
stand-alone feature (not necessarily always used together with =
outbound), I don't see why the STUN keep-alive mechanism is described in =
one draft, and then in another draft we describe how to detect whether a =
proxy supports it or not. Why not put everything in one draft, then?
=20
Regards,
=20
Christer
=20
=20

________________________________

From: Cullen Jennings [mailto:fluffy@cisco.com]
Sent: Tue 04/07/2006 22:37
To: <Erkki.Koivusalo@nokia.com>
Cc: sip@ietf.org; rohan@ekabal.com; dean.willis@softarmor.com
Subject: Re: [Sip] Ready for WGLC on Outbound?




I think people have stopped posting on this because they can't say=20
anything that they have not already said. Don't take that as they agree.

We got very strong feed back on one point - don't send STUN messages=20
to a proxy unless you know it supports them. People felt strongly=20
about that - I don't know why - but there was lots of people that=20
agreed with that.

I imagine people have sent me around 10 options on how to get the=20
stun supported information and the set of outbound proxy URIs -=20
certainly DNS and DHCP are some of them. We can write some drafts on=20
various ways to do it. Right now my concern is making sure that no=20
matter which of these many ways the WG decided to move forward with,=20
outbound will work. So far every one of the possibilities I have=20
heard would be possible with the current outbound. Given that, I'm=20
not sure what we need to change in outbound. I'm not even sure what=20
you are proposing that we change in outbound - I think what you are=20
proposing is we need to hold up outbound until we choose which one of=20
the 10 we will use and furthermore you feel very strongly on which=20
one of the 10 we should use. I'm not keen on this - I think we have=20
shown they are separable problems. The WG decided on this separation=20
of problems awhile back - I think it was in Vancouver - and many=20
people supported it. I think it was a good idea  - we have shown we=20
can move outbound forward before we have determined all the ways that=20
this configuration information is delivered to the UA. I would be=20
very hesitant to reverse this decision now.

I don't think Dean's posts were sent as decries of the requirements=20
from the chair but purely as his opinion in trying to help bring the=20
discussion to consensus. But I do think the draft meets the=20
requirements and ideas he was outlining - I think the current draft=20
does meet the requirements he outlines. It does have a way for the=20
"discovering whether a proxy supports the extensions of outbound" and=20
on the topic of "there are two ways we can establish compatibility=20
with an extensions", the WG rejected option 2 and we have gone with 1.

Cullen (with my outbound author hat on)


On Jul 4, 2006, at 3:50 AM, <Erkki.Koivusalo@nokia.com>=20
<Erkki.Koivusalo@nokia.com> wrote:

>
> Hi Rohan,
>
> Sorry to be persistent with this issue, but what you say
> seems to conflict with what was discussed in March.
>
>> The only time a UAC sends a STUN request as directed by
>> outbound -04 is when the URI in the outbound-proxy-set
>> includes the keepalive=3Dstun parameter. That DHCP does not
>> provide URI is a limitation of the SIP DHCP configuration
>> mechanism, not outbound. If you want to write a draft on
>> how to "shift-up" to using keepalives from some SIP or
>> external stimulus, you can still do this.
>
> What you say seems to suggest that you agree there is a problem
> when using DHCP SIP server option (as it is currently defined)
> for finding out if the discovered proxy supports outbound and
> STUN keepalives, but the solution is not in scope of Outbound
> draft itself.
>
> However I came to a different understanding based on the emails
> seen in the list earlier. For instance this email from Dean
> Willis (SIP WG chair) seems to suggest that the problem should
> be in scope of the outbound draft:
> http://www1.ietf.org/mail-archive/web/sip/current/msg13667.html
>
> "Ok, I think we've boiled this down to a new requirement:
>
> for the "outbound" technique to work, we MUST have a means of
> discovering whether a proxy supports the extensions of outbound.
>
> While we could do that by doing a REGISTER exchange that includes
> an appropriate option-tag in Require:, it would be more efficient
> if the proxy discovery mechanism informed us of this detail. So,
> we posit that the Outbound draft SHOULD select and discuss such a
> mechanism."
>
> And in yet another email
> http://www1.ietf.org/mail-archive/web/sip/current/msg13692.html
> Dean tells this:
>
> "There are two ways we can establish compatibility with an extensions:
>
> 1) By foreknowledge
> 2) By experimentation -- try it and see if it works.
>
> Both have limitations:
>
> In the case of #1, we might have incorrect knowledge. So we'll=20
> probably
> end up testing anyhow . . .
>
> In the case of #2, the experiment might cause a meltdown. So the=20
> "probe
> for discovery" must be made completely safe. Also, we can=20
> eventually end
> up with so many wierd hooks and idiosyncratic implementations that we
> end up doing a BUNCH of probing before we do any real=20
> communicating. And
> it's possible for the conditions to deteriorate so much that probes=20
> for
> two different possible conditions are mutually exclusive -- if one
> condition applies, the probe for another will cause a meltdown.
>
> So what happens if we start STUN-probing a proxy on its SIP port=20
> and it
> doesn't support STUN? My guess is, it's random.
>
> ...
>
> One approach to "forward compatibility" design is to make sure that we
> can use both foreknowledge and experimentation -- the "trust but=20
> verify"
> model.
>
> ...
>
> While it's easy to decide whether a configured registrar URI has a
> keepalive parameter, what happens to proxies we select based on
> something besides direct configuration?
>
> Specifically,
>
> 1) Can the DHCP option for default proxy present the keepalive
> parameter?"
> ----
>
> During these three months after seeing those comments in the list
> I have been expecting the outbound draft to provide answer to
> the question how UA could (within SIP signalling exchange)
> safely to check whether or not a proxy supports STUN keepalive.
> Two mechanims have been suggested how the UA could do this in
> a simple way:
>
> 1. The UA to check that the Path header in 200 OK returned for
>    REGISTER would contain keepalive=3Dstun parameter for the proxy URI
>
> 2. The UA to add Proxy-Reequire: stun-keepalive parameter into
>    the REGISTER request and the proxy to remove this tag from
>    the request before forwarding it towards the registrar
>
> For me, any of those mechanisms would sound feasible and for
> safety reasons I would hope the draft to deny the UA for sending
> STUN keepalives to the proxy which did not pass such a test.
>
>> the same problem exists for the "comp" parameter for example.
>
> Yes but please bear in mind that for comp=3Dsigcomp URI parameter
> RFC 3486 Compressing SIP gives several solutions how the UA
> would be able to check if the proxy supports SigComp while
> outbound does not define any other than relying on configuration.
> A cite from RFC 3486:
>
> 4.1   Obtaining a SIP or SIPS URI with comp=3Dsigcomp
>
>    For requests within a dialog, a next-hop URI with the =
comp=3Dsigcomp
>    parameter is obtained from a Record-Route header field when the
>    dialog is established.  A client sending a request outside a dialog
>    can also obtain SIP URIs with comp=3Dsigcomp in a Contact header=20
> field
>    in a 3xx or 485 response to the request.
>
>    However, clients establishing a session will not typically be=20
> willing
>    to wait until the dialog is established in order to begin=20
> compressing
>    messages.  One of the biggest gains that SigComp can bring to=20
> SIP is
>    the ability to compress the initial INVITE of a dialog, when the=20
> user
>    is waiting for the session to be established.  Therefore, clients
>    need a means to obtain a comp=3Dsigcomp URI from their outbound =
proxy
>    before the user decides to establish a session.
>
>    One solution to this problem is manual configuration.  However,
>    sometimes it is necessary to have clients configured in an=20
> automatic
>    fashion.  Unfortunately, current mechanisms for SIP client
>    configuration (e.g., using DHCP [6]) do not allow to provide the
>    client with URI parameters.  In this case, the client SHOULD=20
> send an
>    uncompressed OPTIONS request to its outbound proxy.  The outbound
>    proxy can provide an alternative SIP URI with the comp=3Dsigcomp
>    parameter in a Contact header field in a 200 OK response to the
>    OPTIONS.  The client can use this URI for subsequent requests that
>    are sent through the same outbound proxy using compression.
>
>    RFC 3261 [1] does not define how a proxy should respond to an=20
> OPTIONS
>    request addressed to itself.  It only describes how servers respond
>    to OPTIONS addressed to a particular user.  Section 11.2 of RFC=20
> 3261
>    says:
>
>       Contact header fields MAY be present in a 200 (OK) response and
>       have the same semantics as in a 3xx response.  That is, they may
>       list a set of alternative names and methods of reaching the=20
> user.
>
>    We extend this behavior to proxy servers responding to OPTIONS
>    addressed to them.  They MAY list a set of alternative URIs to
>    contact the proxy.
>
> What has been called for is that outbound draft would also
> specify one solution how the UA could check the STUN keepalive
> support of a proxy. In my opinion good proposals seem
> to exist how to do it in a reasonable way.
>
> Regards,
>
> Erkki
>
>> -----Original Message-----
>> From: ext Rohan Mahy [mailto:rohan@ekabal.com]
>> Sent: 04.July.2006 03:10
>> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>> Cc: dean.willis@softarmor.com; fluffy@cisco.com; Rohan Mahy;
>> sip@ietf.org; jbemmel@zonnet.nl
>> Subject: Re: [Sip] Ready for WGLC on Outbound?
>>
>> Hi Erkki,
>>
>> The only time a UAC sends a STUN request as directed by
>> outbound -04 is
>> when the URI in the outbound-proxy-set includes the keepalive=3Dstun
>> parameter.  This is 100% backwards safe.  We don't need any text to
>> prevent existing implementations from getting a bunch of STUN=20
>> requests.
>>
>> That DHCP does not provide URI is a limitation of the SIP DHCP
>> configuration mechanism, not outbound.  The same problem
>> exists for the
>> "comp" parameter for example.  If you want to write a draft on how to
>> "shift-up" to using keepalives or using compression from some SIP or
>> external stimulus, you can still do this.  No text in outbound
>> prevents
>> that from happening.
>>
>> thanks,
>> -rohan
>>
>>
>> On Jun 29, 2006, at 23:10, <Erkki.Koivusalo@nokia.com> wrote:
>>>
>>> Hi Rohan,
>>>
>>>>> Also, the issue is that what may be configured (no matter how you
>>>>> configure it) may not be a URI. It may simply be an IP
>> address/FQDN,
>>>>> which is then used to build a SIP-URI. I think we shall
>>>>> allow that, and make it more clear in the spec. [/CHH]
>>>>
>>>> the outbound spec defines a piece of configurable data (the
>>>> 'outbound-proxy-set'), which is a list of URIs.  This list
>> of URIs may
>>>> never have a one-to-one correspondence to a user or administrator
>>>> manually configurable field.  That piece of configurable data=20
>>>> may be
>>>> derived from several fields, manually configurable, discovered via
>>>> DHCP, DNS, the config package, P-Whatchamacallit headers, HTTP
>>>> download, or via a random character generator seeded by the=20
>>>> phase of
>>>> the moon.
>>>>
>>>> The outbound-proxy-set was defined as a list of URIs because the UA
>>>> acts on each item in the list, because each URI is expected to be
>>>> places in a Route header, and because each URI is expected to
>>>> typically have a keepalive URI parameter in it.
>>>
>>> Like mentioned in other recent emails the problem is that if the UA
>>> derives the proxy URI(s) from DHCP SIP server option or e.g. the AOR
>>> of the user, in those cases the derived URI will NOT contain the
>>> keepalive URI parameter in it. DHCP just returns IP addresses or
>>> domain names instead of URIs with parameters.
>>>
>>> Thus in those cases it would be critical for the UA to be able
>>> to ensure in REGISTER - 200 OK sequence that the contacted proxy
>>> supports Outbound and can handle STUN requests sent to its SIP
>>> port (not to cause any damage to an uncompliant proxy). I fully
>>> agree with this proposal from Jeroen:
>>>
>>> --
>>> The text should then say "the UAC MUST wait for a 200 OK to REGISTER
>>> before
>>> it starts sending STUN packets. If the last Path header of the=20
>>> 200 OK
>>> does
>>> not contain keepalive=3Dstun, the UAC MUST NOT use STUN
>> towards this URI.
>>> If
>>> there are no Path headers and no "Supported: outbound", the UAC also
>>> MUST
>>> NOT use STUN towards the registrar.". It could still use TCP
>> keepalive
>>> if it
>>> supports that
>>> --
>>>
>>> Please excuse me if I am just repeating myself and what e.g. Jeroen
>>> has said but I believe the importance of this issue justifies some
>>> redundancy :-)
>>>
>>> Regards,
>>>
>>> Erkki
>>
>>

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Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>, <sip@ietf.org>,
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=20
Hi Miguel,
=20
I agree with your text proposal. It is pretty much the same which was =
proposed weeks ago, when the issue was brought up (eventhough the =
wording may have been a little different).
=20
Regards,
=20
Christer
=20

________________________________

From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
Sent: Wed 05/07/2006 12:44
To: sip@ietf.org; Cullen Jennings; Rohan Mahy
Subject: [Sip] A coment to draft-ietf-sip-outbound-04



Reading Section 4.2 of Outbound-04, there is a sentence that came to my
attention.

    For each outbound proxy URI in the set, the UA MUST send a
    REGISTER in the normal way using this URI as the default outbound
    proxy.

I think it is not appropriate to use a MUST strength in this case. I
suspect that the spirit of the draft is to offer high availability by
making the UA to establish one flow per URI, which is a good spirit.
However, one has to evaluate the cost of establishing one flow per URI,
which might be quite high in terms of battery consumption, for example
(due to STUN refreshes).

I also believe the MUST is misused here because it is not an absolute
requirement for the specification. In other words, the SIP outbound
procedures work OK even if the UA establishes just one flow (even when
it is provisioned with several outbound URIs). Nothing is broken.

So, we should be looking at adding a paragraph that describes the
benefit of establishing one flow per outbound URI, but letting up to the
UA's policy to decide on the absolute number of established flows.

I would suggest to replace the above mentioned sentence with something
around these lines:

    For each outbound proxy URI in the set, the UA MAY send a
    REGISTER in the normal way using this URI as the default outbound
    proxy. Sending one REGISTER per outbound proxy URI decreases the
    likelihood of interrupted service due to dropped connections at the
    cost of maintaining a living flow which each outbound proxy server.

Comments?

/Miguel

--
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Hi Miguel,

>So why would you publish 4 servers with the same Priority in SRV, but=20
>don't want 4 flows, but still want load balancing among what? =20

What I meant is that a service provider might provide e.g.
4 proxies for load balancing (over the whole UA population)=20
while deciding that 2 flows per UA would be sufficient for=20
reliability. Note that DNS SRV allows the UAs to select=20
different servers for their flows, like this:

UA1 would connect to P1 and P2
UA2 would connect to P1 and P3
UA3 would connect to P3 and P4

If the service provider aims to have different SRV weights
for each of those proxies to get bigger amount of load to
servers with more capacity and less amount of load for the
rest of the servers, it would no more be possible with the
solution you outlined. Each of the UAs would create a flow
to each server and assuming that each of the flows will be
treated equally, each of the proxies would get equal amount=20
of the total load.

---
An example with 3 proxies to clarify what I mean:

With legacy TCP SIP the DNS SRV is used so that each=20
UA would create a connection to single proxy only.
So assume we have 100 UAs so that each of those will
generate and recieve equal amount of traffic over 3
proxies.

P1 has SRV weight 5 and 50 UAs will connect to it
P2 has SRV weight 3 and 30 UAs will connect to it
P3 has SRV weight 2 and 20 UAs will connect to it

In this TCP SIP example P1 will get 50%, P2 30%
and P2 20% of the load within the domain.

With your suggestion with outbound the situation=20
would be changed so that each of the 100 UA would
create connections to each of the three proxies.
If those connections are handled in an equal way
the traffic will be distributed equally to each
of the proxies, so each will get 33% of the load.

However if the service provider would decide that
for reliability one connection is sufficient the
result would be the same as without outbound i.e.

P1: 50%   P2: 30%   P3:20%

If the service provider decides that two flows per
UA is needed for reliability and flows are treated
equally the calculation is no that trivial but should
result pretty similar load balancing figures:

50 UAs would select P1 for the primary flow
	30 of them would select P2 for the secondary flow
	20 of them would select P3 for the secondary flow

30 UAs would select P2 for the primary flow
	21 of them would select P1 for the secondary flow
	 9 of them would select P3 for the secondary flow

20 UAs would select P3 for the primary flow
	12 of them would select P1 for the secondary flow
	 8 of them would select P2 for the secondary flow

So the result would be altogether this:

P1 would have 83 flows i.e. 41 % of the load
P2 would have 68 flows i.e. 34 % of the load
P3 would have 49 flows i.e. 25 % of the load

This is why I suggest that your proposal would make
a considerable difference for the load balancing that
service provider can make. It would mean equal load
for each of the servers while SRV RRs usually enable
the provider to share the load unequally to different
proxies based on their capacities.

Regards,

Erkki

>-----Original Message-----
>From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
>Sent: 06.July.2006 10:46
>To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>Cc: Juha Heinanen; sip@ietf.org; Cullen Jennings; Rohan Mahy
>Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
>
>Koivusalo Erkki (Nokia-TP-MSW/Helsinki) wrote:
>
>> What comes to your suggestion on using DNS...
>>=20
>>> The other alternative solution it comes to my mind is to=20
>play with the=20
>>> Priority of the SRV record. So we could say that the UA=20
>establishes a=20
>>> flow to each server defined with the lower Priority.=20
>Servers defined=20
>>> with higher priority will be used in cases of failure.
>>=20
>> ... it seems that there is a drawback. If SRV records would
>> be used like that, then a service provider could not any more
>> use the SRV RRs with same level of priority for balancing the
>> load for the UA population, wishing each UA to maintain lower
>> number of flows than the number of servers in that priority level.
>> E.g. balancing the load to 4 servers would require every UA to
>> maintain 4 flows, which would be impractical and cause extra
>> load (for keepalives and registration refreshes) for all those
>> proxies.
>>=20
>
>So why would you publish 4 servers with the same Priority in SRV, but=20
>don't want 4 flows, but still want load balancing among what?=20
>Strange...
>
>/Miguel
>
>--=20
>Miguel A. Garcia           tel:+358-50-4804586
>sip:miguel.an.garcia@openlaboratory.net
>Nokia Research Center      Helsinki, Finland
>

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Subject: RE: [Sip] Comments on draft-koivusalo-sip-outbound-discovery-02
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Hi,

>> 1) I need to understand whether this updates RFC 3263 or=20
>just extends it. Part of this is what appears on the cover=20
>page and in the RFC index as and when it is finally approved,=20
>but the other part also relates to compatibility, see later comment.

>To answer your first question, I believe this draft updates RFC 3263.

Yesterday we discussed this question with Miguel and our
conclusion was that we are not sure if it extends or updates.
Depends on what these words actually mean. Can someone help
by providing definitions or examples of those ?

My opinion is that my draft only extends RFC 3263, so the
wording in the draft would have to be revised accordingly.
Reasoning behind my opinion is this:

- My draft at least does not make RFC 3263 obsolete.
  RFC 3263 would still be used when the UA tries to
  connect "legacy" SIP proxies without outbound support.

- The procedures in my draft suggest a different way
  for the UA to handle the DNS NAPTR and SRV responses
  like RFC 3263 does. However the UA uses these different
  procedures only when it specifically tries to find and
  contact 1-N proxies supporting outbound, instead of
  resolving SIP URIs for any other purpose.

- My draft is not independent of RFC 3263, it just highlights
  how the processing differs from what is described in 3263=20
  but does not repeat those procedures of the RFC that still=20
  apply.


>> 2) I think you have already acknowledged this, but the=20
>document requires revision to align with the terminology in=20
>the latest version (-04) of draft-ietf-sip-outbound.

I already tried to align the terminology, so please be
more specific on what kind of alignment is needed. If=20
you simply mean usage of word "backup" in certain context,
that will be removed if the corresponding use case and
requirement REQ-PROXY-FARM-03 will be dropped (like I
suggested in another thread). On the other hand it might
still be that my draft would continue promoting the way
of the domain have a pool of idle backup servers. However
it would probably require the outbound draft itself to
be updated like Adam suggested today:
---
>In internal discussions, we came up with realistic scenarios=20
>whereby a client may reasonably end up maintaining two=20
>connections with very different characteristics (UDP versus=20
>TLS would be an extreme example), and be willing to use both=20
>of them, but *strongly* prefer one over the other.
>
>To handle this situation, it might make sense to put stuff in=20
>6.2 that basically says that, for contacts with matching=20
>instance-ids, the proxy MUST NOT try a contact until all=20
>contacts at a higher q-value have been tried first.
---


>> 3) In section 5, as far as I understand, the NAPTR request=20
>is identical for a UA that supports RFC 3263 and for one that=20
>supports this extension. Therefore the difference is in the=20
>response.=20

Correct. What the draft currently says is this:

   The procedures here are invoked when a UA needs to send a REGISTER
   request via a proxy supporting SIP Outbound and set up a persistent
   flow towards that proxy.  The starting point is that the UA has a SIP
   or SIPS (secure SIP) URI of the proxy without port, transport or
   maddr parameters as the next hop.  According to RFC 3263 [2] the UA
   "SHOULD perform a NAPTR query for the domain in the URI."  This
   specification assumes the UA would proceed to perform the NAPTR query
   and related SRV queries as specified in RFC 3263, but with the
   exceptions defined below.

I would like to revise it slightly as follows:

   specification assumes the UA would proceed to perform the NAPTR query
   and related SRV queries as specified in RFC 3263, but processing the
   DNS NAPTR responses using the extensions as defined below.

>> Therefore the paragraph of UA requirements:
>>=20
>>    As per RFC 2915 [4], the client discards any NAPTR records whose
>>    services fields are not applicable.  A client resolving a SIPS URI
>>    MUST discard any services that do not contain "SIPS-O" as the
>>    protocol in the service field.  A client resolving a SIP URI MUST
>>    retain records with "SIP-O" and the transport protocols the client
>>    supports and SHOULD retain records with "SIPS-O", if the client
>>    supports TLS.
>>=20
>> must work with a NAPTR response that has no knowledge of=20
>outbound. Is the above paragraph compatible with this?=20

>And with respect your third question, I agree that the draft does not=20
>consider the case when Outbound is not deployed, so the text has to be=20
>updated to reflect the case.

Right now in the draft there is later a paragraph descriging a=20
fallback to use RFC 3263 procedures if the SIP-O records are=20
not defined for the domain:

   When performing NAPTR queries as specified in this document, if the
   UA is not able to resolve the URI of the proxy successfully to any IP
   address or if connecting to each of the resolved addresses fails,
   then the UA MAY try resolving the URI using the procedures defined in
   RFC 3263 [2].  When doing this fallback UA MUST NOT assume the
   resolved target host to support SIP Outbound.

What I have to do is to integrate this paragraph describing the fallback
behaviour with the paragraph you highlighted. I hope that would make
the story consistent and easily understandable. Also changing the MAY=20
to SHOULD or MUST is probably needed ?


>I have also a couple of other non-nit comments.
>
>1) I think the draft has to be a bit more strict on where this draft=20
>ends and when Outbound begins. At the moment I found this line quite=20
>fuzzy. Assuming this draft is about discovery, but not=20
>execution of the outbound procedures, I believe requirement=20
>REQ-OUTB-DISC-6 has not much to do with discovery, but with outbound.=20

I and Miguel did not reach an agreement about this, so Miguel
started a thread on a mailing list for the number of flows the
UA is expected to create. The resolution for my draft depends on
what happens to outbound draft. In my opinion it would still be
in the scope of the autodiscovery draft to define how many flows
the UA should set up if it discovers N proxies in the domain to
support outbound (and thus be able to derive outbound-proxy-set=20
consisting of N proxies). N flows might not always be the optimal
answer.

>The same applies to several=20
>paragraphs in Section 5, when the draft describes actions=20
>taken upon a reception of a 200 response  for a REGISTER request.

I admit that some of the stuff from my draft would apparently
belong in the scope of the outbound draft and could eventually
be removed from the autodiscovery draft. More specifically all
normative text about how the UA uses the flows (and perhaps
assigns the q values for the related Contacts) could belong
to this category. What I am not sure is whether those pieces
of text (autodiscovery of flows and the usage of those flows)=20
could be separated to two different drafts so that the story=20
would still be understandable and the reader to be able to
map these pieces together. This needs further study and also
cooperation with authors of outbound.


>2) The first paragraph in Section 2.4.4 suggests that a=20
>non-standardized autodiscovery mechanism will require the=20
>UA to insert a "Proxy-Require: outbound". Care must be=20
> taken to avoid that this becomes the de-facto=20
>behavior. It must noted that such Proxy-Require will require=20
>each of the proxies en route to the registrar to support=20
>outbound, which might not be the desired behavior. I would=20
>suggest to add some more text describing this case and=20
>highlighting the problem.

I agree with Miguel that this problem must be added to
make sure that such a solution would not be used. However
the exact formulation depends on if outbound draft will
be revised for Proxy-Require usage or not. This is currently
under discussion.


>3) IANA: something is not clear to me. Is this draft creating a new=20
>registry (as it suggest Sections 4 and 7), named "Registry for=20
>the NAPTR=20
>Resource Record Services Fiedl", or is it updating the=20
>existing registry=20
>at http://www.iana.org/assignments/sip-table
>
>I thought it would be the latter, but the fact that the existing=20
>registry refers to SRV records rather than NAPTR makes me doubt. Can=20
>someone clarify?

We agreed that my draft would update the existing registry.
Unfortunately the current register seems to incorrectly
refer to SRV RRs even if it is for NAPTR RRs. Miguel tries
to get the IANA registry fixed.

Regards,

Erkki

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From sip-bounces@ietf.org Thu Jul 06 06:09:29 2006
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: <ravishankar.shiroor@wipro.com>, <rjsparks@nostrum.com>
References: <532B18E13CF9E64380EF5FDDE265E071166B24@blr-m2-msg.wipro.com>
Subject: Re: [Sip] Update to fork-loop-fix submitted
Date: Thu, 6 Jul 2006 09:50:46 +0100
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Has it been discussed whether the hash function needs to be of cryptographic 
strength?

For example, hash functions used by hash tables are much simpler to compute, 
but typically have reasonable collision avoidance.  For example, one book I 
have [1] suggests a iteration step of a hash function of:

hash = hash * MULTIPLIER + data[n];

where MULTIPLIER is 31 or 37.

(This does assume that multiply is a single cycle operation - or nearly - 
which I believe it is for most modern processors.)

If you were paranoid you could compute two hashes with the different 
MULTIPLIERs and concatenate the result!

Pete.

[1] Kernigan & Pike, "The Practice of Programming", 1999
--
=============================================
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Tech-Know-Ware Ltd
                         for XML to C++ data binding visit
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                         (or http://www.xml2cpp.com)
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----- Original Message ----- 
From: <ravishankar.shiroor@wipro.com>
To: <rjsparks@nostrum.com>
Cc: <sip@ietf.org>
Sent: Thursday, July 06, 2006 5:43 AM
Subject: RE: [Sip] Update to fork-loop-fix submitted


It is always better to reduce the computation in the path of normal
success case and introduce it in the path of a rarer scenario. I think
it will be a fair assumption to make that Spirals will be rarer than the
non-spiraling scenarios.

In terms of increasing logic, the increase in logic would be very small
if it all any.
It would be a trade-off that should be left to the implementer to take
if he considers that hash computation is expensive on his platform. If
this id recommends the computation of hash at the MUST strength in the
first pass of the SIP request, an implementer would not have that
freedom to make that trade-off..

Regards,
Ravi.

--

Ravishankar. G. Shiroor
Wipro Technologies, Bangalore.

ravishankar.shiroor@wipro.com
--


> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> Sent: Wednesday, July 05, 2006 8:21 PM
> To: RaviShankar Shiroor (WT01 - Product Engineering Solutions)
> Cc: sip@ietf.org
> Subject: Re: [Sip] Update to fork-loop-fix submitted
>
> I understand the suggestion. You are trading off letting the
> request through a loop one extra cycle for some lower amount
> of processing the first time the request went through the
> proxy. If you believe that spirals are relatively rare
> things, this change would result in proxies very rarely
> having to build the loop/spiral differentiation hash.
>
> But as Jonathan points out, it's not clear that the hash
> computation is expensive enough to warrant this extra logic
> and the extra network traffic it would allow. I expect that
> the hash calculation expense is small compared to the
> parse/search/compare operations for "Proxy checks to see if
> its address is there in the via already".
>
> RjS
>
> On Jul 3, 2006, at 5:39 AM, <ravishankar.shiroor@wipro.com>
> <ravishankar.shiroor@wipro.com> wrote:
>
> >
> > Hi Robert,
> >
> > I had suggested one more alternative
> >
> > (http://www1.ietf.org/mail-archive/web/sip/current/msg14359.html )
> >
> > Quoted from my previous mail:
> >
> > <quote>
> > I was thinking how to reduce the impact of overhead for
> calculating
> > the
> > digest.
> >
> > If it is suggested as follows, I think it will reduce the
> overhead on
> > normal scenarios:
> >
> > Proxy checks to see if its address is there in the via already.
> > If it is not there, it proceeds normally without calculating the
> > digest
> > and inserting Via to the outgoing request without the digest.
> > If it is there, then it is possible that the request is looping. So,
> > this time around, it calculates a digests and inserts and sends it
> > along.
> > If it is looping, the request will definitely come in
> again, which can
> > be rejected.
> > This way the impact of calculating the digest is required (reduced
> > to a
> > smaller percent of the sessions) only when the proxy is
> suspicious the
> > request might be looping.
> > Of course this results in the scenarios mentioned in the id to be
> > successful in completing the first loop, but the second loop will be
> > detected and arrested. I think this can be applicable to
> general loop
> > detection, and not just fork loop fixes.
> > </quote>
> >
> > Regards,
> > Ravi.
> >
> > --
> >
> > Ravishankar. G. Shiroor
> > Wipro Technologies, Bangalore.
> >
> > ravishankar.shiroor@wipro.com
> > --
> >
> >
> >> -----Original Message-----
> >> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> >> Sent: Sunday, June 25, 2006 8:22 PM
> >> To: IETF SIP List
> >> Subject: [Sip] Update to fork-loop-fix submitted
> >>
> >> Until it appears in the repository, see:
> >>
> http://www.nostrum.com/~rjsparks/draft-ietf-sip-fork-loop-fix-02.txt
> >>
> >> This was a moderately heavy touch - based on list feedback,
> >> I've moved and updated the sections covering the
> >> loopdetection algorithm in from 3261. In the process of
> >> writing the clarifying text, I discovered several places that
> >> need clarification. They are listed as Open Issues in the
> >> draft. Basically there are fields that I can't recall or
> >> reconstruct how they would be useful in the part of the hash
> >> used for loop-detection or spiral differentiation. If anyone
> >> can remember why they need to be included, send a note to the
> >> list and we'll capture that in the draft.
> >>
> >> Please review the updated loop-detection text very carefully.
> >>
> >> Here's the changelog from the draft:
> >>
> >>        Integrated several editorial fixes suggested by
> >> Jonathan Rosenberg
> >>
> >>        Noted that the reduction of the attack to a single
> >> registration
> >>        against a single URI as documented in previous versions,
> >> is, in
> >>        fact, going to be effective against implementations
> >> conforming to
> >>        the standards before this repair.
> >>
> >>        Re-incorporated motivation from the original
> >> maxforwards-problem
> >>        draft into the security considerations section based
> >> on feedback
> >>        from Cullen Jennings
> >>
> >>        Introduced replacement text for the loop detection algorithm
> >>        description in RFC 3261, fixing the bug 648 (the
> >> topmost Via value
> >>        must not be included in the second part) and clarifying the
> >>        algorithm.  Removed several other fields suggested
> by 3261 and
> >>        placed open issues around their presence.
> >>
> >>        Added a Notes to Implementors section capturing the
> >> "common way"
> >>        text and pointing to the interoperability issues that have
> >> been
> >>        observed with loop detection at previous SIPits
> >>
> >>
> >> Thanks,
> >>
> >> RjS
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use
> >> sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >>
> >
> >
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From: "Drage, Keith (Keith)" <drage@lucent.com>
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(As WG chair)

Having looked at some of the comments on outbound being addressed to the list in the last few days, it occurs to me that at least of you have fundamental requirements that impact the solution and these requirements themselves need to be discussed.

Hidden in the current draft you will find section 12 labelled "requirements". This is a somewhat thin section at the moment (not necessarily a bad thing).

Can you all please review this section, and identify whether these requirements encompass all the requirements that you wish to have met in outbound, and then try to motivate what you think is missing.

And as always "send text".

regards

Keith

Keith Drage
Lucent Technologies
drage@lucent.com
tel: +44 1793 776249

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip

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(As WG chair)

Having looked at some of the comments on outbound being addressed to the list in the last few days, it occurs to me that at least of you have fundamental requirements that impact the solution and these requirements themselves need to be discussed.

Hidden in the current draft you will find section 12 labelled "requirements". This is a somewhat thin section at the moment (not necessarily a bad thing).

Can you all please review this section, and identify whether these requirements encompass all the requirements that you wish to have met in outbound, and then try to motivate what you think is missing.

And as always "send text".

regards

Keith

Keith Drage
Lucent Technologies
drage@lucent.com
tel: +44 1793 776249

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

From sip-bounces@ietf.org Thu Jul 06 06:38:20 2006
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Subject: RE: [Sip] Outbound: Validating if the proxy supports outbound or not
Date: Thu, 6 Jul 2006 10:47:38 +0200
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: <Erkki.Koivusalo@nokia.com>,
	<sip@ietf.org>,
	<rohan@ekabal.com>
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=20
Hi,
=20
We talk about "keepalive=3Dstun", and we talk about edge proxies =
supporting outbound.
=20
I think we need to separate the issues.
=20
1. Whether the edge proxy supports STUN for keep-alive, and whether it =
needs to be able to "indicate" that.
=20
2. Whether the edge proxy supports outbound (in which case it also =
supports STUN for keep-alive), and whether it needs to be able to =
"indicate" that.
=20
Regards,
=20
Christer

________________________________

From: Erkki.Koivusalo@nokia.com [mailto:Erkki.Koivusalo@nokia.com]
Sent: Thu 06/07/2006 09:10
To: sip@ietf.org; rohan@ekabal.com
Cc: BPenfield@acmepacket.com; adam@nostrum.com
Subject: RE: [Sip] Outbound: Validating if the proxy supports outbound =
or not




Hi,

The comments sent on the list during last few days seem to
indicate reasonably large concern that it is not sufficient
for the UA to check that the configured proxy URI contains
keepalive=3Dstun parameter, in order for the UA to make sure
that the proxy actually supports outbound. Thus I will
expect that the next version of outbound draft will address
this issue. Note that the authors of the draft have promised
to raise this as an open issue in the WG meeting, so please
be prepared to contribute that discussion.

The comments I am referring to are this (mine included):

Bob Penfield:
>Does that mean that a UA should not send STUN if the
>keepalive=3Dstun is not present?
>
>If so, will outbound be useful prior to the completion of the various
>configuration mechanisms? And is it acceptable for a
>mis-configured UA to cause the proxy or the registrar to crash or
mis-behave?
>
>A mechanism such as the proposed Proxy-Require/Require method
>similar to sec-agree allows the UA to determine that the proxy and
>registrar support STUN before it sends and potentially harmful
messages.

Adam Roach:
>I think this can be fixed with a little more behavior around what
>edge-proxies that support this draft are required to do. Specifically:
>since section 5.1 already has some pretty strong normative
>language that
>requires edge-proxies to insert Path header fields with certain
>properties, it's a rather small leap to dictate that the form of the
>URIs in these Path header fields must include something that the
>registrar can use to verify outbound support.
>
>With that in place, the registrar can now look at an inbound REGISTER
>request and (in addition to checking the Contact header field) look
>through the Path header fields to ensure that all of them contain URIs
>with ";outbound" parameters. If they don't, it canJul 2006 10:49:27 +0200
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: <Erkki.Koivusalo@nokia.com>,
	<sip@ietf.org>,
	<rohan@ekabal.com>
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=20
Hi,
=20
We talk about "keepalive=3Dstun", and we talk about edge proxies =
supporting outbound.
=20
I think we need to separate the issues.
=20
1. Whether the edge proxy supports STUN for keep-alive, and whether it =
needs to be able to "indicate" that.
=20
2. Whether the edge proxy supports outbound (in which case it also =
supports STUN for keep-alive), and whether it needs to be able to =
"indicate" that.
=20
Regards,
=20
Christer

________________________________

From: Erkki.Koivusalo@nokia.com [mailto:Erkki.Koivusalo@nokia.com]
Sent: Thu 06/07/2006 09:10
To: sip@ietf.org; rohan@ekabal.com
Cc: BPenfield@acmepacket.com; adam@nostrum.com
Subject: RE: [Sip] Outbound: Validating if the proxy supports outbound =
or not




Hi,

The comments sent on the list during last few days seem to
indicate reasonably large concern that it is not sufficient
for the UA to check that the configured proxy URI contains
keepalive=3Dstun parameter, in order for the UA to make sure
that the proxy actually supports outbound. Thus I will
expect that the next version of outbound draft will address
this issue. Note that the authors of the draft have promised
to raise this as an open issue in the WG meeting, so please
be prepared to contribute that discussion.

The comments I am referring to are this (mine included):

Bob Penfield:
>Does that mean that a UA should not send STUN if the
>keepalive=3Dstun is not present?
>
>If so, will outbound be useful prior to the completion of the various
>configuration mechanisms? And is it acceptable for a
>mis-configured UA to cause the proxy or the registrar to crash or
mis-behave?
>
>A mechanism such as the proposed Proxy-Require/Require method
>similar to sec-agree allows the UA to determine that the proxy and
>registrar support STUN before it sends and potentially harmful
messages.

Adam Roach:
>I think this can be fixed with a little more behavior around what
>edge-proxies that support this draft are required to do. Specifically:
>since section 5.1 already has some pretty strong normative
>language that
>requires edge-proxies to insert Path header fields with certain
>properties, it's a rather small leap to dictate that the form of the
>URIs in these Path header fields must include something that the
>registrar can use to verify outbound support.
>
>With that in place, the registrar can now look at an inbound REGISTER
>request and (in addition to checking the Contact header field) look
>through the Path header fields to ensure that all of them contain URIs
>with ";outbound" parameters. If they don't, it can take evasive action
>(perhaps not returning the "Supported: outbound" header field in the
>response?) that prevents things from going wrong.

Erkki Koivusalo:
>The specific concerns that I have (and would like the WG to
>consider in the meeting) are closely related to what you told:
>
>>We got very strong feed back on one point - don't send
>>STUN messages to a proxy unless you know it supports them.
>
>Does the WG consider checking the configured URI as a safe enough
>check for the UA to ensure that the SIP proxy support STUN ?
>
>In my opinion it is not. As an example a vendor X could implement
>a compliant but bad UA which provides an option for the user to
>configure the settings via an User Interface e.g. like this:
>"Is the phone behind a NAT: Yes/No". When the user selects
>"Yes" the UA would add the keepalive=3Dstun flag to the proxy URI
>even if the user does not have a clue if the proxy actually
>supports outbound or not. There is nothing in outbound draft
>to prevent such unsafe implementations to appear.

Jeroen van Bemmel:
>The point is that these are really 2 separate issues:
>1. Discovering the set of outbound proxy URIs (with keepalive=3Dstun
>parameters in them)
>2. Confirming that a given proxy in the set actually supports
>outbound (ie validating the keepalive=3Dstun information)
>
>Of these 2 options, (1) is indeed a separable problem.
>(2) - to me - is not, for me outbound is not acceptable
>without something to address (2).

Regards,

Erkki

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





 take evasive action
>(perhaps not returning the "Supported: outbound" header field in the
>response?) that prevents things from going wrong.

Erkki Koivusalo:
>The specific concerns that I have (and would like the WG to
>consider in the meeting) are closely related to what you told:
>
>>We got very strong feed back on one point - don't send
>>STUN messages to a proxy unless you know it supports them.
>
>Does the WG consider checking the configured URI as a safe enough
>check for the UA to ensure that the SIP proxy support STUN ?
>
>In my opinion it is not. As an example a vendor X could implement
>a compliant but bad UA which provides an option for the user to
>configure the settings via an User Interface e.g. like this:
>"Is the phone behind a NAT: Yes/No". When the user selects
>"Yes" the UA would add the keepalive=3Dstun flag to the proxy URI
>even if the user does not have a clue if the proxy actually
>supports outbound or not. There is nothing in outbound draft
>to prevent such unsafe implementations to appear.

Jeroen van Bemmel:
>The point is that these are really 2 separate issues:
>1. Discovering the set of outbound proxy URIs (with keepalive=3Dstun
>parameters in them)
>2. Confirming that a given proxy in the set actually supports
>outbound (ie validating the keepalive=3Dstun information)
>
>Of these 2 options, (1) is indeed a separable problem.
>(2) - to me - is not, for me outbound is not acceptable
>without something to address (2).

Regards,

Erkki

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





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From: <Erkki.Koivusalo@nokia.com>
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Hi,

The q-value scheme Adam suggested would make sense if it
would be used together with the idle backup server scheme
outlined in draft-koivusalo-sip-outbound-discovery-02:

If the domain has SRV records for proxies supporting outbound
with different SRV 'priority' values, then when sending
REGISTER requests the UA would assign the q-values like this:

- For the primary flow created towards a proxy with
  lowest SRV priority value the UA assigns the biggest=20
  q-value (to indicate this is the active flow)

- For the secondary flow created towards a proxy with
  a higher SRV priority value (an idle backup proxy)
  the UA would assign a lower q-value (to indicate
  that this is a backup flow)

Since the DNS SRV records are in control of the service
provider, this scheme would not cause any problems for the
environments like Juha has. He just need to assign the
very same 'priority' value for all his SRV records to
make sure that UA uses same q-value towards all the proxies.

In environments where the location service (registrar) is
on a single server which sees all the flows and the related
contact bindings this usage of q-value could be used in a=20
straightforward way.
 =20
But whether it is in this phase still possible to get
such ideas for q-value usage to be adopted to outbound
draft ... well it is something extra and I am far from
sure if any consensus could be reached about that
quickly enough in order not to delay the draft yet
another four months ...

Regards,

Erkki

>-----Original Message-----
>From: ext Juha Heinanen [mailto:jh@tutpro.com]=20
>Sent: 06.July.2006 09:28
>To: Adam Roach
>Cc: Cullen Jennings; SIP WG
>Subject: [Sip] Outbound Comment: q-value
>
>Adam Roach writes:
>
> > To handle this situation, it might make sense to put stuff in 6.2
> > that basically says that, for contacts with matching instance-ids,
> > the proxy MUST NOT try a contact until all contacts at a higher
> > q-value have been tried first. (I could go along with a weaker
> > strength, but I really do think a prescriptive approach is the only
> > thing that allows the clients to handle the situation I describe
> > above in a manner that is predictable to the users).
>
>i you suggesting that proxies consult each other about the q-value UA
>used when registering with each proxy before forwarding a=20
>request to the
>UA?  if so, your suggesting makes no sense.
>
>-- juha
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
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>

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From sip-bounces@ietf.org Thu Jul 06 08:49:51 2006
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From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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Subject: Re: [Sip] A thought on progressing outbound
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I don't have any problem with the current list of requirements in 
Section 12. The problem is that text in the draft seems to go beyond 
these requirements and impose a certain mandatory redundancy scheme, to 
which I strongly disagree as a "mandatory" feature.

/Miguel

Drage, Keith (Keith) wrote:
> (As WG chair)
> 
> Having looked at some of the comments on outbound being addressed to the list in the last few days, it occurs to me that at least of you have fundamental requirements that impact the solution and these requirements themselves need to be discussed.
> 
> Hidden in the current draft you will find section 12 labelled "requirements". This is a somewhat thin section at the moment (not necessarily a bad thing).
> 
> Can you all please review this section, and identify whether these requirements encompass all the requirements that you wish to have met in outbound, and then try to motivate what you think is missing.
> 
> And as always "send text".
> 
> regards
> 
> Keith
> 
> Keith Drage
> Lucent Technologies
> drage@lucent.com
> tel: +44 1793 776249
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Subject: RE: [Sip] Ready for WGLC on Outbound?
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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	"Jeroen van Bemmel" <jbemmel@zonnet.nl>
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=20
Does a proxy supporting STUN keep-alive have to be an "outbound proxy" =
(assuming you mean an edge proxy), as defind in the outbound spec?
=20
Or, what do you mean by "outbound proxy"? As defined in 3261?
=20
Regards,
=20
Christer
=20

________________________________

From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: Wed 05/07/2006 10:46
To: Jeroen van Bemmel
Cc: Cullen Jennings; sip@ietf.org; dean.willis@softarmor.com; =
rohan@ekabal.com
Subject: Re: [Sip] Ready for WGLC on Outbound?



Jeroen van Bemmel writes:

 > Currently deployed proxies generally do not support STUN, especially =
not on
 > the same ports that they use for SIP. Sending a STUN packet to such a =
proxy
 > could very well be harmless, but it could also cause it to crash.

sure, but those proxies are not outbound proxys and should not be listed
as such.  problem solved.

 > Sure, we can state "any proxy listed as outbound proxy MUST support =
STUN".
 > We might even say "all proxies SHOULD support STUN". But no matter =
what
 > requirements we come up with, they won't be met by currently deployed
 > proxies.

if your currently deployed proxy is not outbound proxy as specified in
the i-d, you don't list it as such.  it is a regular proxy and can do
whatever is specified in rfc3261.

-- juha

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From sip-bounces@ietf.org Thu Jul 06 12:27:25 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: Miguel Garcia <Miguel.An.Garcia@nokia.com>
Subject: RE: [Sip] A thought on progressing outbound
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Firstly I was not necessarily targetting your comments in particular, but lets try and address your issue in terms of some simple tests:

1)	what is currently in the draft fulfils the current requirements
2)	what is currently in the draft does not appear to be technically wrong
3)	we apparently do not need to decide between two alternative technical solutions to a requirement

therefore you have a use case that is different from that of the editors, and ideally we need to reflect the difference between those two use cases either:

-	by clearly stating a need for flexibility to allow those two use cases in the requirements
-	or detailing the use cases that need to be supported.

(assuming that these use cases are considered to be valid)

The first of these is probably correct in this case.

regards

Keith

> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
> Sent: 06 July 2006 12:47
> To: Drage, Keith (Keith)
> Cc: sip@ietf.org
> Subject: Re: [Sip] A thought on progressing outbound
> 
> 
> I don't have any problem with the current list of requirements in 
> Section 12. The problem is that text in the draft seems to go beyond 
> these requirements and impose a certain mandatory redundancy 
> scheme, to 
> which I strongly disagree as a "mandatory" feature.
> 
> /Miguel
> 
> Drage, Keith (Keith) wrote:
> > (As WG chair)
> > 
> > Having looked at some of the comments on outbound being 
> addressed to the list in the last few days, it occurs to me 
> that at least of you have fundamental requirements that 
> impact the solution and these requirements themselves need to 
> be discussed.
> > 
> > Hidden in the current draft you will find section 12 
> labelled "requirements". This is a somewhat thin section at 
> the moment (not necessarily a bad thing).
> > 
> > Can you all please review this section, and identify 
> whether these requirements encompass all the requirements 
> that you wish to have met in outbound, and then try to 
> motivate what you think is missing.
> > 
> > And as always "send text".
> > 
> > regards
> > 
> > Keith
> > 
> > Keith Drage
> > Lucent Technologies
> > drage@lucent.com
> > tel: +44 1793 776249
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
> 

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Subject: RE: [Sip] A thought on progressing outbound
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Drage, Keith \(Keith\)" <drage@lucent.com>,
	"Miguel Garcia" <Miguel.An.Garcia@nokia.com>
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Hi,
=20
In order to progress the work, and avoid discussions later, I think we =
also need to make clear which requirements are related to the STUN =
keep-alive mechanism, but not the rest of outbound, and which =
requirements have nothing to do with the STUN keep-alive mechanism.
=20
Regards,
=20
Christer

________________________________

From: Drage, Keith (Keith) [mailto:drage@lucent.com]
Sent: Thu 06/07/2006 18:25
To: Miguel Garcia
Cc: sip@ietf.org
Subject: RE: [Sip] A thought on progressing outbound



Firstly I was not necessarily targetting your comments in particular, =
but lets try and address your issue in terms of some simple tests:

1)      what is currently in the draft fulfils the current requirements
2)      what is currently in the draft does not appear to be technically =
wrong
3)      we apparently do not need to decide between two alternative =
technical solutions to a requirement

therefore you have a use case that is different from that of the =
editors, and ideally we need to reflect the difference between those two =
use cases either:

-       by clearly stating a need for flexibility to allow those two use =
cases in the requirements
-       or detailing the use cases that need to be supported.

(assuming that these use cases are considered to be valid)

The first of these is probably correct in this case.

regards

Keith

> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
> Sent: 06 July 2006 12:47
> To: Drage, Keith (Keith)
> Cc: sip@ietf.org
> Subject: Re: [Sip] A thought on progressing outbound
>
>
> I don't have any problem with the current list of requirements in
> Section 12. The problem is that text in the draft seems to go beyond
> these requirements and impose a certain mandatory redundancy
> scheme, to
> which I strongly disagree as a "mandatory" feature.
>
> /Miguel
>
> Drage, Keith (Keith) wrote:
> > (As WG chair)
> >
> > Having looked at some of the comments on outbound being
> addressed to the list in the last few days, it occurs to me
> that at least of you have fundamental requirements that
> impact the solution and these requirements themselves need to
> be discussed.
> >
> > Hidden in the current draft you will find section 12
> labelled "requirements". This is a somewhat thin section at
> the moment (not necessarily a bad thing).
> >
> > Can you all please review this section, and identify
> whether these requirements encompass all the requirements
> that you wish to have met in outbound, and then try to
> motivate what you think is missing.
> >
> > And as always "send text".
> >
> > regards
> >
> > Keith
> >
> > Keith Drage
> > Lucent Technologies
> > drage@lucent.com
> > tel: +44 1793 776249
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> --
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>

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Subject: Re: [Sip] Ready for WGLC on Outbound?
Date: Thu, 6 Jul 2006 19:09:33 +0200
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Rohan,

> Jeroen,
>
> I'd like to address this one point.  My recollection from Dallas is that 
> we had a very strong consensus that the existence of keepalive=stun was 
> sufficient to send STUN messages.
>
> Based on this, I think we need to make it a requirement of the 
> configuration mechanism that  you can depend on its result to have 
> accurate keepalive information.
>

No, you cannot shift it off like that. Outbound is introducing binary STUN 
messages that could potentially harm existing systems, so outbound is 
responsible for providing reasonable safeguards. If it were an "ordinary" 
new SIP feature, then standard RFC3261 safeguards would provide sufficient 
protection. But STUN is different

The problem with a requirement like you propose here is twofold:
1) it is too soft, in the sense that you cannot test for it
2) Some configuration mechanisms (like manual configuration) simply cannot 
meet this requirement

Again, what is the reason that we should not include a mechanism for this? 
Do you feel it would make the draft too complex?

Regards,

Jeroen



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Juha Heinanen wrote:
> Adam Roach writes:
>
>  > To handle this situation, it might make sense to put stuff in 6.2
>  > that basically says that, for contacts with matching instance-ids,
>  > the proxy MUST NOT try a contact until all contacts at a higher
>  > q-value have been tried first. (I could go along with a weaker
>  > strength, but I really do think a prescriptive approach is the only
>  > thing that allows the clients to handle the situation I describe
>  > above in a manner that is predictable to the users).
>
> i you suggesting that proxies consult each other about the q-value UA
> used when registering with each proxy before forwarding a request to the
> UA?  if so, your suggesting makes no sense.
>
> -- juha
>   


I have no idea what you're saying here. Let's try getting some common
context.

Open the outbound document. Go to section 6.2. You'll notice that it
talks about the rules that proxy/registrars use to populate the target
set. This is the section to which my comment applies. In particular,
there are two new rules placed on proxy/registrars that implement
outbound, specified in a bullet list.

I'm suggesting a third bullet that reads something like:

    * If multiple bindings have the same instance-id, the proxy MUST NOT
      use a binding until all bindings of greater q-value have been tried.

/a


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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Update to fork-loop-fix submitted
Date: Thu, 6 Jul 2006 12:48:34 -0500
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On Jul 5, 2006, at 11:43 PM, <ravishankar.shiroor@wipro.com>  
<ravishankar.shiroor@wipro.com> wrote:

>
> It is always better to reduce the computation in the path of normal
> success case and introduce it in the path of a rarer scenario. I think
> it will be a fair assumption to make that Spirals will be rarer  
> than the
> non-spiraling scenarios.

But in this case, you are reducing computation (perhaps trivially) at  
the
cost of increased network traffic.

>
> In terms of increasing logic, the increase in logic would be very  
> small
> if it all any.
> It would be a trade-off that should be left to the implementer to take
> if he considers that hash computation is expensive on his platform. If
> this id recommends the computation of hash at the MUST strength in the
> first pass of the SIP request, an implementer would not have that
> freedom to make that trade-off..
>
> Regards,
> Ravi.
>
> --
>
> Ravishankar. G. Shiroor
> Wipro Technologies, Bangalore.
>
> ravishankar.shiroor@wipro.com
> --
>
>
>> -----Original Message-----
>> From: Robert Sparks [mailto:rjsparks@nostrum.com]
>> Sent: Wednesday, July 05, 2006 8:21 PM
>> To: RaviShankar Shiroor (WT01 - Product Engineering Solutions)
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Update to fork-loop-fix submitted
>>
>> I understand the suggestion. You are trading off letting the
>> request through a loop one extra cycle for some lower amount
>> of processing the first time the request went through the
>> proxy. If you believe that spirals are relatively rare
>> things, this change would result in proxies very rarely
>> having to build the loop/spiral differentiation hash.
>>
>> But as Jonathan points out, it's not clear that the hash
>> computation is expensive enough to warrant this extra logic
>> and the extra network traffic it would allow. I expect that
>> the hash calculation expense is small compared to the
>> parse/search/compare operations for "Proxy checks to see if
>> its address is there in the via already".
>>
>> RjS
>>
>> On Jul 3, 2006, at 5:39 AM, <ravishankar.shiroor@wipro.com>
>> <ravishankar.shiroor@wipro.com> wrote:
>>
>>>
>>> Hi Robert,
>>>
>>> I had suggested one more alternative
>>>
>>> (http://www1.ietf.org/mail-archive/web/sip/current/msg14359.html )
>>>
>>> Quoted from my previous mail:
>>>
>>> <quote>
>>> I was thinking how to reduce the impact of overhead for
>> calculating
>>> the
>>> digest.
>>>
>>> If it is suggested as follows, I think it will reduce the
>> overhead on
>>> normal scenarios:
>>>
>>> Proxy checks to see if its address is there in the via already.
>>> If it is not there, it proceeds normally without calculating the
>>> digest
>>> and inserting Via to the outgoing request without the digest.
>>> If it is there, then it is possible that the request is looping. So,
>>> this time around, it calculates a digests and inserts and sends it
>>> along.
>>> If it is looping, the request will definitely come in
>> again, which can
>>> be rejected.
>>> This way the impact of calculating the digest is required (reduced
>>> to a
>>> smaller percent of the sessions) only when the proxy is
>> suspicious the
>>> request might be looping.
>>> Of course this results in the scenarios mentioned in the id to be
>>> successful in completing the first loop, but the second loop will be
>>> detected and arrested. I think this can be applicable to
>> general loop
>>> detection, and not just fork loop fixes.
>>> </quote>
>>>
>>> Regards,
>>> Ravi.
>>>
>>> --
>>>
>>> Ravishankar. G. Shiroor
>>> Wipro Technologies, Bangalore.
>>>
>>> ravishankar.shiroor@wipro.com
>>> --
>>>
>>>
>>>> -----Original Message-----
>>>> From: Robert Sparks [mailto:rjsparks@nostrum.com]
>>>> Sent: Sunday, June 25, 2006 8:22 PM
>>>> To: IETF SIP List
>>>> Subject: [Sip] Update to fork-loop-fix submitted
>>>>
>>>> Until it appears in the repository, see:
>>>>
>> http://www.nostrum.com/~rjsparks/draft-ietf-sip-fork-loop-fix-02.txt
>>>>
>>>> This was a moderately heavy touch - based on list feedback,
>>>> I've moved and updated the sections covering the
>>>> loopdetection algorithm in from 3261. In the process of
>>>> writing the clarifying text, I discovered several places that
>>>> need clarification. They are listed as Open Issues in the
>>>> draft. Basically there are fields that I can't recall or
>>>> reconstruct how they would be useful in the part of the hash
>>>> used for loop-detection or spiral differentiation. If anyone
>>>> can remember why they need to be included, send a note to the
>>>> list and we'll capture that in the draft.
>>>>
>>>> Please review the updated loop-detection text very carefully.
>>>>
>>>> Here's the changelog from the draft:
>>>>
>>>>        Integrated several editorial fixes suggested by
>>>> Jonathan Rosenberg
>>>>
>>>>        Noted that the reduction of the attack to a single
>>>> registration
>>>>        against a single URI as documented in previous versions,
>>>> is, in
>>>>        fact, going to be effective against implementations
>>>> conforming to
>>>>        the standards before this repair.
>>>>
>>>>        Re-incorporated motivation from the original
>>>> maxforwards-problem
>>>>        draft into the security considerations section based
>>>> on feedback
>>>>        from Cullen Jennings
>>>>
>>>>        Introduced replacement text for the loop detection algorithm
>>>>        description in RFC 3261, fixing the bug 648 (the
>>>> topmost Via value
>>>>        must not be included in the second part) and clarifying the
>>>>        algorithm.  Removed several other fields suggested
>> by 3261 and
>>>>        placed open issues around their presence.
>>>>
>>>>        Added a Notes to Implementors section capturing the
>>>> "common way"
>>>>        text and pointing to the interoperability issues that have
>>>> been
>>>>        observed with loop detection at previous SIPits
>>>>
>>>>
>>>> Thanks,
>>>>
>>>> RjS
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use
>>>> sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>
>>>
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> -----Original Message-----
> From: Juha Heinanen [mailto:jh@tutpro.com]
> Sent: Wednesday, July 05, 2006 4:18 PM
> To: Jeroen van Bemmel
> Cc: Miguel Garcia; sip@ietf.org; Cullen Jennings; Rohan Mahy
> Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
> 
> Jeroen van Bemmel writes:
> 
>  > For flows beyond the maximum the network "suffers" (just like the
> registrar
>  > determines the expiration time). But by imposing a minimum number of
> flows,
>  > the UA potentially "suffers" (spending scarce resources such as battery
>  > power). And anyway, when the UA sets up less flows the consequences are
> for
>  > its user.
> 
> in the above situation, assuming that network has two proxies, if UA
> decides to establish only one flow instead of two, then the likely hood
> is 50% that incoming call to the UA cannot be establishes.  it thus
> makes no sense for the UA to violate network policy on number of flows.

That doesn't make any sense to me.  If a UA only establishes 1 flow, then
the registrar only has one registered contact/flow, which points through 1
edge proxy if there is one, and 100% of the incoming calls to the UA will go
over that 1 flow path.  Obviously if the flow fails, there will be some
amount of service-outage to the UA until it detects the loss and connects
through an alternate path, but hopefully that's not happening 50% of the
time. :)  

-hadriel


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inline.

Jeroen van Bemmel wrote:

> Cullen,
> 
> The point is that these are really 2 separate issues:
> 1. Discovering the set of outbound proxy URIs (with keepalive=stun 
> parameters in them)
> 2. Confirming that a given proxy in the set actually supports outbound 
> (ie validating the keepalive=stun information)
> 
> Of these 2 options, (1) is indeed a separable problem. (2) - to me - is 
> not, for me outbound is not acceptable without something to address (2). 
> It really should say "the UAC MUST NOT use STUN until { mechanism for 
> (2) says its supported }"

I personally think that the presence of the keepalive=stun attribute in 
the URI is sufficient of a "confirmation". The examples you and others 
have presented for why this is needed are (1) misconfiguration (someone 
gives a URI to the client which is incorrect in that it contains this 
flag when it shouldn't), (2) malicious users, (3) constructed URIs. On 
(2), let me point out that an attacker that can modify DNS or modify SIP 
messages (or whatever is used for the discovery function) can launch far 
more effective attacks than just adding keepalive=stun in the hopes that 
it causes a low-quality proxy to crash.

On (1), misconfugration could potentially produce all kinds of things 
that would cause a client to send things to the server that might make 
it crash, including non-SIP URI, URI with invalid header parameters, 
etc. I don't see keepalive=stun as special.

On (3), this is definitely a possibility. However I think the best way 
to mitigate is to say 'don't do that'. SIP-outbound can have some words 
which warn implementations that they MUST NOT insert this parameter into 
URI that they construct from a domain name.

Finally, let me say that if you have a proxy that crashes on receipt of 
binary messages on its sip port, you are in for real trouble. I think a 
far better solution is for proxy vendors to fix their proxies, rather 
than for us to bend over backwards to design new mechanisms that help 
protect such fragile servers from ever seeing these messages. No matter 
what we put in our specs, attackers can still send binary messages that 
will result in the same result you are trying to prevent.

Thanks,
Jonathan R.




-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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On Jul 6, 2006, at 12:09 PM, Jeroen van Bemmel wrote:

> Rohan,
>
>> Jeroen,
>>
>> I'd like to address this one point.  My recollection from Dallas  
>> is that we had a very strong consensus that the existence of  
>> keepalive=stun was sufficient to send STUN messages.
>>
>> Based on this, I think we need to make it a requirement of the  
>> configuration mechanism that  you can depend on its result to have  
>> accurate keepalive information.
>>
>
> No, you cannot shift it off like that. Outbound is introducing  
> binary STUN messages that could potentially harm existing systems,  
> so outbound is responsible for providing reasonable safeguards. If  
> it were an "ordinary" new SIP feature, then standard RFC3261  
> safeguards would provide sufficient protection. But STUN is different

I've been watching this train of reasoning for a while now, and am  
not sure I buy it. Do we really expect SIP devices to melt down if  
they get stun and don't expect it? If so, they are big targets for  
DoS attacks.


>
> The problem with a requirement like you propose here is twofold:
> 1) it is too soft, in the sense that you cannot test for it
> 2) Some configuration mechanisms (like manual configuration) simply  
> cannot meet this requirement
>
> Again, what is the reason that we should not include a mechanism  
> for this? Do you feel it would make the draft too complex?
>
> Regards,
>
> Jeroen
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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> Use sip-implementors@cs.columbia.edu for questions on current sip
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inline.

Miguel Garcia wrote:

> Cullen Jennings wrote:
> 
>>
>>
>> I expect you are right in that this MUST is too strong here but I 
>> think there is an argument for SHOULD. The problem is that if the 
>> domain assume the UA is doing two flows and the UA does one, things 
>> are broken.
>>
> 
> I would like to understand which part of the outbound mechanism is 
> broken if the domain assumes the UE will do two flows, but the UA does 
> one flow. Yes, it isn't reliable, but the mechanism works nicely... or not?
> 
> Then... why can't a domain indicate just one outbound URI? At the moment 
> the draft has an artificial lower limit for all the domains to provide 
> two URIs and impose a reliability that might not be required/needed.

Thats not my read.

The draft says that a client has to support at least 2 outbound URI, and 
that it MUST send a REGISTER for each. However, that doesn't mean a 
domain *has* to provide two; it can also provide one. A domain would 
only provide one if it knows that it doesn't need two for purposes of 
reliability, or if it doesn't require reliability of service.

As such, I support the current text which says that a client MUST send a 
REGISTER to each outbound URI.

Thanks,
Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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My interpretation was the opposite - if a Register comes in with the same
{instance-id,reg-id} then it should indeed replace/overwrite the previous
binding of the same tuple.  I know there are several sections in the draft
which imply otherwise, but I don't understand why.  There is no real value
I can see to keeping a binding for an old flow of the same reg-id, since by
definition the UA would only "re-use" a reg-id if the old flow failed (or
even if it didn't think it had failed, but something rebooted and changed
the flow).  

I can see no realistic way in which an old flow of the same
{instance-id,reg-id} would succeed in reaching the UA.  Am I missing
something?

-hadriel
p.s. it's theoretically possible, if a sip-udp UA rebooted and resolved a
new proxy address and sent a register to it with the same reg-id, that it
may still be possible to reach the UA through the old "flow" for a time.  
But is that a realistic case worth caring about?



> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Thursday, July 06, 2006 1:12 AM
> To: Cullen Jennings
> Cc: SIP WG
> Subject: [Sip] Outbound Comment: binding key
> 
>  From section 6.1:
> 
>    When a reg-id header parameter is present in a Contact header field
>    value in a REGISTER request, the corresponding binding is between an
>    AOR and the combination of instance-id and reg-id.
> 
> I don't think this is quite correct. This statement appears to be
> asserting that {instance-id,reg-id} is a unique key to a binding
> associated with an AOR. This would, in turn, imply that any successful
> registration attempt with an {instance-id,reg-id} that matches an
> existing binding would overwrite the existing record.
> 
> I don't think that is what's intended, since section 6.2 contains the
> following text:
> 
>    o  If two bindings have the same instance-id and reg-id, the proxy
>       SHOULD prefer the contact that was most recently updated.
> 
> If the binding is only to {instance-id,reg-id}, then that text is
> nonsensical: you can't have two identical bindings to the same AOR.
> 
> I suspect that the text in 6.1 should actually read something more like:
> 
>    When a reg-id header parameter is present in a Contact header field
>    value in a REGISTER request, the corresponding binding is between an
>    AOR and the combination of instance-id, reg-id, and network flow.
> 
> /a
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Jonathan Rosenberg wrote:
> Finally, let me say that if you have a proxy that crashes on receipt of 
> binary messages on its sip port, you are in for real trouble. I think a 
> far better solution is for proxy vendors to fix their proxies, rather 
> than for us to bend over backwards to design new mechanisms that help 
> protect such fragile servers from ever seeing these messages. No matter 
> what we put in our specs, attackers can still send binary messages that 
> will result in the same result you are trying to prevent.

I would agree with this in the case of UDP or in other cases where the 
transport does the framing. It seems that for stream based transports 
unrecognized binary data could mess up framing. Proxies could try to 
recover but I don't know if it's reasonable to expect proxies to keep 
connections open for an extended period of time when all they see is a 
some garbage every 30 seconds or so. It's not necessarily that the proxy 
crashes but that it gives up on a particular connection at some point.

Thanks,
Anders

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From sip-bounces@ietf.org Thu Jul 06 15:48:35 2006
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Subject: RE: [Sip] Outbound-04 and "max-flow" in 200 OK for REGISTER
Date: Thu, 6 Jul 2006 10:02:35 +0200
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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Hi,
=20
Even if we would be able to provide this information in DNS, I still =
think it would be useful to be able to provide it in the REGISTER =
response.
=20
The number of flows the network wants the user(s) to register may be =
much more "dynamic" than the DNS update.
=20
In theory the number of flows could even be different for different =
users.
=20
Regards,
=20
Christer
=20
=20

________________________________

From: Cullen Jennings [mailto:fluffy@cisco.com]
Sent: Wed 05/07/2006 16:26
To: <Erkki.Koivusalo@nokia.com>
Cc: rohan@ekabal.com; dean.willis@softarmor.com; Christer Holmberg =
(JO/LMF); sip@ietf.org
Subject: Re: [Sip] Outbound-04 and "max-flow" in 200 OK for REGISTER




On Jun 29, 2006, at 10:55 PM, <Erkki.Koivusalo@nokia.com>=20
<Erkki.Koivusalo@nokia.com> wrote:

>
>> I can think of a number of ways to make this work where
>> the intended semantics are entirely embedded in DNS.
>
> If you have any ideas like that, could you please pass them
> to me since I am working on a related I-D where no such ideas
> have been introduced. I would be happy to receive those to be
> able to check if they work with the rest of the DNS scheme
> currently proposed. If yes, I could take them into my draft.

Well long and short of it is, if you are embedding the config=20
information in DNS, just put it in DNS - look at how much information=20
a ENUM or SRV record provides. I'm not sure if it is a good idea to=20
do this or not but I'm just pointing out ways you might consider.




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Subject: Re: [Sip] Ready for WGLC on Outbound?
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> Jonathan Rosenberg wrote:
>> Finally, let me say that if you have a proxy that crashes on receipt of 
>> binary messages on its sip port, you are in for real trouble. I think a 
>> far better solution is for proxy vendors to fix their proxies, rather 
>> than for us to bend over backwards to design new mechanisms that help 
>> protect such fragile servers from ever seeing these messages. No matter 
>> what we put in our specs, attackers can still send binary messages that 
>> will result in the same result you are trying to prevent.
>
> I would agree with this in the case of UDP or in other cases where the 
> transport does the framing. It seems that for stream based transports 
> unrecognized binary data could mess up framing. Proxies could try to 
> recover but I don't know if it's reasonable to expect proxies to keep 
> connections open for an extended period of time when all they see is a 
> some garbage every 30 seconds or so. It's not necessarily that the proxy 
> crashes but that it gives up on a particular connection at some point.
>
> Thanks,
> Anders

Yes, exactly! And for TLS a proxy might interpret it as an encryption 
failure, and trigger some anomaly detection system, and then blacklist the 
client. All of this can be prevented through a simple measure: have the 
proxy insert some flag. As mechanisms go, this to me hardly feels like 
bending over backwards.

None of this will help against malicious attackers, but it will work for UAs 
that are simply unaware. And it will also make debugging easier

Regards,

Jeroen


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From sip-bounces@ietf.org Thu Jul 06 17:14:42 2006
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Subject: Re: [Sip] Ready for WGLC on Outbound?
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[ snap ]
> Jonathan wrote
>
> I'm not objecting to the concept of this check. I just think its not 
> necessary, something that might be optional at best. As such, if folks 
> decide they want something, it can go in another draft. I dont think we 
> need to burden sip-outbound with this.
>

The trouble is, once outbound is "out" any mechanism defined in a separate 
draft won't work as good. Because then there will be clients that simply 
ignore the flag. We either MUST now, or MAY never fix it.

For the benefit of those amongst us with live systems, I am including a 
simple Java test program below so you can test the effect of a STUN binding 
request on your unsuspecting proxy (currently UDP only, trivial to adapt)

Regards,

Jeroen

import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.InetSocketAddress;
import java.net.SocketException;
import java.net.UnknownHostException;

/**
 * Tests what happens when a client sends a STUN Binding request over UDP
 * @author Jeroen
 */
public final class STUNClient {

 /**
  * @param args
  * @throws UnknownHostException
  * @throws SocketException
  */
 public static void main(String[] args) throws Exception {


  byte [] bindingRequest = new byte[ 20 ];
  bindingRequest[0] = 0x00;
  bindingRequest[1] = 0x01; // type: 0x0001
  bindingRequest[2] = 0x00;
  bindingRequest[3] = 0x00; // length: 0 (excl header)
  // 4..19: random 128-bit transaction id, lets take 0

  DatagramPacket p = new DatagramPacket( bindingRequest, 
bindingRequest.length,
   new InetSocketAddress(InetAddress.getLocalHost(),5060)
  );
  DatagramSocket s = new DatagramSocket();
  s.send( p );
  s.receive( p );
  System.out.println( p.getSocketAddress() );
 }

}




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Subject: RE: [Sip] Outbound Comment: negotiation and edge proxies
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Adam Roach" <adam@nostrum.com>, "Cullen Jennings" <fluffy@cisco.com>,
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=20
Hi Adam,
=20
>However, negotiation of the outbound mechanism is between the client =
and the registrar.=20
=20
Negotiation of the STUN for keep-alive mechanism is between the client =
and the edge proxy.
=20
And, again, I hope we can agree on the fact that the STUN for keep-alive =
is a stand-alone function, so the "outbound" option tag may not be =
present at all.
=20
Regards,
=20
Christer
=20
=20

________________________________

From: Adam Roach [mailto:adam@nostrum.com]
Sent: Thu 06/07/2006 07:49
To: Cullen Jennings; SIP WG
Subject: [Sip] Outbound Comment: negotiation and edge proxies



This one, I might just be confused about, but I combed the draft and
can't figure out how it is supposed to work.

Section 5 of the outbound document places a number of requirements on
edge proxies that must be implemented for outbound to work. However,
negotiation of the outbound mechanism is between the client and the
registrar. An edge-proxy that doesn't understand outbound will blithely
shuttle messages back and forth between the client while they signal
support for the outbound mechanism.

During registration, all a non-outbound edge-proxy will see and care
about is "Supported: path," which makes it fat and happy. It won't care
about the funky bits on the end of the Contact. It won't care about the
"Supported: outbound" header field in the REGISTER response. And it
won't do anything suspicious that might give the client or the registrar
some idea that things are about to go quite wrong. The client and
registrar have done the secret handshake, and are absolutely certain
that outbound handling is alive and well.

I think this can be fixed with a little more behavior around what
edge-proxies that support this draft are required to do. Specifically:
since section 5.1 already has some pretty strong normative language that
requires edge-proxies to insert Path header fields with certain
properties, it's a rather small leap to dictate that the form of the
URIs in these Path header fields must include something that the
registrar can use to verify outbound support.

For example:

   When an Edge Proxy receives a registration request with a reg-id
   header parameter in the Contact header field, it MUST form a flow
   identifier token that is unique to this network flow.  The Edge Proxy
   MUST insert this token into a URI referring to this proxy and place
   this URI into a Path header field as described in RFC 3327 [RFC3327].
   The token MAY be placed in the userpart of the URI. The URI MUST
   include an ";outbound" parameter.

With that in place, the registrar can now look at an inbound REGISTER
request and (in addition to checking the Contact header field) look
through the Path header fields to ensure that all of them contain URIs
with ";outbound" parameters. If they don't, it can take evasive action
(perhaps not returning the "Supported: outbound" header field in the
response?) that prevents things from going wrong.

/a

----

P.S. There is one corner case that this doesn't seem to cover: the
situation in which the client has a locally provisioned default outbound
proxy. This locally provisioned default outbound proxy won't use Path at
all, but it will remain in the signaling for all requests from the
client. I'm not certain how often this will become a problem, and
working around it would require something similar to the Path indication
I describe above, but on Via header fields. That feels like a really
ugly hack, and I'm not sure it's worth the inelegance to handle cases
that will be rare, if they exist at all.

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From sip-bounces@ietf.org Thu Jul 06 17:39:20 2006
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Subject: RE: [Sip] Ready for WGLC on Outbound?
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I think we all agree every device _should_ handle a STUN message on the same
port just fine, and I assume vendors active in the IETF or going to SIPit
would do just that.  (whatever "handle fine" means to them)

However the difference between a malicious attacker or benevolent tester
doing it, and this draft, is this will be an IETF RFC from the sip WG
specifying the "attack" as part of a legit SIP mechanism. (and it's not RFC
4475 :)  Whereas this WG has added a Required header or some other check
mechanism for virtually any other extension I can think of where the
proxy/registrar is really required to support something, this one does not
have such a mechanism.  Why?

And it's not the same as just a new option-tag or header - this is a
backwards-compatibility breaker.  There is no defined action for a
3261-compliant device to take when receiving such a binary message, is
there?  It cannot "reject" it with a sip response, cannot make the UA stop
sending it, nor can the UA detect the proxy doesn't support it even if the
UA wanted to check.  The UA will detect a keepalive loss, and try
"re-connecting" by re-registering and getting another keepalive loss,
indefinitely (although at least with back-off timers).

Meanwhile the operator of this 3261-compliant proxy just got a headache
courtesy of the IETF.  For each user who mistakenly checks some little
check-box which says whatever the vendor thought it should say for a
keepalive=stun semantics the operator must take action to correct - if
they're lucky.  If they're unlucky, and their proxy silently discards the
stun, they'll wonder why some UAs can make/receive calls for ~30 seconds,
then disappear, then come back, ad infinitum.  And this is a bad kind of
failure to detect/fix, because a reboot makes it "work" for a short time.

-hadriel
p.s. a STUN request should really be added to the next
sipping-torture-rfc4475bis.


> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: Thursday, July 06, 2006 2:07 PM
> To: Jeroen van Bemmel
> Cc: Cullen Jennings; sip@ietf.org; dean.willis@softarmor.com;
> rohan@ekabal.com
> Subject: Re: [Sip] Ready for WGLC on Outbound?
> 
> inline.
> 
> Jeroen van Bemmel wrote:
> 
> > Cullen,
> >
> > The point is that these are really 2 separate issues:
> > 1. Discovering the set of outbound proxy URIs (with keepalive=stun
> > parameters in them)
> > 2. Confirming that a given proxy in the set actually supports outbound
> > (ie validating the keepalive=stun information)
> >
> > Of these 2 options, (1) is indeed a separable problem. (2) - to me - is
> > not, for me outbound is not acceptable without something to address (2).
> > It really should say "the UAC MUST NOT use STUN until { mechanism for
> > (2) says its supported }"
> 
> I personally think that the presence of the keepalive=stun attribute in
> the URI is sufficient of a "confirmation". The examples you and others
> have presented for why this is needed are (1) misconfiguration (someone
> gives a URI to the client which is incorrect in that it contains this
> flag when it shouldn't), (2) malicious users, (3) constructed URIs. On
> (2), let me point out that an attacker that can modify DNS or modify SIP
> messages (or whatever is used for the discovery function) can launch far
> more effective attacks than just adding keepalive=stun in the hopes that
> it causes a low-quality proxy to crash.
> 
> On (1), misconfugration could potentially produce all kinds of things
> that would cause a client to send things to the server that might make
> it crash, including non-SIP URI, URI with invalid header parameters,
> etc. I don't see keepalive=stun as special.
> 
> On (3), this is definitely a possibility. However I think the best way
> to mitigate is to say 'don't do that'. SIP-outbound can have some words
> which warn implementations that they MUST NOT insert this parameter into
> URI that they construct from a domain name.
> 
> Finally, let me say that if you have a proxy that crashes on receipt of
> binary messages on its sip port, you are in for real trouble. I think a
> far better solution is for proxy vendors to fix their proxies, rather
> than for us to bend over backwards to design new mechanisms that help
> protect such fragile servers from ever seeing these messages. No matter
> what we put in our specs, attackers can still send binary messages that
> will result in the same result you are trying to prevent.
> 
> Thanks,
> Jonathan R.
> 
> 
> 
> 
> --
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Subject: [Sip] an alternative approach for outbound proxy discovery
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I want to suggest a possible alternative approach for outbound proxy 
discovery than the one in draft-koivusalo-sip-outbound-discovery. So, 
before I get to it, two caveats:

1. I am NOT proposing that this be added to sip-outbound, and I'm not 
suggesting that a normative reference to this or another mechanism 
should be in outbound. I believe that any discovery mecanism should be 
orthogonal, and what I am about to propose below is orthogonal as well.

2. I have not been able to follow all of the nearly countless messages 
on sip-outbound, so I'm not sure if what I'm about to propose has been 
proposed. So, please point me to relevant threads if it has been


The idea is to not use DNS, but rather use SIP mechanisms for the 
discovery. I much prefer a non-DNS solution for several reasons which 
I'll enumerate below. The basic mechanism combines two things:

1. usage of the 305/service-route as I have proposed in 
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-route-construct-01.txt

2. usage of a new header field in REGISTER responses which specifies 
additional URI to try REGISTERing to

So, here is the idea. UA starts with its AOR (sip:user@example.com) for 
example, or with a domain name learned from DHCP. In either case, it 
applies RFC3263 and gets the URI for a proxy. This is going to be the 
home proxy, and there is no keepalive=stun and thus no STUN keepalives 
here yet. UA sends a REGISTER request. The proxy receives this, and 
knows that it doesn't accept REGISTER directly like this; it needs to be 
sent via an edge proxy. There are many ways the proxy can know it got 
the REGISTER directly and not through an EP. For example, requests from 
EP come to a different port, or better yet, via TLS mutual auth from 
edge proxy.

So, home proxy redirects the request with a 305. The URI in the contact 
header field is an edge proxy, and has the keepalive=stun parameter in 
it. Now, this URI can be a specific IP address, or more interestingly, a 
hostname that resolves to any number of proxies. This basically becomes 
the first proxy in the outbound proxy set. If its a hostname, client 
performs regular old RFC3263 and picks one. Client sends a request 
there, now with the topmost Route set equal to this URI (this is what I 
have proposed for 305 handling in route-construct).

This arrives at the edge proxy, which inserts itself into Path, which is 
subsequently reflected into Service-Route, which the proxy can modify on 
its way towards the client (again this is what I have proposed in 
route-construct). This Service-Route points to that edge proxy.

Now, the UA has a single proxy (so far) in that outbound proxy set. 
However, that 200 OK to the REGISTER, in addition to having a 
Service-Route, would also have another header field, called 
Additional-Proxies or something like that. This header field would 
contain one or more URI pointing to proxies that the client should 
register additional flows through. For each of those, the client 
performs a registration (each of those URI would be the topmost route 
header field in a REGISTER message, each for a different reg-id). Now, 
it a multiplicity of flows and outbound proxies all set up.

The primary benefit of this mechanism over a DNS one is its dynamic 
nature. The edge proxy which receives the original REGISTER request and 
response is the one that inserts the Additional-Proxies header field. 
Thus, it can purposefully choose a specific proxy acting as its backup, 
or choose a proxy in a different geographic region, or whatever. These 
choices can differ from user to user. The original REGISTER requests can 
be load balanced across all of the proxies in a cluster as a consequence 
of DNS, but the second flow is selected explicitly by the first proxy.

This also lends itself extremely well to mid-dialog failover solutions, 
since the backup proxy is selected by the primary. Thus, it can choose 
one to which it is replicating state. It also allows for a variety of 
algorithms in selecting a record-route value.

With this mechanism, it is clear how the provider chooses the number of 
flows that a UA makes - its an additional one for each value of the 
Additional-Proxies header field.

It also allows us to keep the existing meaning of DNS queries, which are 
meant to choose a single server and not learn a set of servers that all 
must be contacted.

There is a cost of a single extra round trip for the very first REGISTER 
to be redirected. However, I do not view this as particularly onerous.

Thanks,
Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Jeroen van Bemmel wrote:

>> Jonathan Rosenberg wrote:
>>
>>> Finally, let me say that if you have a proxy that crashes on receipt 
>>> of binary messages on its sip port, you are in for real trouble. I 
>>> think a far better solution is for proxy vendors to fix their 
>>> proxies, rather than for us to bend over backwards to design new 
>>> mechanisms that help protect such fragile servers from ever seeing 
>>> these messages. No matter what we put in our specs, attackers can 
>>> still send binary messages that will result in the same result you 
>>> are trying to prevent.
>>
>>
>> I would agree with this in the case of UDP or in other cases where the 
>> transport does the framing. It seems that for stream based transports 
>> unrecognized binary data could mess up framing. Proxies could try to 
>> recover but I don't know if it's reasonable to expect proxies to keep 
>> connections open for an extended period of time when all they see is a 
>> some garbage every 30 seconds or so. It's not necessarily that the 
>> proxy crashes but that it gives up on a particular connection at some 
>> point.
>>
>> Thanks,
>> Anders
> 
> 
> Yes, exactly! And for TLS a proxy might interpret it as an encryption 
> failure, and trigger some anomaly detection system,

There wouldn't be a TLS decryption failure - this is all transparent 
data as far as TLS is concerned. Anders is correct that you might lose 
framing on the TCP connection, in whcih case the client gets no service. 
No other clients are affected. However, keep in mind that the use case 
here is a misconfiguration. Any number of misconfigurations, such as an 
invalid IP address, for example, will cause the client to not receive 
service.

I'm not objecting to the concept of this check. I just think its not 
necessary, something that might be optional at best. As such, if folks 
decide they want something, it can go in another draft. I dont think we 
need to burden sip-outbound with this.

Indeed, I'll note that, in the outbound proxy discovery mechanism I just 
proposed, the flag comes for free with the mechanism since the outbound 
proxies are confirmed via service-routes, which can contain the 
stun=keepalive flag. This would argue strongly that any kind of check, 
if needed, goes with the discovery mechanism and not sip outbound itself.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Hadriel:

I agree with your assessment, and would be _much_ happier if we simply
removed all the text that implied that the proxy has to maintain
multiple bindings with the same instance-id and reg-id.

I think, at the very least, the draft needs some motivating text around
why the proxy/registrar is expected to keep tabs of stale registration
state.

/a

Hadriel Kaplan wrote:
> My interpretation was the opposite - if a Register comes in with the same
> {instance-id,reg-id} then it should indeed replace/overwrite the previous
> binding of the same tuple.  I know there are several sections in the draft
> which imply otherwise, but I don't understand why.  There is no real value
> I can see to keeping a binding for an old flow of the same reg-id, since by
> definition the UA would only "re-use" a reg-id if the old flow failed (or
> even if it didn't think it had failed, but something rebooted and changed
> the flow).  
>
> I can see no realistic way in which an old flow of the same
> {instance-id,reg-id} would succeed in reaching the UA.  Am I missing
> something?
>
> -hadriel
> p.s. it's theoretically possible, if a sip-udp UA rebooted and resolved a
> new proxy address and sent a register to it with the same reg-id, that it
> may still be possible to reach the UA through the old "flow" for a time.  
> But is that a realistic case worth caring about?
>
>
>
>   
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Thursday, July 06, 2006 1:12 AM
>> To: Cullen Jennings
>> Cc: SIP WG
>> Subject: [Sip] Outbound Comment: binding key
>>
>>  From section 6.1:
>>
>>    When a reg-id header parameter is present in a Contact header field
>>    value in a REGISTER request, the corresponding binding is between an
>>    AOR and the combination of instance-id and reg-id.
>>
>> I don't think this is quite correct. This statement appears to be
>> asserting that {instance-id,reg-id} is a unique key to a binding
>> associated with an AOR. This would, in turn, imply that any successful
>> registration attempt with an {instance-id,reg-id} that matches an
>> existing binding would overwrite the existing record.
>>
>> I don't think that is what's intended, since section 6.2 contains the
>> following text:
>>
>>    o  If two bindings have the same instance-id and reg-id, the proxy
>>       SHOULD prefer the contact that was most recently updated.
>>
>> If the binding is only to {instance-id,reg-id}, then that text is
>> nonsensical: you can't have two identical bindings to the same AOR.
>>
>> I suspect that the text in 6.1 should actually read something more like:
>>
>>    When a reg-id header parameter is present in a Contact header field
>>    value in a REGISTER request, the corresponding binding is between an
>>    AOR and the combination of instance-id, reg-id, and network flow.
>>
>> /a
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>     
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>   



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>  such detection takes time and leaves a window of opportunity for
>    missed incoming requests, this mechanism allows the UA to use
>    multiple flows to the proxy or registrar.  This mechanism also uses a

multiple flows to multiple proxies or registrars

> Edge Proxy: An Edge Proxy is any proxy that is located topologically
>       between the registering User Agent and the registrar.

nit: topological in what sense? Certainly not in the IP sense. Its only 
in the SIP sense. SHould be clarified.

>   outbound-proxy-set A set of SIP URIs (Uniform Resource Identifiers)

Missing colon after outbound-proxy-set.

>  is power cycled.  Each UA can register multiple times over different
>    connections for the same SIP Address of Record (AOR) to achieve high

over different flows for the same...

> The registrar can use the reg-id label to
>    recognize that a UA is registering after a reboot or a network
>    failure.

reg-id is not just needed for reboots; its needed to differentiate 
different registrations to the same instance for purposes of high 
availability

> When a proxy goes to route a message to a UA for which it has a
> 
> 
> 
> Jennings & Mahy         Expires December 27, 2006               [Page 5]
> 
> Internet-Draft     Client Initiated Connections in SIP         June 2006
> 
> 
>    binding, it can use any one of the flows on which a successful
>    registration has been completed. 

this paragraph really applies to a home proxy. I think you should 
clarify that, and then summarize edge proxy procedures too.

> The registrar saves the instance-id
>    ("urn:uuid:00000000-0000-0000-0000-000A95A0E128") and reg-id ("1")
>    along with the rest of the Contact header field.  If the instance-id
>    and reg-id are the same as a previous registration for the same AOR,
>    the proxy uses the most recently created registration first. 

it uses it first when processing incoming requests. The rest of the 
paragraph is talking about processing register requests, and this 
summary sounds like its implying that you would delete the previous 
registration.

> When the proxy goes to forward a request to a given target,
>    it looks and finds the flows over which it received the registration.

this uses flows as plural, but the use case is talking about a single 
flow being registered at a time. So, I think you mean that the proxy 
looks and finds the flow over which it has received the most recent 
registration from that instance.

> If the proxy has multiple flows that all go to
>    this UA, it can choose any one of registration bindings for this AOR
>    that has the same instance-id as the selected UA.  

sure - but in this example section you've been talking about a single flow.

> hosts that can perform the outbound proxy/registrar function for the
>    domain.  Reliability is achieved by having the UA form two TCP
>    connections to the domain.

two flows

> The UA is configured with multiple outbound proxy registration URIs.
>    These URIs are configured into the UA through whatever the normal
>    mechanism is to configure the proxy or registrar address in the UA.

I think we are looking at several options which aren't the 'normal 
mechanism' for configuration. Indeed I think it shouldn't be the normal 
configuration mechanism at all. Can we just say that the UA is 
configured with multiple URI and the means is outside the scope of this 
specification.

> The domain also needs to ensure that a request for the UA sent to
> 
> 
> 
> Jennings & Mahy         Expires December 27, 2006               [Page 8]
> 
> Internet-Draft     Client Initiated Connections in SIP         June 2006
> 
> 
>    host1 or host2 is then sent across the appropriate flow to the UA.
>    The domain might choose to use the Path header approach (as described
>    in the next section) to store this internal routing information on
>    host1 or host2.

I don't understand the Path reference here. Path is useful if there is a 
physically distinct EP, which there isn't. For this particular use case 
(co-located outbound proxy/registrar and dual flows), I think that the 
proxy, when doing the registration lookup, will find two registrations. 
It needs to select the one corresponding to the flow that it knows 
about. Somewhere in the normative text also we need to say this.

> When a single server fails, all the UAs that have a flow through it
>    will detect a flow failure and try to reconnect. 

No, they won't try to reconnect, at least not immediately. You point out 
later that this is done gradually to avoid avalance restart, but this 
sentence in particular contradicts that. I would suggest:

When a single server fails, the UAs that have a flow through it will 
eventually detect failure of that flow. The clients themselves will need 
to reconnect in order to re-establish the flow. If clients attempted to 
reconnect immediately, this would cause a flood of connection attempts 
at around the same time, a situation known as avalanche restart. To 
avoid this condition, if a UA has multiple flows, and one of the servers 
fails.....

>   restart.  If a UA has multiple flows, and one of the servers fails,
>    the UA delays the specified time before trying to form a new
>    connection to replace the flow to the server that failed.  By

new flow

>   When used in this fashion to achieve high reliability, the operator
>    will need to configure DNS such that the various URIs in the outbound
>    proxy set do not resolve to the same host.

this statement is fine but doesn't belong in an overview of operation 
section i think.

>  Some SIP deployments use edge proxies such that the UA sends the
>    REGISTER to an Edge Proxy that then forwards the REGISTER to the
>    Registrar.  The Edge Proxy includes a Path header [RFC3327] so that
>    when the registrar later forwards a request to this UA, the request
>    is routed through the Edge Proxy.  There could be a NAT or firewall
>    between the UA and the Edge Proxy.

you don't show the NAT in any of the other two use cases and certainly 
they can be there too. THis makes it sound kind of like you need an EP 
in cases where there is a NAT, which is not the case.

>   These systems can use effectively the same mechanism as described in
>    the previous sections but need to use the Path header.  When the Edge

this is kind of weasely text. I'd suggest:

In this topology, the same mechanism as described in the previous 
section is used, with the addition of the Path header.

>  registrar succeeds, it allows the Edge Proxy to indirectly check the
>    user's authentication information via the registrar.  The identifier
>    SHOULD be placed in the user portion of a loose route in the Path
>    header.  If the registration succeeds, the Edge Proxy needs to map

normative statements do not belong in overview sections. see rfc4485.

>  For connection-less transports, a flow definition could change
>    because a NAT device in the network path reboots and the resulting
>    public IP address or port mapping for the UA changes.  To detect
>    this, requests are sent over the same flow that is being used for the
>    SIP traffic. 

stun requests are sent over the same flow

>  cycles of the device.  The Instance ID MUST NOT change as the device
>    moves from one network to another.

suggest also saying that the instance ID MUST NOT vary across different 
network interfaces that the device has. There has been some confusion 
about whether a multi-homed device would have different instance IDs on 
each network interface.

>   The UA SHOULD include a "sip.instance" media feature tag as a UA
>    characteristic [RFC3840] in requests and responses.  

in the very next section, you say that inclusion of sip.instance media 
feature tag in a registration is a MUST. So, these are not consistent.

Also, you never actually say that this value is the UA instance URN 
you've been talking about in the previous paragraphs.

>  The value of this parameter MUST be a URN [RFC2141].  One case where
>    a UA may not want to include the URN in the sip.instance media
>    feature tag is when it is making an anonymous request or some other
>    privacy concern requires that the UA not reveal its identity.

..may not want to include its instance identifier URN in the 
sip.instance....

>    UAs obtain at configuration time one or more SIP URIs representing
>    the default outbound-proxy-set. This specification assumes the set
>    is determined via any of a number of configuration mechanisms and
>    future specifications may define additional mechanisms such as using
>    DNS to discover this set.  How the UA is configured is outside the
>    scope of this specification. 

I think the term 'configuration' is a bit loaded here, and others I think
have commented on this too. I'd suggest, instead:


UAs obtain, through some means outside the scope of this specification, 
one or more SIP URIs representing the default outbound-proxy-set.

>  URIs.  For each outbound proxy URI in the set, the UA MUST send a
>    REGISTER in the normal way using this URI as the default outbound
>    proxy. 

the word 'normal' concerns me here. What does this mean? Per rfc3261 
procedures? Thats not really right, since we are updating those 
procedures here. Suggest:

For each outbound proxy URI in the set, the UA MUST send a REGISTER 
using the procedures defined here (which augment those defined in 
RFC3261) using this URI as the default outbound proxy.

>    These ordinary registration requests MUST also add a distinct reg-id
>    parameter to the Contact header field. 

remove 'ordinary'

You also need to define 'distinct'. What you mean is that the value must 
be distinct across all other flows to be concurrently registered by the 
same instance.

> The
>    reg-id sequence does not have to be sequential but MUST be exactly
>    the same reg-id sequence each time the device power cycles or reboots
>    so that the reg-id values will collide with the previously used
>    reg-id values.  This is so the proxy can realize that the older
>    registrations are probably not useful.

This text is very confusing, especially the first sentence which says 
that a sequence of something doesn't have to be sequential. I'd suggest:

Furthermore, if the UA will be using N simultaneous flows, the N values 
of the reg-id that are used MUST be reused in subsequent refreshes or 
brand new registrations made at a later time. As a consequence, the 
reg-id forms a sort of sequence number, giving a persistent identifer to 
each of the N concurrent registrations that the UA will make. Indeed, 
one perfectly valid choice for the reg-id is to use a monotonically 
increasing sequence number from 1 to N, however this is not required.

>  The UAC MAY examine successful registrations for the presence of an
>    'outbound' option-tag in a Supported header field value.  Presence of
>    this option-tag indicates that the registrar is compliant with this
>    specification.

this is meaningless unless you specify something that the client does 
differently based on the presence of this header field.


Also, this section doesn't talk about how a UA determines whether or not 
the registrar supports the contact header field extensions and new 
mechanisms defined here. It needs to do that.

> The UA performs normal DNS resolution on the next hop URI (as
>    described in RFC 3263 [RFC3263]) to find a protocol, IP address, and
>    port.  For non-TLS protocols, if the UA has an existing flow to this
>    IP address, and port with the correct protocol, then the UA MUST use
>    the existing connection.  For TLS protocols, there must also be a
>    match between the host production in the next hop and one of the URIs
>    contained in the subjectAltName in the peer certificate.  If the UA
>    cannot use one of the existing flows, then it SHOULD form a new flow
>    by sending a datagram or opening a new connection to the next hop, as
>    appropriate for the transport protocol.

I'm pretty sure that this text would fundamentally require UA changes 
for any mid-dialog failover solution to work. I'm still noodling over 
whether there is a better matching algorithm that is more likely to work 
with future mid-dialog failover mechanisms.

> The time between keepalive requests when using UDP-based transports
> 
> 
> 
> Jennings & Mahy         Expires December 27, 2006              [Page 14]
> 
> Internet-Draft     Client Initiated Connections in SIP         June 2006
> 
> 
>    SHOULD be a random number between 24 and 29 seconds while for TCP-
>    based transports it SHOULD be a random number between 95 and 120
>    seconds.  These times MAY be configurable.

you need to clarify that you redraw the random number each time you send 
a keepalive, not just once on startup. ALso, the bit thats configurable 
are the ranges (24-29, 95-120). Also should specify that the random 
number is uniformly distributed.

> If the parameter is present, the UA needs
>    to periodically perform keepalive checks by sending a STUN [I-D.ietf-
>    behave-rfc3489bis] Binding Requests over the flow.

you should point to the binding keepalive usage specifically in 3489bis.

>  These three times MAY be configurable in the UA.  The three times
>    are:
>    o  max-time with a default of 1800 seconds
>    o  base-time-all-fail with a default of 30 seconds
>    o  base-time-not-failed with a default of 90 seconds

the formula doesn't include the variables 'base-time-all-fail' or 
'base-time-not-failed'. It just has 'base-time'. Suggest:

      o base time with a default of 90 seconds if there are other 
non-failed flows, and 30 seconds if all other flows have failed


>   When an Edge Proxy receives a registration request with a reg-id
>    header parameter in the Contact header field, it MUST form a flow
>    identifier token that is unique to this network flow.  The Edge Proxy
>    MUST insert this token into a URI referring to this proxy and place
>    this URI into a Path header field as described in RFC 3327 [RFC3327].
>    The token MAY be placed in the userpart of the URI.

It also needs to remember the flow and associated it with this token.

>  Algorithm 1: The proxy base64 decodes the user part of the Route
>       header.  For a TCP-based transport, if a connection specified by
>       the file descriptor is present and the creation time of the file
>       descriptor matches the creation time encoded in the Route header,
>       the proxy forwards the request over that connection.  For a UDP-
>       based transport, the proxy forwards the request from the encoded
>       file descriptor to the source IP address and port.

you need to specify the conditions under which a 410 is sent. You do 
that for algorithm 2 but not here.

> For a binding
>    with an instance-id, the registrar still stores the Contact header
>    field value URI with the binding, but does not consider the Contact
>    URI for comparison purposes. 

I think you mean:

For a binding to an instance-ID and reg-ID, the registrar still stores....

Also, you may want to clarify handling when there is an instance-ID but 
no reg-id, and reg-id but no instance ID.


You also need to point out here quite explicitly that it is very likely 
that there can be registrations for the same AOR that have difference 
instance ID and reg-IDs, but the same contact URI. This happens when the 
client is behind NAT and includes a private IP address in the contact 
URI, such as sip:10.0.1.1:5060.

>  In addition to the normal information stored in the binding record,
>    some additional information MUST be stored for any registration that
> 
> 
> 
> Jennings & Mahy         Expires December 27, 2006              [Page 18]
> 
> Internet-Draft     Client Initiated Connections in SIP         June 2006
> 
> 
>    contains a reg-id header parameter in the Contact header field value.
>    The registrar MUST store enough information to uniquely identify the
>    network flow over which the request arrived.  For common operating
>    systems with TCP, this would typically just be the file descriptor.
>    For common operating systems with UDP this would typically be the
>    file descriptor for the local socket that received the request, the
>    local interface, and the IP address and port number of the remote
>    side that sent the request.

this is only for registrars co-located with edge proxies. Need to 
clarify that.

>    o  The proxy MUST NOT populate the target set with more than one
>       contact with the same AOR and instance-id at a time.  If a request
>       for a particular AOR and instance-id fails with a 410 response,
>       the proxy SHOULD replace the failed branch with another target (if
>       one is available) with the same AOR and instance-id, but a
>       different reg-id.
>    o  If two bindings have the same instance-id and reg-id, the proxy
>       SHOULD prefer the contact that was most recently updated.

here is where you need to add words that in the co-located EP/home proxy 
case, the reg-id corresponding to the flow held by that EP is used. Also 
I think generally more guidance is needed on selection of registrations 
when there are multiple reg-id. For example, if you are using 
geo-redundancy, the home proxy would use the reg-id received from teh EP 
in its own region.


> Additionally, when the
>    proxy forwards a request to a binding that contains a reg-id, the
>    proxy MUST send the request over the same network flow that was saved
>    with the binding.

only applies to the EP -should be clear on that.

> 7.1.  STUN Processing
> 
>    This document defines a new STUN usage for connectivity checks.

no no. Firstly its not 'connectivity check', as thats something 
different entirely. This is binding keepalive usage. I think we should 
define the usage in rfc3489bis, and merely specify here that this usage 
is applied to sip.

The only thing that needs to be in here is the keepalive=stun bit. 
Everything else in this section is already in rfc3489bis.

> The simple case is when there are no edge proxies.  In this case, the
>    only time an entry can be added to the routing for a given AOR is
>    when the registration succeeds.  SIP already protects against
>    attackers being able to successfully register, and this scheme relies
>    on that security. 

attackers can be authenticated too. The scheme in here does not assume 
that attackers are unauthenticated.


>  This section registers a new media feature tag, per the procedures
>    defined in RFC 2506 [RFC2506].  The tag is placed into the sip tree,
>    which is defined in RFC 3840 [RFC3840].
> 
>    Media feature tag name:  sip.instance
> 
>    ASN.1 Identifier:  New assignment by IANA.

If memory serves me correctly, in the end with rfc3840 I had to pick the 
ASN.1 identifiers.




Thanks,
Jonathan R.






-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Thu Jul 06 18:43:10 2006
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From: Jonathan Rosenberg <jdrosen@cisco.com>
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Kevin,

Thanks for taking a crack at this. The mid-dialog case is a nasty hard 
problem and one we really do need to solve.

I'm not sure the mechanism you propose has the necessary security 
properties, however. The attack I'm worried about is where an attacker, 
who is a valid but malicious user in the system, registers to the 
network and includes an instance ID and reg-id matching that of another 
user, the target. In this case, the edge proxy creates the same flow ID, 
and probably replaces the binding between that flow ID and the flow on 
which the register was received. Thus, an attacker will hijack incoming 
calls targeted for the target.

Another big part of the problem is how the edge proxies construct a URI 
whose domain part happens to resolve to the actual two proxies which 
happen to be the ones that the client registered through. With current 
sip-outbound neither edge proxy knows what the other edge proxy is, so 
how could you construct such a URI? (note I've just posted an 
alternative solution for outbound discovery which will make this possible).

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Thu Jul 06 19:10:39 2006
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>, "IETF SIP List" <sip@ietf.org>
Subject: Re: [Sip] an alternative approach for outbound proxy discovery
Date: Fri,  7 Jul 2006 02:08:56 +0300
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Hi Jonathan,

There are a couple of issues/deployment restrictions with your =
proposal.

1. The home proxy has, as you indicate, to know that an edge proxy is =
needed. If the edge proxy is used only for STUN keepalive that may be =
difficult, because the home proxy would need to know when keepalive is =
needed in the first place. Or, of course it could force all registrations =
to go via the edge proxy.

2. The home proxy and the edge proxy may be in different administrative =
domains, so the home proxy may not know the address of the edge proxy =
(maybe not even in which domain it's located).

3. The request may traverse a numbet of proxies in any case, so the home =
proxy would have to know whether one of those is the edge proxy or =
not.

Regards,

Christer


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From sip-bounces@ietf.org Thu Jul 06 19:10:50 2006
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Ben Campbell" <ben@estacado.net>, "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Subject: Re: [Sip] Ready for WGLC on Outbound?
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Hi,

I agree with Bell. A server shouldn't crash if someone sends "non-SIP" =
stuff to it. In that case people will be in trouble - with or without =
outbound and STUN.

The main reasons I want to know whether the proxy supports STUN are:

1. I don't want to send STUN request for nothing. It's not about whether =
the receiver will crash or not - it's just bad and unoptimized behavior. =
And, as Jeroen mentioned, the proxy could blacklist a client sending =
"unknown stuff".

2. A client may choose another edge proxy, if the current does not support =
STUN.

Regards,

Christer


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Subject: Re: [Sip] Comments on draft-gurbani-sip-domain-certs-01
Date: Thu, 6 Jul 2006 19:35:22 -0400
To: "Vijay K. Gurbani" <vkg@lucent.com>
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On Jul 5, 2006, at 4:01 PM, Vijay K. Gurbani wrote:

Quick question:
> on the same outcome you mention above; i.e., it suffices
> that when contacting "sips:alice@atlanta.com", the certificate
> presented contains "atlanta.com."
>
Do you mean "the name in the certificate contains the string" or  
"ends with the string" or something else. In other words, does a  
certificate for badboy.atlanta.com.chaoscomputerclub.org match a sip  
uri of sip:alice@atlanta.com.

One would think not...

Or do you mean "the name in the certificate is precisely the string  
'atlanta.com'", in which case does the certificate match sip uri's  
registered in a subdomain, e.g. alice@downtown.atlanta.com.

The "contains" terminology ought to be a bit tighter, no?

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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00 Issue
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 Hi Eric,

   Thx for the your comments and insights. Pl find my replies below

<< Removed Unrelated stuff>

No, that's not correct either, as I showed in an earlier diagram, of
exactly this case, repeated here.

Alice        Atlanta         Biloxi
-----        -------         ------
AES-SHA      AES-SHA         3DES-SHA
AES-SHA256   AES-SHA256      AES-SHA

In this case, Alice and Atlanta will (most likely) negotiate AES-SHA256.
Atlanta will offer AES-SHA and AES-SHA256. Biloxi will select AES-SHA.=20

Now, it's certainly true that if Alice configures her client to be
AES-SHA256 only then she won't be able to get a call through, but that's
a feature because it's a deliberate advertisement that she's unwilling
to do other algorithms.=20


1) Here you think, after the upgrade Alice still advertises the older
cipher-suites. Where I think, it will use the latest cipher-suites in
the TLS. And say there are some cipher-suites from the last versions,
and some more from the second last version etc in a header. It can
enumerate any number of cipher-suite with the preference. Any proxy down
the line can use from the preferential order.  In your algorithm, Take
the case of 40 bit cipher-suites being advertised by Alice, and it is
being selected  due to commonality in the hop way ahead of UAC. Though
there is another more secure cipher-suite available and honoured by the
UAC, but that is not carried forward. Let us take the following generic
case

          UAC            P1       P2       P3        P4         UAS
          ----          ----     ----     ----      ----        ----
Ciphers  A,B,C,D,E       B,C     C,D      D,B       B,E         B,E   =20

   UAC advertises all possible cipher-suites. Here hop P1 to P2 will be
established as C is the common cipher-suite between supported and
offered. Now as the advertised cipher-suites are not carried forward,
Call will fail between Hop P2 and P3, as D is not the cipher-suite used
on the earlier hop and P2 has no knowledge that UAC is willing to honour
D. P2 will reject the call.=20

2) You seem to focus on the mandatory cipher-suite being a common
denominator always to setup the call. I found following for Mandatory
cipher-suites=20

RFC 2246 in Mandatory Cipher Suite section says=20

   "In the absence of an application profile standard specifying
   otherwise, a TLS compliant application MUST implement the cipher
   suite TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA. "

RFC 4346 in Mandatory Cipher Suites says=20
   "In the absence of an application profile standard specifying
   otherwise, a TLS compliant application MUST implement the cipher
   suite TLS_RSA_WITH_3DES_EDE_CBC_SHA."
=20
RFC 3261 mentions that TLS_RSA_WITH_AES_128_CBC_SHA MUST be supported,
but others can be supported too. It asks for supporting 3DES also.
=20
Going forward, we might say AES256 MUST be supported, and others can be
too. Your expertise will be helpful for commenting on chances of change
of MUST for SIP.=20


Now let us take a generic case with moving mandatories


     UAC      P1         P2           P3          UAS
     ---      --         -----        -----       -----
 ciphers      A1,A2       A1          A1,A2       A1,A2

Assume A1 is older mandatory cipher, and A2 is new mandatory cipher.=20

Your scheme doesn't allow the usage of A2 on the further hops (P3-UAS)
down the line, though being supported and acknowdged by UAC, as
somewhere in between older cipher is used.=20



Thx
Samir


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Subject: RE: [Sip] Review of draft-srivastava-sip-e2e-ciphersuites-00 Issue
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Hi Eric,

   This is another thread, to discuss content in the response.

<<Removed Unrelated stuff>>

The whole concept of "least strong" is a mistake. What should be
reported back is the entire list of proxies the message went through and
the relevant crypto algorithms.

1. From the another threads, it looks that you are in agreement for a
reverse channel for indicating back the cipher etc used at each hop.

2. I am okay with sending the complete via trails with ciphers etc used,
and let UAC decide how it wants to interpret it. It will be having more
number of bytes. If they are in the same packet, it doesn't cost much.=20

Thx
Samir


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   From: Jonathan Rosenberg <jdrosen@cisco.com>

   Thanks for taking a stab at this. The 'hidden semantic' behind REFER is 
   a looming potential problem that I don't think we've fully captured. 
   Your draft takes a good step towards uncovering it, but I don't think it 
   quite gets there.

Thanks!  In the next revision, I'll take care of these issues.  At the
moment, my interest was just to list all the distinct meanings and
determine if it was *possible* for a UA to determine at all times
which was intended.  I don't think we've made that mistake, but until
it's pointed out that the mistake can be made, it's accidental that we
haven't.

I'm also curious if there are cases I've missed.  I'd be surprised if
I've found all of them.

Dale

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Subject: RE: [Sip] Update to fork-loop-fix submitted
Date: Fri, 7 Jul 2006 09:12:54 +0530
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Comment Inline.

Regards,
Ravi.

--
=20
Ravishankar. G. Shiroor
Wipro Technologies, Bangalore.
=20
ravishankar.shiroor@wipro.com
--
=20

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: Thursday, July 06, 2006 11:19 PM
> To: RaviShankar Shiroor (WT01 - Product Engineering Solutions)
> Cc: sip@ietf.org
> Subject: Re: [Sip] Update to fork-loop-fix submitted
>=20
>=20
> On Jul 5, 2006, at 11:43 PM, <ravishankar.shiroor@wipro.com>=20
> <ravishankar.shiroor@wipro.com> wrote:
>=20
> >
> > It is always better to reduce the computation in the path of normal
> > success case and introduce it in the path of a rarer=20
> scenario. I think
> > it will be a fair assumption to make that Spirals will be rarer =20
> > than the
> > non-spiraling scenarios.
>=20
> But in this case, you are reducing computation (perhaps=20
> trivially) at =20
> the
> cost of increased network traffic.

True, but it is not going to make it increase the traffic exponentially
which is the reason why this draft is coming out.
If the traffic were not to increase exponentially, max-forwards would be
considered sufficient to handle it right?

>=20
> >
> > In terms of increasing logic, the increase in logic would be very =20
> > small
> > if it all any.
> > It would be a trade-off that should be left to the=20
> implementer to take
> > if he considers that hash computation is expensive on his=20
> platform. If
> > this id recommends the computation of hash at the MUST=20
> strength in the
> > first pass of the SIP request, an implementer would not have that
> > freedom to make that trade-off..
> >
> > Regards,
> > Ravi.
> >
> > --
> >
> > Ravishankar. G. Shiroor
> > Wipro Technologies, Bangalore.
> >
> > ravishankar.shiroor@wipro.com
> > --
> >
> >
> >> -----Original Message-----
> >> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> >> Sent: Wednesday, July 05, 2006 8:21 PM
> >> To: RaviShankar Shiroor (WT01 - Product Engineering Solutions)
> >> Cc: sip@ietf.org
> >> Subject: Re: [Sip] Update to fork-loop-fix submitted
> >>
> >> I understand the suggestion. You are trading off letting the
> >> request through a loop one extra cycle for some lower amount
> >> of processing the first time the request went through the
> >> proxy. If you believe that spirals are relatively rare
> >> things, this change would result in proxies very rarely
> >> having to build the loop/spiral differentiation hash.
> >>
> >> But as Jonathan points out, it's not clear that the hash
> >> computation is expensive enough to warrant this extra logic
> >> and the extra network traffic it would allow. I expect that
> >> the hash calculation expense is small compared to the
> >> parse/search/compare operations for "Proxy checks to see if
> >> its address is there in the via already".
> >>
> >> RjS
> >>
> >> On Jul 3, 2006, at 5:39 AM, <ravishankar.shiroor@wipro.com>
> >> <ravishankar.shiroor@wipro.com> wrote:
> >>
> >>>
> >>> Hi Robert,
> >>>
> >>> I had suggested one more alternative
> >>>
> >>> (http://www1.ietf.org/mail-archive/web/sip/current/msg14359.html )
> >>>
> >>> Quoted from my previous mail:
> >>>
> >>> <quote>
> >>> I was thinking how to reduce the impact of overhead for
> >> calculating
> >>> the
> >>> digest.
> >>>
> >>> If it is suggested as follows, I think it will reduce the
> >> overhead on
> >>> normal scenarios:
> >>>
> >>> Proxy checks to see if its address is there in the via already.
> >>> If it is not there, it proceeds normally without calculating the
> >>> digest
> >>> and inserting Via to the outgoing request without the digest.
> >>> If it is there, then it is possible that the request is=20
> looping. So,
> >>> this time around, it calculates a digests and inserts and sends it
> >>> along.
> >>> If it is looping, the request will definitely come in
> >> again, which can
> >>> be rejected.
> >>> This way the impact of calculating the digest is required (reduced
> >>> to a
> >>> smaller percent of the sessions) only when the proxy is
> >> suspicious the
> >>> request might be looping.
> >>> Of course this results in the scenarios mentioned in the id to be
> >>> successful in completing the first loop, but the second=20
> loop will be
> >>> detected and arrested. I think this can be applicable to
> >> general loop
> >>> detection, and not just fork loop fixes.
> >>> </quote>
> >>>
> >>> Regards,
> >>> Ravi.
> >>>
> >>> --
> >>>
> >>> Ravishankar. G. Shiroor
> >>> Wipro Technologies, Bangalore.
> >>>
> >>> ravishankar.shiroor@wipro.com
> >>> --
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> >>>> Sent: Sunday, June 25, 2006 8:22 PM
> >>>> To: IETF SIP List
> >>>> Subject: [Sip] Update to fork-loop-fix submitted
> >>>>
> >>>> Until it appears in the repository, see:
> >>>>
> >>=20
> http://www.nostrum.com/~rjsparks/draft-ietf-sip-fork-loop-fix-02.txt
> >>>>
> >>>> This was a moderately heavy touch - based on list feedback,
> >>>> I've moved and updated the sections covering the
> >>>> loopdetection algorithm in from 3261. In the process of
> >>>> writing the clarifying text, I discovered several places that
> >>>> need clarification. They are listed as Open Issues in the
> >>>> draft. Basically there are fields that I can't recall or
> >>>> reconstruct how they would be useful in the part of the hash
> >>>> used for loop-detection or spiral differentiation. If anyone
> >>>> can remember why they need to be included, send a note to the
> >>>> list and we'll capture that in the draft.
> >>>>
> >>>> Please review the updated loop-detection text very carefully.
> >>>>
> >>>> Here's the changelog from the draft:
> >>>>
> >>>>        Integrated several editorial fixes suggested by
> >>>> Jonathan Rosenberg
> >>>>
> >>>>        Noted that the reduction of the attack to a single
> >>>> registration
> >>>>        against a single URI as documented in previous versions,
> >>>> is, in
> >>>>        fact, going to be effective against implementations
> >>>> conforming to
> >>>>        the standards before this repair.
> >>>>
> >>>>        Re-incorporated motivation from the original
> >>>> maxforwards-problem
> >>>>        draft into the security considerations section based
> >>>> on feedback
> >>>>        from Cullen Jennings
> >>>>
> >>>>        Introduced replacement text for the loop=20
> detection algorithm
> >>>>        description in RFC 3261, fixing the bug 648 (the
> >>>> topmost Via value
> >>>>        must not be included in the second part) and=20
> clarifying the
> >>>>        algorithm.  Removed several other fields suggested
> >> by 3261 and
> >>>>        placed open issues around their presence.
> >>>>
> >>>>        Added a Notes to Implementors section capturing the
> >>>> "common way"
> >>>>        text and pointing to the interoperability issues that have
> >>>> been
> >>>>        observed with loop detection at previous SIPits
> >>>>
> >>>>
> >>>> Thanks,
> >>>>
> >>>> RjS
> >>>>
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use
> >>>> sip-implementors@cs.columbia.edu for questions on current sip
> >>>> Use sipping@ietf.org for new developments on the=20
> application of sip
> >>>>
> >>>
> >>>
> >>> The information contained in this electronic message and any
> >>> attachments to this message are intended for the exclusive use of
> >>> the addressee(s) and may contain proprietary, confidential or
> >>> privileged information. If you are not the intended recipient, you
> >>> should not disseminate, distribute or copy this e-mail. Please
> >>> notify the sender immediately and destroy all copies of this
> >>> message and any attachments.
> >>>
> >>> WARNING: Computer viruses can be transmitted via email. The
> >>> recipient should check this email and any attachments for the
> >>> presence of viruses. The company accepts no liability for any
> >>> damage caused by any virus transmitted by this email.
> >>>
> >>> www.wipro.com
> >>
> >>
> >
> >
> > The information contained in this electronic message and any =20
> > attachments to this message are intended for the exclusive use of =20
> > the addressee(s) and may contain proprietary, confidential or =20
> > privileged information. If you are not the intended recipient, you =20
> > should not disseminate, distribute or copy this e-mail. Please =20
> > notify the sender immediately and destroy all copies of this =20
> > message and any attachments.
> >
> > WARNING: Computer viruses can be transmitted via email. The =20
> > recipient should check this email and any attachments for the =20
> > presence of viruses. The company accepts no liability for any =20
> > damage caused by any virus transmitted by this email.
> >
> > www.wipro.com
>=20
>=20

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>   the connection.  SIP entities that can only open a connection in a
>    specific direction -- perhaps because of Network Address Translation
>    (NAT) and firewall reasons -- reuse their connections using the
>    mechanism described in [1].

another important reason is that, in the case of TLS, only one side will
offer a certificate. In other words, connect-reuse only applies in truly
symmetric models.

>  Opening an extra connection where an existing one is sufficient can
>    result in potential scaling and performance problems.  Consider the
>    call flow shown below where Proxy A and Proxy B use the Record-Route
>    mechanism to stay involved in a dialog.  Proxy B will establish a new
>    TLS connection just to send a BYE request.

this presumes in general that there aren't persistent, long-lived
connections already set up.

>  Opening an extra connection where an existing one is sufficient can
>    result in potential scaling and performance problems.  Consider the
>    call flow shown below where Proxy A and Proxy B use the Record-Route
>    mechanism to stay involved in a dialog.  Proxy B will establish a new
>    TLS connection just to send a BYE request.

the bottom of section 3 already talked about benefits; you should
combine together into one place.

> This effectively doubles the load on each
>    entity. 

definitely not.

It doubles the amount of work for connection establishment, but its far
from obvious to me in most use cases that the overall work of the proxy
is dominated by connection establishment. Today, for the vast majority
of cases, there are fairly fixed relationships between proxies and this
mechanism won't impact the workload at all. This only starts to show up
as an issue in cases of broad interconnection across numerous domains
(i.e., every enterprise connects to every other enterprise). Again
though I wonder if the connection establishment is really the dominant
CPU cost.

>  ReINVITEs or UPDATE [7] requests are expected to be handled
>    automatically and rapidly in order to avoid media and session state
>    from being out of step.  If a reINVITE requires a new TLS connection,
>    the reINVITE could be delayed by several extra round-trip times.
>    Depending on the round-trip time, this combined delay could be
>    perceptible or even annoying to a human user. 

Can you give a specific use case? I don't think there are established
guidelines for how long it takes to add video, for example.

>  The act of reusing a connection is initiated by an user agent client
>    (UAC, or the proxy half of the UAC) when it adds an "alias" parameter
>    to the added Via header (the parameter itself is defined later).
>    When a user agent server (UAS, or the proxy half of the UAS) receives
>    the request, it examines the topmost Via header. 

I think you mean just 'client' and 'server' which are well defiend in
rfc3261, and are what you want. 'proxy half of a UAC' makes no sense.

> Without any loss of
>    generality, it should be clear that the same technique can be used
>    for connection reuse between a UAC and an edge proxy, or between an
>    edge proxy and a UAS, or between an UAC and an UAS.

Just talk about clients and servers.

> The presence of the "alias" parameter indicates that P1 does support
>    aliasing.  P2 now authenticates the connection and if the
>    authentication was successful, P2 creates an alias to P1 using the
>    advertised address in the topmost Via.

Firstly, you need to define 'successful' authentication. You need to
match the domain in the certificate with the via address. Also, this
last sentence is very confusing, since the table is NOT populated with
the advertised address, its populated with the received address, as
described in the text below.

Also how do you populate destination port? The overview section doesn't
say. I believe it is NOT the received port (as IP address is) but rather
the one advertised in the sent-by field, or 5061 (the default) if no
port is listed.

> To implement connection aliases for resolved addresses, a SIP node
>    could (for example) search an additional data structure (the alias
>    table) prior to opening a new connection, or could modify the data
> 
> 
> 
> Mahy, et al.             Expires August 5, 2006                 [Page 7]
> 
> Internet-Draft            SIP Connection Reuse             February 2006
> 
> 
>    structure in which it keeps active connection state so that aliases,
>    active connections, and blacklisted nodes are all discovered when
>    looking for an active connection.

you've not defined 'an additional data structure' or blacklisted nodes;
I couldn't follow this paragraph.

> 6.  Requirements

you may want to have an intro paragraph which says that these are the
requirements which motivated this specification

> 7.  Formal Syntax
> 
>    The following syntax specification uses the augmented Backus-Naur
>    Form (BNF) as described in RFC 4234 [6].  This document extends the
>    via-params to include a new via-alias defined below.

please use the =/ syntax notation that we've started using for sip
specs. Its much better.

>  request.  Whenever the RFC3263 server selection mechanism executed at
>    the receiver results in the choice of this IP address, port, and
>    transport tuple, the alias MUST be used instead.

I think a SHOULD is fine here; and indeed if the connection has gone
away you obviously won't be reusing the alias.

> An alias is formed at the sender of the request when it executes the
>    RFC3263 server selection mechanism to arrive at an IP address, port,
>    and transport tuple to send a request to.  Subsequent requests going
>    to the same destination address MUST use the alias instead.

Hmm this MUST is stronger than in rfc3261, which actually says SHOULD. I
think the existing SHOULD is fine.


>  Only one alias SHOULD exist for the resolved address.  If more than
>    one alias is requested because of race conditions (or any other

I don't think the SHOULD here is capitalized; you're not specifying an
action to take.

You need to define 'resolved address'. I don't know what that is.

> Because an alias connection might be reclaimed during a transaction,
>    clients SHOULD NOT enforce the RFC 3261 requirement of sending CANCEL
>    and ACK (for non 2xx responses) to the same port.  If the alias
>    connection no longer exists, the client SHOULD open a new connection
>    to the resolved address and send the CANCEL or ACK there instead.
>    The newly opened connection MAY be inserted into the alias table.

Hmm. I think this rule only applies to aliases formed through the
mechanism defined here, NOT through the ones obtained from rfc3263.

> And furthermore,
>    it MUST reuse the connection when subsequent requests in the same or
>    different transactions are destined to the same resolved address.

Again per above, I think this is SHOULD. Generally with all of these
reuse requirements SHOULD is OK.

>  Whether or not to allow an aliased connection ultimately depends on
> 
> 
> 
> Mahy, et al.             Expires August 5, 2006                 [Page 9]
> 
> Internet-Draft            SIP Connection Reuse             February 2006
> 
> 
>    the recepient of the request.  

I think you mean that a client gets no confirmation that its downstream
peer created the alias, or indeed that it even supports this specification.

> Clients must authenticate the connection before forming an alias.
>    Section 9.1 discusses the authentication steps in more detail.

You need to integrate that processing as part of the core client and
server behaviors.


> A TLS server conformant to this specification MUST ask for a client
>    certificate; if the client possesses a certificate, it will be
>    presented to the server for mutual authentication.  The server MUST
>    ensure that the canonical host name of the client is present either

where does one obtain the canonical hostname?

It seems to me, that in a proxy to proxy case, the receiving proxy (the 
server side) will need to check the record-route hostname against the 
client certificate.


> This document adds a parameter to the SIP header field parameters
>    registry:

reference rfc3968

>  Reference:  This document

Should say: RFC XXXX [[NOTE TO RFC-EDITOR: please replace with the final 
RFC number of this specification]]


> 13.1  Normative References
> 
>    [1]  Jennings, C. and R. Mahy, "Managing Client Initiated Connections
>         in the Session Initiation  Protocol (SIP)",
>         draft-ietf-sip-outbound-01.txt (work in progress), October 2005.

I dont think this is a normative reference.


Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Having just done a detailed review of the document, I will say that I am 
skeptical of the value of this specification at this time.

Firstly, it has fairly limited deployment scenarios:

1. no NATs between the elements
2. mutual TLS auth only

assuming these are true, its valuable only when there is generally a lot 
of churn in TLS connection setups. For example, in an environment where 
there is a fixed and known-ahead-of-time peering relationship between 
providers, connect-reuse doesn't do much for you anyway, since there are 
relatively few TLS connections and they are set up when the system 
boots, or perhaps just once.

Even if the connections aren't set up at startup, but on demand, over 
time the value of connect-reuse is really zero if the total number of 
TLS connections that are needed by the proxy is below the capacity of 
the server. In other words, if a proxy only ever connects to 100 other 
proxies, then at some point the proxy gets its 200 TLS connections set 
up (assuming no connect-reuse) and thats it. It never tears them down 
since it never runs out of resources, so TLS handshaking becomes a 
fixed, one-time cost. That makes it uninteresting to optimize.

The value, then, comes only when the set of proxies that get connected 
to is constantly changing, such that TLS handshaking between proxy peers 
is an ongoing and non-trivial percentage of the overall CPU effort.

To be frank, I am not aware of a single deployment situation today that 
meets this constraint. Can someone point me to a real deployment 
scenario where connect-reuse is applicable and has a non-trivial 
improvement in the performance of the proxy?

Thanks,
Jonathan R.


Nils Ohlmeier wrote:

> On Friday 23 June 2006 23:08, Dean Willis wrote:
> 
>>We've been following the connection reuse draft, currently draft-ietf-
>>sip-connect-reuse-05.txt, for a long long time.
>>
>>We've slipped the charter delivery date several times.
>>
>>Are we ever going to finish this work? Do we need to finish this work?
>>
>>Is this something that we could just not do?
>>
>>
>>Otherwise said: "Does the SIP working group still have a consensus to
>>expend working group effort on connection reuse?"
>>
>>If you have an opinion, for or against, please let it be known on
>>this list.
> 
> 
> I think their are two options left:
> 
> - either push it out as "TLS-connection-reuse-between-proxies" (if nobody 
> comes up with any issue any more)
> - or drop it because the real world usage is too limited
> 
> I have little preference for the first alternative, but wouldn't mind to go 
> for the second option.
> 
> Regards
>   Nils
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
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Vijay,

A few comments on the draft.

Firstly, I think the draft mixes authentication and authorization. You 
spend a bunch of time talking about checks that a server makes when 
receiving a request from a client, comparing bits of the SIP message to 
the TLS certificate. To me, all of those checks are authorization, not 
authentication. TLS mutual auth has done what its name suggests - it has 
already authenticated the other side. Now you are trying to figure out 
whether they are authorized to send you a SIP request. I think these are 
quite separable.

For example, if my proxy has a policy about only connecting to specific 
other domains, then it can take the identities determined from TLS 
mutual auth and use those to decide whether to accept requests coming 
from that proxy. No need to every look at the via, record-route or whatever.

More importantly though, I'm not at all sure that the Via based check 
you are suggesting adds any value. The value in the Via header is almost 
never used for anything. Its certainly not used for sending requests in 
the reverse direction. Its only used if the server needs to open a 
connection to the client to send a response - an extremely rare case. 
Thus, it seems futile to verify the value of a field that is never used 
for anything.

Indeed, I'm not at all sure that this whole concept of verifying the 
client certificate against fields in the SIP request message makes any 
sense at all. Validaing the Via is almost completely pointless as I 
mention above. Record-route, which at first glance seems better, won't 
work either since the proxy might not have record-routed. From doesn't 
work due to retargeting and also in networks where there are multiple 
proxies in the terminating domain.

Thinking about this some more, I think connect-reue has got the 
validation model wrong. Its not about validating that the client is 
authorized to send this request. Rather, what you want to demonstrate is 
that connect-reuse has security that is equivalent to the 'normal' 
connection-in-each-direction model. The way you do that is as follows.
Consider proxies A and B. A opens a connection to B, and provides a 
certificate. B caches the certificate and associates it with that 
connection. B also adds the entries to the alias table as your draft 
suggests. However, it does not attempt to validate the incoming SIP 
request against the certificate at all.

Now, when B needs to send a request in the reverse direction, it looks 
up the r-uri/route in DNS, gets the IP/port, and checks the alias table 
as you have described. If it finds a match, it checks the cached 
certificate for that connection. The trick is - that certificate had 
better look exactly like what it would have looked like if B had opened 
a new TLS connection towards A to send that request. In particular, the 
r-uri/route domain had better be a match for the SAN in the cached 
certificate. If its a match, the request is sent over the connection. If 
not, a fresh connection is opened.

This validation model has the desired property; a request gets delivered 
in the reverse direction with connect-reuse only if it would have been 
delivered had connect-reuse not been used.

This has the benefit of making certificate validation in SIP always look 
like the existing web-model of client to server connections only. This 
simplifies the work that domain-certs has to do. It merely needs to give 
guidance on how the server certificate should look like, addressing the 
two main things EKR mentioned:

1. what field does the identity get coded into
2. what does that identity represent - the specific host or the domain

My two cents is, I think it should always just be the domain 
(example.com and not proxy2.example.com). Thus if you connect to 
proxy2.example.com, and the server provided a certificate for 
example.com, that would be OK. This handles the record-routing use case 
you mention in the draft and works for server farms and TLS offloaders.

I also think its highly unlikely that a cerificate authority would hand 
out certs that verify that a *host* is proxy2.example.com. How would a 
CA verify that this is true?


Hope this makes sense,
Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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inline.

Scott Lawrence wrote:

> On Mon, 2006-06-26 at 09:00 -0700, Cullen Jennings wrote:
> 
>>On Jun 16, 2006, at 2:59 PM, Scott Lawrence wrote:
>>
>>
>>>There are, I think, two issues remaining for discussion:
>>
>>I disagree - I don't think the draft of the discussion has addressed  
>>the issue I brought up.
> 
> 
> I'm going to guess here - do you mean the issue of how large the
> response should be?
> 
> There is text in the draft on _how_ to reduce the size of the response.
> 
> 3261 does not provide guidance clear guidance on response size limits.
> It _does_ say that all implementations MUST be able to handle messages
> up to the maximum UDP packet size, which can probably handle even 70 hop
> messages.  If you think that SIP needs better definition of _when_ to
> limit response sizes and _what_ size they should be limited to, then I
> think that's an interesting discussion, but it's a bigger question than
> this error response and a problem that already exists whether this draft
> is accepted or not.
> 

Well, here is the history.

The transport design of SIP was based on the model that the response 
used the same transport as the request. We had the somewhat naive view 
that the response wouldn't be much larger than the request. So, we would 
take the MTU, subtract off a few hundred bytes to account for any 
possible increase in the size of the response, and that should be OK.

So, the implied limit is the path MTU itself, or 1500 bytes if unknown. 
But this is not stated explicitly, its implicit from the design.

One idea on dealing with the size problem is, instead of including the 
sip-frag in a response, the resposne just indicates that the hop limit 
was exceeded. The server which generated the response then sends a 
request in a totally new transaction, towards the contact of the 
request. That request contains the sip-frag with all of the information, 
and could be sent over TCP if needed.

That idea doesn't work for requests which omit contact, and also means 
that intermediate proxies on the path of the failed request might not 
see the diagnostic information, since it would come over a different 
transaction.

Anyway, just a thought.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Fri Jul 07 01:55:50 2006
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Well, then I would say that what is missing is a mechanism that allows 
the domain to indicate the number of flows the UA should establish, 
independently of the number of provided URIs. At the moment both are 
tied together, one URI equals one flow. I think those should be decoupled.

/Miguel

Drage, Keith (Keith) wrote:
> Firstly I was not necessarily targetting your comments in particular, but lets try and address your issue in terms of some simple tests:
> 
> 1)	what is currently in the draft fulfils the current requirements
> 2)	what is currently in the draft does not appear to be technically wrong
> 3)	we apparently do not need to decide between two alternative technical solutions to a requirement
> 
> therefore you have a use case that is different from that of the editors, and ideally we need to reflect the difference between those two use cases either:
> 
> -	by clearly stating a need for flexibility to allow those two use cases in the requirements
> -	or detailing the use cases that need to be supported.
> 
> (assuming that these use cases are considered to be valid)
> 
> The first of these is probably correct in this case.
> 
> regards
> 
> Keith
> 
>> -----Original Message-----
>> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
>> Sent: 06 July 2006 12:47
>> To: Drage, Keith (Keith)
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] A thought on progressing outbound
>>
>>
>> I don't have any problem with the current list of requirements in 
>> Section 12. The problem is that text in the draft seems to go beyond 
>> these requirements and impose a certain mandatory redundancy 
>> scheme, to 
>> which I strongly disagree as a "mandatory" feature.
>>
>> /Miguel
>>
>> Drage, Keith (Keith) wrote:
>>> (As WG chair)
>>>
>>> Having looked at some of the comments on outbound being 
>> addressed to the list in the last few days, it occurs to me 
>> that at least of you have fundamental requirements that 
>> impact the solution and these requirements themselves need to 
>> be discussed.
>>> Hidden in the current draft you will find section 12 
>> labelled "requirements". This is a somewhat thin section at 
>> the moment (not necessarily a bad thing).
>>> Can you all please review this section, and identify 
>> whether these requirements encompass all the requirements 
>> that you wish to have met in outbound, and then try to 
>> motivate what you think is missing.
>>> And as always "send text".
>>>
>>> regards
>>>
>>> Keith
>>>
>>> Keith Drage
>>> Lucent Technologies
>>> drage@lucent.com
>>> tel: +44 1793 776249
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>> -- 
>> Miguel A. Garcia           tel:+358-50-4804586
>> sip:miguel.an.garcia@openlaboratory.net
>> Nokia Research Center      Helsinki, Finland
>>

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Subject: RE: [Sip] Outbound Comment: binding key
Date: Fri, 7 Jul 2006 08:56:27 +0300
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From: <Erkki.Koivusalo@nokia.com>
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Hi,

In March before Dallas meeting Jeroen raised the same question:

> Some comments on this draft:
>=20
> Section 3.2 (and elsewhere): in addition to stating that a
proxy/registrar
> should prefer the most recently registered flow, I would suggest that
the
> registrar explicitly invalidates an existing registration when it
receives a
> new registration with the same instance-id / reg-id. This situation
would
> most likely be due to a reboot of the client. For TCP and other
> connection-oriented transports it would be detected, but a
registration over
> UDP could survive a client crash

And this is what Cullen answered:

> This has been a topic of large debate and we (with many but not most
people
> saying exactly what you are saying). We asked about it in WG meeting
and on
> list and came to agreement that the text that does not allow "explicit
> invalidation" was better. I'm scared to reopen this.=20

I do not have a clue what kind of arguments were there in that old
debate,
but I admit I have been a couple of times wondering this same issue. It
would be nice if someone would bring up the thread Cullen is referrng
to.

If changing the draft like you and Jeroen suggested I believe the draft
SHOULD be more explicit about what happens to the old flows ? Will they
stay in the registrar database forever ?

After one year of operation, does the registrar have e.g. 27 mappings
stored for one UA which has rebooted 27 times, even if the most recent
of them is used ?

Regards,

Erkki

>-----Original Message-----
>From: ext Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
>Sent: 06.July.2006 22:08
>To: 'Adam Roach'; 'Cullen Jennings'
>Cc: 'SIP WG'
>Subject: RE: [Sip] Outbound Comment: binding key
>
>My interpretation was the opposite - if a Register comes in=20
>with the same
>{instance-id,reg-id} then it should indeed replace/overwrite=20
>the previous
>binding of the same tuple.  I know there are several sections=20
>in the draft
>which imply otherwise, but I don't understand why.  There is=20
>no real value
>I can see to keeping a binding for an old flow of the same=20
>reg-id, since by
>definition the UA would only "re-use" a reg-id if the old flow=20
>failed (or
>even if it didn't think it had failed, but something rebooted=20
>and changed
>the flow). =20
>
>I can see no realistic way in which an old flow of the same
>{instance-id,reg-id} would succeed in reaching the UA.  Am I missing
>something?
>
>-hadriel
>p.s. it's theoretically possible, if a sip-udp UA rebooted and=20
>resolved a
>new proxy address and sent a register to it with the same=20
>reg-id, that it
>may still be possible to reach the UA through the old "flow"=20
>for a time. =20
>But is that a realistic case worth caring about?
>
>
>
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Thursday, July 06, 2006 1:12 AM
>> To: Cullen Jennings
>> Cc: SIP WG
>> Subject: [Sip] Outbound Comment: binding key
>>=20
>>  From section 6.1:
>>=20
>>    When a reg-id header parameter is present in a Contact=20
>header field
>>    value in a REGISTER request, the corresponding binding is=20
>between an
>>    AOR and the combination of instance-id and reg-id.
>>=20
>> I don't think this is quite correct. This statement appears to be
>> asserting that {instance-id,reg-id} is a unique key to a binding
>> associated with an AOR. This would, in turn, imply that any=20
>successful
>> registration attempt with an {instance-id,reg-id} that matches an
>> existing binding would overwrite the existing record.
>>=20
>> I don't think that is what's intended, since section 6.2 contains the
>> following text:
>>=20
>>    o  If two bindings have the same instance-id and reg-id, the proxy
>>       SHOULD prefer the contact that was most recently updated.
>>=20
>> If the binding is only to {instance-id,reg-id}, then that text is
>> nonsensical: you can't have two identical bindings to the same AOR.
>>=20
>> I suspect that the text in 6.1 should actually read=20
>something more like:
>>=20
>>    When a reg-id header parameter is present in a Contact=20
>header field
>>    value in a REGISTER request, the corresponding binding is=20
>between an
>>    AOR and the combination of instance-id, reg-id, and network flow.
>>=20
>> /a
>>=20
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

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From sip-bounces@ietf.org Fri Jul 07 02:01:30 2006
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Inline

Jonathan Rosenberg wrote:
> inline.
> 
> Miguel Garcia wrote:
> 
>> Cullen Jennings wrote:
>>
>>>
>>>
>>> I expect you are right in that this MUST is too strong here but I 
>>> think there is an argument for SHOULD. The problem is that if the 
>>> domain assume the UA is doing two flows and the UA does one, things 
>>> are broken.
>>>
>>
>> I would like to understand which part of the outbound mechanism is 
>> broken if the domain assumes the UE will do two flows, but the UA does 
>> one flow. Yes, it isn't reliable, but the mechanism works nicely... or 
>> not?
>>
>> Then... why can't a domain indicate just one outbound URI? At the 
>> moment the draft has an artificial lower limit for all the domains to 
>> provide two URIs and impose a reliability that might not be 
>> required/needed.
> 
> Thats not my read.
> 
> The draft says that a client has to support at least 2 outbound URI, and 
> that it MUST send a REGISTER for each. However, that doesn't mean a 
> domain *has* to provide two; it can also provide one. A domain would 
> only provide one if it knows that it doesn't need two for purposes of 
> reliability, or if it doesn't require reliability of service.

True, but then the UA does not have an alternative outbound server, in 
case the supplied one fails. What I don't like is the one to one 
relation between the number of supplied outbound URIs and the number of 
flows (and REGISTER).

Additionally I suspect that users may still impose local policies in the 
UAs as for the number of established flows, independently of the number 
of learned URIs. So, the MUST in the spec is always subject to local policy.

And a third thing is that, according to RFC 2119:

1. MUST   This word, or the terms "REQUIRED" or "SHALL", mean that the
    definition is an absolute requirement of the specification.

I don't see the need for a MUST in that paragraph, because the spec is 
not broken even if the UA does not obey the MUST.

/Miguel


> As such, I support the current text which says that a client MUST send a 
> REGISTER to each outbound URI.
> 
> Thanks,
> Jonathan R.
> 
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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major issues:

Well, no change here in these. I think we need more discussion on them:

1. I still think this mechanism is applicable outside of max-forwards 
violations. The current text limits it to 483.

2. I agree with Cullen that we do need to specify a size limit for the 
response. Having thought about it more, I'm worried that the sipfrag 
body can't be more than 200 bytes! Consider a request that loops a lot. 
At some point, the via header itself will grow sufficiently to cause the 
mesage to switch from udp to tcp. That happens at the point at which the 
udp request exceeds 1300 bytes. For a response to that request to be 
also sent over UDP with no fragmentation, the response cannot be larger 
than the request by more than 200 bytes.

Now, you might argue that proxies often strip Via's, and that a small 
max-forwards might never get the request to pop over to tcp. However if 
you are looking for a safe value, its 200  bytes.

Thsi might motivate us to look at something different to return the data 
to the client, rather than in that response. Another option is content 
indirection.





minor issues:

> The SIP protocol imposes a limit on the number of hops a request can
>    transit on the way to its destination.  The number of hops remaining
>    for the request is carried in the Max-Forwards header, and is

header field

>    request has traversed a series of proxies, the response follows the
>    Vias back to the requester - in the case of a typical 483 response it
>    can be difficult to determine even what server the response came

same comment as on -02. Difficult implies that its possible, just hard. 
Its NOT possible.

>     A Warning header with a warn-code of 399 that identifies the
>          system returning the error.

header field


>  If including all Via and Route headers is still too large, the
>    implementation SHOULD remove the oldest Vias (those nearest the
>    message originator) until the size is acceptable; the only exception
>    to this rule is when . 

when....?



Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Fri Jul 07 02:50:48 2006
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Date: Fri, 07 Jul 2006 08:48:49 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] Outbound Comment: binding key
To: Erkki.Koivusalo@nokia.com, HKaplan@acmepacket.com, adam@nostrum.com,
	fluffy@cisco.com
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Inline

Regards,

jeroen

-----Oorspronkelijk bericht -----
Van: Erkki.Koivusalo@nokia.com
Aan: HKaplan@acmepacket.com; adam@nostrum.com; fluffy@cisco.com
CC: sip@ietf.org
Verzonden: 7-7-06 07:56
Onderwerp: RE: [Sip] Outbound Comment: binding key


Hi,

In March before Dallas meeting Jeroen raised the same question:

> Some comments on this draft:
> 
> Section 3.2 (and elsewhere): in addition to stating that a
proxy/registrar
> should prefer the most recently registered flow, I would suggest that
the
> registrar explicitly invalidates an existing registration when it
receives a
> new registration with the same instance-id / reg-id. This situation
would
> most likely be due to a reboot of the client. For TCP and other
> connection-oriented transports it would be detected, but a
registration over
> UDP could survive a client crash

And this is what Cullen answered:

> This has been a topic of large debate and we (with many but not most
people
> saying exactly what you are saying). We asked about it in WG meeting
and on
> list and came to agreement that the text that does not allow "explicit
> invalidation" was better. I'm scared to reopen this. 

I do not have a clue what kind of arguments were there in that old
debate,
but I admit I have been a couple of times wondering this same issue. It
would be nice if someone would bring up the thread Cullen is referrng
to.

If changing the draft like you and Jeroen suggested I believe the draft
SHOULD be more explicit about what happens to the old flows ? Will they
stay in the registrar database forever ?

After one year of operation, does the registrar have e.g. 27 mappings
stored for one UA which has rebooted 27 times, even if the most recent
of them is used ?
[jvb] well,it would not get quite so bad;lifetimes are still bound by the expiry interval. But I don't see any benefit in keeping the mapping around.What is wrong with explicit invalidation?

Regards,

Erkki

>-----Original Message-----
>From: ext Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
>Sent: 06.July.2006 22:08
>To: 'Adam Roach'; 'Cullen Jennings'
>Cc: 'SIP WG'
>Subject: RE: [Sip] Outbound Comment: binding key
>
>My interpretation was the opposite - if a Register comes in 
>with the same
>{instance-id,reg-id} then it should indeed replace/overwrite 
>the previous
>binding of the same tuple.  I know there are several sections 
>in the draft
>which imply otherwise, but I don't understand why.  There is 
>no real value
>I can see to keeping a binding for an old flow of the same 
>reg-id, since by
>definition the UA would only "re-use" a reg-id if the old flow 
>failed (or
>even if it didn't think it had failed, but something rebooted 
>and changed
>the flow).  
>
>I can see no realistic way in which an old flow of the same
>{instance-id,reg-id} would succeed in reaching the UA.  Am I missing
>something?
>
>-hadriel
>p.s. it's theoretically possible, if a sip-udp UA rebooted and 
>resolved a
>new proxy address and sent a register to it with the same 
>reg-id, that it
>may still be possible to reach the UA through the old "flow" 
>for a time.  
>But is that a realistic case worth caring about?
>
>
>
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Thursday, July 06, 2006 1:12 AM
>> To: Cullen Jennings
>> Cc: SIP WG
>> Subject: [Sip] Outbound Comment: binding key
>> 
>>  From section 6.1:
>> 
>>    When a reg-id header parameter is present in a Contact 
>header field
>>    value in a REGISTER request, the corresponding binding is 
>between an
>>    AOR and the combination of instance-id and reg-id.
>> 
>> I don't think this is quite correct. This statement appears to be
>> asserting that {instance-id,reg-id} is a unique key to a binding
>> associated with an AOR. This would, in turn, imply that any 
>successful
>> registration attempt with an {instance-id,reg-id} that matches an
>> existing binding would overwrite the existing record.
>> 
>> I don't think that is what's intended, since section 6.2 contains the
>> following text:
>> 
>>    o  If two bindings have the same instance-id and reg-id, the proxy
>>       SHOULD prefer the contact that was most recently updated.
>> 
>> If the binding is only to {instance-id,reg-id}, then that text is
>> nonsensical: you can't have two identical bindings to the same AOR.
>> 
>> I suspect that the text in 6.1 should actually read 
>something more like:
>> 
>>    When a reg-id header parameter is present in a Contact 
>header field
>>    value in a REGISTER request, the corresponding binding is 
>between an
>>    AOR and the combination of instance-id, reg-id, and network flow.
>> 
>> /a
>> 
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
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>

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From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
To: Miguel Garcia <Miguel.An.Garcia@nokia.com>,
	Jonathan Rosenberg <jdrosen@cisco.com>
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Inline

> Thats not my read.
> 
> The draft says that a client has to support at least 2 outbound URI, and 
> that it MUST send a REGISTER for each. However, that doesn't mean a 
> domain *has* to provide two; it can also provide one. A domain would 
> only provide one if it knows that it doesn't need two for purposes of 
> reliability, or if it doesn't require reliability of service.

True, but then the UA does not have an alternative outbound server, in 
case the supplied one fails. 

[jvb] you could still have multiple alternatives from DNS (ie traditional DNS that is)

What I don't like is the one to one 
relation between the number of supplied outbound URIs and the number of 
flows (and REGISTER).

[jvb] indeed, especially since the domain may have no control over how many URIs the client discovers. Manual entry is one example. And what happens  if one URI comes via manual provisioning and 2 more via DNS as per outbound-discovery?

Additionally I suspect that users may still impose local policies in the 
UAs as for the number of established flows, independently of the number 
of learned URIs. So, the MUST in the spec is always subject to local policy

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Subject: RE: [Sip] comments on sip-outbound-04
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Hi,

>> When a UA is about to send a request, it first=20
>> performs normal processing to select the next hop URI.

>> The UA performs normal DNS resolution on the next=20
>> hop URI (as described in RFC 3263 [RFC3263]) to=20
>> find a protocol, IP address, and port. For non-TLS=20
>> protocols, if the UA has an existing flow to this
>> IP address, and port with the correct protocol, then=20
>> the UA MUST use the existing connection. =20
>
>I'm pretty sure that this text would fundamentally require=20
>UA changes for any mid-dialog failover solution to work.=20

Mid-dialog failover is not the only case for which the
referred piece of text has a problem. It also contains
a clear bug at least when used literally for registration
refreshes.

For outbound to work correctly, UA must send the registration
refresh over the same flow that was created when sending
the initial registration to be refreshed. Currently the
draft fails to mention this as I have earlier pointed out.

RFC 3263 procedures DO NOT guarantee that when UA resolves
once again the URI of the next hop (of the outbound proxy),
it would result the same IP address which was received
when resolving the same URI previously. If it is different
IP address then another flow will be used, perhaps towards
a totally different proxy as used earlier.

UA should cache the DNS RRs it received when it resolved the
URI the first time, so the problem is not whether the DNS
server returns the same values or not. The failure can=20
happen if the domain has a number of SRV records with
the same transport and priority values, but different
weights (for load balancing reasons).

a) RFC 3263 states that to resolve a SIP URI without a
   transport or port parameter, the UA will have to use
   NAPTR, SRV and address queries to get the IP address.

b) RFC 2782 defines the weight for SRV RR as follows:

        A server selection mechanism.  The weight field specifies a
        relative weight for entries with the same priority. Larger
        weights SHOULD be given a proportionately higher probability of
        being selected. The range of this number is 0-65535.  This is a
        16 bit unsigned integer in network byte order.  Domain
        administrators SHOULD use Weight 0 when there isn't any server
        selection to do, to make the RR easier to read for humans (less
        noisy).  In the presence of records containing weights greater
        than 0, records with weight 0 should have a very small chance of
        being selected.

        In the absence of a protocol whose specification calls for the
        use of other weighting information, a client arranges the SRV
        RRs of the same Priority in the order in which target hosts,
        specified by the SRV RRs, will be contacted. The following
        algorithm SHOULD be used to order the SRV RRs of the same
        priority:

        To select a target to be contacted next, arrange all SRV RRs
        (that have not been ordered yet) in any order, except that all
        those with weight 0 are placed at the beginning of the list.

        Compute the sum of the weights of those RRs, and with each RR
        associate the running sum in the selected order. Then choose a
        uniform random number between 0 and the sum computed
        (inclusive), and select the RR whose running sum value is the
        first in the selected order which is greater than or equal to
        the random number selected. The target host specified in the
        selected SRV RR is the next one to be contacted by the client.
        Remove this SRV RR from the set of the unordered SRV RRs and
        apply the described algorithm to the unordered SRV RRs to select
        the next target host.  Continue the ordering process until there
        are no unordered SRV RRs.  This process is repeated for each
        Priority.

When processing the SRV records as per RFC 3263, the UA may select
different record each time, due to the randomness involved for
selecting the order of SRV records to try. Even if the UA would
cache the DNS records itself, nothing requires it to cache the
ultimate result of RFC 3263 processing, so the result may be
different IP address each time.=20

For legacy SIP this has not been a problem. The registrar (not to
mention the proxy) could not care less if the registration refresh
was routed over different edge proxy each time. However for outbound
this would not cause the desired behaviour. I can identify at least
two issues:

1. It is not clear how the UA should handle the flow related=20
   to a registration if it swaps from one proxy to another.

2. If the UA has one primary and one secondary flow, it might
   happen that when refreshing the primary flow, the UA would
   accidentally move it to the same proxy towards which the
   secondary flow is going to. So redundancy would be lost.

P.S. I tried to raise this issue already twice but because I
did not get any clear response, I now do it once again as I will
be out of office for the next few weeks. I hope that this time
I was able to formulate the problem statement clearly enough
for you to get the point.

Regards,

Erkki

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Subject: RE: [Sip] Ready for WGLC on Outbound?
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Hi,

What Hadriel says here is in my opinion a clear reason why the
outbound draft (instead of any other optional or additional draft)
must have the safe check for the UA to ensure whether or not the=20
contacted proxy supports outbound:
=20
>And it's not the same as just a new option-tag or header - this is a
>backwards-compatibility breaker.  There is no defined action for a
>3261-compliant device to take when receiving such a binary message, is
>there?  It cannot "reject" it with a sip response, cannot make=20
>the UA stop
>sending it, nor can the UA detect the proxy doesn't support it=20
>even if the
>UA wanted to check.  The UA will detect a keepalive loss, and try
>"re-connecting" by re-registering and getting another keepalive loss,
>indefinitely (although at least with back-off timers).
>
>Meanwhile the operator of this 3261-compliant proxy just got a headache
>courtesy of the IETF.  For each user who mistakenly checks some little
>check-box which says whatever the vendor thought it should say for a
>keepalive=3Dstun semantics the operator must take action to correct - =
if
>they're lucky.  If they're unlucky, and their proxy silently=20
>discards the
>stun, they'll wonder why some UAs can make/receive calls for=20
>~30 seconds,
>then disappear, then come back, ad infinitum.  And this is a=20
>bad kind of
>failure to detect/fix, because a reboot makes it "work" for a=20
>short time.

Another reason is embedded into this statement from Jonathan:

> (3) constructed URIs.=20
>
> On (3), this is definitely a possibility. However I think the best way

> to mitigate is to say 'don't do that'. SIP-outbound can have some
words=20
> which warn implementations that they MUST NOT insert this parameter
into=20
> URI that they construct from a domain name.

This suggestion seems to prevent the UA to use DHCP to discover
and use a local SIP proxy even if such an outbound compliant
proxy would exist. This does not sound a good idea to me.

Regards,

Erkki

>> -----Original Message-----
>> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
>> Sent: Thursday, July 06, 2006 2:07 PM
>> To: Jeroen van Bemmel
>> Cc: Cullen Jennings; sip@ietf.org; dean.willis@softarmor.com;
>> rohan@ekabal.com
>> Subject: Re: [Sip] Ready for WGLC on Outbound?
>>=20
>> inline.
>>=20
>> Jeroen van Bemmel wrote:
>>=20
>> > Cullen,
>> >
>> > The point is that these are really 2 separate issues:
>> > 1. Discovering the set of outbound proxy URIs (with =
keepalive=3Dstun
>> > parameters in them)
>> > 2. Confirming that a given proxy in the set actually=20
>supports outbound
>> > (ie validating the keepalive=3Dstun information)
>> >
>> > Of these 2 options, (1) is indeed a separable problem. (2)=20
>- to me - is
>> > not, for me outbound is not acceptable without something=20
>to address (2).
>> > It really should say "the UAC MUST NOT use STUN until {=20
>mechanism for
>> > (2) says its supported }"
>>=20
>> I personally think that the presence of the keepalive=3Dstun=20
>attribute in
>> the URI is sufficient of a "confirmation". The examples you=20
>and others
>> have presented for why this is needed are (1)=20
>misconfiguration (someone
>> gives a URI to the client which is incorrect in that it contains this
>> flag when it shouldn't), (2) malicious users, (3)=20
>constructed URIs. On
>> (2), let me point out that an attacker that can modify DNS=20
>or modify SIP
>> messages (or whatever is used for the discovery function)=20
>can launch far
>> more effective attacks than just adding keepalive=3Dstun in=20
>the hopes that
>> it causes a low-quality proxy to crash.
>>=20
>> On (1), misconfugration could potentially produce all kinds of things
>> that would cause a client to send things to the server that=20
>might make
>> it crash, including non-SIP URI, URI with invalid header parameters,
>> etc. I don't see keepalive=3Dstun as special.
>>=20
>> On (3), this is definitely a possibility. However I think=20
>the best way
>> to mitigate is to say 'don't do that'. SIP-outbound can have=20
>some words
>> which warn implementations that they MUST NOT insert this=20
>parameter into
>> URI that they construct from a domain name.
>>=20
>> Finally, let me say that if you have a proxy that crashes on=20
>receipt of
>> binary messages on its sip port, you are in for real=20
>trouble. I think a
>> far better solution is for proxy vendors to fix their proxies, rather
>> than for us to bend over backwards to design new mechanisms that help
>> protect such fragile servers from ever seeing these=20
>messages. No matter
>> what we put in our specs, attackers can still send binary=20
>messages that
>> will result in the same result you are trying to prevent.
>>=20
>> Thanks,
>> Jonathan R.
>>=20
>>=20
>>=20
>>=20
>> --
>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>> Cisco Fellow                                   Parsippany,=20
>NJ 07054-2711
>> Cisco Systems
>> jdrosen@cisco.com                              FAX:   (973) 952-5050
>> http://www.jdrosen.net                         PHONE: (973) 952-5000
>> http://www.cisco.com
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
>_______________________________________________
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>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

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From: Dean Willis <dean.willis@softarmor.com>
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To: Juha Heinanen <jh@tutpro.com>
Subject: Re: [Sip] Ready for WGLC on Outbound?
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Juha Heinanen wrote:
> Jeroen van Bemmel writes:
> 
>  > 2. Confirming that a given proxy in the set actually supports outbound (ie 
>  > validating the keepalive=stun information)
> 
> i don't understand this one.  if a proxy is listed as an outbound proxy
> (for example in the result returned from DNS), why can't we require that it
> supports stun keepalives?

What do you mean "require"? Use a proxy-require option tag?

Or just hope it won't melt when you send it a STUN packet, meaning that 
you "require the operator to get it right".

The problem as I see it is we have no way of easily detecting an error 
if somebody mistakenly lists a non-compliant proxy as an outbound proxy 
in DHCP or some other discovery mechanism. I hate leaping off a trapeze 
without a safety net . . .

--
Dean

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Subject: Re: [Sip] Ready for WGLC on Outbound?
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Jeroen van Bemmel wrote:
> Cullen,
> 
> The point is that these are really 2 separate issues:
> 1. Discovering the set of outbound proxy URIs (with keepalive=stun 
> parameters in them)
> 2. Confirming that a given proxy in the set actually supports outbound 
> (ie validating the keepalive=stun information)
> 
> Of these 2 options, (1) is indeed a separable problem. (2) - to me - is 
> not, for me outbound is not acceptable without something to address (2). 
> It really should say "the UAC MUST NOT use STUN until { mechanism for 
> (2) says its supported }"
> 
> There have been several options mentioned to do this, a simple way is to 
> have the edge proxy insert a flag. Question is: why not?

I find that I've developed a great deal of sympathy for the above 
position. Discovery is indeed separable from error-detection (or 
confirmation, detecting "absence of error").

--
Dean

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Hi,

> True, but then the UA does not have an alternative outbound server, in 
> case the supplied one fails. What I don't like is the one to one 
> relation between the number of supplied outbound URIs and the number of 
> flows (and REGISTER).

using a SHOULD instead of a MUST would break this coupling, which would 
give more freedom to the UA in order to decide how many flows it wants 
to establish (maybe the terminal does not want high reliability).

If things do not break, a SHOULD is more appropriate than a MUST.

Cheers,

Gonzalo

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Subject: RE: [Sip] an alternative approach for outbound proxy discovery
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Hi Jonathan,

>The idea is to not use DNS, but rather use SIP mechanisms for the=20
>discovery. I much prefer a non-DNS solution for several reasons which=20
>I'll enumerate below. The basic mechanism combines two things:
>
>1. usage of the 305/service-route as I have proposed in=20
>http://www.ietf.org/internet-drafts/draft-rosenberg-sip-route-c
>onstruct-01.txt
>
>2. usage of a new header field in REGISTER responses which specifies=20
>additional URI to try REGISTERing to

Sounds a nice idea as it has all those good properties
that you mentioned, but it is not without some drawbacks
and open issues.

One difficulty is that such a solution would require the service=20
provider to upgrade also the home proxies when starting to=20
deploy outbound, instead of just upgrading the edge proxies=20
and registrar. There would be no hope of autodiscovery without
such a network upgrade. For the DNS based solution what would
be needed is just a reconfiguration of the DNS server together
with the UAs which implement the procedures. No software upgrade
would be needed for any server in the network.

For the technical point of view I have the following questions:

Most importantly, how can the UA make sure what is the situation
if it does not receive 305 response or 200 OK with Additional-
Proxies for the REGISTER ?

- Was it so that the contacted proxy supports outbound and
  the network does not require the UA to contact any additional
  proxy, so that everything is fine when receiving 200 OK that=20
  does not contain Additional-Proxies header ? In this case
  the UA can perfectly well start sending STUN keepalives
  towards the proxy.

- Or was it so that the proxy supports neither outbound
  nor this new optional draft and therefore did not return 305=20
  or Additional-Proxies header ? In this case UA could
  make a false assumption and continue using outbound even if
  that would not work.

So I assume that 200 OK header would always have to contain=20
some indication whether or not the proxy supports outbound,
right ? This is already being discussed in another thread
but your idea would add one another reason for mandating
such an indication within outbound draft.

>So, here is the idea. UA starts with its AOR (sip:user@example.com)=20
>for example, or with a domain name learned from DHCP. In either case,=20
>it applies RFC3263 and gets the URI for a proxy. This is going to=20
>be the home proxy.

In my understanding the main motivation for using DHCP for the
SIP proxy discovery is to find a local proxy instead of the home
proxy. Home proxy could always be found by resolving the AOR.
I am not totally sure if your idea was relying on an assumption=20
that the UA would always try contacting the home proxy at first.
Anyway that is not always the case.

Mobile clients like laptop PCs and mobile phones could be attached
to a network anywhere in the world, even in different continent
than the one where the home proxy or edge proxies for a domain
are located. In those cases the UA would probably prefer using
a local proxy, if available, rather than the remote edge proxy
of user's home domain. DHCP is used to find such a local proxy.

The discovered local proxy may or may not support outbound:

- In the case it supports the proxy has to indicate the=20
  support to the UA in 200 OK as told above.

- In case it does not support the proxy should redirect.
  However is it certain for such a proxy that the UA needs
  outbound support ? The UA does not send Proxy-Require
  header or anything like that to tell the local proxy about
  what it needs. Can such a local proxy rely on this
  kind of hints what you outlined below ?

>UA sends a REGISTER request. The proxy receives this, and=20
>knows that it doesn't accept REGISTER directly like this;=20
>it needs to be sent via an edge proxy. There are many ways=20
>the proxy can know it got the REGISTER directly and not=20
>through an EP. For example, requests from EP come to a=20
>different port, or better yet, via TLS mutual auth from=20
>edge proxy.

Regards,

Erkki=20

>-----Original Message-----
>From: ext Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
>Sent: 06.July.2006 23:11
>To: IETF SIP List
>Subject: [Sip] an alternative approach for outbound proxy discovery
>
>I want to suggest a possible alternative approach for outbound proxy=20
>discovery than the one in draft-koivusalo-sip-outbound-discovery. So,=20
>before I get to it, two caveats:
>
>1. I am NOT proposing that this be added to sip-outbound, and I'm not=20
>suggesting that a normative reference to this or another mechanism=20
>should be in outbound. I believe that any discovery mecanism should be=20
>orthogonal, and what I am about to propose below is orthogonal as well.
>
>2. I have not been able to follow all of the nearly countless messages=20
>on sip-outbound, so I'm not sure if what I'm about to propose has been=20
>proposed. So, please point me to relevant threads if it has been
>
>
>The idea is to not use DNS, but rather use SIP mechanisms for the=20
>discovery. I much prefer a non-DNS solution for several reasons which=20
>I'll enumerate below. The basic mechanism combines two things:
>
>1. usage of the 305/service-route as I have proposed in=20
>http://www.ietf.org/internet-drafts/draft-rosenberg-sip-route-c
>onstruct-01.txt
>
>2. usage of a new header field in REGISTER responses which specifies=20
>additional URI to try REGISTERing to
>
>So, here is the idea. UA starts with its AOR=20
>(sip:user@example.com) for=20
>example, or with a domain name learned from DHCP. In either case, it=20
>applies RFC3263 and gets the URI for a proxy. This is going to be the=20
>home proxy, and there is no keepalive=3Dstun and thus no STUN =
keepalives=20
>here yet. UA sends a REGISTER request. The proxy receives this, and=20
>knows that it doesn't accept REGISTER directly like this; it=20
>needs to be=20
>sent via an edge proxy. There are many ways the proxy can know it got=20
>the REGISTER directly and not through an EP. For example,=20
>requests from=20
>EP come to a different port, or better yet, via TLS mutual auth from=20
>edge proxy.
>
>So, home proxy redirects the request with a 305. The URI in=20
>the contact=20
>header field is an edge proxy, and has the keepalive=3Dstun parameter =
in=20
>it. Now, this URI can be a specific IP address, or more=20
>interestingly, a=20
>hostname that resolves to any number of proxies. This=20
>basically becomes=20
>the first proxy in the outbound proxy set. If its a hostname, client=20
>performs regular old RFC3263 and picks one. Client sends a request=20
>there, now with the topmost Route set equal to this URI (this=20
>is what I=20
>have proposed for 305 handling in route-construct).
>
>This arrives at the edge proxy, which inserts itself into=20
>Path, which is=20
>subsequently reflected into Service-Route, which the proxy can=20
>modify on=20
>its way towards the client (again this is what I have proposed in=20
>route-construct). This Service-Route points to that edge proxy.
>
>Now, the UA has a single proxy (so far) in that outbound proxy set.=20
>However, that 200 OK to the REGISTER, in addition to having a=20
>Service-Route, would also have another header field, called=20
>Additional-Proxies or something like that. This header field would=20
>contain one or more URI pointing to proxies that the client should=20
>register additional flows through. For each of those, the client=20
>performs a registration (each of those URI would be the topmost route=20
>header field in a REGISTER message, each for a different reg-id). Now,=20
>it a multiplicity of flows and outbound proxies all set up.
>
>The primary benefit of this mechanism over a DNS one is its dynamic=20
>nature. The edge proxy which receives the original REGISTER=20
>request and=20
>response is the one that inserts the Additional-Proxies header field.=20
>Thus, it can purposefully choose a specific proxy acting as=20
>its backup,=20
>or choose a proxy in a different geographic region, or whatever. These=20
>choices can differ from user to user. The original REGISTER=20
>requests can=20
>be load balanced across all of the proxies in a cluster as a=20
>consequence=20
>of DNS, but the second flow is selected explicitly by the first proxy.
>
>This also lends itself extremely well to mid-dialog failover=20
>solutions,=20
>since the backup proxy is selected by the primary. Thus, it can choose=20
>one to which it is replicating state. It also allows for a variety of=20
>algorithms in selecting a record-route value.
>
>With this mechanism, it is clear how the provider chooses the=20
>number of=20
>flows that a UA makes - its an additional one for each value of the=20
>Additional-Proxies header field.
>
>It also allows us to keep the existing meaning of DNS queries,=20
>which are=20
>meant to choose a single server and not learn a set of servers=20
>that all=20
>must be contacted.
>
>There is a cost of a single extra round trip for the very=20
>first REGISTER=20
>to be redirected. However, I do not view this as particularly onerous.
>
>Thanks,
>Jonathan R.
>
>
>
>--=20
>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>Cisco Fellow                                   Parsippany, NJ=20
>07054-2711
>Cisco Systems
>jdrosen@cisco.com                              FAX:   (973) 952-5050
>http://www.jdrosen.net                         PHONE: (973) 952-5000
>http://www.cisco.com
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

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Subject: RE: [Sip] FW: I-D ACTION:draft-audet-sip-sips-guidelines-02.txt
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Comment on content of Section 3:

      TLS MAY be used between Proxy B and UA b@B, depending on local
      policy.

   Because B is responsible for it's own security policy inside it's own
   domain, it could use some other security mechanism.  For example, it
   could be using IPsec, or some lower-layer security mechanism.

<my_comment>
It is not just B that needs security, it might be A who wants the
security too.
A might decide not to call B if there were no SIPS.
Thus it might not be ok to leave the decision to B to decide if he wants
to use TLS.
</my_comment>


--
=20
Ravishankar. G. Shiroor
Wipro Technologies, Bangalore.
=20
ravishankar.shiroor@wipro.com
--
=20

> -----Original Message-----
> From: Francois Audet [mailto:audet@nortel.com]=20
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> To: IETF SIP List
> Subject: [Sip] FW: I-D ACTION:draft-audet-sip-sips-guidelines-02.txt
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> >=20
> >=20
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> >=20
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From sip-bounces@ietf.org Fri Jul 07 09:22:06 2006
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We'd also get pretty much the same thing by using the "by reference" mechanism in draft-ietf-sip-location-conveyance.

The server sends a URI to some location where the diagnostic information is placed. All the issues that exist have to be handled for location-conveyance with that mechanism.

Out of interest, for Jonathan's proposal, on networks where we're running on the verge of hop limit expiries for all requests, don't we tend to some form of amplification situation. The server sends the 4xx response plus a new request with the diagnostic information. A certain percentage (because they took slightly longer paths) of the new requests hit the hop limit in the reverse direction and send 4xx back to the original server plus a request containing the diagnostic inforation, etc, etc. Of course, on properly configured networks...

regards

Keith


> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: 07 July 2006 06:45
> To: Scott Lawrence
> Cc: Cullen Jennings; SIP WG List
> Subject: Re: [Sip] draft-ietf-sip-hop-limit-diagnostics-03
> 
> 
> inline.
> 
> Scott Lawrence wrote:
> 
> > On Mon, 2006-06-26 at 09:00 -0700, Cullen Jennings wrote:
> > 
> >>On Jun 16, 2006, at 2:59 PM, Scott Lawrence wrote:
> >>
> >>
> >>>There are, I think, two issues remaining for discussion:
> >>
> >>I disagree - I don't think the draft of the discussion has 
> addressed  
> >>the issue I brought up.
> > 
> > 
> > I'm going to guess here - do you mean the issue of how large the
> > response should be?
> > 
> > There is text in the draft on _how_ to reduce the size of 
> the response.
> > 
> > 3261 does not provide guidance clear guidance on response 
> size limits.
> > It _does_ say that all implementations MUST be able to 
> handle messages
> > up to the maximum UDP packet size, which can probably 
> handle even 70 hop
> > messages.  If you think that SIP needs better definition of 
> _when_ to
> > limit response sizes and _what_ size they should be limited 
> to, then I
> > think that's an interesting discussion, but it's a bigger 
> question than
> > this error response and a problem that already exists 
> whether this draft
> > is accepted or not.
> > 
> 
> Well, here is the history.
> 
> The transport design of SIP was based on the model that the response 
> used the same transport as the request. We had the somewhat 
> naive view 
> that the response wouldn't be much larger than the request. 
> So, we would 
> take the MTU, subtract off a few hundred bytes to account for any 
> possible increase in the size of the response, and that should be OK.
> 
> So, the implied limit is the path MTU itself, or 1500 bytes 
> if unknown. 
> But this is not stated explicitly, its implicit from the design.
> 
> One idea on dealing with the size problem is, instead of 
> including the 
> sip-frag in a response, the resposne just indicates that the 
> hop limit 
> was exceeded. The server which generated the response then sends a 
> request in a totally new transaction, towards the contact of the 
> request. That request contains the sip-frag with all of the 
> information, 
> and could be sent over TCP if needed.
> 
> That idea doesn't work for requests which omit contact, and 
> also means 
> that intermediate proxies on the path of the failed request might not 
> see the diagnostic information, since it would come over a different 
> transaction.
> 
> Anyway, just a thought.
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> _______________________________________________
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> 

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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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Miguel,


On Jul 6, 2006, at 22:58, Miguel Garcia wrote:
>>> Then... why can't a domain indicate just one outbound URI? At the 
>>> moment the draft has an artificial lower limit for all the domains 
>>> to provide two URIs and impose a reliability that might not be 
>>> required/needed.
>> Thats not my read.
>> The draft says that a client has to support at least 2 outbound URI, 
>> and that it MUST send a REGISTER for each. However, that doesn't mean 
>> a domain *has* to provide two; it can also provide one. A domain 
>> would only provide one if it knows that it doesn't need two for 
>> purposes of reliability, or if it doesn't require reliability of 
>> service.
>
> True, but then the UA does not have an alternative outbound server, in 
> case the supplied one fails. What I don't like is the one to one 
> relation between the number of supplied outbound URIs and the number 
> of flows (and REGISTER).

There is a difference from a flow failing to a particular server and an 
outbound URI failing.  If you only provide a single URI, its trivial to 
make the single URI have several SRV entries pointing to separate 
actual servers.

> Additionally I suspect that users may still impose local policies in 
> the UAs as for the number of established flows, independently of the 
> number of learned URIs. So, the MUST in the spec is always subject to 
> local policy.

The spec currently says that UAs MUST support a minimum of 2 flows.  If 
they don't open the two flows, the mechanism is broken since the UA can 
lose inbound messages for several minutes.

thanks,
-rohan


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Subject: Re: [Sip] Ready for WGLC on Outbound?
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Jeroen,

On Jul 6, 2006, at 14:12, Jeroen van Bemmel wrote:
> [ snap ]
>> Jonathan wrote
>>
>> I'm not objecting to the concept of this check. I just think its not 
>> necessary, something that might be optional at best. As such, if 
>> folks decide they want something, it can go in another draft. I dont 
>> think we need to burden sip-outbound with this.
>
> The trouble is, once outbound is "out" any mechanism defined in a 
> separate draft won't work as good. Because then there will be clients 
> that simply ignore the flag.

Why is this anymore likely for clients that implement outbound?  What 
is the motivation for clients to ignore the flag?  If they ignore the 
flag and send STUN requests anyway against the they trash their own 
proxy.  That is not in the interest of the client vendor.

thanks,
-rohan


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Dean Willis wrote:
> The problem as I see it is we have no way of easily detecting an error 
> if somebody mistakenly lists a non-compliant proxy as an outbound 
> proxy in DHCP or some other discovery mechanism. I hate leaping off a 
> trapeze without a safety net . . .

You mean like sending SIP to an address and port just because DNS 
resolution leads you to believe that address and port is ready to 
receive SIP? But what if someone misconfigures their DNS and 
accidentally resolves a request for the SIP service to their SMTP port?

At some point, we need to trust that system administrators have a couple 
of brain cells to rub together; that they can notice when they've 
misconfigured their systems; and that they will take corrective measures 
when things fail to work.

/a

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From sip-bounces@ietf.org Fri Jul 07 11:54:43 2006
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Subject: RE: [Sip] comments on draft-johns-sip-outbound-middialog-draft-00
Date: Fri, 7 Jul 2006 09:26:18 -0600
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Jonathan,

Thanks for taking the time to read the draft and provide comments.

To your first point, I agree this is a problem but do not think it is
limited to the approach I propose. Thinking about it some more, I think
the same security risk is present in the current outbound procedures.
For your use case, the edge proxy will create a new flow II however, the
registrar will replace the existing registration with the new one given
that the instance ID and reg-ID match an existing registration. This
will essentially have the same effect of hijacking incoming calls.

The only way I could think of to mitigate this is through the
authentication framework. If the registrar challenges the registration
requests it can verify that a user is allowed to register a given AOR.
If it is allowed, then the malicious user is hijacking the registration
from themselves. If they are not allowed to register against a given AOR
then the registration request should fail and protect from malicious
users.

As for your second point, I will need to go back and read your proposal.
My initial thought was that this was configured and managed via the
service provider.

Thanks again for the comments.
Kevin

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
Sent: Thursday, July 06, 2006 4:41 PM
To: IETF SIP List
Subject: [Sip] comments on draft-johns-sip-outbound-middialog-draft-00

Kevin,

Thanks for taking a crack at this. The mid-dialog case is a nasty hard
problem and one we really do need to solve.

I'm not sure the mechanism you propose has the necessary security
properties, however. The attack I'm worried about is where an attacker,
who is a valid but malicious user in the system, registers to the
network and includes an instance ID and reg-id matching that of another
user, the target. In this case, the edge proxy creates the same flow ID,
and probably replaces the binding between that flow ID and the flow on
which the register was received. Thus, an attacker will hijack incoming
calls targeted for the target.

Another big part of the problem is how the edge proxies construct a URI
whose domain part happens to resolve to the actual two proxies which
happen to be the ones that the client registered through. With current
sip-outbound neither edge proxy knows what the other edge proxy is, so
how could you construct such a URI? (note I've just posted an
alternative solution for outbound discovery which will make this
possible).

Thanks,
Jonathan R.
--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Fri Jul 07 12:35:25 2006
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From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] Ready for WGLC on Outbound?
To: Adam Roach <adam@nostrum.com>
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Adam,

I hear what you're saying - one cannot even begin to list every possible way in which things could go wrong, let alone implement preventive measures

However, this case is different. 1) there is a simple solution that can avoid much pain and 2) What you describe below is relatively easy to diagnose (mail server receiving SIP). For outbound, what you will see is UAs that re-register every 90 seconds or so. Try detecting that, and then relating that to a DNS configuration error which is not apparent from only inspecting the DNS config
3) There are multiple config mechanisms. DNS is setup by (relative) professionals, but eg manual config is much more error prone (and nothing can be assumed about future mechanisms)

People used to think that 32 bits would be plenty,and look where that's gotten us. Now is the moment to say "if the topmost Route URI contains a 'keepalive=stun' parameter, an edge proxy MUST copy this parameter into the Path URI that it inserts". One simple sentence to prevent all sorts of trouble

Regards,

Jeroen

-----Oorspronkelijk bericht -----
Van: "Adam Roach" <adam@nostrum.com>
Aan: "Dean Willis" <dean.willis@softarmor.com>
CC: "Cullen Jennings" <fluffy@cisco.com>; sip@ietf.org; "Juha Heinanen" <jh@tutpro.com>; rohan@ekabal.com
Verzonden: 7-7-06 17:28
Onderwerp: Re: [Sip] Ready for WGLC on Outbound?

Dean Willis wrote:
> The problem as I see it is we have no way of easily detecting an error 
> if somebody mistakenly lists a non-compliant proxy as an outbound 
> proxy in DHCP or some other discovery mechanism. I hate leaping off a 
> trapeze without a safety net . . .

You mean like sending SIP to an address and port just because DNS 
resolution leads you to believe that address and port is ready to 
receive SIP? But what if someone misconfigures their DNS and 
accidentally resolves a request for the SIP service to their SMTP port?

At some point, we need to trust that system administrators have a couple 
of brain cells to rub together; that they can notice when they've 
misconfigured their systems; and that they will take corrective measures 
when things fail to work.

/a

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From sip-bounces@ietf.org Fri Jul 07 12:49:15 2006
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From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] Ready for WGLC on Outbound?
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Rohan,

Clients ignoring the flag would not do so on purpose, but simply because they would not know about it (ie only implementing outbound and not the separate extension). By making it part of outbound it becomes atomic: either the client does both STUN and "crash prevention", or neither of these. That is the only way to make it safe

Regards ,

Jeroen


-----Oorspronkelijk bericht -----
Van: "Rohan Mahy" <rohan@ekabal.com>
Aan: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
CC: dean.willis@softarmor.com; "Cullen Jennings" <fluffy@cisco.com>; "Rohan Mahy" <rohan@ekabal.com>; "Jonathan Rosenberg" <jdrosen@cisco.com>; "Anders Kristensen" <andersk@cisco.com>; sip@ietf.org
Verzonden: 7-7-06 17:25
Onderwerp: Re: [Sip] Ready for WGLC on Outbound?

Jeroen,

On Jul 6, 2006, at 14:12, Jeroen van Bemmel wrote:
> [ snap ]
>> Jonathan wrote
>>
>> I'm not objecting to the concept of this check. I just think its not 
>> necessary, something that might be optional at best. As such, if 
>> folks decide they want something, it can go in another draft. I dont 
>> think we need to burden sip-outbound with this.
>
> The trouble is, once outbound is "out" any mechanism defined in a 
> separate draft won't work as good. Because then there will be clients 
> that simply ignore the flag.

Why is this anymore likely for clients that implement outbound?  What 
is the motivation for clients to ignore the flag?  If they ignore the 
flag and send STUN requests anyway against the they trash their own 
proxy.  That is not in the interest of the client vendor.

thanks,
-rohan



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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Rohan Mahy" <rohan@ekabal.com>, "Miguel Garcia"
	<Miguel.An.Garcia@nokia.com>
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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IHi,

What if the client only gets IP address(es) of the edge proxy? Is each =
address treated as a URI?

As I've said earlier, a SIP URI is something the client creates, but it =
does not necessarily be what is configured (given as input, manually or =
using some other mechanism).

I've also indicated that there may be other mechanisms than "multi flow" to =
provide reliability. So, in those cases multiple flows may not be needed, =
but one may still want to use outbound.

Note tha I am NOT arguing against saying that a client must support the =
establishment of two flows.


Also, if a client uses STUN keepalive, but not the rest of outbound =
(instance-id etc) it would not make sense trying to send multiple =
registrations, since the registrar would not be able to differentiate the =
"flows". I think that needs to be clear in the spec, a part of defining the =
STUN keepalive as a standalone function.
Regards,

Christer


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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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Rohan,

>> Additionally I suspect that users may still impose local policies in the 
>> UAs as for the number of established flows, independently of the number 
>> of learned URIs. So, the MUST in the spec is always subject to local 
>> policy.
>
> The spec currently says that UAs MUST support a minimum of 2 flows.  If 
> they don't open the two flows, the mechanism is broken since the UA can 
> lose inbound messages for several minutes.
>

OK, let me try to see if I understand where this "several minutes" comes 
from, and if we could quantify "broken" here.

Assuming that the client uses the default keepalive timer values (UDP: 
24..29 seconds, TCP: 95..120 seconds), I deduce that worst case a client 
using only one flow could loose its registration status for a period of 120 
seconds (on average, this would be only 60 seconds). So your "several" 
equals 2 ?

One could probably calculate the chance that in this window a request would 
come in (given message rate assumptions, etc). Multiply that with the chance 
at flow failure (given component MTBF figures etc), and you would have the 
kind of reliability one would have for one flow only. Call that reliability 
R

My question now is: what are reasonable assumptions for these figures, and 
could it be that for some clients, R could be sufficient?
Intuitively I say: yes, that's possible. There exist clients and users that 
would be happy with a scheme that is broken for (1-R) part (I know at least 
one - me).

There is currently nothing in the requirements regarding reliability. 
Section 4.4 mentions the relationship between timer values and server load, 
and has some text on "the expected time that a user may not receive inbound 
communications"; but not in relation to the number of flows (nor does it 
mention to de-synchronize keepalive timers across multiple flows). Do we 
need that, perhaps with some sample quantitative analysis?

Regards,

Jeroen 


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Overall, this is a HUGE improvement over draft-tschofenig-sip-saml-04, 
which is the last version I read (I see that -05 was similar to 
draft-ietf-sip-saml). This looks like a viable solution overall, and I 
like how it dovetails with sip-identity.



Major comments:

* I'm a little worried that the SAML assertion itself doesn't include 
any content from the SIP message which clearly limits its scope of 
applicability to the request. You are relying on authentication via 
sip-identity for that, but there are attacks where this won't work.

For example, consider a rogue proxy or MITM which sees a SIP request fly 
by. The AS was executed by an upstream proxy, so this request has an 
Identity and Identity-Info header field. The proxy modifies the 
Indentity-Info header field, which contains a URL to an older assertion 
generated by the AS for the same subject, say, for a previous call it 
just made.

This is not a major threat, but there may be others like it. Is there 
harm in including things like call-id, contact and other parameters into 
the saml assertion so they are signed too?

* I'm not clear on how this mechanism works with older sip-identity 
implementations. The main issue is whether the URI in the identity-info 
header field will return just a certificate, as specified in 
sip-identity, or whether it returns a saml assertion. An older 
sip-identity verifier will expect a certificate but it will get an 
assertion if the AS implements sip-saml. Perhaps you are assuming that 
the Accept headers or something are used for this? Somehow this needs to 
be handled.


 > 6.1.4.1.3.  Element: <Conditions>
 >
 >    The <Conditions> element SHOULD contain an <AudienceRestriction>
 >    element, which itself SHOULD contain an <Audience> element.  The
 >    value of the <Audience> element SHOULD be the same as the addr-spec
 >    of the SIP request's To header field.

Hmm. Here is our infamous retargeting problem again. If the request is 
retargeted, does this mean that the assertion cannot be used by the 
forwarded party? This restriction doesn't exist for normal sip-identity.


Minor comments:

> The scope of this specification is:
> 
>    o  Specify a SIP profile of SAML -- aka a "SIP SAML profile" -- such
>       that a subject's profile attributes, and their domain's
>       certificate, can be conveyed to a relying party using SAML.  In
>       doing so, satisfy the requirements outlined in [I-D.ietf-sipping-
>       trait-authz], and compose with [I-D.ietf-sip-identity].

perhaps I've misunderstood, but don't we also need to specify a SIP SAML 
binding?

>  SIP-specific SAML profile(s) and binding(s).  This is technically
>    straigh-forward as both SAML and SIP are explicitly extensible.

straightforward

> [RFC3261].  Depending on the chosen SIP security mechanism
>            either digest authentication, S/MIME or Transport Layer
>            Security is used to provide the AS with a strong assurance
>            about the identity of Alice.

well, TLS in this use case is not normally used to provide the AS with 
the identity of ALice. Perhaps you are referring to mutual 
authentication with tls? Or maybe you just mean digest over tls?

>  signature" for the message and add Identity and Identity-Info
>            headers to the message.  The AS also at this time constructs

I feel like a broken record, but its header field and not header


* sections 6.1.3, 6.1.2 and 5 all seem pretty redundant, as they contain 
the same flow with nearly the same levels of detail. Are all three needed?


>  Attribute: NotOnOrAfter
> 
>       The value of the NotOnOrAfter XML attribute MUST be set to a time
>       instant later than the value for NotBefore.

guidance on how to set this?

>   This section specifies the "SAML HTTP-URI-based SIP Binding",
>    (SHUSB).

SHUSB? Geez, this acronym is awful.

> Section 3.7.5.4 Error Reporting:
> 
>       All SHOULDs in this section are to be interpreted as MUSTs.
> 

This is somewhat mysterious. Is there something SIP specific which would 
cause you to want to do this, or just a beef with the fact that 
everything there is a SHOULD and not a MUST? Also perhaps a few words 
about whats in that section would be helpful.

Indeed its not obvious to me why we can't just use the existing SAML URI 
binding if we're just doing a vanilla http query to retrieve the assertion.

> The value of the <NameID> element MUST be the same as the Address of
>    Record (AoR) value used in computing the "signed-identity-digest"
>    which forms the value of the Identity header.  See Section 9 of
>    [I-D.ietf-sip-identity].

I read this as being that this is the literal AoR, i.e., 
sip:user@domain. But, in the example later on:

> 1 <Assertion ID="_a75adf55-01d7-40cc-929f-dbd8372ebdfc"
>     2    IssueInstant="2003-04-17T00:46:02Z" Version="2.0"
>     3    xmlns="urn:oasis:names:tc:SAML:2.0:assertion">
>     4    <Issuer>
>     5       example.com
>     6    </Issuer>
>     7    <Subject>
>     8       <NameID
>     9         Format=
>    10         "urn:oasis:names:tc:SAML:1.1:nameid-format:emailAddress">
>    11         Alice@example.com
>    12       </NameID>

there is no sip:, and the format attribute has a URN which explicitly 
says emailAddress and not SIPURI. I think this should be a SIP URI.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Fri Jul 07 16:04:28 2006
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I wasn't able to find any disucssion on this draft so far in the 
archives; my apologies if I missed it.

This draft concerns me greatly.

The first thing that worries me is the general idea that 3gpp or tispan 
get to 'choose' which SIP header field contains what type of identity. 
They get to do no such thing. These header fields have defined meanings. 
Having either organization decide to specify their own interpretation 
for the type of identity that goes in there is a guarnateed receipe for 
huge, massive, and unending interoperability failures.

On the technical front, I strongly disagree that the From and 
P-Asserted-Identity represent different types of identity in any way. 
They both have always identified the user that is the sender of the 
request. In cases where you had an enterprise user Joe making a SIP 
call, the From would contain sip:joe@enterprise.com and this would be 
authenticated by the proxy in the enterprise, followed by insertion of a 
P-Asserted-ID which says the same thing. The From, in this use case, 
should not be sip:enterprise.com.

Next, I am puzzled by the requirement to deliver separate identities to 
the calling party for these two things. I understand the concept of a 
billing identity being separate from the identity of the caller. I do 
not, however, understand why the billing ID is of any interest to the 
called party whatsoever. It seems that this would, at best, be of 
interest only within the domain of the calling party. Since its also 
directly inferrable from the identity of the caller (presumably the 
database contains a linkage between the two), I also don't see why it 
would need to be carried in the sip message itself. Is there a use case 
where there is not a provisioned relationship between the user making 
the call and the billing entity? Certainly there is a clear direct 
relationship in the enterprise example in the draft.

Finally, even if I buy the idea that there are two identities and that 
they both need to be conveyed, I strongly disagree with the notion that 
one of them can be conveyed with SIP-identity, while P-A-ID is OK for 
the other. There are certain well-known technical limitations with 
P-A-ID which sip-identity addresses. There is nothing about the nature 
of the identity which invalidates any of those issues.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Sorry - haven't been following this and don't have a position on the 
merits. But one thing Jonathan didn't mention below that might be relevant:

With two connections, the traffic is split across them. This potentially 
affects the flow control. I can see that there could be both pros and 
cons to this.

	Paul

Jonathan Rosenberg wrote:
> Having just done a detailed review of the document, I will say that I am 
> skeptical of the value of this specification at this time.
> 
> Firstly, it has fairly limited deployment scenarios:
> 
> 1. no NATs between the elements
> 2. mutual TLS auth only
> 
> assuming these are true, its valuable only when there is generally a lot 
> of churn in TLS connection setups. For example, in an environment where 
> there is a fixed and known-ahead-of-time peering relationship between 
> providers, connect-reuse doesn't do much for you anyway, since there are 
> relatively few TLS connections and they are set up when the system 
> boots, or perhaps just once.
> 
> Even if the connections aren't set up at startup, but on demand, over 
> time the value of connect-reuse is really zero if the total number of 
> TLS connections that are needed by the proxy is below the capacity of 
> the server. In other words, if a proxy only ever connects to 100 other 
> proxies, then at some point the proxy gets its 200 TLS connections set 
> up (assuming no connect-reuse) and thats it. It never tears them down 
> since it never runs out of resources, so TLS handshaking becomes a 
> fixed, one-time cost. That makes it uninteresting to optimize.
> 
> The value, then, comes only when the set of proxies that get connected 
> to is constantly changing, such that TLS handshaking between proxy peers 
> is an ongoing and non-trivial percentage of the overall CPU effort.
> 
> To be frank, I am not aware of a single deployment situation today that 
> meets this constraint. Can someone point me to a real deployment 
> scenario where connect-reuse is applicable and has a non-trivial 
> improvement in the performance of the proxy?
> 
> Thanks,
> Jonathan R.
> 
> 
> Nils Ohlmeier wrote:
> 
>> On Friday 23 June 2006 23:08, Dean Willis wrote:
>>
>>> We've been following the connection reuse draft, currently draft-ietf-
>>> sip-connect-reuse-05.txt, for a long long time.
>>>
>>> We've slipped the charter delivery date several times.
>>>
>>> Are we ever going to finish this work? Do we need to finish this work?
>>>
>>> Is this something that we could just not do?
>>>
>>>
>>> Otherwise said: "Does the SIP working group still have a consensus to
>>> expend working group effort on connection reuse?"
>>>
>>> If you have an opinion, for or against, please let it be known on
>>> this list.
>>
>>
>> I think their are two options left:
>>
>> - either push it out as "TLS-connection-reuse-between-proxies" (if 
>> nobody comes up with any issue any more)
>> - or drop it because the real world usage is too limited
>>
>> I have little preference for the first alternative, but wouldn't mind 
>> to go for the second option.
>>
>> Regards
>>   Nils
>>
>> _______________________________________________
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>>
> 

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From sip-bounces@ietf.org Fri Jul 07 20:51:41 2006
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This draft attempts to clarify the underspecification of SIPS in RFC
3261. This seems like a good idea in principle, but at least from the
security guy perspective, I worry that this draft isn't completely
baked yet. I do hasten to add that this is the security guy
perspective, so there are probably SIP complexities I'm missing.

That said, ISTM that there are a number of problems that need to be
disentangled:

1. What a UA which is trying to contact a given URI 
   can expect in terms of how their messages will be
   handled.
2. What a UA can safely publish as their AOR and how
   to do it.
3. What a UA which receives a message can determine
   about how the message was handled.

This draft handles all of these issues together and I think
it would be best to handle them separately. Taking a step back,
I think we want the following:

1. A UA which tries to dereference a SIPS URI knows that
   as long as its dealing with compliant proxies its messages
   will be transmitted securely all the way to the peer.

2. A UA can make itself available in the following ways:
   - Secure only
   - Secure or insecure but hopefully if you're dealing
     with a secure peer then you both get secure.
   - Insecure only.
   So, AOR advertisement should support this.

3. A UA can determine (again assuming it's dealing with compliant
   proxies) whether the path from the peer was secure.

Do I have these requirements right?


Issue (1) needs to be dealt with by imposing rules about 
what the proxies do. I.e., that when presented with SIPS
they use TLS in the outgoing direction. The edge case--and
the one that's creating most of the heartbreak--appears to
be what happens inside of trust domains. 

Based on the discussion in this draft, Issue (2) breaks down
into two cases:

(a) What you advertise when you register.
(b) What you put in your own headers (i.e., the To).

Case (a) seems like it's relatively straightforward using the rules
described in S 5. It's case (b) that's tricky, due to the asymmetry of
"SIPS means you must use TLS to your first hop proxy" and "SIPS means
you may or may not use TLS between the last hop proxy and the final
destination".  Since that last link is about to become the first
hop, this seems highly problematic.

ISTM that there are two natural fixes for this:

(a) Decree that SIPS means TLS all the way (this is a lot less
    problematic in the face of outbound than it otherwise
    would be.)
(b) Decree that it's none of our business what happens inside
    people's trust boundaries and so SIPS only means secure
    in between them.

This draft, however, takes neither approach and instead attempts
to thread the needle between these. I'm not sure that this is
practical--and it's certainly creating a lot of complexity.


Issue (3) seems like it ought to be solved the same way that it's
ntended to be solved in RFC 3207 (STARTTLS): examine the Via lines
or something like them. I would ignore the URI entirely here, especially
in view of the confusion noted above.


DETAILED COMMENTS:
S 5.1:
I'm not sure why you need TLS transport to do a registration
of a SIPS contact.

S 6.

   If a UAS does not support SIPS, it MUST reject a request over TLS
   with response code 416. 

I don't understand how this can happen. Can you explain?
  
   Upon receiveing a 416 a UAC SHOULD not re-
   attempt the request with a SIP URI by automatically replacing the
   SIPS scheme with a SIP scheme.  If the UAC does re-attempt the call
   with a SIP URI, it SHOULD inform to the user that the security level
   is downgraded.

Shouldn't this last SHOULD be a MUST? It seems like a security condition.


   The presence of a SIPS Request-URI does not necessarily indicate that
   the request was sent end-to-end securely. 

The presence by the time you receive it, right? CAuse it should mean
that at the sender....



   26.4.4/RFC 3261 also explains that S/MIME may also be used by the
   originating UAC to ensure that the original form of the To header
   field is carried end-to-end.  While not specifically mentioned in
   26.4.4/RFC 3261, this is meant to imply that [RFC3893] would be used
   to "tunnel" important headers (such as To and From) in an encrypted
   S/MIME body, replicating the information in the SIP message, and
   allowing the UAS to validate the content of those important headers.
   While this approach is certainly legal, another approach is to use
   the SIP Identity mechanism defined in [I-D.ietf-sip-identity].  SIP
   Identity creates a signed identity digest which includes, amongst
   other things, the AOR of the sender (from the From header) and the
   AOR of the original destination (from the To header).  It is
   RECOMMENDED that a UAC use the mechanism in [I-D.ietf-sip-identity]
   instead of the one defined in RFC 3893.

You say "encrypted" here, but you really mean signed, right?


S 13.
I'm not sure I find this that helpful.

S 14.
It seems to me that most of these rules have security implications.

-Ekr













    




   

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From sip-bounces@ietf.org Fri Jul 07 21:23:33 2006
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From: "Dan Wing" <dwing@cisco.com>
To: "'Elwell, John'" <john.elwell@siemens.com>, <sip@ietf.org>
Subject: RE: [Sip] Issue 4 in draft-ietf-sip-connected-identity-00.txt
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> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Tuesday, May 02, 2006 1:24 PM
> To: sip@ietf.org
> Subject: [Sip] Issue 4 in draft-ietf-sip-connected-identity-00.txt
> 
> The fourth issue is not actually in draft-00 but was raised
> subsequently.  If a UA supports a change of From header field URI
> mid-dialog, as indicated by the "id-change" option tag, we could
> mandate that it also support the SIP UPDATE method.  Then the peer
> UA in a dialog would be free to use the UPDATE method to report
> connected identity, rather than the less suitable re-INVITE method.
> The use of UPDATE means 2 messages instead of 3 and avoids the need
> for an offer-answer exchange.  I heard no objections when this was
> proposed a few days ago.  Do we have consensus?

I prefer UPDATE.  I am considering the needs of some proposed 
SRTP keying techniques which could benefit from strong authentication
of identity and of the fingerprints or certificates in SDP.  For
those cases it's helpful to include SDP in the UPDATE so that it
gets signed with Identity, even if that SDP is exactly the same as
the SDP in the answer.  An example of where Identity could help is 
with a pure DH mode (such as draft-baugher-mmusic-sdp-dh).  It can 
also help provide additional protection of a UA's self-signed 
certificate (various MIKEY modes) or certificate fingerprint 
(mmusic-comedia-tls, mcgrew-tls-srtp).  

Even without SRTP and without comedia-tls, the reasons that 
Identity chose to include the offerer's SDP in its signature 
should be applicable to the answerer's SDP.  Otherwise we're
saying the answerer's SDP doesn't need authentication or
integrity protection.

-d

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From sip-bounces@ietf.org Fri Jul 07 21:23:38 2006
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From: "Dan Wing" <dwing@cisco.com>
To: "'Elwell, John'" <john.elwell@siemens.com>, <sip@ietf.org>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
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I hadn't noticed any followups to this issue.  I have a suggestion:

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Tuesday, May 02, 2006 1:21 PM
> To: sip@ietf.org
> Subject: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
>
> The second issue concerns the fact that an UPDATE request in the
> backward direction following dialog establishment can be wasted if
> the peer UA (the UAC for the INVITE transaction) does not support
> it.  It states:
>
>       "OPEN ISSUE: If the identity has has not changed compared with
>       the URI in the To header field and the UAC (for the INVITE
>       request) does not support verification of an Identity header,
>       this UPDATE or re-INVITE transaction would fulfil no useful
>       purpose.  However, the UAS (for the INVITE transaction) does
>       not know this.  In order to save this round trip, should we
>       have an additional option tag to indicate support for receipt
>       of the Identity header on mid- dialog requests?  Then if this
>       option tag is not present in the INVITE request the additional
>       transaction could be omitted.  Note that the id-change option
>       tag does not seem appropriate for this purpose, since a UA
>       could support a change of URI without supporting Identity."
>
> Whilst I do not wish to proliferate option tags, I do see some
> benefit in such a tag.  Any opinions?

If an Invite arrives with both Identity and the id-change tag,
it indicates the offerer understands both Identity and connected-
identity, so the answerer should generate an request (UPDATE or 
re-INVITE).

-d


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From sip-bounces@ietf.org Sat Jul 08 01:06:37 2006
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Hi Jonathan;

My comments inline.

Jonathan Rosenberg wrote:
> Overall, this is a HUGE improvement over draft-tschofenig-sip-saml-04,
> which is the last version I read (I see that -05 was similar to
> draft-ietf-sip-saml). This looks like a viable solution overall, and I
> like how it dovetails with sip-identity.
>
>
>
> Major comments:
>
> * I'm a little worried that the SAML assertion itself doesn't include
> any content from the SIP message which clearly limits its scope of
> applicability to the request. You are relying on authentication via
> sip-identity for that, but there are attacks where this won't work.
>
> For example, consider a rogue proxy or MITM which sees a SIP request
> fly by. The AS was executed by an upstream proxy, so this request has
> an Identity and Identity-Info header field. The proxy modifies the
> Indentity-Info header field, which contains a URL to an older
> assertion generated by the AS for the same subject, say, for a
> previous call it just made.
>
> This is not a major threat, but there may be others like it. Is there
> harm in including things like call-id, contact and other parameters
> into the saml assertion so they are signed too?
>
> * I'm not clear on how this mechanism works with older sip-identity
> implementations. The main issue is whether the URI in the
> identity-info header field will return just a certificate, as
> specified in sip-identity, or whether it returns a saml assertion. An
> older sip-identity verifier will expect a certificate but it will get
> an assertion if the AS implements sip-saml. Perhaps you are assuming
> that the Accept headers or something are used for this? Somehow this
> needs to be handled.
>
I asked the exact question on the SIP-SAML design team ML as well as in
the last meeting.
Currently as you mentioned, there is no indication on whether the URI
referred by identity-info
header points to a certificate or sip-identity.

As far as I understood from the reply I got was that because SAML is
based on federation or
bi-lateral agreement, AS should know whether the target will understand
the SAML. So there is
no problems.

Although, if call is forwarded to a domain which is not part of the
federation, it may cause
some issues.

There are few other issues with the current SIP-SAML spec if it was to
be used to convey
information that would affect a routing decision on the signaling path.
The issues are addressed
on the draft which evaluates the profile defined in the SIP-SAML spec to
convey CPC related
information.

http://www.ietf.org/internet-drafts/draft-schubert-sipping-saml-cpc-01.txt

BTW it's not currently mentioned in the draft, but if more than one
asserting party exists on
the signaling path that needs to add artifacts(URI to the assertion),
the current SIP-SAML
might not work as sip-identity only allows one uri to be referred AFAIK.

> > 6.1.4.1.3. Element: <Conditions>
> >
> > The <Conditions> element SHOULD contain an <AudienceRestriction>
> > element, which itself SHOULD contain an <Audience> element. The
> > value of the <Audience> element SHOULD be the same as the addr-spec
> > of the SIP request's To header field.
>
> Hmm. Here is our infamous retargeting problem again. If the request is
> retargeted, does this mean that the assertion cannot be used by the
> forwarded party? This restriction doesn't exist for normal sip-identity.
Agreed.
Same can be said if the assertion is to be used by the intermediaries to
alter the
routing of the call.
>
>
> Minor comments:
>
>> The scope of this specification is:
>>
>> o Specify a SIP profile of SAML -- aka a "SIP SAML profile" -- such
>> that a subject's profile attributes, and their domain's
>> certificate, can be conveyed to a relying party using SAML. In
>> doing so, satisfy the requirements outlined in [I-D.ietf-sipping-
>> trait-authz], and compose with [I-D.ietf-sip-identity].
>
> perhaps I've misunderstood, but don't we also need to specify a SIP
> SAML binding?
>
>> SIP-specific SAML profile(s) and binding(s). This is technically
>> straigh-forward as both SAML and SIP are explicitly extensible.
>
> straightforward
>
>> [RFC3261]. Depending on the chosen SIP security mechanism
>> either digest authentication, S/MIME or Transport Layer
>> Security is used to provide the AS with a strong assurance
>> about the identity of Alice.
>
> well, TLS in this use case is not normally used to provide the AS with
> the identity of ALice. Perhaps you are referring to mutual
> authentication with tls? Or maybe you just mean digest over tls?
>
>> signature" for the message and add Identity and Identity-Info
>> headers to the message. The AS also at this time constructs
>
> I feel like a broken record, but its header field and not header
>
>
> * sections 6.1.3, 6.1.2 and 5 all seem pretty redundant, as they
> contain the same flow with nearly the same levels of detail. Are all
> three needed?
>
>
>> Attribute: NotOnOrAfter
>>
>> The value of the NotOnOrAfter XML attribute MUST be set to a time
>> instant later than the value for NotBefore.
>
> guidance on how to set this?
>
>> This section specifies the "SAML HTTP-URI-based SIP Binding",
>> (SHUSB).
>
> SHUSB? Geez, this acronym is awful.
>
>> Section 3.7.5.4 Error Reporting:
>>
>> All SHOULDs in this section are to be interpreted as MUSTs.
>>
>
> This is somewhat mysterious. Is there something SIP specific which
> would cause you to want to do this, or just a beef with the fact that
> everything there is a SHOULD and not a MUST? Also perhaps a few words
> about whats in that section would be helpful.
>
> Indeed its not obvious to me why we can't just use the existing SAML
> URI binding if we're just doing a vanilla http query to retrieve the
> assertion.
>
>> The value of the <NameID> element MUST be the same as the Address of
>> Record (AoR) value used in computing the "signed-identity-digest"
>> which forms the value of the Identity header. See Section 9 of
>> [I-D.ietf-sip-identity].
>
> I read this as being that this is the literal AoR, i.e.,
> sip:user@domain. But, in the example later on:
>
>> 1 <Assertion ID="_a75adf55-01d7-40cc-929f-dbd8372ebdfc"
>> 2 IssueInstant="2003-04-17T00:46:02Z" Version="2.0"
>> 3 xmlns="urn:oasis:names:tc:SAML:2.0:assertion">
>> 4 <Issuer>
>> 5 example.com
>> 6 </Issuer>
>> 7 <Subject>
>> 8 <NameID
>> 9 Format=
>> 10 "urn:oasis:names:tc:SAML:1.1:nameid-format:emailAddress">
>> 11 Alice@example.com
>> 12 </NameID>
>
> there is no sip:, and the format attribute has a URN which explicitly
> says emailAddress and not SIPURI. I think this should be a SIP URI.

I addressed this a while ago as well.

Regards
Shida

>
> Thanks,
> Jonathan R.


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From sip-bounces@ietf.org Sat Jul 08 01:47:11 2006
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Ready for WGLC on Outbound?
Date: Fri, 7 Jul 2006 22:45:06 -0700
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Jeroen,

Clients which implement outbound-04 as written already have a flag: the 
presence of keepalive=stun in the outbound-proxy-set.  The draft is 
pretty clear that clients send STUN messages to their outbound proxies 
when this parameter is present.

Some *other* flag that lets a client with a derived outbound-proxy-set 
figure out it is also safe to send STUN to a proxy is a perfectly fine 
future extension which does no harm.  Clients that just implement 
outbound-04 will never send STUN messages to the proxies with the 
future extension but no keepalive parameter, nor will they send STUN 
messages to unsuspecting proxies.

thanks,
-rohan


On Jul 7, 2006, at 9:47, Jeroen Van Bemmel wrote:

> Rohan,
>
> Clients ignoring the flag would not do so on purpose, but simply 
> because they would not know about it (ie only implementing outbound 
> and not the separate extension). By making it part of outbound it 
> becomes atomic: either the client does both STUN and "crash 
> prevention", or neither of these. That is the only way to make it safe
>
> Regards ,
>
> Jeroen
>
>
> -----Oorspronkelijk bericht -----
> Van: "Rohan Mahy" <rohan@ekabal.com>
> Aan: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
> CC: dean.willis@softarmor.com; "Cullen Jennings" <fluffy@cisco.com>; 
> "Rohan Mahy" <rohan@ekabal.com>; "Jonathan Rosenberg" 
> <jdrosen@cisco.com>; "Anders Kristensen" <andersk@cisco.com>; 
> sip@ietf.org
> Verzonden: 7-7-06 17:25
> Onderwerp: Re: [Sip] Ready for WGLC on Outbound?
>
> Jeroen,
>
> On Jul 6, 2006, at 14:12, Jeroen van Bemmel wrote:
>> [ snap ]
>>> Jonathan wrote
>>>
>>> I'm not objecting to the concept of this check. I just think its not
>>> necessary, something that might be optional at best. As such, if
>>> folks decide they want something, it can go in another draft. I dont
>>> think we need to burden sip-outbound with this.
>>
>> The trouble is, once outbound is "out" any mechanism defined in a
>> separate draft won't work as good. Because then there will be clients
>> that simply ignore the flag.
>
> Why is this anymore likely for clients that implement outbound?  What
> is the motivation for clients to ignore the flag?  If they ignore the
> flag and send STUN requests anyway against the they trash their own
> proxy.  That is not in the interest of the client vendor.
>
> thanks,
> -rohan
>


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From sip-bounces@ietf.org Sat Jul 08 02:13:06 2006
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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Fri, 7 Jul 2006 23:11:16 -0700
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Hi Christer,

It is near the end of a long day today, so I am having trouble 
understanding why we are still having this discussion.  I really do 
want to understand what you are trying to accomplish.  I am just 
relating a humorous thought that popped into my punchy brain as I sit 
here.

I have to admit that my initial reaction is that this is something like 
the following vignette.  A coach tells a runner about the new track 
about to be installed with black spongy rubber.  The runner asks can 
you please add a sprinkler system to it before it is installed?  The 
coach asks why and the runner responds that the black track will be hot 
to run on in bare feet.

Of course running on the track with bare feet is a perfectly 
acceptable, but running with shoes is certainly more common, possibly 
even recommended.  The new track will be a great improvement for 
everyone and will work with and without shoes.  However its unclear 
that the sprinkler system is really a good idea, that it will work, how 
long it will take to install, and how much it will cost.  It could be 
installed later if it turns out to be needed.  In the interim, folks 
with shoes and tough feet can use the track immediately.

On Jul 7, 2006, at 10:06, Christer Holmberg wrote:
> Hi,
>
> What if the client only gets IP address(es) of the edge proxy? Is each 
> address treated as a URI?

the client needs to figure out how to derive a URI to place in its 
Route header or to use as a target for RFC 3263.  how this is done is 
up to you, but getting an IP address alone feels like getting running 
shorts and no shoes.

> As I've said earlier, a SIP URI is something the client creates, but 
> it does not necessarily be what is configured (given as input, 
> manually or using some other mechanism).
>
> I've also indicated that there may be other mechanisms than "multi 
> flow" to provide reliability. So, in those cases multiple flows may 
> not be needed, but one may still want to use outbound.

I still don't see the advantage of trying to do one flow "reliability" 
with outbound proxies only addressed by an IP address.  Why is going 
barefoot better than wearing sneakers?  I think I would need to see a 
full use case (a few paragraphs and maybe a simple drawing) to get what 
you have in mind.

> Note that I am NOT arguing against saying that a client must support 
> the establishment of two flows.

ok, good to know.

> Also, if a client uses STUN keepalive, but not the rest of outbound 
> (instance-id etc) it would not make sense trying to send multiple 
> registrations, since the registrar would not be able to differentiate 
> the "flows". I think that needs to be clear in the spec, a part of 
> defining the STUN keepalive as a standalone function.

I actually understand the motivation for this one.  The issue is how 
much of the STUN keepalive mechanism belongs in 3489bis, how much could 
go in outbound in a separate section (or a separate document), and how 
much is specific to outbound.  I think all the text about what you do 
when you try to setup a flow and it doesn't work (retry after x 
seconds...) is really very specific to multiple flows.  There is the 
binding request/response that belongs in 3489bis.  What is unclear to 
me is what else is left.  Certainly you need some part of a doc that 
defines the keepalive URI parameter and updates 3261 to say it ok to 
send STUN on the same port as SIP when you have positive knowledge 
(such as the keepalive parameter) that the server is ok with that.

thanks,
-rohan


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Jonathan: Thanks for a detailed read.  A lot of your comments
were incorporated as requested; below are the remaining
comments that warranted a discussion.

Jonathan Rosenberg wrote:
> another important reason is that, in the case of TLS, only one side 
> will offer a certificate.

Is that always true?  After all, a server can ask a client of
a certificate as well.  If the client is a proxy, it will have one,
and it will give it to the server.  In connect-reuse, if
mutual authentication is to be done, the server has to authenticate
the client as well.  An attempt on how it does that is  is teased
out in domain-certs (more on that in a separate email thread).

> this presumes in general that there aren't persistent, long-lived 
> connections already set up.

Which is reasonable today, since there are no guidelines for
persistent long-lived connections in rfc3261.  Instead it talks
about general reuse of connections without specifying anything else.
Maybe some will say that how to do this is implementation defined,
but I think that we ought to be able to provide some more stricter
guidelines to implementors, especially as TCP and TLS become the
dominant transports.  Otherwise it is already a mess for TCP,
where some implementations keep the connections open across
transactions and dialogs, some do not, and yet others open a new
connection to send a response!  Let's not create the same mess for
TLS.

>> This effectively doubles the load on each entity.
> 
> definitely not.
> 
> It doubles the amount of work for connection establishment, [...] 
> Again though I wonder if the connection establishment is really the 
> dominant CPU cost.

In "SSL and TLS" by EKR, Section 6.19 analyzes TLS connection
establishment.  Servers will build up considerable queues as
the server CPU saturates by the SSL handshakes it is already
performing.

> Can you give a specific use case? I don't think there are established
>  guidelines for how long it takes to add video, for example.

No, they are not.  However, it is appropriate that a re-INV
over an existing TLS connection will complete faster than one
that requires SSL handshake.

>> The presence of the "alias" parameter indicates that P1 does 
>> support aliasing.  P2 now authenticates the connection and if the 
>> authentication was successful, P2 creates an alias to P1 using the
>>  advertised address in the topmost Via.
> 
> Firstly, you need to define 'successful' authentication. You need to
> match the domain in the certificate with the via address.

Good point about defining 'successful' authentication.  I have
tried to do highlight the issue of a server authenticating a
client in domain-certs.  Via is the only obvious choice here.

> Also, this last sentence is very confusing, since the table is NOT
> populated with the advertised address, its populated with the
> received address, as described in the text below.

I will clarify.

> Also how do you populate destination port? The overview section 
> doesn't say. I believe it is NOT the received port (as IP address is)
>  but rather the one advertised in the sent-by field, or 5061 (the 
> default) if no port is listed.

Yes; I will clarify.

>> structure in which it keeps active connection state so that 
>> aliases, active connections, and blacklisted nodes are all 
>> discovered when looking for an active connection.
> 
> you've not defined 'an additional data structure' or blacklisted 
> nodes; I couldn't follow this paragraph.

I will rephrase and clarify.

>> 6.  Requirements
> 
> you may want to have an intro paragraph which says that these are the
>  requirements which motivated this specification

Sure.

>> 7.  Formal Syntax
[...]
> please use the =/ syntax notation that we've started using for sip 
> specs. Its much better.

Sure.

>> request.  Whenever the RFC3263 server selection mechanism executed 
>> at the receiver results in the choice of this IP address, port, and
>>  transport tuple, the alias MUST be used instead.
> 
> I think a SHOULD is fine here; and indeed if the connection has gone
>  away you obviously won't be reusing the alias.

I can clarify the corner case.  But I really think that the
normative strength should be MUST.  The more choices, the
more complexity.  Think of it from an implementation point
of view: if an alias exists, why not use it instead of
opening up a new connection?  If you open up a new connection,
then do you put it in the alias table?  If you do, then you
have two aliases.  How do you choose which one to use on
subsequent requests?

>> An alias is formed at the sender of the request when it executes 
>> the RFC3263 server selection mechanism to arrive at an IP address, 
>> port, and transport tuple to send a request to.  Subsequent 
>> requests going to the same destination address MUST use the alias 
>> instead.
> 
> Hmm this MUST is stronger than in rfc3261, which actually says 
> SHOULD. I think the existing SHOULD is fine.

Yes, rfc3261 says SHOULD (actually RECOMMENDED is the word used
there).  But for the same reason I wrote above, it will be
good to reuse the same alias.

>> A TLS server conformant to this specification MUST ask for a client
>> certificate; if the client possesses a certificate, it will be 
>> presented to the server for mutual authentication.  The server MUST
>> ensure that the canonical host name of the client is present 
>> either
> 
> where does one obtain the canonical hostname?
> 
> It seems to me, that in a proxy to proxy case, the receiving proxy 
> (the server side) will need to check the record-route hostname 
> against the client certificate.

R-R is optional.  The only header that fits the bill is Via.

Thanks so much for a detailed read.  Much appreciated.

Ciao.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)


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From sip-bounces@ietf.org Sat Jul 08 03:09:26 2006
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Jonathan Rosenberg wrote:
> Having just done a detailed review of the document, I will say that I am 
> skeptical of the value of this specification at this time.
> 
> Firstly, it has fairly limited deployment scenarios:
> 
> 1. no NATs between the elements
> 2. mutual TLS auth only

Jonathan: This is imposed by other realities.  The NAT issue is
discussed in outbound, and a decision has already been made not
to do connect reuse of TCP connections.

> assuming these are true, its valuable only when there is generally a lot 
> of churn in TLS connection setups. For example, in an environment where 
> there is a fixed and known-ahead-of-time peering relationship between 
> providers, connect-reuse doesn't do much for you anyway, since there are 
> relatively few TLS connections and they are set up when the system 
> boots, or perhaps just once.

True, but there is nothing to stop UACs in an enterprise to do
connect reuse to their default outbound proxy.

> Even if the connections aren't set up at startup, but on demand, over 
> time the value of connect-reuse is really zero if the total number of 
> TLS connections that are needed by the proxy is below the capacity of 
> the server. In other words, if a proxy only ever connects to 100 other 
> proxies, then at some point the proxy gets its 200 TLS connections set 
> up (assuming no connect-reuse) and thats it. It never tears them down 
> since it never runs out of resources, so TLS handshaking becomes a 
> fixed, one-time cost. That makes it uninteresting to optimize.

RFC3261 is rather silent on how to do persistent connections.
This has let to a mess for TCP, where some implementations keep the
connections open across transactions and dialogs, some do not, and
yet others open a new connection to send a response!  Let's not
create the same mess for TLS.

> The value, then, comes only when the set of proxies that get connected 
> to is constantly changing, such that TLS handshaking between proxy peers 
> is an ongoing and non-trivial percentage of the overall CPU effort.

...Or the number of UACs using a default outbound proxy is
constantly changing...

> To be frank, I am not aware of a single deployment situation today that 
> meets this constraint. Can someone point me to a real deployment 
> scenario where connect-reuse is applicable and has a non-trivial 
> improvement in the performance of the proxy?

Connection reuse is underspecified in rfc3261, which takes the
position that connections accepted by the transport layer
will frequently not be reused.  Connect-reuse solves this
problem.  Without it, there is *absolutely* no guidance in how
to send requests in the backwards direction on the same
connection if outbound is not used.  Given the state of
TCP connection reuse today, I think we can do better with
TLS.  The draft itself has seen implementations, so it
appears that it has some utility.

The draft can be wrapped up in a couple of weeks after
Montreal.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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From sip-bounces@ietf.org Sat Jul 08 03:51:41 2006
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Rohan Mahy" <rohan@ekabal.com>, "Jeroen Van Bemmel" <jbemmel@zonnet.nl>
Subject: Re: [Sip] Ready for WGLC on Outbound?
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Hi,

Maybe I've missed it, but what is the purpose of sending the keepalive =
parameter to the edge proxy in the first place?

To me it is only used to tell the client to use STUN keepalive, but what =
will the proxy do with that information? Should it reject a reuest if it =
does expect STUN, but the parameter is not present? Or?

Regards,

Christer

Regards,

Christer


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From sip-bounces@ietf.org Sat Jul 08 07:30:35 2006
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I have one concern and one comment.

The concern in the use of the From header value to compare to the 
p-asserted-identity value. In IMS, those might not be the same.

The comment I have is about the behaviour of the ingress proxy. I think 
it would be a really good idea for the ingress proxy to re-populate the 
Identity header and Identity-Info header using its own certificate 
after verifying the Identity header value from the remote egress proxy. 
This helps the UAS because it will most likely have the certificate of 
its ingress proxy and does not have to find the certificate of the 
remote end.

Regards,
Hisham


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From sip-bounces@ietf.org Sat Jul 08 09:15:27 2006
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 From section 6.1:

   When a reg-id header parameter is present in a Contact header field
   value in a REGISTER request, the corresponding binding is between an
   AOR and the combination of instance-id and reg-id.

I don't think this is quite correct. This statement appears to be 
asserting that {instance-id,reg-id} is a unique key to a binding 
associated with an AOR. This would, in turn, imply that any successful 
registration attempt with an {instance-id,reg-id} that matches an 
existing binding would overwrite the existing record.

I don't think that is what's intended, since section 6.2 contains the 
following text:

   o  If two bindings have the same instance-id and reg-id, the proxy
      SHOULD prefer the contact that was most recently updated.

If the binding is only to {instance-id,reg-id}, then that text is 
nonsensical: you can't have two identical bindings to the same AOR.

I suspect that the text in 6.1 should actually read something more like:

   When a reg-id header parameter is present in a Contact header field
   value in a REGISTER request, the corresponding binding is between an
   AOR and the combination of instance-id, reg-id, and network flow.

/a

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   From: Ben Campbell <ben@estacado.net>

   I've been watching this train of reasoning for a while now, and am
   not sure I buy it. Do we really expect SIP devices to melt down if
   they get stun and don't expect it? If so, they are big targets for
   DoS attacks.

The problem (I think) is whether the device can recover framing
adequately to receive the next message on the "channel" in question.
Clearly, this isn't a problem with UDP or SCTP, which frame messages.
But if a UA establishes a TCP connection and sends a STUN message that
the recipient doesn't understand, we need to know that the recipient
can recover framing fast enough to process the SIP that the UA sends
afterward.  Otherwise, the UA's STUN messages would be performing a
DoS attack on its own SIP connection!

Dale

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Juha Heinanen wrote:
> Adam Roach writes:
>
>  > To handle this situation, it might make sense to put stuff in 6.2
>  > that basically says that, for contacts with matching instance-ids,
>  > the proxy MUST NOT try a contact until all contacts at a higher
>  > q-value have been tried first. (I could go along with a weaker
>  > strength, but I really do think a prescriptive approach is the only
>  > thing that allows the clients to handle the situation I describe
>  > above in a manner that is predictable to the users).
>
> i you suggesting that proxies consult each other about the q-value UA
> used when registering with each proxy before forwarding a request to the
> UA?  if so, your suggesting makes no sense.
>
> -- juha
>   


I have no idea what you're saying here. Let's try getting some common 
context.

Open the outbound document. Go to section 6.2. You'll notice that it 
talks about the rules that proxy/registrars use to populate the target 
set. This is the section to which my comment applies. In particular, 
there are two new rules placed on proxy/registrars that implement 
outbound, specified in a bullet list.

I'm suggesting a third bullet that reads something like:

    * If multiple bindings have the same instance-id, the proxy MUST NOT
      use a binding until all bindings of greater q-value have been tried.

/a

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From sip-bounces@ietf.org Sat Jul 08 09:15:32 2006
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Hadriel:

I agree with your assessment, and would be _much_ happier if we simply 
removed all the text that implied that the proxy has to maintain 
multiple bindings with the same instance-id and reg-id.

I think, at the very least, the draft needs some motivating text around 
why the proxy/registrar is expected to keep tabs of stale registration 
state.

/a

Hadriel Kaplan wrote:
> My interpretation was the opposite - if a Register comes in with the same
> {instance-id,reg-id} then it should indeed replace/overwrite the previous
> binding of the same tuple.  I know there are several sections in the draft
> which imply otherwise, but I don't understand why.  There is no real value
> I can see to keeping a binding for an old flow of the same reg-id, since by
> definition the UA would only "re-use" a reg-id if the old flow failed (or
> even if it didn't think it had failed, but something rebooted and changed
> the flow).  
>
> I can see no realistic way in which an old flow of the same
> {instance-id,reg-id} would succeed in reaching the UA.  Am I missing
> something?
>
> -hadriel
> p.s. it's theoretically possible, if a sip-udp UA rebooted and resolved a
> new proxy address and sent a register to it with the same reg-id, that it
> may still be possible to reach the UA through the old "flow" for a time.  
> But is that a realistic case worth caring about?
>
>
>
>   
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Thursday, July 06, 2006 1:12 AM
>> To: Cullen Jennings
>> Cc: SIP WG
>> Subject: [Sip] Outbound Comment: binding key
>>
>>  From section 6.1:
>>
>>    When a reg-id header parameter is present in a Contact header field
>>    value in a REGISTER request, the corresponding binding is between an
>>    AOR and the combination of instance-id and reg-id.
>>
>> I don't think this is quite correct. This statement appears to be
>> asserting that {instance-id,reg-id} is a unique key to a binding
>> associated with an AOR. This would, in turn, imply that any successful
>> registration attempt with an {instance-id,reg-id} that matches an
>> existing binding would overwrite the existing record.
>>
>> I don't think that is what's intended, since section 6.2 contains the
>> following text:
>>
>>    o  If two bindings have the same instance-id and reg-id, the proxy
>>       SHOULD prefer the contact that was most recently updated.
>>
>> If the binding is only to {instance-id,reg-id}, then that text is
>> nonsensical: you can't have two identical bindings to the same AOR.
>>
>> I suspect that the text in 6.1 should actually read something more like:
>>
>>    When a reg-id header parameter is present in a Contact header field
>>    value in a REGISTER request, the corresponding binding is between an
>>    AOR and the combination of instance-id, reg-id, and network flow.
>>
>> /a
>>
>>
>> _______________________________________________
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>> This list is for NEW development of the core SIP Protocol
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>>     
>
>
> _______________________________________________
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>
>   


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Erkki.Koivusalo@nokia.com writes:

 > The comments sent on the list during last few days seem to
 > indicate reasonably large concern that it is not sufficient
 > for the UA to check that the configured proxy URI contains
 > keepalive=stun parameter, in order for the UA to make sure
 > that the proxy actually supports outbound. 

erkki,

i don't understand this concern.  your own outbound discovery draft
allows finding out outbound and non-outbound proxies of a domain.  if
proxy is outbound proxy, it of course support stun keepalives.  if not,
it doesn't.

-- juha

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Christer Holmberg (JO/LMF) writes:

 > Does a proxy supporting STUN keep-alive have to be an "outbound
 > proxy" (assuming you mean an edge proxy), as defind in the outbound
 > spec?

i don't know or care.  see subject line.  this thread is about outbound
proxies.

-- juha

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Jeroen van Bemmel writes:

 > No, you cannot shift it off like that. Outbound is introducing binary STUN 
 > messages that could potentially harm existing systems, so outbound is 
 > responsible for providing reasonable safeguards. 

i don't buy that.  stun port can be configured on many UAs.  it is thus
easy to set it to 5060.  proxies should be build to last any non-sip
packets.

-- juha

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The subjects says "Ready for WGLC on Outbound", and currently the usage =
of STUN for keep-alive is defined in that draft. Where do you see the =
word "proxy" in the subject line???

________________________________

From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: Sat 08/07/2006 17:46
To: Christer Holmberg (JO/LMF)
Cc: Jeroen van Bemmel; Cullen Jennings; sip@ietf.org; =
dean.willis@softarmor.com; rohan@ekabal.com
Subject: RE: [Sip] Ready for WGLC on Outbound?



Christer Holmberg (JO/LMF) writes:

 > Does a proxy supporting STUN keep-alive have to be an "outbound
 > proxy" (assuming you mean an edge proxy), as defind in the outbound
 > spec?

i don't know or care.  see subject line.  this thread is about outbound
proxies.

-- juha



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Juha, how do you define an "oubound proxy"?
=20
Is a proxy that supports STUN for keep-alive, but nothing else, an =
outbound proxy?

________________________________

From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: Sat 08/07/2006 17:39
To: Erkki.Koivusalo@nokia.com
Cc: sip@ietf.org; rohan@ekabal.com; adam@nostrum.com; =
BPenfield@acmepacket.com
Subject: RE: [Sip] Outbound: Validating if the proxy supports outbound =
or not



Erkki.Koivusalo@nokia.com writes:

 > The comments sent on the list during last few days seem to
 > indicate reasonably large concern that it is not sufficient
 > for the UA to check that the configured proxy URI contains
 > keepalive=3Dstun parameter, in order for the UA to make sure
 > that the proxy actually supports outbound.

erkki,

i don't understand this concern.  your own outbound discovery draft
allows finding out outbound and non-outbound proxies of a domain.  if
proxy is outbound proxy, it of course support stun keepalives.  if not,
it doesn't.

-- juha

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Adam Roach writes:

 > I'm suggesting a third bullet that reads something like:
 > 
 >     * If multiple bindings have the same instance-id, the proxy MUST NOT
 >       use a binding until all bindings of greater q-value have been tried.

if outbound set has two proxies and UA registers with one using q value
0.5 and with the other using q value 0.6, there would be only one
binding per proxy.  in that case, i hope you are not suggesting that one
proxy asks the other if it has higher q value for the same instance id.

-- juha

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Hadriel Kaplan writes:

 > > in the above situation, assuming that network has two proxies, if UA
 > > decides to establish only one flow instead of two, then the likely hood
 > > is 50% that incoming call to the UA cannot be establishes.  it thus
 > > makes no sense for the UA to violate network policy on number of flows.
 > 
 > That doesn't make any sense to me.  

obviously you didn't notice that in the example there was two proxies,
one flow to each.

-- juha

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Jonathan Rosenberg writes:

 > Finally, let me say that if you have a proxy that crashes on receipt of 
 > binary messages on its sip port, you are in for real trouble. 

fully agree.  let's close this tread.  

-- juha

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Dean Willis writes:

 > What do you mean "require"? Use a proxy-require option tag?

no, we simply say that outbound proxy MUST support stun keepalives.

 > Or just hope it won't melt when you send it a STUN packet, meaning that 
 > you "require the operator to get it right".

 > The problem as I see it is we have no way of easily detecting an error 
 > if somebody mistakenly lists a non-compliant proxy as an outbound proxy 
 > in DHCP or some other discovery mechanism. I hate leaping off a trapeze 
 > without a safety net . . .

no properly implemented proxy melts no matter what you sent to it.  this
kind of arguments are completely bogus.

-- juha

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From: Juha Heinanen <jh@tutpro.com>
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Subject: RE: [Sip] Ready for WGLC on Outbound?
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Christer Holmberg (JO/LMF) writes:

 > The subjects says "Ready for WGLC on Outbound", and currently the
 > usage of STUN for keep-alive is defined in that draft. 

yes, but only in connection with outbound proxies.

-- juha

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From: Juha Heinanen <jh@tutpro.com>
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Subject: RE: [Sip] Outbound: Validating if the proxy supports outbound or not
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Christer Holmberg (JO/LMF) writes:

 > Is a proxy that supports STUN for keep-alive, but nothing else, an
 > outbound proxy?

no. why you ask me?  see section 5 of outbound. but i will give up
this tread now.

-- juha

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Juha Heinanen wrote:
> Adam Roach writes:
>
>  > I'm suggesting a third bullet that reads something like:
>  > 
>  >     * If multiple bindings have the same instance-id, the proxy MUST NOT
>  >       use a binding until all bindings of greater q-value have been tried.
>
> if outbound set has two proxies and UA registers with one using q value
> 0.5 and with the other using q value 0.6, there would be only one
> binding per proxy.  in that case, i hope you are not suggesting that one
> proxy asks the other if it has higher q value for the same instance id.
>   

I'm not adding any new requirements here. The following bullet has
identical implications:

o  If two bindings have the same instance-id and reg-id, the proxy
      SHOULD prefer the contact that was most recently updated.

/a



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From sip-bounces@ietf.org Sat Jul 08 14:03:32 2006
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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Juha,

Hadriel is not the only one who was puzzled by this, so was I. I've been 
trying to follow your reasoning, and I think I now know where this confusion 
comes from

The reason outbound introduces multiple flows is stated in the introduction: 
"[...] Since such detection takes time and leaves a window of opportunity 
for missed incoming requests, this mechanism allows the UA to use multiple 
flows to the proxy or registrar."

I read this as "multiple flows to the *same, single* proxy or registrar". 
When reading this, I think "ah, that makes sense, because NAT devices can be 
unstable and drop flows all the time, and keepalive messages could get 
lost". So I interpret this as protection against failing NAT devices, and 
*not* against failing proxies.

This is why I initially did not understand why things would break if a UA 
sets up only 1 flow, instead of 2. This makes sense only in case of flows to 
*distinct* proxies, ie your 50% scheme below.

What you are doing, is saying: "ah, outbound gives me multiple flows, and 
that is its main benefit (and not NAT traversal)". So you take something 
that is only a side effect of the main intention of the draft, and turn it 
around by assuming that flows go to different proxies. Probably inspired by 
section 3.3 which does this too.

My conclusion is that all this reliability stuff (mainly section 3.3 and 
part of section 3.4) is a *separate issue*, and has nothing to do with the 
core of outbound. It should not be in this draft, because it is confusing 
and - as the text already points out - only one of many ways to provide 
reliability.

Section 3.4 should be rewritten to consider the case of a UA, a *single* 
edge proxy, and a registrar. By including two edge proxies here, two 
different issues are introduced: 1) how does one do outbound in case of an 
edge proxy and 2) how does one deal with multiple edge proxies. Again, 
separate that please!

Regards,

Jeroen

----- Original Message ----- 
From: "Juha Heinanen" <jh@tutpro.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>
Cc: <sip@ietf.org>
Sent: Saturday, July 08, 2006 7:11 PM
Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04


> Hadriel Kaplan writes:
>
> > > in the above situation, assuming that network has two proxies, if UA
> > > decides to establish only one flow instead of two, then the likely 
> > > hood
> > > is 50% that incoming call to the UA cannot be establishes.  it thus
> > > makes no sense for the UA to violate network policy on number of 
> > > flows.
> >
> > That doesn't make any sense to me.
>
> obviously you didn't notice that in the example there was two proxies,
> one flow to each.
>
> -- juha
>
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jeroen,

multiple flows to the same proxy is not interesting.  the only added
value of outbound is multiple flows to more than one proxy.

-- juha

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Dear outbound authors,

Could you share your thoughts on the view Juha is expressing below?

Regards,

Jeroen

----- Original Message ----- 
From: "Juha Heinanen" <jh@tutpro.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Cc: "Hadriel Kaplan" <HKaplan@acmepacket.com>; <sip@ietf.org>
Sent: Saturday, July 08, 2006 8:28 PM
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04


> jeroen,
> 
> multiple flows to the same proxy is not interesting.  the only added
> value of outbound is multiple flows to more than one proxy.
> 
> -- juha

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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Jeroen van Bemmel writes:

 > Could you share your thoughts on the view Juha is expressing below?

jeroen,

are you saying that if outbound-proxy-set consists of two sip uris, it
would be ok if the UA sets up two flows to one of them and no flow to
the other one?

if this is allowed by outbound i-d, then i have wasted my (and other
people's) time on it.

-- juha

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Subject: RE: [Sip] Ready for WGLC on Outbound?
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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>> The subjects says "Ready for WGLC on Outbound", and currently the
>> usage of STUN for keep-alive is defined in that draft.
>yes, but only in connection with outbound proxies.

There is an ongoing discussion related to documenting it as a standalone =
feature, and one of the options IS to document that standalone feature =
in the outbound draft.
=20
I fully agree on the comments saying that a proxy should be able to =
receive STUN without crashing (it may enable some black-list functions =
etc, though, as described by Jeroen). However, I DO think it's a bad =
standard when you send stuff without knowing if the receiver supports it =
or not. In SIP we do use mechanisms (error codes, headers etc) to =
indicate what entities support and don't support, and I don't see why we =
shouldn't be able to do that also in this case. =20
=20
Regards,
=20
Christer
=20

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Subject: STUN keep-alive: Where to document it? (was [Sip] Re: A coment to
	draft-ietf-sip-outbound-04)
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Moving the discussion from the "[Sip] Re: A coment to =
draft-ietf-sip-outbound-04" thread:
=20
Rohan wrote:
=20
>> Also, if a client uses STUN keepalive, but not the rest of outbound
>> (instance-id etc) it would not make sense trying to send multiple
>> registrations, since the registrar would not be able to differentiate
>> the "flows". I think that needs to be clear in the spec, a part of
>> defining the STUN keepalive as a standalone function.
>I actually understand the motivation for this one.  The issue is how
>much of the STUN keepalive mechanism belongs in 3489bis, how much could
>go in outbound in a separate section (or a separate document), and how
>much is specific to outbound.
=20
My proposal would be:
=20
3489bis:
-----------
=20
- We pretty much leave the text as it is. 3489bis is not SIP specific, =
but we could say how it can be used to detect whether the NAT has =
re-booted (checking the XOR-xxx value in the STUN response). I believe =
that is currently said in the outbound spec, but it's not outbound =
specific. I don't have the specs in front of me, so I may be wrong...
=20
=20
Outbound spec:
----------------------
=20
- The outbound spec refers to the SIP STUN keep-alive mechanism (no =
matter if described in the outbound draft or a separate draft), says =
that it MUST be used with outbound.
- That outbound related actions taken if the client detects (from the =
STUN response) that the NAT has re-booted.
=20
=20
Separate document OR separate section in the outbound spec:
-------------------------------------------------------------------------=
----------
=20
- Here we define whatever option-tags and URI parameters are needed for =
the SIP usage of  STUN keep-alive (NOTE: outbound related option-tags =
etc are not defined here)
- Here we define how to use it for SIP, e.g. saying that the STUN =
messages are sent to the same port as the SIP requests
- Here we MAY say something about "generic SIP actions" if the client =
detects NAT re-boot, if there is something we don't think is outbound =
specific (need to think about that).
=20
=20
>I think all the text about what you do when you try to setup a flow and =
it doesn't work (retry after x seconds...) is really very >specific to =
multiple flows.
=20
I agree. "Flow" is an outbound specific term, and has nothing to do with =
STUN keep-alive as such.
=20
>There is the binding request/response that belongs in 3489bis.  What is =
unclear to
>me is what else is left.  Certainly you need some part of a doc that =
defines the keepalive URI parameter and updates 3261 to >say it ok to =
send STUN on the same port as SIP when you have positive knowledge
>(such as the keepalive parameter) that the server is ok with that.
=20
Yes. That is pretty much what I suggested above.
=20
And, IF some new parameter is needed for the proxy to indicate whether =
it supports STUN or not (please let us have that discussion elsewhere - =
not in this thread :), it should also be defined here.=20
=20
But, IF we think that the =
indicate-what-a-proxy-support-and-does-not-support mechanism itself is =
GENERIC, and not only related to STUN (in future there may be other =
extensions where it's needed), that should of course then be described =
in a dedicated document. But, let's have that discussion elsewhere also =
:)
=20
Regards,
=20
Christer




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From sip-bounces@ietf.org Sat Jul 08 17:02:52 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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	"Rohan Mahy" <rohan@ekabal.com>, <sip@ietf.org>
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Subject: Re: STUN keep-alive: Where to document it? (was [Sip] Re: A coment
	todraft-ietf-sip-outbound-04)
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Christer,

And don't forget UA-UA keep-alives (and thus the need for a UAC/UAS to 
indicate support). Once the UA registers using outbound, at one point it's 
going to want to make a call (or subscribe, or whatever). I'd like to avoid 
a situation in which some proxy must always record-route for this to work 
(think eg about SIPSEC)

Also, I've been considering the following: what if we would reverse the 
client/server roles for STUN in outbound? Have the proxy initiate binding 
requests, and the UA answer them. Then the proxy can have more control over 
its load, and there is no risk that a non-outbound proxy gets STUNned. Just 
a thought...

Regards,

Jeroen

----- Original Message ----- 
From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Rohan Mahy" <rohan@ekabal.com>; <sip@ietf.org>
Cc: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>; "Cullen Jennings" 
<fluffy@cisco.com>
Sent: Saturday, July 08, 2006 9:47 PM
Subject: STUN keep-alive: Where to document it? (was [Sip] Re: A coment 
todraft-ietf-sip-outbound-04)


Moving the discussion from the "[Sip] Re: A coment to 
draft-ietf-sip-outbound-04" thread:

Rohan wrote:

>> Also, if a client uses STUN keepalive, but not the rest of outbound
>> (instance-id etc) it would not make sense trying to send multiple
>> registrations, since the registrar would not be able to differentiate
>> the "flows". I think that needs to be clear in the spec, a part of
>> defining the STUN keepalive as a standalone function.
>I actually understand the motivation for this one.  The issue is how
>much of the STUN keepalive mechanism belongs in 3489bis, how much could
>go in outbound in a separate section (or a separate document), and how
>much is specific to outbound.

My proposal would be:

3489bis:
-----------

- We pretty much leave the text as it is. 3489bis is not SIP specific, but 
we could say how it can be used to detect whether the NAT has re-booted 
(checking the XOR-xxx value in the STUN response). I believe that is 
currently said in the outbound spec, but it's not outbound specific. I don't 
have the specs in front of me, so I may be wrong...


Outbound spec:
----------------------

- The outbound spec refers to the SIP STUN keep-alive mechanism (no matter 
if described in the outbound draft or a separate draft), says that it MUST 
be used with outbound.
- That outbound related actions taken if the client detects (from the STUN 
response) that the NAT has re-booted.


Separate document OR separate section in the outbound spec:
-----------------------------------------------------------------------------------

- Here we define whatever option-tags and URI parameters are needed for the 
SIP usage of  STUN keep-alive (NOTE: outbound related option-tags etc are 
not defined here)
- Here we define how to use it for SIP, e.g. saying that the STUN messages 
are sent to the same port as the SIP requests
- Here we MAY say something about "generic SIP actions" if the client 
detects NAT re-boot, if there is something we don't think is outbound 
specific (need to think about that).


>I think all the text about what you do when you try to setup a flow and it 
>doesn't work (retry after x seconds...) is really very >specific to 
>multiple flows.

I agree. "Flow" is an outbound specific term, and has nothing to do with 
STUN keep-alive as such.

>There is the binding request/response that belongs in 3489bis.  What is 
>unclear to
>me is what else is left.  Certainly you need some part of a doc that 
>defines the keepalive URI parameter and updates 3261 to >say it ok to send 
>STUN on the same port as SIP when you have positive knowledge
>(such as the keepalive parameter) that the server is ok with that.

Yes. That is pretty much what I suggested above.

And, IF some new parameter is needed for the proxy to indicate whether it 
supports STUN or not (please let us have that discussion elsewhere - not in 
this thread :), it should also be defined here.

But, IF we think that the indicate-what-a-proxy-support-and-does-not-support 
mechanism itself is GENERIC, and not only related to STUN (in future there 
may be other extensions where it's needed), that should of course then be 
described in a dedicated document. But, let's have that discussion elsewhere 
also :)

Regards,

Christer




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From sip-bounces@ietf.org Sat Jul 08 18:11:44 2006
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I have been watching this ongoing discussion with some bemusement.
It appears that there are two very different views of the purpose for 
the multiple connections specified in 
draft-ietf-sip-outbound-04.txt.  There also seems to be some 
discrepancies about where the multiple connections go.

With regard to purpose, the question I see is whether the reliability 
is for the clients benefit, or the networks.  If it is for the 
client's benefit, then as has been suggested the client should be 
able to choose whether to take advantage of it or not.  The multiple 
connections ought not be MUST strength.  On the other hand, if it is 
a mechanism to enable the network to offer better service 
(particularly, better service to incoming requests) then the network 
would reasonably expect the multiple connections, and so the usage 
should be a MUST.
--Personally, I think that network techniques for providing 
reliability are the network's job / business, and so the multiple 
connections are about the clients desire for reliability, and ought 
to be SHOULD, or even MAY strength.

I believe that the above alternatives lead to confusion about the 
draft.  Further, I think that there is related confusion about the 
usage of the multiple connections.

I note that the wording of the draft is very fuzzy as to whether the 
multiple connections are to the same server, multiple "closely 
coupled" servers, or distinct servers.  I had assumed that the 
connections would be to distinct servers.  However, re-reading the 
draft I see wording like (3.1) "it [proxy] can use any one of the 
flows on which a successful registration has been completed."  For a 
proxy to have several flows would imply that the client connected to 
the same server multiple times.  is in order.  Wording in section 3.2 
also talks about the proxy, which is the registrar, having multiple 
connections to the target.
I suspect that all the wording about a single registrar is to be 
interpreted in conjunction with the example case (3.3) of a single 
registrar that is actually two "closely coupled" hosts.  This is a 
perfectly reasonable deployment.
However, I think we would be much better off describing this as if it 
were two registrars, and indicating that coordination among the 
registrars is outside of the scope of this draft.  And we ought not 
require that they be closely coupled.
Whatever the actual intent of the authors of 3.1, 3.2, and 3.3, we 
need to decide what we want it to mean, and clarify the wording so 
that it says that, clearly.

Yours,
Joel M. Halpern

PS: I believe the reference in section 4.4 to "Section 4.2 to form a 
new flow" is actually a reference to 4.4.2?



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From sip-bounces@ietf.org Sat Jul 08 18:53:50 2006
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>, "Rohan Mahy"
	<rohan@ekabal.com>, sip@ietf.org
Subject: Re: STUN keep-alive: Where to document it? (was [Sip] Re: A coment
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Hi,

I guess we would need an option tag for the UAC/UAS to indicate the =
support.

When it comes to the edge proxy sending the STUN requests, that is probably =
the same thing as the "network initiated keepalive" mechanism I proposed =
while we were still discussing whether we are going to use STUN in the =
first place. I did not propose STUN at that point, but anything that =
triggers a message (a response in the STUN case) to be sent back from the =
client towards the edge proxy (some NATs require keepalive traffic in that =
direction in order to keep the bindings open).

One issue with that proposal was that the client doesn't know if e.g. a NAT =
is rebooting. It would require some timer, so that the client can assume a =
reboot has occured if it doesn't receive a STUN request before the timer =
expires.

Regards,

Christer


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From sip-bounces@ietf.org Sat Jul 08 19:13:43 2006
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Hi Christer,

>One issue with that proposal was that the client doesn't know if e.g. a NAT 
>is rebooting. It would require some timer, so that the client can assume a 
>reboot has >occured if it doesn't receive a STUN request before the timer 
>expires.

Also client-initiated STUN requires such a periodic timer. Furthermore, 
sending a STUN request and then waiting for a response also implicitly uses 
a timer (typical implementation is the sending thread blocking on some event 
with a timeout, reception of the response setting this event)

I just scanned briefly through RFC3489bis section 7.1. It appears that much 
more needs to be said on exactly how STUN is to be done. For example,
RFC3489bis suggests a retransmission scheme with timeout after 9500ms for 
UDP; for TCP it says "SHOULD NOT retransmit", but what should be used for 
timeout? Surely not 9.5 seconds? And compared to SIP standard timings, a 
retransmission after 100ms (UDP) seems rather quick...

Regards,

Jeroen


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From sip-bounces@ietf.org Sat Jul 08 22:01:53 2006
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I generally like the draft, and I think the work is very important.

Anyhow, AFAIK there is no distinct way to request SIP-Identity to
AS, so just because one receives a sip-identity/id-change tag does
not necessary mean that sender of an UPDATE will be able to send
the sip-identity back(Even if the UAS supports sip-identity/id-change,
the domain of the UAS might not have AS in place to provide sip-identity).

This is because there is no option tag to require proxy to add
sip-identity,
and because sip-identity is generally performed at AS, UA is generally
unaware of the domian's support for sip-identity.

Furthermore, because its the AS that adds sip-identity(except when
UA has its own certificate which I think is unlikely), just because UA
receives the sip-identity, does not necessary mean that UA sending
the original INVITE(offerer that you have described in your comment)
with identity-info header supports sip-identity.

Last note, looking at the sentence in 6.1 which is shown below, I am not
sure if
mandating UA to send an UPDATE to update its identity is aligned
with the semantics of Supported header field. I don't really have a strong
opinion on this point, but it felt a little funny to mandate a behavior
with
a presence of "Supported" header.

Thanks & Regards
Shida

Dan Wing wrote:
> I hadn't noticed any followups to this issue.  I have a suggestion:
>
>   
>> -----Original Message-----
>> From: Elwell, John [mailto:john.elwell@siemens.com] 
>> Sent: Tuesday, May 02, 2006 1:21 PM
>> To: sip@ietf.org
>> Subject: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
>>
>> The second issue concerns the fact that an UPDATE request in the
>> backward direction following dialog establishment can be wasted if
>> the peer UA (the UAC for the INVITE transaction) does not support
>> it.  It states:
>>
>>       "OPEN ISSUE: If the identity has has not changed compared with
>>       the URI in the To header field and the UAC (for the INVITE
>>       request) does not support verification of an Identity header,
>>       this UPDATE or re-INVITE transaction would fulfil no useful
>>       purpose.  However, the UAS (for the INVITE transaction) does
>>       not know this.  In order to save this round trip, should we
>>       have an additional option tag to indicate support for receipt
>>       of the Identity header on mid- dialog requests?  Then if this
>>       option tag is not present in the INVITE request the additional
>>       transaction could be omitted.  Note that the id-change option
>>       tag does not seem appropriate for this purpose, since a UA
>>       could support a change of URI without supporting Identity."
>>
>> Whilst I do not wish to proliferate option tags, I do see some
>> benefit in such a tag.  Any opinions?
>>     
>
> If an Invite arrives with both Identity and the id-change tag,
> it indicates the offerer understands both Identity and connected-
> identity, so the answerer should generate an request (UPDATE or 
> re-INVITE).
>
> -d
>
>
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>
>
>   


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From sip-bounces@ietf.org Sat Jul 08 22:11:57 2006
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Subject: RE: [Sip] Proposals for review process to conclude WGLCs
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You can add another vote in support of option 1), as I think it
increases the level of commitment for the reviewers, along with the
document editor/authors.  It might also prove quite useful in terms of
resolving some issues in real-time, thus reducing the amount of
thrashing on specific issues on the list due to misunderstandings via
email.  As you've already identified, a recorder would be imperative for
the process.=20

I also support the idea of SIP doing something different than SIPPING
and at year end we can evaluate both options and see what seems to work
best. I'm beginning to think the SIPPING approach might be much better
applied early in the cycle (i.e, for the first few iterations), but that
the conference calls might be the most effective way to handle the
WGLCs.   =20

Obviously, any of the approaches requires committed and dedicated
reviewers for the documents.  The response in the SIPPING WG has been
pretty good, but I think it would be optimal to add the requirement that
a document isn't accepted as a WG document, unless you also have the
reviewers committed.    This might delay some things, but in my mind, if
we don't have WG members committed to the work, along with the
editor/authors, we don't have the resources to effectively complete the
work (in a timely manner). =20

Mary

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Friday, June 30, 2006 3:26 PM
To: IETF SIP List
Subject: [Sip] Proposals for review process to conclude WGLCs



One of the questions we've struggled with is how to get sufficient =20
review on a document under WGLC to make sure it's really done.


SIPPING is following a new procedure with designated reviewers, doing =20
review by email, and sending stuff back to the. Apparently this =20
suffers from the expected issues, with designated reviewers sometimes =20
not quite meeting their review commitments.

Cisco made me take a peer-review refresher class the other day, so =20
this started me thinking about some alternatives.

In the model taught in this class, each peer review  includes 1) a =20
"moderator" or "reader" who frames the questions and keeps things on =20
track, 2) a "recorder" who records conclusions and issues raised =20
during the review, 3) the author(s), who neither presents no =20
interprets the work, but does respond to specific questions about it, =20
and 4) the reviewers, who hopefully have expertise and have committed =20
time to the review.


I'd like to propose a couple of alternative models to what SPPING has =20
been doing and seeing if either resonates with our group.


1) Real-time review sessions via conference call: When we get ready =20
to do a final review, the chairs assemble a review team. Presumably, =20
this review team is assembled by polling the list for volunteers, and =20
by pushing extra hard for stakeholders and other people with specific =20
expertise. The moderator builds up a checklist and set of basic =20
questions to consider, and distributes it to the review team. At a =20
tiem agreed to by enough of the participants, we have a conference =20
call, not to exceed two hours, and work through the checklist in a =20
moderated fashion. At the conclusion of that two hours, we either =20
reach a positive decision, a negative decision (with list of issues =20
and actions), or decide we need more review.


2) Moderated email-review, using the entire list as the "review =20
team". In this model, the chairs pick a moderator and recorder for =20
the review. The moderator sets up a checklist and sends it to the =20
list, along with an announcement of the review. The moderator then =20
goes through the checklist item-by-item on list, polling for =20
comments. The recorder records any issues or conclusions. Each item =20
might take two or three days for primary comments, but could be =20
revisited during the review process if needed. The author could =20
answer specific questions about the draft, but not paraphrase or =20
interpret.  At the conclusion of the review, the moderator and =20
reviewer would report to the chairs, who would decide whether we have =20
issues, a conclusion, or need further review.


What do y'all think?

--
Dean


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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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From: "Dan Wing" <dwing@cisco.com>
To: <shida@ntt-at.com>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
Date: Sun, 9 Jul 2006 00:27:17 -0400
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> I generally like the draft, and I think the work is very important.

Me, too.

> Anyhow, AFAIK there is no distinct way to request SIP-Identity to
> AS, so just because one receives a sip-identity/id-change tag does
> not necessary mean that sender of an UPDATE will be able to send
> the sip-identity back(Even if the UAS supports sip-identity/id-change,
> the domain of the UAS might not have AS in place to provide
> sip-identity).

Agreed; if we agree we need to optimize that case, we will need
an additional option tag.  But, still, there isn't a way for
a UAS to "know" that his request will be protected with Identity.
So I'm not sure of the value such an additional tag would bring
over my suggestion.

> This is because there is no option tag to require proxy to add
> sip-identity,
> and because sip-identity is generally performed at AS, UA is generally
> unaware of the domian's support for sip-identity.
>
> Furthermore, because its the AS that adds sip-identity(except when
> UA has its own certificate which I think is unlikely), just because UA
> receives the sip-identity, does not necessary mean that UA sending
> the original INVITE(offerer that you have described in your comment)
> with identity-info header supports sip-identity.

Even if the original UA itself doesn't support Identity, one of the
domain's proxies added the Identity signature and it's likely that
proxy may validate incoming requests containing Identity signatures.

-d

> Last note, looking at the sentence in 6.1 which is shown
> below, I am not sure if
> mandating UA to send an UPDATE to update its identity is aligned
> with the semantics of Supported header field. I don't really
> have a strong
> opinion on this point, but it felt a little funny to mandate
> a behavior with
> a presence of "Supported" header.
>
> Thanks & Regards
> Shida
>
> Dan Wing wrote:
> > I hadn't noticed any followups to this issue.  I have a suggestion:
> >
> >  
> >> -----Original Message-----
> >> From: Elwell, John [mailto:john.elwell@siemens.com]
> >> Sent: Tuesday, May 02, 2006 1:21 PM
> >> To: sip@ietf.org
> >> Subject: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
> >>
> >> The second issue concerns the fact that an UPDATE request in the
> >> backward direction following dialog establishment can be wasted if
> >> the peer UA (the UAC for the INVITE transaction) does not support
> >> it.  It states:
> >>
> >>       "OPEN ISSUE: If the identity has has not changed
> compared with
> >>       the URI in the To header field and the UAC (for the INVITE
> >>       request) does not support verification of an Identity header,
> >>       this UPDATE or re-INVITE transaction would fulfil no useful
> >>       purpose.  However, the UAS (for the INVITE transaction) does
> >>       not know this.  In order to save this round trip, should we
> >>       have an additional option tag to indicate support for receipt
> >>       of the Identity header on mid- dialog requests?  Then if this
> >>       option tag is not present in the INVITE request the
> additional
> >>       transaction could be omitted.  Note that the id-change option
> >>       tag does not seem appropriate for this purpose, since a UA
> >>       could support a change of URI without supporting Identity."
> >>
> >> Whilst I do not wish to proliferate option tags, I do see some
> >> benefit in such a tag.  Any opinions?
> >>    
> >
> > If an Invite arrives with both Identity and the id-change tag,
> > it indicates the offerer understands both Identity and connected-
> > identity, so the answerer should generate an request (UPDATE or
> > re-INVITE).
> >
> > -d
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >   

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At 09:34 AM 6/30/2006 +0200, Hannes Tschofenig wrote:
>Hi James
>
>thanks for your quick response and your clarifications.

I'm slower this time, sorry

>Please find my response below:
>
>
>James M. Polk wrote:
>>At 11:03 PM 6/29/2006 +0200, Hannes Tschofenig wrote:
>>
>>>Hi Brian,
>>>Hi James
>>>
>>>I have a few questions concerning the latest SIP location conveyance 
>>>document.
>>
>>I moved this discussion to the SIP list, as that's the home of this ID.
>>
>>>* Usage Cases
>>>
>>>I think we need to consider the following scenarios:
>>>
>>>(a) Location transport per-value (a1) and per-reference (a2)
>>>
>>>(b) Location transport in the header (b1) and in the body (b2)
>>>
>>>(a) and (b) has to be combined to produce a reasonable combination.
>>>
>>>Let me go through the individual cases:
>>>
>>>- If SIP UA Alice wants to convey location information to SIP UA Bob 
>>>then it should put the PIDF-LO into the body (and apply S/MIME to it). 
>>>(Case a1+b2)
>>
>>this is covered
>>
>>>- If a proxy wants to add location information to a SIP message then it 
>>>creates the PIDF-LO and a reference to it. The reference is placed into 
>>>the SIP header. (Case a2+b1)
>>
>>this is covered
>>
>>>Would the proxy use M2M/M2E security?
>>
>>this is orthogonal
>
>I thought that this issue is important and would deserve a discussion in 
>the document.
>I could only find discussions about S/MIME usage for end-to-end security.

so far, a UAC shouldn't use S/MIME on a PIDF-LO message body if it knows a 
proxy will need to read that message body to route the message.  911/112 
calling is a good example of this, that can be coded in a phone to *not* 
use S/MIME (or any message body encryption to hide a PIDF-LO, knowing that 
info will be needed in transit).

Now, there has been a suggestion that every time a UAC sends location, it 
uses S/MIME (meaning location is always sent by-value from an endpoint). 
That only when the UAC receives an error to that request will it consider 
sending a subsequent request without S/MIME.  This will cause an extra 
round-trip to have the UAC send the request without S/MIME for the Proxy to 
be able to read the necessary location information to route the request to 
the appropriate destination.

What do you think of this language in the doc?



>>>- If SIP UA Alice has to perform location-based routing (e.g., as used 
>>>in emergency scenarios) then it has to place something into the SIP 
>>>header since the proxy is not supposed to inspect the body (particularly 
>>>since it can be encrypted).
>>
>>not true
>>Nothing in SIP prevents a proxy from inspecting a body.  A proxy just 
>>cannot add or modify a body.
>>
>>>What is SIP UA Alice supposed todo?
>>
>>Alice can include location by-value or by-reference.
>>
>>>I think it is not realistic to assume that the SIP UA creates a 
>>>reference and makes it available by running a webserver on the end host.
>>
>>I tend to agree, therefore I believe Alice includes location by-value if 
>>she has it
>>
>>>It
>>
>>what is "it" in this context?
>>
>>>cannot add location information to the SIP header (as a base64 encoded 
>>>PIDF-LO) since this functionality is not provided by the draft.
>
>It=the SIP proxy
>
>The SIP proxy cannot add location information to the SIP header (as a 
>base64 encoded PIDF-LO) since this functionality is not provided by the draft.

This is not true.  see this table that's been in the doc for some time:

    The following table extends the values in Table 2&3 of RFC3261
    [RFC3261].

       Header field             where proxy INV ACK CAN BYE REG OPT PRA
       ----------------------------------------------------------------
       Location                 Rr    ar     o   -   -   o   o   o   -

       Header field             where proxy SUB NOT UPD MSG REF INF PUB
       ----------------------------------------------------------------
       Location                 Rr    ar     -   -   o   o   o   o   -

adding a header value to a SIP message is standard procedure for SIP 
proxies. The above (unlabelled) table grants this ability explicitly to 
Location header usage.



>>I think you're operating under the impression that proxies cannot read 
>>bodies, which they can.  Any B2BUA or SBC can modify bodies, or add 
>>bodies - but we cannot assume there will be one in the message path.
>
>The SIP proxy cannot read the body if it is encrypted.

fair

If we are to have an error response sent from a proxy that wants to read 
the PIDF-LO message body (it first has to assume there is one in the 
message), we can use the existing 424 (Bad Location Information) response 
message, and add a RFC 3326 Reason header with a 493 (Undecipherable) 
indication (the 493 is defined in RFC 3261). This could explicitly state to 
the UAC that a proxy cannot read the assumed PIDF-LO by-value message body, 
and to send a new request without encrypting that message body.

Is this desired, and do we want to have this text in this document, or 
ECRIT's phoneBCP (and maybe the Framework)?



>>>This would be case a1+b1.
>>>
>>>- What happens if the SIP UA Alice only has a location reference and she 
>>>wants to send it to SIP UA Bob (where intermediaries must not process 
>>>the location information)? Is she able to place the reference into the 
>>>body of the message? I don't think you cover this option in your draft.
>>
>>SIP Headers can be placed in a message body already.  Do you want that 
>>specified here too?
>
>Ok. That's fine.
>
>>
>>>This would be case a2+b2.
>>>
>>>- What happens if you want to perform location-based routing (i.e., 
>>>processing of the PIDF-LO by intermediaries) and conveyance to the other 
>>>end point?
>>
>>by-value covers this I believe
>
>I suspect that you want to put the location header into the message to let 
>the proxy know where to find the PIDF-LO and that it actually has to look 
>for one.

most likely


>If you don't want intermediaries to process it then you want to apply 
>S/MIME and if not available (=quite likely) then you wouldn't put the 
>location header into the message. Right?

right now, this spec states if location is conveyed in a message, that 
message MUST have a Location header (with a cid to the message body part, 
or a URI for the by-reference).



>>>Do you include the information only in the header and not in the body or 
>>>in the header and the body?
>>
>>either a header or body
>>The proxies can read and route based on the body, and
>>Bob can dereference a header if that's all that's in the message.
>>
>>
>>>* I think that the following text is already covered in the PIDF-LO 
>>>profile draft:
>>>
>>>"
>>>    More than one location representation or format MAY be included in
>>>    the same message body part, but all MUST point at the same position
>>>    on the earth (altitude not withstanding), as this would confuse the
>>>    recipient by pointing at more than one position within the same
>>>    message body part.  There MAY be a case in which part or parts of
>>>    one location format and part or parts of another format exist in the
>>>    same message body part.  These complementary pieces of information
>>>    MUST point at the same position on the earth, yet are incomplete
>>>    within their own format. For example, there maybe be a latitude and
>>>    longitude in coordinate format and a civic altitude value to
>>>    complete a 3-dimensional position of a thing (i.e. which floor of a
>>>    building the UA is on in a building at a particular lat/long
>>>    coordinates pair).
>>>"
>>
>>This needs to be in the "Using Protocol's" document too IMO.  I do not 
>>believe this is inconsistent with the PIDF-LO profile doc.
>
>I think the PIDF-LO profile document is more precise on these issues 
>and  repeating it does not add value.

but this doc doesn't reference the PIDF-LO profile doc, so SIP folks will 
not necessarily find it anywhere else.





>>>* PIDF-LO in SIP Header
>>>
>>>You write:
>>>"
>>>    The amount
>>>    of information that is necessary to appropriately transmit location
>>>    information in a format that is understandable is larger than a SIP
>>>    header could realistically include.
>>>"
>>>
>>>What is the upper limit?
>>
>>I do not know, but I do not believe it is 27 CATypes as separate fields 
>>in a Location header.  I think Henning worked out a 500 byte 
>><location-info> element, and while that's allowable I believe, I think 
>>it's hard to read.
>
>Why is it a problem if you assume that the SIP proxies will inspect the 
>body? He has to read the same XML document.

for one, it makes reading the SIP message harder to parse with a massive 
header in the middle of it (that currently only can be a message body).



>>>* Scenario Description
>>>
>>>In the introduction you list three scenarios (actually four).
>>>
>>>I assume that Scenario #1 refers to the scenario in Figure 2 of Section 
>>>4 where the end point performs location-based routing. Correct?
>>
>>agreed
>>
>>>Thinking about the service URN draft one could get the impression that a 
>>>solution scenario according to Scenario #3B could also be designed to 
>>>accomplish the goal of Scenario #1.
>>
>>yes, but I choose to call them out separately to have a reader understand 
>>the different scenarios
>
>Ok. That's fine.
>
>>
>>>I might be good to combine the scenario description of Section 1 with 
>>>the text and the figures in Section 4.
>>
>>I was told explicitly to reduce my figures in section 4.  Therefore, I 
>>have a basic request/response transaction (figure 1), a request/response 
>>transaction involving an intermediary performing routing based on the 
>>location contents of the request (figure 2) and then articulate how 
>>Location can be conveyed in a specific SIP Request type (INVITE here).
>>I have been told by the WG chairs in SIP to not have more than this.
>>This doesn't prevent a user-cases ID from being generated giving these 
>>examples.  I think that would be appropriate once this ID reaches RFC status.
>
>I am fine with the number of message flows but Section 1 talks about 
>scenarios without figures and later you talk again about scenarios with 
>figure and there is no link between them.

ok. I need to link the two somehow.


>Ciao
>Hannes
>
>>
>>
>>>Ciao
>>>Hannes
>>>
>>>
>>>_______________________________________________
>>>Geopriv mailing list
>>>Geopriv@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>

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From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Bell Labs Security Technology Research Group
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To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [Sip] comments on draft-gurbani-sip-domain-certs-01
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Jonathan Rosenberg wrote:
> Vijay,
> 
> A few comments on the draft.

Jonathan: Thanks for reviewing the draft.  More inline.

> Firstly, I think the draft mixes authentication and authorization. 
> You spend a bunch of time talking about checks that a server makes 
> when receiving a request from a client, comparing bits of the SIP 
> message to the TLS certificate. To me, all of those checks are 
> authorization, not authentication. TLS mutual auth has done what its
>  name suggests - it has already authenticated the other side.

To me, the term "mutual authentication" implies that the
client authenticates the server, and the server authenticates
the client.  As it exists today, TLS mutual authentication
in SIP does only the former, not the latter.

> Now you are trying to figure out whether they are authorized to send
> you a SIP request. I think these are quite separable.

They may be separable, but rfc3261 conflates these.  When
the server asks a certificate of a client, rfc3261 expects
the server to match the domain in the certificate to the
"domainname" portion of the From header.  I am trying to
tease this out and better specify it from an implementation
point of view.  Due to retargetting, using the From is
pointless.  R-R is optional, so that only leaves the Via
header.  However, the identity of the client in the Via header
is usually in the form of a FQDN, whereas the certificate
from the client will normally only contain a domain.

> For example, if my proxy has a policy about only connecting to 
> specific other domains, then it can take the identities determined 
> from TLS mutual auth and use those to decide whether to accept 
> requests coming from that proxy. No need to every look at the via, 
> record-route or whatever.

But what do you do absent such a policy?  A proxy put up on
the Internet by a service provider cannot be expected to know
all the domainsthat will send it a request.  Thus, in the worst
case, it will do minimal authentication on the client certificate:
probably the CA check and the expiry check.  If that iFrom sip-bounces@ietf.org Sun Jul 09 01:46:03 2006
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Date: Sun, 09 Jul 2006 00:44:06 -0500
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Bell Labs Security Technology Research Group
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Jonathan Rosenberg wrote:
> Vijay,
> 
> A few comments on the draft.

Jonathan: Thanks for reviewing the draft.  More inline.

> Firstly, I think the draft mixes authentication and authorization. 
> You spend a bunch of time talking about checks that a server makes 
> when receiving a request from a client, comparing bits of the SIP 
> message to the TLS certificate. To me, all of those checks are 
> authorization, not authentication. TLS mutual auth has done what its
>  name suggests - it has already authenticated the other side.

To me, the term "mutual authentication" implies that the
client authenticates the server, and the server authenticates
the client.  As it exists today, TLS mutual authentication
in SIP does only the former, not the latter.

> Now you are trying to figure out whether they are authorized to send
> you a SIP request. I think these are quite separable.

They may be separable, but rfc3261 conflates these.  When
the server asks a certificate of a client, rfc3261 expects
the server to match the domain in the certificate to the
"domainname" portion of the From header.  I am trying to
tease this out and better specify it from an implementation
point of view.  Due to retargetting, using the From is
pointless.  R-R is optional, so that only leaves the Via
header.  However, the identity of the client in the Via header
is usually in the form of a FQDN, whereas the certificate
from the client will normally only contain a domain.

> For example, if my proxy has a policy about only connecting to 
> specific other domains, then it can take the identities determined 
> from TLS mutual auth and use those to decide whether to accept 
> requests coming from that proxy. No need to every look at the via, 
> record-route or whatever.

But what do you do absent such a policy?  A proxy put up on
the Internet by a service provider cannot be expected to know
all the domainsthat will send it a request.  Thus, in the worst
case, it will do minimal authentication on the client certificate:
probably the CA check and the expiry check.  If that is the expected
behavior, then we ought to proscribe it in a form that
implementors can understand and code to.  Today, no such text
or guideline exist.

> Thinking about this some more, I think connect-reue has got the 
> validation model wrong. [...] This validation model has the desired 
> property; a request gets delivered in the reverse direction with 
> connect-reuse only if it would have been delivered had connect-reuse 
> not been used.

Thanks; this'll be added to connect-reuse.

> This simplifies the work that domain-certs has to do. It merely
> needs to give guidance on how the server certificate should look 
> like, addressing the two main things EKR mentioned:
> 
> 1. what field does the identity get coded into
> 2. what does that identity represent - the specific host or the
> domain
> 
> My two cents is, I think it should always just be the domain 
> (example.com and not proxy2.example.com). Thus if you connect to 
> proxy2.example.com, and the server provided a certificate for 
> example.com, that would be OK. 

This works fine for majority of the cases where the client
authenticates the server, i.e., client has a R-URI of
"sips:alice@example.com" and the server certificate asserts
an SAN identity of "DNS:example.com".  The corner case where
there is a problem is the case where the client follows a R-URI of
"sips:alice@downtown.example.com" and gets a certificate
saying "DNS:example.com".  Is downtown a subdomain?  If so,
is example.com authorized to present a certificate for
downtown.example.com?  Or is downtown a host?  In which
case, it would be trivial for the certificate to have an
additional DNS name matching that host.

The same problem occurs when a client follows a Route
header (sips:proxy2.example.com) and gets a certificate
saying "DNS:example.com".  Is proxy2 a host in example.com?
Or is it a subordinate domain of example.com?

> I also think its highly unlikely that a cerificate authority would 
> hand out certs that verify that a *host* is proxy2.example.com. How 
> would a CA verify that this is true?

The CA will not verify this; I would think it would be upto
the holder of the example.com domain to ensure the integrity
of any additional identities stored in the certificate.

I am curious: my understanding is that a certificate for a
virtual servers may contain multiple domain names in it
(c.f. Identity, Section 13.4: "certificates have varying
ways of describing their subjects, and may indeed have
multiple subjects, especially in 'virtual hosting' cases
where multiple domains are managed by a single application.").
If so, how do CAs do validation of these individual domain
names?

> Hope this makes sense.

Yes, a lot of good comments driving some discussion above...

Thanks.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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	ihmail.ih.lucent.com (8.11.7p1+Sun/EMS-1.5 ss the expected
behavior, then we ought to proscribe it in a form that
implementors can understand and code to.  Today, no such text
or guideline exist.

> Thinking about this some more, I think connect-reue has got the 
> validation model wrong. [...] This validation model has the desired 
> property; a request gets delivered in the reverse direction with 
> connect-reuse only if it would have been delivered had connect-reuse 
> not been used.

Thanks; this'll be added to connect-reuse.

> This simplifies the work that domain-certs has to do. It merely
> needs to give guidance on how the server certificate should look 
> like, addressing the two main things EKR mentioned:
> 
> 1. what field does the identity get coded into
> 2. what does that identity represent - the specific host or the
> domain
> 
> My two cents is, I think it should always just be the domain 
> (example.com and not proxy2.example.com). Thus if you connect to 
> proxy2.example.com, and the server provided a certificate for 
> example.com, that would be OK. 

This works fine for majority of the cases where the client
authenticates the server, i.e., client has a R-URI of
"sips:alice@example.com" and the server certificate asserts
an SAN identity of "DNS:example.com".  The corner case where
there is a problem is the case where the client follows a R-URI of
"sips:alice@downtown.example.com" and gets a certificate
saying "DNS:example.com".  Is downtown a subdomain?  If so,
is example.com authorized to present a certificate for
downtown.example.com?  Or is downtown a host?  In which
case, it would be trivial for the certificate to have an
additional DNS name matching that host.

The same problem occurs when a client follows a Route
header (sips:proxy2.example.com) and gets a certificate
saying "DNS:example.com".  Is proxy2 a host in example.com?
Or is it a subordinate domain of example.com?

> I also think its highly unlikely that a cerificate authority would 
> hand out certs that verify that a *host* is proxy2.example.com. How 
> would a CA verify that this is true?

The CA will not verify this; I would think it would be upto
the holder of the example.com domain to ensure the integrity
of any additional identities stored in the certificate.

I am curious: my understanding is that a certificate for a
virtual servers may contain multiple domain names in it
(c.f. Identity, Section 13.4: "certificates have varying
ways of describing their subjects, and may indeed have
multiple subjects, especially in 'virtual hosting' cases
where multiple domains are managed by a single application.").
If so, how do CAs do validation of these individual domain
names?

> Hope this makes sense.

Yes, a lot of good comments driving some discussion above...

Thanks.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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To: oran@cisco.com
Subject: Re: [Sip] Comments on draft-gurbani-sip-domain-certs-01
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	<44AC1A8E.4000707@lucent.com>
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David R Oran wrote:
> Quick question:
> 
>> on the same outcome you mention above; i.e., it suffices
>> that when contacting "sips:alice@atlanta.com", the certificate
>> presented contains "atlanta.com."
>>
> Do you mean "the name in the certificate contains the string" or  "ends 
> with the string" or something else. In other words, does a  certificate 
> for badboy.atlanta.com.chaoscomputerclub.org match a sip  uri of 
> sip:alice@atlanta.com.
> 
> One would think not...

I would hope so as well.

> Or do you mean "the name in the certificate is precisely the string  
> 'atlanta.com'", in which case does the certificate match sip uri's  
> registered in a subdomain, e.g. alice@downtown.atlanta.com.

Yes, exactly!  You hit the nail right on the head, Dave.

What should an implementor of a client expect when a
R-URI contains sips:alice@downtown.atlanta.com and
the certificate that is received contains "atlanta.com"?
It could, of course, simply close the connection.  But then
again, "downtown" could have been a certain host in the
atlanta.com domain versus a subordinate domain, in which
case, the client may have torn the connection down prematurely.

Are the guidelines that implementors must match exactly
the domainname in the cerificate to the domainname in
the R-URI?  Anything else is considered suspect?

If so, then how do we handle this case of a mid-dialog
request:

     INVITE sips:alice@alicehost.atlanta.com SIP/2.0
     Route: <sips:proxy1.atlanta.com;lr>
     ...

The proxy (proxy1.atlanta.com) presents a certificate that
asserts "atlanta.com".  What should the UAC do now?
Close the connection?

I think any guidance we provide around these issues is
well worth it if we are to have a semblance of security
in the stacks.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





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Subject: Re: [Sip] Comments on draft-gurbani-sip-domain-certs-01
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David R Oran wrote:
> Quick question:
> 
>> on the same outcome you mention above; i.e., it suffices
>> that when contacting "sips:alice@atlanta.com", the certificate
>> presented contains "atlanta.com."
>>
> Do you mean "the name in the certificate contains the string" or  "ends 
> with the string" or something else. In other words, does a  certificate 
> for badboy.atlanta.com.chaoscomputerclub.org match a sip  uri of 
> sip:alice@atlanta.com.
> 
> One would think not...

I would hope so as well.

> Or do you mean "the name in the certificate is precisely the string  
> 'atlanta.com'", in which case does the certificate match sip uri's  
> registered in a subdomain, e.g. alice@downtown.atlanta.com.

Yes, exactly!  You hit the nail right on the head, Dave.

What should an implementor of a client expect when a
R-URI contains sips:alice@downtown.atlanta.com and
the certificate that is received contains "atlanta.com"?
It could, of course, simply close the connection.  But then
again, "downtown" could have been a certain host in the
atlanta.com domain versus a subordinate domain, in which
case, the client may have torn the connection down prematurely.

Are the guidelines that implementors must match exactly
the domainname in the cerificate to the domainname in
the R-URI?  Anything else is considered suspect?

If so, then how do we handle this case of a mid-dialog
request:

     INVITE sips:alice@alicehost.atlanta.com SIP/2.0
     Route: <sips:proxy1.atlanta.com;lr>
     ...

The proxy (proxy1.atlanta.com) presents a certificate that
asserts "atlanta.com".  What should the UAC do now?
Close the connection?

I think any guidance we provide around these issues is
well worth it if we are to have a semblance of security
in the stacks.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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From sip-bounces@ietf.org Sun Jul 09 02:36:47 2006
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>, "Rohan Mahy"
	<rohan@ekabal.com>, sip@ietf.org
Subject: Re: STUN keep-alive: Where to document it? (was [Sip] Re: A coment
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Hi Jeroen,

Could we move this discussion to a separate thread? I would like this =
thread to simply focus on the where-will-we-document-things issue.

Regards,

Christer


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Moved from the "RE: STUN keep-alive: Where to document it? (was [Sip] =
Re: A coment  todraft-ietf-sip-outbound-04)" thread:
=20
Jeroen wrote:
=20
>I just scanned briefly through RFC3489bis section 7.1. It appears that =
much
>more needs to be said on exactly how STUN is to be done. For example,
>RFC3489bis suggests a retransmission scheme with timeout after 9500ms =
for
>UDP; for TCP it says "SHOULD NOT retransmit", but what should be used =
for
>timeout? Surely not 9.5 seconds? And compared to SIP standard timings, =
a
>retransmission after 100ms (UDP) seems rather quick...=20

I agree that we probably need to document that. I found the same issue =
with RTSP at one point - there were no =
transaction/re-transmission/timeout timers etc defined (I haven't been =
working with RTSP for a while now, so I don't know what the situation is =
at the moment).

The question is whether we want "SIP specific" behavior, in which case =
it should be in the SIP STUN keep-alive document (or dedicated chapter =
in the outbound spec), or if we want generic behavior in which case it =
should be in 3489bis.

(...OR, if we want outbound specific behavior, in which case it should =
be in the outbound spec).

Regards,
=20
Christer

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Moved from the "STUN keep-alive: Where to document it? (was [Sip] Re: A =
coment  todraft-ietf-sip-outbound-04)" thread:

Jeroen wrote:


>>One issue with that proposal was that the client doesn't know if e.g. =
a NAT
>>is rebooting. It would require some timer, so that the client can =
assume a
>>reboot has >occured if it doesn't receive a STUN request before the =
timer
>>expires.

>Also client-initiated STUN requires such a periodic timer. Furthermore,
>sending a STUN request and then waiting for a response also implicitly =
uses
>a timer (typical implementation is the sending thread blocking on some =
event
>with a timeout, reception of the response setting this event)

Correct. But, these timers a "local" client timers aren't they, i.e. how =
often it sends STUN requests and for how long it should wait for a =
reply?=20

In case the server generates the STUN request the client must know which =
timers the server uses. That can of course be a standardized value, OR =
it can be provided in some header, but the client needs to know about =
it.

Regards,

Christer

=20


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Rohan Mahy wrote:
> 
> The spec currently says that UAs MUST support a minimum of 2 flows.  If 
> they don't open the two flows, the mechanism is broken since the UA can 
> lose inbound messages for several minutes.

I think I have a different opinion on whether the mechanism is broken or 
not. The mechanism is not broken, but obviously the server can loose 
inbound connections. This scenario, although not perfect, may be much 
desirable than draining battery of portable devices at a fast speed.

/Miguel

> 
> thanks,
> -rohan
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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I think the system is "broken" in the sense that it does not behave as the 
authors of the draft had intended
There are 2 questions here:
1) Is that intention expressed sufficiently clear and consistent in the 
draft?
2) Is the intended behavior (and associated system architecture) 
appropriate/desirable (in all/most cases)?

The draft seems to be written under the assumption that "reliability" (not 
well defined yet what that means) can only be provided through multiple 
flows. However, "outbound" (ie enabling UAs behind NAT to register and 
receive incoming requests) can be done using only a single flow, albeit with 
a somewhat higher chance at missing incoming requests.

If the system is architected as Juha was describing (2 independent proxies, 
UA missing 50% of its calls when only setting up 1 flow), the chance at 
missed requests for such a UA is much higher than double that of a single 
flow system. I feel that outbound should certainly not mandate such an 
architecture.

Currently in SIP the assumption is that a single registration suffices to 
receive all incoming requests for a given AoR. If we break that, it means 
that current UAs won't be able to register with "outbound-enhanced" 
registrars and edge proxies. That to me is not acceptable.

In other words, a requirement is missing: "Support UAs that do not implement 
this extension"

Regards,

jeroen

----- Original Message ----- 
From: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>
To: "ext Rohan Mahy" <rohan@ekabal.com>
Cc: "Cullen Jennings" <fluffy@cisco.com>; <sip@ietf.org>
Sent: Sunday, July 09, 2006 1:25 PM
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04


> Rohan Mahy wrote:
>>
>> The spec currently says that UAs MUST support a minimum of 2 flows.  If 
>> they don't open the two flows, the mechanism is broken since the UA can 
>> lose inbound messages for several minutes.
>
> I think I have a different opinion on whether the mechanism is broken or 
> not. The mechanism is not broken, but obviously the server can loose 
> inbound connections. This scenario, although not perfect, may be much 
> desirable than draining battery of portable devices at a fast speed.
>
> /Miguel
>
>>
>> thanks,
>> -rohan
>>
>
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


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From: Juha Heinanen <jh@tutpro.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: [Sip] an alternative approach for outbound proxy discovery
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i liked jonathan's SIP based discovery mechanism and didn't see any
problems with it.

 > 1. I am NOT proposing that this be added to sip-outbound, and I'm not 
 > suggesting that a normative reference to this or another mechanism 
 > should be in outbound. 

however, i can't agree with the above statement, because i don't see any
difference with outbound discovery and rfc3263 discovery referenced in
rfc3261.  it would have been impossible to set up any vendor independent
sip services unless rfc3263 had existed and mandated by rfc3261.  or
perhaps people only want closed, vendor specific outbound
implementations?

-- juha

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Issue 4 in draft-ietf-sip-connected-identity-00.txt
To: Dan Wing <dwing@cisco.com>, sip@ietf.org
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Dan,

Yes, I agree with your motivation for being able to have signed SDP in the
reverse direction. This doesn't necessarily mean UPDATE has to be used -
re-INVITE would do, but I prefer to use UPDATE unless there are other
reasons for using re-INVITE.

The question is whether we should mandate including an SDP offer in the
UPDATE so that we always have the benefit of this signature.

John 

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: 08 July 2006 02:21
> To: Elwell, John; sip@ietf.org
> Subject: RE: [Sip] Issue 4 in draft-ietf-sip-connected-identity-00.txt
> 
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com] 
> > Sent: Tuesday, May 02, 2006 1:24 PM
> > To: sip@ietf.org
> > Subject: [Sip] Issue 4 in draft-ietf-sip-connected-identity-00.txt
> > 
> > The fourth issue is not actually in draft-00 but was raised
> > subsequently.  If a UA supports a change of From header field URI
> > mid-dialog, as indicated by the "id-change" option tag, we could
> > mandate that it also support the SIP UPDATE method.  Then the peer
> > UA in a dialog would be free to use the UPDATE method to report
> > connected identity, rather than the less suitable re-INVITE method.
> > The use of UPDATE means 2 messages instead of 3 and avoids the need
> > for an offer-answer exchange.  I heard no objections when this was
> > proposed a few days ago.  Do we have consensus?
> 
> I prefer UPDATE.  I am considering the needs of some proposed 
> SRTP keying techniques which could benefit from strong authentication
> of identity and of the fingerprints or certificates in SDP.  For
> those cases it's helpful to include SDP in the UPDATE so that it
> gets signed with Identity, even if that SDP is exactly the same as
> the SDP in the answer.  An example of where Identity could help is 
> with a pure DH mode (such as draft-baugher-mmusic-sdp-dh).  It can 
> also help provide additional protection of a UA's self-signed 
> certificate (various MIKEY modes) or certificate fingerprint 
> (mmusic-comedia-tls, mcgrew-tls-srtp).  
> 
> Even without SRTP and without comedia-tls, the reasons that 
> Identity chose to include the offerer's SDP in its signature 
> should be applicable to the answerer's SDP.  Otherwise we're
> saying the answerer's SDP doesn't need authentication or
> integrity protection.
> 
> -d
> 

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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Sat, 8 Jul 2006 17:16:26 -0700
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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On Jul 7, 2006, at 10:47, Jeroen van Bemmel wrote:
> Rohan,
>
>>> Additionally I suspect that users may still impose local policies in 
>>> the UAs as for the number of established flows, independently of the 
>>> number of learned URIs. So, the MUST in the spec is always subject 
>>> to local policy.
>>
>> The spec currently says that UAs MUST support a minimum of 2 flows.  
>> If they don't open the two flows, the mechanism is broken since the 
>> UA can lose inbound messages for several minutes.
>
> OK, let me try to see if I understand where this "several minutes" 
> comes from, and if we could quantify "broken" here.
>
> Assuming that the client uses the default keepalive timer values (UDP: 
> 24..29 seconds, TCP: 95..120 seconds), I deduce that worst case a 
> client using only one flow could loose its registration status for a 
> period of 120 seconds (on average, this would be only 60 seconds). So 
> your "several" equals 2 ?

Yes, you are right. The UA could lose inbound messages for 2 minutes  I 
went back and rechecked all the timers.  I was still (wrongly) thinking 
about the problem based on timer values that resulted in a ~10 minute 
window of potential downtime.  While 10 minutes was not acceptable to 
me, I agree that 2 minutes is in a gray area that will be fine for some 
and not fine for others.

So we are back to a pure semantic question.  Does the service get to 
decide how many flows the UA opens (up to an implementation supported 
maximum), does the UA decide how many flows to open, or is there some 
negotiation?   I still think the current semantics are fine, because 
they allow a single flow to a URI that can still load-share and 
failover to a large number of actual servers.  Of course there are 
others who don't like that because it relies on DNS for failover.

thanks,
-rohan


> One could probably calculate the chance that in this window a request 
> would come in (given message rate assumptions, etc). Multiply that 
> with the chance at flow failure (given component MTBF figures etc), 
> and you would have the kind of reliability one would have for one flow 
> only. Call that reliability R
>
> My question now is: what are reasonable assumptions for these figures, 
> and could it be that for some clients, R could be sufficient?
> Intuitively I say: yes, that's possible. There exist clients and users 
> that would be happy with a scheme that is broken for (1-R) part (I 
> know at least one - me).
>
> There is currently nothing in the requirements regarding reliability. 
> Section 4.4 mentions the relationship between timer values and server 
> load, and has some text on "the expected time that a user may not 
> receive inbound communications"; but not in relation to the number of 
> flows (nor does it mention to de-synchronize keepalive timers across 
> multiple flows). Do we need that, perhaps with some sample 
> quantitative analysis?
>
> Regards,
>
> Jeroen


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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
Date: Sat, 8 Jul 2006 17:25:32 -0700
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Hi,

I do not understand the motivation for wanting multiple flows to the 
same actual server.

At some point we need to ask ourselves as a group whether we are a 
building a mechanism that solves a problem in every conceivable 
topology we can come up with , or whether it is sufficient for us to 
build a mechanism that solves a variety of important problems using one 
or two plausible topologies.

thanks,
-rohan


On Jul 8, 2006, at 11:43, Jeroen van Bemmel wrote:
> Dear outbound authors,
>
> Could you share your thoughts on the view Juha is expressing below?
>
> Regards,
>
> Jeroen
>
> ----- Original Message ----- From: "Juha Heinanen" <jh@tutpro.com>
> To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
> Cc: "Hadriel Kaplan" <HKaplan@acmepacket.com>; <sip@ietf.org>
> Sent: Saturday, July 08, 2006 8:28 PM
> Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
>
>
>> jeroen,
>> multiple flows to the same proxy is not interesting.  the only added
>> value of outbound is multiple flows to more than one proxy.
>> -- juha


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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Sat, 8 Jul 2006 17:16:26 -0700
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On Jul 7, 2006, at 10:47, Jeroen van Bemmel wrote:
> Rohan,
>
>>> Additionally I suspect that users may still impose local policies in 
>>> the UAs as for the number of established flows, independently of the 
>>> number of learned URIs. So, the MUST in the spec is always subject 
>>> to local policy.
>>
>> The spec currently says that UAs MUST support a minimum of 2 flows.  
>> If they don't open the two flows, the mechanism is broken since the 
>> UA can lose inbound messages for several minutes.
>
> OK, let me try to see if I understand where this "several minutes" 
> comes from, and if we could quantify "broken" here.
>
> Assuming that the client uses the default keepalive timer values (UDP: 
> 24..29 seconds, TCP: 95..120 seconds), I deduce that worst case a 
> client using only one flow could loose its registration status for a 
> period of 120 seconds (on average, this would be only 60 seconds). So 
> your "several" equals 2 ?

Yes, you are right. The UA could lose inbound messages for 2 minutes  I 
went back and rechecked all the timers.  I was still (wrongly) thinking 
about the problem based on timer values that resulted in a ~10 minute 
window of potential downtime.  While 10 minutes was not acceptable to 
me, I agree that 2 minutes is in a gray area that will be fine for some 
and not fine for others.

So we are back to a pure semantic question.  Does the service get to 
decide how many flows the UA opens (up to an implementation supported 
maximum), does the UA decide how many flows to open, or is there some 
negotiation?   I still think the current semantics are fine, because 
they allow a single flow to a URI that can still load-share and 
failover to a large number of actual servers.  Of course there are 
others who don't like that because it relies on DNS for failover.

thanks,
-rohan


> One could probably calculate the chance that in this window a request 
> would come in (given message rate assumptions, etc). Multiply that 
> with the chance at flow failure (given component MTBF figures etc), 
> and you would have the kind of reliability one would have for one flow 
> only. Call that reliability R
>
> My question now is: what are reasonable assumptions for these figures, 
> and could it be that for some clients, R could be sufficient?
> Intuitively I say: yes, that's possible. There exist clients and users 
> that would be happy with a scheme that is broken for (1-R) part (I 
> know at least one - me).
>
> There is currently nothing in the requirements regarding reliability. 
> Section 4.4 mentions the relationship between timer values and server 
> load, and has some text on "the expected time that a user may not 
> receive inbound communications"; but not in relation to the number of 
> flows (nor does it mention to de-synchronize keepalive timers across 
> multiple flows). Do we need that, perhaps with some sample 
> quantitative analysis?
>
> Regards,
>
> Jeroen


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From sip-bounces@ietf.org Sun Jul 09 13:54:54 2006
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Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
Date: Sat, 8 Jul 2006 17:25:32 -0700
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Hi,

I do not understand the motivation for wanting multiple flows to the 
same actual server.

At some point we need to ask ourselves as a group whether we are a 
building a mechanism that solves a problem in every conceivable 
topology we can come up with , or whether it is sufficient for us to 
build a mechanism that solves a variety of important problems using one 
or two plausible topologies.

thanks,
-rohan


On Jul 8, 2006, at 11:43, Jeroen van Bemmel wrote:
> Dear outbound authors,
>
> Could you share your thoughts on the view Juha is expressing below?
>
> Regards,
>
> Jeroen
>
> ----- Original Message ----- From: "Juha Heinanen" <jh@tutpro.com>
> To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
> Cc: "Hadriel Kaplan" <HKaplan@acmepacket.com>; <sip@ietf.org>
> Sent: Saturday, July 08, 2006 8:28 PM
> Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
>
>
>> jeroen,
>> multiple flows to the same proxy is not interesting.  the only added
>> value of outbound is multiple flows to more than one proxy.
>> -- juha


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Date: Sun, 09 Jul 2006 22:27:09 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
To: Dan Wing <dwing@cisco.com>, sip@ietf.org
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Dan,

I am not sure that would work. The problem is the UAC typically does not
insert the Identity header field in a request, this is typically done by an
authentication service at the outbound proxy. So the fact that the request
went through an authentication service and had an Identity header field
added does not mean that the response will go through a verification
service. However, I am open to opinions on that.

John

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: 08 July 2006 02:21
> To: Elwell, John; sip@ietf.org
> Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
> 
> I hadn't noticed any followups to this issue.  I have a suggestion:
> 
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com] 
> > Sent: Tuesday, May 02, 2006 1:21 PM
> > To: sip@ietf.org
> > Subject: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
> >
> > The second issue concerns the fact that an UPDATE request in the
> > backward direction following dialog establishment can be wasted if
> > the peer UA (the UAC for the INVITE transaction) does not support
> > it.  It states:
> >
> >       "OPEN ISSUE: If the identity has has not changed compared with
> >       the URI in the To header field and the UAC (for the INVITE
> >       request) does not support verification of an Identity header,
> >       this UPDATE or re-INVITE transaction would fulfil no useful
> >       purpose.  However, the UAS (for the INVITE transaction) does
> >       not know this.  In order to save this round trip, should we
> >       have an additional option tag to indicate support for receipt
> >       of the Identity header on mid- dialog requests?  Then if this
> >       option tag is not present in the INVITE request the additional
> >       transaction could be omitted.  Note that the id-change option
> >       tag does not seem appropriate for this purpose, since a UA
> >       could support a change of URI without supporting Identity."
> >
> > Whilst I do not wish to proliferate option tags, I do see some
> > benefit in such a tag.  Any opinions?
> 
> If an Invite arrives with both Identity and the id-change tag,
> it indicates the offerer understands both Identity and connected-
> identity, so the answerer should generate an request (UPDATE or 
> re-INVITE).
> 
> -d
> 

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From sip-bounces@ietf.org Sun Jul 09 17:43:40 2006
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> Hi,
>
> I do not understand the motivation for wanting multiple flows to the same 
> actual server.
>

Not multiple *connections*, but multiple flows via different edge proxies: 
yes (see section 3.4). Of course, the single server itself will need to be 
built properly to provide adequate reliability (eg storage that survives 
reboots for registration state, etc).

Perhaps we should let edge proxies refuse a REGISTER that comes in when 
another flow already exists through it though.

> At some point we need to ask ourselves as a group whether we are a 
> building a mechanism that solves a problem in every conceivable topology 
> we can come up with , or whether it is sufficient for us to build a 
> mechanism that solves a variety of important problems using one or two 
> plausible topologies.
>

The problem is: in the current outbound draft there are *several* mechanisms 
( multiple flows, keepalive, outbound request handling, reliability schemes) 
defined that solve a variety of *separate* problems, while making 
assumptions that rule out certain other, equally plausible, topologies.

For example, topologies involving mobile devices are important too. Such 
networks are quite capable of providing certain degrees of reliability 
themselves, but that won't do any good once the battery is drained. For 
these devices, "MUST setup 2 flows" can be too much. So we cannot (only) 
have a scheme that requires that

Regards,

Jeroen

> thanks,
> -rohan
>
>
> On Jul 8, 2006, at 11:43, Jeroen van Bemmel wrote:
>> Dear outbound authors,
>>
>> Could you share your thoughts on the view Juha is expressing below?
>>
>> Regards,
>>
>> Jeroen
>>
>> ----- Original Message ----- From: "Juha Heinanen" <jh@tutpro.com>
>> To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
>> Cc: "Hadriel Kaplan" <HKaplan@acmepacket.com>; <sip@ietf.org>
>> Sent: Saturday, July 08, 2006 8:28 PM
>> Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
>>
>>
>>> jeroen,
>>> multiple flows to the same proxy is not interesting.  the only added
>>> value of outbound is multiple flows to more than one proxy.
>>> -- juha
> 


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From sip-bounces@ietf.org Sun Jul 09 17:57:25 2006
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	<sip@ietf.org>
Subject: RE: [Sip] STUN keep-alive: Client vs server generating requests
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I would strongly suggest you keep with the client-side initiating STUN
requests.  It's been our experience that the device behind the NAT (on the
private/internal side) is always at a distinct advantage over the one on the
outside of it for keepalives.  If you want the reasons I can list them, but
it's probably not worth the time and I think this has been said before by
others on this mailing list in the past.

It would be nice though if the server could give the client a stun-interval
value to use.  Like many solutions do with Registers and expires/min-expires
today.  Some regions of the world seem to have insanely short NAT cache
timers, and others have very long ones, and there's a penalty to be paid for
the former.

-hadriel


> -----Original Message-----
> From: Christer Holmberg (JO/LMF) [mailto:christer.holmberg@ericsson.com]
> Sent: Sunday, July 09, 2006 4:26 AM
> To: Jeroen van Bemmel; Rohan Mahy; sip@ietf.org
> Cc: Miguel Garcia; Cullen Jennings
> Subject: [Sip] STUN keep-alive: Client vs server generating requests
> 
> Moved from the "STUN keep-alive: Where to document it? (was [Sip] Re: A
> coment  todraft-ietf-sip-outbound-04)" thread:
> 
> Jeroen wrote:
> 
> 
> >>One issue with that proposal was that the client doesn't know if e.g. a
> NAT
> >>is rebooting. It would require some timer, so that the client can assume
> a
> >>reboot has >occured if it doesn't receive a STUN request before the
> timer
> >>expires.
> 
> >Also client-initiated STUN requires such a periodic timer. Furthermore,
> >sending a STUN request and then waiting for a response also implicitly
> uses
> >a timer (typical implementation is the sending thread blocking on some
> event
> >with a timeout, reception of the response setting this event)
> 
> Correct. But, these timers a "local" client timers aren't they, i.e. how
> often it sends STUN requests and for how long it should wait for a reply?
> 
> In case the server generates the STUN request the client must know which
> timers the server uses. That can of course be a standardized value, OR it
> can be provided in some header, but the client needs to know about it.
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> _______________________________________________
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From sip-bounces@ietf.org Sun Jul 09 18:05:15 2006
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Hadriel,

OK, I'll take your word for it. If we then interpret such a 
'stun-interval=xxx' as an indication of support for STUN (and say 'no STUN 
unless interval is provided'), I'm happy

Only trouble is: in many cases, the proxy would not know which type of NAT 
there is in between, let alone how long it caches...

Regards,

Jeroen

----- Original Message ----- 
From: "Hadriel Kaplan" <HKaplan@acmepacket.com>
To: "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>; <sip@ietf.org>
Sent: Sunday, July 09, 2006 11:55 PM
Subject: RE: [Sip] STUN keep-alive: Client vs server generating requests


>I would strongly suggest you keep with the client-side initiating STUN
> requests.  It's been our experience that the device behind the NAT (on the
> private/internal side) is always at a distinct advantage over the one on 
> the
> outside of it for keepalives.  If you want the reasons I can list them, 
> but
> it's probably not worth the time and I think this has been said before by
> others on this mailing list in the past.
>
> It would be nice though if the server could give the client a 
> stun-interval
> value to use.  Like many solutions do with Registers and 
> expires/min-expires
> today.  Some regions of the world seem to have insanely short NAT cache
> timers, and others have very long ones, and there's a penalty to be paid 
> for
> the former.
>
> -hadriel
>
>
>> -----Original Message-----
>> From: Christer Holmberg (JO/LMF) [mailto:christer.holmberg@ericsson.com]
>> Sent: Sunday, July 09, 2006 4:26 AM
>> To: Jeroen van Bemmel; Rohan Mahy; sip@ietf.org
>> Cc: Miguel Garcia; Cullen Jennings
>> Subject: [Sip] STUN keep-alive: Client vs server generating requests
>>
>> Moved from the "STUN keep-alive: Where to document it? (was [Sip] Re: A
>> coment  todraft-ietf-sip-outbound-04)" thread:
>>
>> Jeroen wrote:
>>
>>
>> >>One issue with that proposal was that the client doesn't know if e.g. a
>> NAT
>> >>is rebooting. It would require some timer, so that the client can 
>> >>assume
>> a
>> >>reboot has >occured if it doesn't receive a STUN request before the
>> timer
>> >>expires.
>>
>> >Also client-initiated STUN requires such a periodic timer. Furthermore,
>> >sending a STUN request and then waiting for a response also implicitly
>> uses
>> >a timer (typical implementation is the sending thread blocking on some
>> event
>> >with a timeout, reception of the response setting this event)
>>
>> Correct. But, these timers a "local" client timers aren't they, i.e. how
>> often it sends STUN requests and for how long it should wait for a reply?
>>
>> In case the server generates the STUN request the client must know which
>> timers the server uses. That can of course be a standardized value, OR it
>> can be provided in some header, but the client needs to know about it.
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 


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From sip-bounces@ietf.org Sun Jul 09 18:11:02 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: [Sip] outbound: recovery in case of flow failure
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This is a multi-part message in MIME format.

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Suggestion for section 5.3 (Edge proxy forwarding) and 6.2 (registrar =
forwarding): When a flow is found but fails upon trying, the registrar / =
edge proxy could send back a '480 Temporarily Unavailable' response, =
with a Retry-After equal to the maximum time a flow recovery would take =
(ie about 29 seconds for UDP, 120 seconds for TCP)

This could help the caller in deciding to retry after giving the callee =
enough time to restore its flow

Regards,

Jeroen
------=_NextPart_000_00A5_01C6A3B5.0E9D8490
Content-Type: text/html;
	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5450.4" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Suggestion for section 5.3 (Edge proxy =
forwarding)=20
and 6.2 (registrar forwarding): When a flow is found but fails upon =
trying, the=20
registrar / edge proxy could send back a '480 Temporarily Unavailable' =
response,=20
with a Retry-After equal to the maximum time a flow recovery would take =
(ie=20
about 29 seconds for UDP, 120 seconds for TCP)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>This could help the caller in deciding =
to retry=20
after giving the callee enough time to restore its flow</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Regards,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Jeroen</FONT></DIV></BODY></HTML>

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_______________________________________________
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--===============0528211146==--





From sip-bounces@ietf.org Sun Jul 09 18:14:49 2006
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From: "Hadriel Kaplan" <HKaplan@acmepacket.com>
To: "'Juha Heinanen'" <jh@tutpro.com>
Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04
Date: Sun, 9 Jul 2006 18:12:57 -0400
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I'm really confused now.  If there are 2 proxies, with "one flow to each",
then clearly the UAC created a flow to each, and registered through each.
Jeroen's case is one where the operator configures 2 outbound proxies, but
the UAC only USES one - i.e, it only registers and stuns through/to one.
Not that it somehow registers to it but rejects requests from it. (right
Jeroen?)

Whether you as an operator configure all UACs with 2 proxies, or more, does
not make a difference for his case - if the UAC only establishes 1 flow,
then it only registers through one proxy, and only one proxy adds a path
header, and the registrar only has one flow and registration contact info to
use to reach the UAC, and 100% of the requests go that 1 proxy path.  The
other proxy(s) know nothing about the UAC, short of being able to find out
from the location server that it's through the 1 proxy that the UAC actually
registered through.

To suggest otherwise, that somehow the secondary proxy gets 50% of the
requests (and they fail) even if the UAC did not register through it, means
that in a failure mode where the UAC can only reach one proxy, then it only
gets 50% service.  That would be a significant step BACKWARDS from today's
fairly common reliability mechanisms and availability models, IMO. 

-hadriel


> -----Original Message-----
> From: Juha Heinanen [mailto:jh@tutpro.com]
> Sent: Saturday, July 08, 2006 1:11 PM
> To: Hadriel Kaplan
> Cc: sip@ietf.org
> Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04
> 
> Hadriel Kaplan writes:
> 
>  > > in the above situation, assuming that network has two proxies, if UA
>  > > decides to establish only one flow instead of two, then the likely
> hood
>  > > is 50% that incoming call to the UA cannot be establishes.  it thus
>  > > makes no sense for the UA to violate network policy on number of
> flows.
>  >
>  > That doesn't make any sense to me.
> 
> obviously you didn't notice that in the example there was two proxies,
> one flow to each.
> 
> -- juha


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From sip-bounces@ietf.org Sun Jul 09 18:36:30 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Hi Hadriel,

> I'm really confused now.  If there are 2 proxies, with "one flow to each",
> then clearly the UAC created a flow to each, and registered through each.
> Jeroen's case is one where the operator configures 2 outbound proxies, but
> the UAC only USES one - i.e, it only registers and stuns through/to one.
> Not that it somehow registers to it but rejects requests from it. (right
> Jeroen?)
>

Right, it establishes only 1 registration (like most UACs do today) and 
keeps this flow alive

> Whether you as an operator configure all UACs with 2 proxies, or more, 
> does
> not make a difference for his case - if the UAC only establishes 1 flow,
> then it only registers through one proxy, and only one proxy adds a path
> header, and the registrar only has one flow and registration contact info 
> to
> use to reach the UAC, and 100% of the requests go that 1 proxy path.  The
> other proxy(s) know nothing about the UAC, short of being able to find out
> from the location server that it's through the 1 proxy that the UAC 
> actually
> registered through.
>
> To suggest otherwise, that somehow the secondary proxy gets 50% of the
> requests (and they fail) even if the UAC did not register through it, 
> means
> that in a failure mode where the UAC can only reach one proxy, then it 
> only
> gets 50% service.  That would be a significant step BACKWARDS from today's
> fairly common reliability mechanisms and availability models, IMO.
>

In my mind, how you would end up with this "50% model" is to have 2 
registrar instances configured in DNS as load-balancing alternatives for the 
domain. With or without edge proxies, this model would indeed require the UA 
to setup 2 registrations/flows, or miss 50% of the requests (on average).

I am no great fan of this model, but if people want to do this, the UA needs 
to know how many registrations it is expected to create. This is *not* equal 
to the number of URIs in the magically discovered outbound proxy set, 
because the UA cannot tell how many distinct registrar instances are 
reachable through this set. The number of registrations cannot be larger 
than the number of URIs in the set, but it could be less (i.e. value is in 
range 1..#URIs)

Regards,

Jeroen

> -hadriel
>
>
>> -----Original Message-----
>> From: Juha Heinanen [mailto:jh@tutpro.com]
>> Sent: Saturday, July 08, 2006 1:11 PM
>> To: Hadriel Kaplan
>> Cc: sip@ietf.org
>> Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04
>>
>> Hadriel Kaplan writes:
>>
>>  > > in the above situation, assuming that network has two proxies, if UA
>>  > > decides to establish only one flow instead of two, then the likely
>> hood
>>  > > is 50% that incoming call to the UA cannot be establishes.  it thus
>>  > > makes no sense for the UA to violate network policy on number of
>> flows.
>>  >
>>  > That doesn't make any sense to me.
>>
>> obviously you didn't notice that in the example there was two proxies,
>> one flow to each.
>>
>> -- juha
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


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From sip-bounces@ietf.org Sun Jul 09 19:52:46 2006
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From: "Hadriel Kaplan" <HKaplan@acmepacket.com>
To: "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>,
	"'Juha Heinanen'" <jh@tutpro.com>
Subject: RE: [Sip] A coment to draft-ietf-sip-outbound-04
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Hi Jeroen, inline...

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> 
> Hi Hadriel,
> 
> In my mind, how you would end up with this "50% model" is to have 2
> registrar instances configured in DNS as load-balancing alternatives for
> the
> domain. With or without edge proxies, this model would indeed require the
> UA
> to setup 2 registrations/flows, or miss 50% of the requests (on average).

I would think any model using 2 registrars in this way has to use a shared
registrar database, a database exchange/sync mechanism, or a "if not on this
registrar then try the other" mechanism.  There is simply no way to
guarantee every UAC has reach-ability to both registrars all the time.
There are just too many failure modes, some of which are long-lived.  They
would just be asking for trouble, imho.  :)

-hadriel 


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From: "Dan Wing" <dwing@cisco.com>
To: "'Elwell, John'" <john.elwell@siemens.com>, <sip@ietf.org>
Subject: RE: [Sip] Issue 4 in draft-ietf-sip-connected-identity-00.txt
Date: Sun, 9 Jul 2006 21:04:45 -0400
Keywords: direct-to-dwing
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> Yes, I agree with your motivation for being able to have 
> signed SDP in the reverse direction. This doesn't necessarily mean 
> UPDATE has to be used - re-INVITE would do, but I prefer to use 
> UPDATE unless there are other reasons for using re-INVITE.
> 
> The question is whether we should mandate including an SDP 
> offer in the UPDATE so that we always have the benefit of 
> this signature.

Agreed.

And if the SDP is exactly the same, I see a lot of value and little harm.

-d

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From: "Dan Wing" <dwing@cisco.com>
To: "'Elwell, John'" <john.elwell@siemens.com>, <sip@ietf.org>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
Date: Sun, 9 Jul 2006 21:12:26 -0400
Keywords: direct-to-dwing
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> I am not sure that would work. The problem is the UAC 
> typically does not insert the Identity header field in a 
> request, this is typically done by an authentication service 
> at the outbound proxy. So the fact that the request went 
> through an authentication service and had an Identity header 
> field added does not mean that the response will go through 
> a verification service. 

I agree there is nothing a UA can do to ensure its request
gets an Identity header added.  

All I'm saying is that if you receive a request with an Identity
headers, there is a pretty darn high likelyhood that the requestor
is interested in having your response authenticated (by a backwards
request using connected-identity).  All you can do is send a request
back to him and hope that your request will get an Identity header
added by your authentication service.  If it doesn't get an Identity
header added, the requestor's local policy or implementation will
decide what to do next: accept the call or drop it (because the
connected-identity response lacks an Identity header).

I argue that an additional tag wouldn't convey any additional 
meaning over my proposal.

-d

> However, I am open to opinions on that.
> 
> John
> 
> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com] 
> > Sent: 08 July 2006 02:21
> > To: Elwell, John; sip@ietf.org
> > Subject: RE: [Sip] Issue 2 in 
> draft-ietf-sip-connected-identity-00.txt
> > 
> > I hadn't noticed any followups to this issue.  I have a suggestion:
> > 
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens.com] 
> > > Sent: Tuesday, May 02, 2006 1:21 PM
> > > To: sip@ietf.org
> > > Subject: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
> > >
> > > The second issue concerns the fact that an UPDATE request in the
> > > backward direction following dialog establishment can be wasted if
> > > the peer UA (the UAC for the INVITE transaction) does not support
> > > it.  It states:
> > >
> > >       "OPEN ISSUE: If the identity has has not changed 
> compared with
> > >       the URI in the To header field and the UAC (for the INVITE
> > >       request) does not support verification of an 
> Identity header,
> > >       this UPDATE or re-INVITE transaction would fulfil no useful
> > >       purpose.  However, the UAS (for the INVITE transaction) does
> > >       not know this.  In order to save this round trip, should we
> > >       have an additional option tag to indicate support 
> for receipt
> > >       of the Identity header on mid- dialog requests?  
> Then if this
> > >       option tag is not present in the INVITE request the 
> additional
> > >       transaction could be omitted.  Note that the 
> id-change option
> > >       tag does not seem appropriate for this purpose, since a UA
> > >       could support a change of URI without supporting Identity."
> > >
> > > Whilst I do not wish to proliferate option tags, I do see some
> > > benefit in such a tag.  Any opinions?
> > 
> > If an Invite arrives with both Identity and the id-change tag,
> > it indicates the offerer understands both Identity and connected-
> > identity, so the answerer should generate an request (UPDATE or 
> > re-INVITE).
> > 
> > -d
> > 

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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
Date: Mon, 10 Jul 2006 03:41:26 +0200
To: "Elwell, John" <john.elwell@siemens.com>
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What is the point of supporting id-change without supporting Identity. 
If you support connected-identity, then you MUST support Identity RFC. 
That's it.

Hisham

On May 2, 2006, at 10:21 PM, Elwell, John wrote:

> The second issue concerns the fact that an UPDATE request in the 
> backward
> direction following dialog establishment can be wasted if the peer UA 
> (the
> UAC for the INVITE transaction) does not support it. It states:
>
>       "OPEN ISSUE: If the identity has has not changed compared with 
> the
>       URI in the To header field and the UAC (for the INVITE request)
>       does not support verification of an Identity header, this UPDATE
>       or re-INVITE transaction would fulfil no useful purpose.  
> However,
>       the UAS (for the INVITE transaction) does not know this.  In 
> order
>       to save this round trip, should we have an additional option tag
>       to indicate support for receipt of the Identity header on mid-
>       dialog requests?  Then if this option tag is not present in the
>       INVITE request the additional transaction could be omitted.  Note
>       that the id-change option tag does not seem appropriate for this
>       purpose, since a UA could support a change of URI without
>       supporting Identity."
>
> Whilst I do not wish to proliferate option tags, I do see some benefit 
> in
> such a tag. Any opinions?
>
> John
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Issue 4 in draft-ietf-sip-connected-identity-00.txt
Date: Mon, 10 Jul 2006 03:45:15 +0200
To: "Elwell, John" <john.elwell@siemens.com>
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I don't think we should mandate UPDATE. INVITE does the job if an 
implementer chooses not to do UPDATE.

Hisham

On May 2, 2006, at 10:23 PM, Elwell, John wrote:

> The fourth issue is not actually in draft-00 but was raised 
> subsequently. If
> a UA supports a change of From header field URI mid-dialog, as 
> indicated by
> the "id-change" option tag, we could mandate that it also support the 
> SIP
> UPDATE method. Then the peer UA in a dialog would be free to use the 
> UPDATE
> method to report connected identity, rather than the less suitable 
> re-INVITE
> method. The use of UPDATE means 2 messages instead of 3 and avoids the 
> need
> for an offer-answer exchange. I heard no objections when this was 
> proposed a
> few days ago. Do we have consensus?
>
> John
>
>
>
> _______________________________________________
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> Use sipping@ietf.org for new developments on the application of sip
>


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Date: Mon, 10 Jul 2006 02:53:38 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
To: Dan Wing <dwing@cisco.com>, sip@ietf.org
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Dan,

That was not what I was trying to say. The UAC (for the INVITE request) has
no control over whether an Identity header field is added by its domain
proxy. So just because the UAS receives an INVITE request with an Identity
header field, does not mean that the UAC put it there and would be
interested in receiving an UPDATE request with an Identity header field.

However, I don't have a strong objection to the UAS (for the INVITE) always
sending back an UPDATE even if the From header URI would be unchanged from
that in the To header URI of the INVITE request.

Regards,

John 

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: 10 July 2006 02:12
> To: Elwell, John; sip@ietf.org
> Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
> 
> > I am not sure that would work. The problem is the UAC 
> > typically does not insert the Identity header field in a 
> > request, this is typically done by an authentication service 
> > at the outbound proxy. So the fact that the request went 
> > through an authentication service and had an Identity header 
> > field added does not mean that the response will go through 
> > a verification service. 
> 
> I agree there is nothing a UA can do to ensure its request
> gets an Identity header added.  
> 
> All I'm saying is that if you receive a request with an Identity
> headers, there is a pretty darn high likelyhood that the requestor
> is interested in having your response authenticated (by a backwards
> request using connected-identity).  All you can do is send a request
> back to him and hope that your request will get an Identity header
> added by your authentication service.  If it doesn't get an Identity
> header added, the requestor's local policy or implementation will
> decide what to do next: accept the call or drop it (because the
> connected-identity response lacks an Identity header).
> 
> I argue that an additional tag wouldn't convey any additional 
> meaning over my proposal.
> 
> -d
> 
> > However, I am open to opinions on that.
> > 
> > John
> > 
> > > -----Original Message-----
> > > From: Dan Wing [mailto:dwing@cisco.com] 
> > > Sent: 08 July 2006 02:21
> > > To: Elwell, John; sip@ietf.org
> > > Subject: RE: [Sip] Issue 2 in 
> > draft-ietf-sip-connected-identity-00.txt
> > > 
> > > I hadn't noticed any followups to this issue.  I have a 
> suggestion:
> > > 
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:john.elwell@siemens.com] 
> > > > Sent: Tuesday, May 02, 2006 1:21 PM
> > > > To: sip@ietf.org
> > > > Subject: [Sip] Issue 2 in 
> draft-ietf-sip-connected-identity-00.txt
> > > >
> > > > The second issue concerns the fact that an UPDATE request in the
> > > > backward direction following dialog establishment can 
> be wasted if
> > > > the peer UA (the UAC for the INVITE transaction) does 
> not support
> > > > it.  It states:
> > > >
> > > >       "OPEN ISSUE: If the identity has has not changed 
> > compared with
> > > >       the URI in the To header field and the UAC (for the INVITE
> > > >       request) does not support verification of an 
> > Identity header,
> > > >       this UPDATE or re-INVITE transaction would fulfil 
> no useful
> > > >       purpose.  However, the UAS (for the INVITE 
> transaction) does
> > > >       not know this.  In order to save this round trip, 
> should we
> > > >       have an additional option tag to indicate support 
> > for receipt
> > > >       of the Identity header on mid- dialog requests?  
> > Then if this
> > > >       option tag is not present in the INVITE request the 
> > additional
> > > >       transaction could be omitted.  Note that the 
> > id-change option
> > > >       tag does not seem appropriate for this purpose, since a UA
> > > >       could support a change of URI without supporting 
> Identity."
> > > >
> > > > Whilst I do not wish to proliferate option tags, I do see some
> > > > benefit in such a tag.  Any opinions?
> > > 
> > > If an Invite arrives with both Identity and the id-change tag,
> > > it indicates the offerer understands both Identity and connected-
> > > identity, so the answerer should generate an request (UPDATE or 
> > > re-INVITE).
> > > 
> > > -d
> > > 
> 

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
To: Hisham Khartabil <hisham.khartabil@telio.no>
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Hisham,

Well that was how I had it originally, but I was asked to consider
separating it. I don't have a strong opinion.

John 

> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no] 
> Sent: 10 July 2006 02:41
> To: Elwell, John
> Cc: sip@ietf.org
> Subject: Re: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
> 
> What is the point of supporting id-change without supporting 
> Identity. 
> If you support connected-identity, then you MUST support 
> Identity RFC. 
> That's it.
> 
> Hisham
> 
> On May 2, 2006, at 10:21 PM, Elwell, John wrote:
> 
> > The second issue concerns the fact that an UPDATE request in the 
> > backward
> > direction following dialog establishment can be wasted if 
> the peer UA 
> > (the
> > UAC for the INVITE transaction) does not support it. It states:
> >
> >       "OPEN ISSUE: If the identity has has not changed 
> compared with 
> > the
> >       URI in the To header field and the UAC (for the 
> INVITE request)
> >       does not support verification of an Identity header, 
> this UPDATE
> >       or re-INVITE transaction would fulfil no useful purpose.  
> > However,
> >       the UAS (for the INVITE transaction) does not know this.  In 
> > order
> >       to save this round trip, should we have an additional 
> option tag
> >       to indicate support for receipt of the Identity header on mid-
> >       dialog requests?  Then if this option tag is not 
> present in the
> >       INVITE request the additional transaction could be 
> omitted.  Note
> >       that the id-change option tag does not seem 
> appropriate for this
> >       purpose, since a UA could support a change of URI without
> >       supporting Identity."
> >
> > Whilst I do not wish to proliferate option tags, I do see 
> some benefit 
> > in
> > such a tag. Any opinions?
> >
> > John
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] concerns on draft-elwell-sip-tispan-connected-identity-	01
To: Jonathan Rosenberg <jdrosen@cisco.com>, IETF SIP List <sip@ietf.org>
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Jonathan,

Thanks for reading this and commenting. I will leave the TISPAN guys to
reply on the TISPAN reasons for needing two identities (although this was
discussed at length on an old thread) and their rationale for using PAI for
one / Identity for the other. 

All I can say is that the document lays down a concern that some TISPAN
people have about the connected-identity draft and possible ways to overcome
that concern. One solution, of course, would be to do nothing - leave the
connected-identity draft as it is. But in taking that course we will need to
show that TISPAN requirements can be met by some other means.

John

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
> Sent: 07 July 2006 20:50
> To: IETF SIP List
> Subject: [Sip] concerns on 
> draft-elwell-sip-tispan-connected-identity-01
> 
> I wasn't able to find any disucssion on this draft so far in the 
> archives; my apologies if I missed it.
> 
> This draft concerns me greatly.
> 
> The first thing that worries me is the general idea that 3gpp 
> or tispan 
> get to 'choose' which SIP header field contains what type of 
> identity. 
> They get to do no such thing. These header fields have 
> defined meanings. 
> Having either organization decide to specify their own interpretation 
> for the type of identity that goes in there is a guarnateed 
> receipe for 
> huge, massive, and unending interoperability failures.
> 
> On the technical front, I strongly disagree that the From and 
> P-Asserted-Identity represent different types of identity in any way. 
> They both have always identified the user that is the sender of the 
> request. In cases where you had an enterprise user Joe making a SIP 
> call, the From would contain sip:joe@enterprise.com and this would be 
> authenticated by the proxy in the enterprise, followed by 
> insertion of a 
> P-Asserted-ID which says the same thing. The From, in this use case, 
> should not be sip:enterprise.com.
> 
> Next, I am puzzled by the requirement to deliver separate 
> identities to 
> the calling party for these two things. I understand the concept of a 
> billing identity being separate from the identity of the caller. I do 
> not, however, understand why the billing ID is of any interest to the 
> called party whatsoever. It seems that this would, at best, be of 
> interest only within the domain of the calling party. Since its also 
> directly inferrable from the identity of the caller (presumably the 
> database contains a linkage between the two), I also don't see why it 
> would need to be carried in the sip message itself. Is there 
> a use case 
> where there is not a provisioned relationship between the user making 
> the call and the billing entity? Certainly there is a clear direct 
> relationship in the enterprise example in the draft.
> 
> Finally, even if I buy the idea that there are two identities 
> and that 
> they both need to be conveyed, I strongly disagree with the 
> notion that 
> one of them can be conveyed with SIP-identity, while P-A-ID is OK for 
> the other. There are certain well-known technical limitations with 
> P-A-ID which sip-identity addresses. There is nothing about 
> the nature 
> of the identity which invalidates any of those issues.
> 
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From sip-bounces@ietf.org Sun Jul 09 23:51:57 2006
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Subject: Re: [Sip] Proposals for review process to conclude WGLCs
Date: Sun, 9 Jul 2006 22:48:27 -0500
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Time zones are going to be an issue with any real-time review  
process. Maybe it is an issue we can deal with, but I suspect some  
number people who would otherwise participate may be locked out, or  
at least discouraged, by the idea of doing conference calls in their  
normal sleep cycle.

On Jul 8, 2006, at 9:10 PM, Mary Barnes wrote:

> You can add another vote in support of option 1), as I think it
> increases the level of commitment for the reviewers, along with the
> document editor/authors.  It might also prove quite useful in terms of
> resolving some issues in real-time, thus reducing the amount of
> thrashing on specific issues on the list due to misunderstandings via
> email.  As you've already identified, a recorder would be  
> imperative for
> the process.
>
> I also support the idea of SIP doing something different than SIPPING
> and at year end we can evaluate both options and see what seems to  
> work
> best. I'm beginning to think the SIPPING approach might be much better
> applied early in the cycle (i.e, for the first few iterations), but  
> that
> the conference calls might be the most effective way to handle the
> WGLCs.
>
> Obviously, any of the approaches requires committed and dedicated
> reviewers for the documents.  The response in the SIPPING WG has been
> pretty good, but I think it would be optimal to add the requirement  
> that
> a document isn't accepted as a WG document, unless you also have the
> reviewers committed.    This might delay some things, but in my  
> mind, if
> we don't have WG members committed to the work, along with the
> editor/authors, we don't have the resources to effectively complete  
> the
> work (in a timely manner).
>
> Mary
>
> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Friday, June 30, 2006 3:26 PM
> To: IETF SIP List
> Subject: [Sip] Proposals for review process to conclude WGLCs
>
>
>
> One of the questions we've struggled with is how to get sufficient
> review on a document under WGLC to make sure it's really done.
>
>
> SIPPING is following a new procedure with designated reviewers, doing
> review by email, and sending stuff back to the. Apparently this
> suffers from the expected issues, with designated reviewers sometimes
> not quite meeting their review commitments.
>
> Cisco made me take a peer-review refresher class the other day, so
> this started me thinking about some alternatives.
>
> In the model taught in this class, each peer review  includes 1) a
> "moderator" or "reader" who frames the questions and keeps things on
> track, 2) a "recorder" who records conclusions and issues raised
> during the review, 3) the author(s), who neither presents no
> interprets the work, but does respond to specific questions about it,
> and 4) the reviewers, who hopefully have expertise and have committed
> time to the review.
>
>
> I'd like to propose a couple of alternative models to what SPPING has
> been doing and seeing if either resonates with our group.
>
>
> 1) Real-time review sessions via conference call: When we get ready
> to do a final review, the chairs assemble a review team. Presumably,
> this review team is assembled by polling the list for volunteers, and
> by pushing extra hard for stakeholders and other people with specific
> expertise. The moderator builds up a checklist and set of basic
> questions to consider, and distributes it to the review team. At a
> tiem agreed to by enough of the participants, we have a conference
> call, not to exceed two hours, and work through the checklist in a
> moderated fashion. At the conclusion of that two hours, we either
> reach a positive decision, a negative decision (with list of issues
> and actions), or decide we need more review.
>
>
> 2) Moderated email-review, using the entire list as the "review
> team". In this model, the chairs pick a moderator and recorder for
> the review. The moderator sets up a checklist and sends it to the
> list, along with an announcement of the review. The moderator then
> goes through the checklist item-by-item on list, polling for
> comments. The recorder records any issues or conclusions. Each item
> might take two or three days for primary comments, but could be
> revisited during the review process if needed. The author could
> answer specific questions about the draft, but not paraphrase or
> interpret.  At the conclusion of the review, the moderator and
> reviewer would report to the chairs, who would decide whether we have
> issues, a conclusion, or need further review.
>
>
> What do y'all think?
>
> --
> Dean
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
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> Use sipping@ietf.org for new developments on the application of sip
>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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Hi;


Dan Wing wrote:
>> I generally like the draft, and I think the work is very important.
>>     
>
> Me, too.
>
>   
>> Anyhow, AFAIK there is no distinct way to request SIP-Identity to
>> AS, so just because one receives a sip-identity/id-change tag does
>> not necessary mean that sender of an UPDATE will be able to send
>> the sip-identity back(Even if the UAS supports sip-identity/id-change,
>> the domain of the UAS might not have AS in place to provide
>> sip-identity).
>>     
>
> Agreed; if we agree we need to optimize that case, we will need
> an additional option tag.  But, still, there isn't a way for
> a UAS to "know" that his request will be protected with Identity.
> So I'm not sure of the value such an additional tag would bring
> over my suggestion.
>   
In past, there has been RFC which proposed the use of proxy-require for
the next hop only(First hop proxy removes the relevant option-tag in the
proxy-require). RFC 3329 is one of the example for this, and proposal
similar to that may be considered.
>   
>> This is because there is no option tag to require proxy to add
>> sip-identity,
>> and because sip-identity is generally performed at AS, UA is generally
>> unaware of the domian's support for sip-identity.
>>
>> Furthermore, because its the AS that adds sip-identity(except when
>> UA has its own certificate which I think is unlikely), just because UA
>> receives the sip-identity, does not necessary mean that UA sending
>> the original INVITE(offerer that you have described in your comment)
>> with identity-info header supports sip-identity.
>>     
>
> Even if the original UA itself doesn't support Identity, one of the
> domain's proxies added the Identity signature and it's likely that
> proxy may validate incoming requests containing Identity signatures.
>
>   
Agree.

Regards
Shida Schubert

> -d
>
>   
>> Last note, looking at the sentence in 6.1 which is shown
>> below, I am not sure if
>> mandating UA to send an UPDATE to update its identity is aligned
>> with the semantics of Supported header field. I don't really
>> have a strong
>> opinion on this point, but it felt a little funny to mandate
>> a behavior with
>> a presence of "Supported" header.
>>
>> Thanks & Regards
>> Shida
>>
>> Dan Wing wrote:
>>     
>>> I hadn't noticed any followups to this issue.  I have a suggestion:
>>>
>>>  
>>>       
>>>> -----Original Message-----
>>>> From: Elwell, John [mailto:john.elwell@siemens.com]
>>>> Sent: Tuesday, May 02, 2006 1:21 PM
>>>> To: sip@ietf.org
>>>> Subject: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
>>>>
>>>> The second issue concerns the fact that an UPDATE request in the
>>>> backward direction following dialog establishment can be wasted if
>>>> the peer UA (the UAC for the INVITE transaction) does not support
>>>> it.  It states:
>>>>
>>>>       "OPEN ISSUE: If the identity has has not changed
>>>>         
>> compared with
>>     
>>>>       the URI in the To header field and the UAC (for the INVITE
>>>>       request) does not support verification of an Identity header,
>>>>       this UPDATE or re-INVITE transaction would fulfil no useful
>>>>       purpose.  However, the UAS (for the INVITE transaction) does
>>>>       not know this.  In order to save this round trip, should we
>>>>       have an additional option tag to indicate support for receipt
>>>>       of the Identity header on mid- dialog requests?  Then if this
>>>>       option tag is not present in the INVITE request the
>>>>         
>> additional
>>     
>>>>       transaction could be omitted.  Note that the id-change option
>>>>       tag does not seem appropriate for this purpose, since a UA
>>>>       could support a change of URI without supporting Identity."
>>>>
>>>> Whilst I do not wish to proliferate option tags, I do see some
>>>> benefit in such a tag.  Any opinions?
>>>>    
>>>>         
>>> If an Invite arrives with both Identity and the id-change tag,
>>> it indicates the offerer understands both Identity and connected-
>>> identity, so the answerer should generate an request (UPDATE or
>>> re-INVITE).
>>>
>>> -d
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>>   
>>>       
>
>
>   


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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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On Jul 9, 2006, at 8:37, Jeroen van Bemmel wrote:

> I think the system is "broken" in the sense that it does not behave as 
> the authors of the draft had intended
> There are 2 questions here:
> 1) Is that intention expressed sufficiently clear and consistent in 
> the draft?
> 2) Is the intended behavior (and associated system architecture) 
> appropriate/desirable (in all/most cases)?
>
> The draft seems to be written under the assumption that "reliability" 
> (not well defined yet what that means) can only be provided through 
> multiple flows. However, "outbound" (ie enabling UAs behind NAT to 
> register and receive incoming requests) can be done using only a 
> single flow, albeit with a somewhat higher chance at missing incoming 
> requests.

the draft assumes that *if* the outbound-proxy-set contained <n> URIs, 
then whoever provided the outbound-proxy-set believes that <n> flows 
are needed to provide reliability.  If only a single URI is provided in 
the outbound-proxy-set, then a single flow is sufficient.

> If the system is architected as Juha was describing (2 independent 
> proxies, UA missing 50% of its calls when only setting up 1 flow), the 
> chance at missed requests for such a UA is much higher than double 
> that of a single flow system. I feel that outbound should certainly 
> not mandate such an architecture.

The UA doesn't miss 50% of its calls in this case.  If only one flow is 
registered, the registrar/home proxy will forward incoming requests 
only over that flow.  Its just that the UA is not playing nicely with 
the proxies, as they would like to load balance over two flows, but 
they have to forward all the traffic for the UA over one flow instead.

> Currently in SIP the assumption is that a single registration suffices 
> to receive all incoming requests for a given AoR. If we break that, it 
> means that current UAs won't be able to register with 
> "outbound-enhanced" registrars and edge proxies. That to me is not 
> acceptable.

there is no requirement in outbound that pre-outbound registrations 
will not work.

> In other words, a requirement is missing: "Support UAs that do not 
> implement this extension"

What amount of "support" do you envision?

thanks,
-rohan


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On Jul 8, 2006, at 12:23 PM, Juha Heinanen wrote:

>
>> Or just hope it won't melt when you send it a STUN packet, meaning  
>> that
>> you "require the operator to get it right".
>
>> The problem as I see it is we have no way of easily detecting an  
>> error
>> if somebody mistakenly lists a non-compliant proxy as an outbound  
>> proxy
>> in DHCP or some other discovery mechanism. I hate leaping off a  
>> trapeze
>> without a safety net . . .
>
> no properly implemented proxy melts no matter what you sent to it.   
> this
> kind of arguments are completely bogus.

Dale has noted in a separate thread that STUN breaks TCP framing,  
possibly creating a self-inflicted DoS attack that the UA cannot  
detect to recover from.

So let's say my UA comes up on a strange network and I use nmap to  
find nodes responding on 5060/tcp that might serve as an outbound proxy.

I find one. Should I STUN it first, or ask it to process a SIP  
message request with a "Proxy-Require: outbound" before stunning it?

The current outbound draft, although it defines such an option tag,  
seems to define its use only in responses to REGISTER requests issued  
by a registrar. It says nothing about its use by a proxy that is in- 
between the registrar and the UA.

--
Dean



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Hi;

I think they may have separate semantics. I see "id-change" tag
marely defines the ability to express ones ability to withstand the
change of From.

Usage of sip-identity along with id-change to provide authenticated
connected identity is defining a usage of id-change. I think it's important
to describe the usage somewhere. Where? I don't know.. .

As discussed on this thread, id-change is used by the end-device
and sip-identity as I can foresee will be likely to be deployed on the
proxy.

So support "id-change" also means support "sip-identity" may not work
too well.

Regards
Shida

Elwell, John wrote:
> Hisham,
>
> Well that was how I had it originally, but I was asked to consider
> separating it. I don't have a strong opinion.
>
> John 
>
>   
>> -----Original Message-----
>> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no] 
>> Sent: 10 July 2006 02:41
>> To: Elwell, John
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
>>
>> What is the point of supporting id-change without supporting 
>> Identity. 
>> If you support connected-identity, then you MUST support 
>> Identity RFC. 
>> That's it.
>>
>> Hisham
>>
>> On May 2, 2006, at 10:21 PM, Elwell, John wrote:
>>
>>     
>>> The second issue concerns the fact that an UPDATE request in the 
>>> backward
>>> direction following dialog establishment can be wasted if 
>>>       
>> the peer UA 
>>     
>>> (the
>>> UAC for the INVITE transaction) does not support it. It states:
>>>
>>>       "OPEN ISSUE: If the identity has has not changed 
>>>       
>> compared with 
>>     
>>> the
>>>       URI in the To header field and the UAC (for the 
>>>       
>> INVITE request)
>>     
>>>       does not support verification of an Identity header, 
>>>       
>> this UPDATE
>>     
>>>       or re-INVITE transaction would fulfil no useful purpose.  
>>> However,
>>>       the UAS (for the INVITE transaction) does not know this.  In 
>>> order
>>>       to save this round trip, should we have an additional 
>>>       
>> option tag
>>     
>>>       to indicate support for receipt of the Identity header on mid-
>>>       dialog requests?  Then if this option tag is not 
>>>       
>> present in the
>>     
>>>       INVITE request the additional transaction could be 
>>>       
>> omitted.  Note
>>     
>>>       that the id-change option tag does not seem 
>>>       
>> appropriate for this
>>     
>>>       purpose, since a UA could support a change of URI without
>>>       supporting Identity."
>>>
>>> Whilst I do not wish to proliferate option tags, I do see 
>>>       
>> some benefit 
>>     
>>> in
>>> such a tag. Any opinions?
>>>
>>> John
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>>       
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
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> Use sipping@ietf.org for new developments on the application of sip
>
>
>   


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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Proposals for review process to conclude WGLCs
Date: Mon, 10 Jul 2006 00:22:28 -0500
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On Jul 9, 2006, at 10:48 PM, Ben Campbell wrote:

> Time zones are going to be an issue with any real-time review  
> process. Maybe it is an issue we can deal with, but I suspect some  
> number people who would otherwise participate may be locked out, or  
> at least discouraged, by the idea of doing conference calls in  
> their normal sleep cycle.

You people sleep? No wonder we're missing milestones! Keith, we need  
to recalculate all the milestones to accommodate this sleep thing.

Seriously, one would have to schedule an individual review to meet  
the schedules of the selected review team. We'd have to be careful  
not to mix people in very bad configurations. This can usually be  
sorted out for groups of 3-5 people selected from a large pool,  
assuming they're willing to work between 7:00 AM and midnight in  
their home time zones. Of course, that does leave out at least one  
area director I know of . . .

--
Dean



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From: Juha Heinanen <jh@tutpro.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Subject: Re: [Sip] A coment to draft-ietf-sip-outbound-04
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Jeroen van Bemmel writes:

 > For example, topologies involving mobile devices are important too. Such 
 > networks are quite capable of providing certain degrees of reliability 
 > themselves, but that won't do any good once the battery is drained. For 
 > these devices, "MUST setup 2 flows" can be too much. So we cannot (only) 
 > have a scheme that requires that

i don't remember anybody suggesting "MUST setup 2 flows", but "MUST be
capable of setting at least 2 flows".  the domain will then decide how
many flows need to be set up (by that particular device if we adopts
jonathan's discovery proposal).

-- juha

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From sip-bounces@ietf.org Mon Jul 10 07:04:49 2006
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Date: Mon, 10 Jul 2006 11:59:47 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Review of draft-ietf-sip-connected-identity-00
To: Eric Rescorla <ekr@networkresonance.com>, sip@ietf.org
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Eric,

Thanks for reading this draft. See my responses below.

John 

> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@networkresonance.com] 
> Sent: 04 July 2006 04:39
> To: sip@ietf.org
> Subject: [Sip] Review of draft-ietf-sip-connected-identity-00
> 
> BACKGROUND
> SIP allows "retargeting" in which the endpoints (on either end) of
> a call can change. So, for instance, Alice might be calling Bob
> but actually get forwarded to Charlie. This draft describes a
> mechanism for the new endpoint to notify their peer that their
> identity is not the same as the peer expected.
> 
> 
> BASIC MESSAGE FLOW
> The basic message flow without this draft looks something like this:
> 
> Alice                         Proxy                       Carol
> -----                         -----                       -----
> INVITE (To=Bob) -> 
>                                 INVITE (To=Bob) ->     
>                                                  <- OK (To=Bob)
>                    <- OK (To=Bob)                
> ACK (To=Bob) ->                
>                                 ACK (To=Bob) ->
> 
> 
> This draft adds a new set of messages for Carol to tell Alice about
> the retargeting. Like this:
> 
> 
> Alice                         Proxy                       Carol
> -----                         -----                       -----
> INVITE (To=Bob) -> 
>                                 INVITE (To=Bob) ->     
>                                                  <- OK (To=Bob)
>                    <- OK (To=Bob)                
> ACK (To=Bob) ->                
>                                 ACK (To=Bob) ->
> 
>                                          <- UPDATE (From=Carol)
>             <- UPDATE (From=Carol)
> ACK (From=Carol) ->                
>                                 ACK (From=Carol) ->
> 
> 
> 
> 
> SECURITY ISSUES
> This draft relies on the ordinary SIP authentication mechanisms
> (primarily Identity, I would imagine) in order to validate the update
> message. Unfortunately, this just tells the other side that they're
> now talking to someone new securely. They have no way of determining
> whether the retargeting was authorized or not. This is explicitly
> acknowledged in the Security Considerations section. Clearly 
> something along these lines is needed. Perhaps it could
> go in a different draft, but this does seem like the obvious one.
[JRE] There has been a lot of discussion on this point in the past, and it
is proving quite hard to come up with a solution for determining whether
retargeting was authorised or not. Whilst there are some people still
looking into this (e.g., draft-cao-sip-response-identity-00 and
draft-cao-sip-response-auth-01), the SIP group felt that it would be
appropriate to proceed with the connected-identity work, since this is well
understood and easily achievable. This does not prevent continued
investigations into authorised retargeting, but any mechanism to achieve
that will be in a separate document.

> 
> 
> EDITORIAL
> 1. I found this document fairly hard to read until I figured 
> out what was
> going on, perhaps due to my lack of familiarity with SIP. ISTM that a
> diagram like the one above would be appropriate here--some diagram
> definitely seems advisable.
[JRE] We do have this in the examples section.

> 
> 2. It would be helpful to have unique and distinctive terms for the
> UA that isn't being retargeted (Alice above) and the one that is
> (Bob/Carol). I found myself trying to figure that out. These
> terms should be called out explicitly early on.
[JRE] I get your point - the use of the terms UAC and UAS are not helpful,
since roles are reversed when the UPDATE request is sent. I will think about
it further when I work on the next draft after the meeting.

> 
> 
> S 3. 
> You should explicitly state that To doesn't change during retargeting.
> That is an unobvious behavior of SIP.
[JRE] OK

> 
> I think some diagrams of these scenarios would help.
> 
> The following paragraph is a good example of what I mean above about
> distinctive terms:
> 
>    However, call re-arrangements are not always carried out 
> using a new
>    INVITE request.  Sometimes a B2BUA performs call re-arrangements
>    using third party call control (3PCC) techniques.  With such
>    techniques the UA involved in the original call and still 
> involved in
>    the re-arranged call receives only a re-INVITE or UPDATE request in
>    the context of the original dialog between that UA and the B2BUA.
>    This forces the UA to re-negotiate the session with the new remote
>    party, but introduces a need to convey the identity of the 
> new remote
>    party to the UA.  Because there is no new INVITE request 
> (outside the
>    context of the existing dialog), techniques applicable to new calls
>    do not apply.
> 
> "the UA involved in the original call and still involved in
> the re-arranged call" is too long to process as one chunk.
[JRE] I will write instead: "For example, when call transfer is carried out
by a B2BUA the transferred UA With such techniques the UA involved in the
original call and still involved in the re-arranged call receives only a
re-INVITE or UPDATE request in the context of the original dialog between
that UA and the B2BUA."

> 
> 
> S 4.
> It should be made clear that this is background material that
> will be used later. It's not an alternative to the current
> approach, which I kind of got the impression from "Existing".
[JRE] OK
> 
> It would probably halp to name these mechanisms in some easy
> to remember way. I'm sure they already have names....
[JRE] OK

> 
> S 6.2.
> Just for clarity, you don't need to send an UPDATE/re-INVITEif you
> don't support id-change. So, the peers already need to be able to deal
> with a non-UPDATE-sending situation, right? So, I'm not sure
> I understandw hy you need to send the UPDATE/re-INVITE if nothing
> has changed...
[JRE] In order to an Authentication Service to insert an Identity header
field, to provide authentication of the fact that nothing has changed. The
source XML has this explanation, but due to an error in the XML tags it got
dropped by xmlToRfc.

> 
> 7.2.
> There is a very bad break in this table. The headers need to be on
> the same page as the columns.
[JRE] I will investigate whether xmlToRfc allows me to correct this.

> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
To: shida@ntt-at.com, Dan Wing <dwing@cisco.com>
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Shida,

I don't think it would work with edge proxies. 

John

> -----Original Message-----
> From: Shida Schubert [mailto:shida@ntt-at.com] 
> Sent: 10 July 2006 05:16
> To: Dan Wing
> Cc: Elwell, John; sip@ietf.org
> Subject: Re: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
> 
> 
> Hi;
> 
> 
> Dan Wing wrote:
> >> I generally like the draft, and I think the work is very important.
> >>     
> >
> > Me, too.
> >
> >   
> >> Anyhow, AFAIK there is no distinct way to request SIP-Identity to
> >> AS, so just because one receives a sip-identity/id-change tag does
> >> not necessary mean that sender of an UPDATE will be able to send
> >> the sip-identity back(Even if the UAS supports 
> sip-identity/id-change,
> >> the domain of the UAS might not have AS in place to provide
> >> sip-identity).
> >>     
> >
> > Agreed; if we agree we need to optimize that case, we will need
> > an additional option tag.  But, still, there isn't a way for
> > a UAS to "know" that his request will be protected with Identity.
> > So I'm not sure of the value such an additional tag would bring
> > over my suggestion.
> >   
> In past, there has been RFC which proposed the use of 
> proxy-require for
> the next hop only(First hop proxy removes the relevant 
> option-tag in the
> proxy-require). RFC 3329 is one of the example for this, and proposal
> similar to that may be considered.
> >   
> >> This is because there is no option tag to require proxy to add
> >> sip-identity,
> >> and because sip-identity is generally performed at AS, UA 
> is generally
> >> unaware of the domian's support for sip-identity.
> >>
> >> Furthermore, because its the AS that adds sip-identity(except when
> >> UA has its own certificate which I think is unlikely), 
> just because UA
> >> receives the sip-identity, does not necessary mean that UA sending
> >> the original INVITE(offerer that you have described in 
> your comment)
> >> with identity-info header supports sip-identity.
> >>     
> >
> > Even if the original UA itself doesn't support Identity, one of the
> > domain's proxies added the Identity signature and it's likely that
> > proxy may validate incoming requests containing Identity signatures.
> >
> >   
> Agree.
> 
> Regards
> Shida Schubert
> 
> > -d
> >
> >   
> >> Last note, looking at the sentence in 6.1 which is shown
> >> below, I am not sure if
> >> mandating UA to send an UPDATE to update its identity is aligned
> >> with the semantics of Supported header field. I don't really
> >> have a strong
> >> opinion on this point, but it felt a little funny to mandate
> >> a behavior with
> >> a presence of "Supported" header.
> >>
> >> Thanks & Regards
> >> Shida
> >>
> >> Dan Wing wrote:
> >>     
> >>> I hadn't noticed any followups to this issue.  I have a 
> suggestion:
> >>>
> >>>  
> >>>       
> >>>> -----Original Message-----
> >>>> From: Elwell, John [mailto:john.elwell@siemens.com]
> >>>> Sent: Tuesday, May 02, 2006 1:21 PM
> >>>> To: sip@ietf.org
> >>>> Subject: [Sip] Issue 2 in 
> draft-ietf-sip-connected-identity-00.txt
> >>>>
> >>>> The second issue concerns the fact that an UPDATE request in the
> >>>> backward direction following dialog establishment can be 
> wasted if
> >>>> the peer UA (the UAC for the INVITE transaction) does not support
> >>>> it.  It states:
> >>>>
> >>>>       "OPEN ISSUE: If the identity has has not changed
> >>>>         
> >> compared with
> >>     
> >>>>       the URI in the To header field and the UAC (for the INVITE
> >>>>       request) does not support verification of an 
> Identity header,
> >>>>       this UPDATE or re-INVITE transaction would fulfil no useful
> >>>>       purpose.  However, the UAS (for the INVITE 
> transaction) does
> >>>>       not know this.  In order to save this round trip, should we
> >>>>       have an additional option tag to indicate support 
> for receipt
> >>>>       of the Identity header on mid- dialog requests?  
> Then if this
> >>>>       option tag is not present in the INVITE request the
> >>>>         
> >> additional
> >>     
> >>>>       transaction could be omitted.  Note that the 
> id-change option
> >>>>       tag does not seem appropriate for this purpose, since a UA
> >>>>       could support a change of URI without supporting Identity."
> >>>>
> >>>> Whilst I do not wish to proliferate option tags, I do see some
> >>>> benefit in such a tag.  Any opinions?
> >>>>    
> >>>>         
> >>> If an Invite arrives with both Identity and the id-change tag,
> >>> it indicates the offerer understands both Identity and connected-
> >>> identity, so the answerer should generate an request (UPDATE or
> >>> re-INVITE).
> >>>
> >>> -d
> >>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol
> >>> Use sip-implementors@cs.columbia.edu for questions on current sip
> >>> Use sipping@ietf.org for new developments on the 
> application of sip
> >>>
> >>>
> >>>   
> >>>       
> >
> >
> >   
> 

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   From: Dean Willis <dean.willis@softarmor.com>

   Dale has noted in a separate thread that STUN breaks TCP framing,  
   possibly creating a self-inflicted DoS attack that the UA cannot  
   detect to recover from.

Actually, Anders Kristensen raised the framing problem earlier, I was
just echoing him in the later context.

Dale

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From sip-bounces@ietf.org Mon Jul 10 09:27:26 2006
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From: "Dan Wing" <dwing@cisco.com>
To: "'Elwell, John'" <john.elwell@siemens.com>, <sip@ietf.org>
Subject: RE: [Sip] Issue 2 in draft-ietf-sip-connected-identity-00.txt
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> That was not what I was trying to say. The UAC (for the 
> INVITE request) has no control over whether an Identity header 
> field is added by its domain proxy. So just because the UAS 
> receives an INVITE request with an Identity header field, does 
> not mean that the UAC put it there and would be interested in 
> receiving an UPDATE request with an Identity header field.
>
> However, I don't have a strong objection to the UAS (for the 
> INVITE) always sending back an UPDATE even if the From header 
> URI would be unchanged from that in the To header URI of 
> the INVITE request.

Cool.  

We could define another Supported tag to optimize this slightly,
(so the UAC indicates it desires connected-identity from the 
UAS) but we're still stuck with the problem that the UAS can't 
ensure its response will get an Identity header.  I don't see
another tag being that beneficial to that slight optimization.

-d


> Regards,
> 
> John 
> 
> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com] 
> > Sent: 10 July 2006 02:12
> > To: Elwell, John; sip@ietf.org
> > Subject: RE: [Sip] Issue 2 in 
> draft-ietf-sip-connected-identity-00.txt
> > 
> > > I am not sure that would work. The problem is the UAC 
> > > typically does not insert the Identity header field in a 
> > > request, this is typically done by an authentication service 
> > > at the outbound proxy. So the fact that the request went 
> > > through an authentication service and had an Identity header 
> > > field added does not mean that the response will go through 
> > > a verification service. 
> > 
> > I agree there is nothing a UA can do to ensure its request
> > gets an Identity header added.  
> > 
> > All I'm saying is that if you receive a request with an Identity
> > headers, there is a pretty darn high likelyhood that the requestor
> > is interested in having your response authenticated (by a backwards
> > request using connected-identity).  All you can do is send a request
> > back to him and hope that your request will get an Identity header
> > added by your authentication service.  If it doesn't get an Identity
> > header added, the requestor's local policy or implementation will
> > decide what to do next: accept the call or drop it (because the
> > connected-identity response lacks an Identity header).
> > 
> > I argue that an additional tag wouldn't convey any additional 
> > meaning over my proposal.
> > 
> > -d
> > 
> > > However, I am open to opinions on that.
> > > 
> > > John
> > > 
> > > > -----Original Message-----
> > > > From: Dan Wing [mailto:dwing@cisco.com] 
> > > > Sent: 08 July 2006 02:21
> > > > To: Elwell, John; sip@ietf.org
> > > > Subject: RE: [Sip] Issue 2 in 
> > > draft-ietf-sip-connected-identity-00.txt
> > > > 
> > > > I hadn't noticed any followups to this issue.  I have a 
> > suggestion:
> > > > 
> > > > > -----Original Message-----
> > > > > From: Elwell, John [mailto:john.elwell@siemens.com] 
> > > > > Sent: Tuesday, May 02, 2006 1:21 PM
> > > > > To: sip@ietf.org
> > > > > Subject: [Sip] Issue 2 in 
> > draft-ietf-sip-connected-identity-00.txt
> > > > >
> > > > > The second issue concerns the fact that an UPDATE 
> request in the
> > > > > backward direction following dialog establishment can 
> > be wasted if
> > > > > the peer UA (the UAC for the INVITE transaction) does 
> > not support
> > > > > it.  It states:
> > > > >
> > > > >       "OPEN ISSUE: If the identity has has not changed 
> > > compared with
> > > > >       the URI in the To header field and the UAC (for 
> the INVITE
> > > > >       request) does not support verification of an 
> > > Identity header,
> > > > >       this UPDATE or re-INVITE transaction would fulfil 
> > no useful
> > > > >       purpose.  However, the UAS (for the INVITE 
> > transaction) does
> > > > >       not know this.  In order to save this round trip, 
> > should we
> > > > >       have an additional option tag to indicate support 
> > > for receipt
> > > > >       of the Identity header on mid- dialog requests?  
> > > Then if this
> > > > >       option tag is not present in the INVITE request the 
> > > additional
> > > > >       transaction could be omitted.  Note that the 
> > > id-change option
> > > > >       tag does not seem appropriate for this purpose, 
> since a UA
> > > > >       could support a change of URI without supporting 
> > Identity."
> > > > >
> > > > > Whilst I do not wish to proliferate option tags, I do see some
> > > > > benefit in such a tag.  Any opinions?
> > > > 
> > > > If an Invite arrives with both Identity and the id-change tag,
> > > > it indicates the offerer understands both Identity and 
> connected-
> > > > identity, so the answerer should generate an request (UPDATE or 
> > > > re-INVITE).
> > > > 
> > > > -d
> > > > 
> > 

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From sip-bounces@ietf.org Mon Jul 10 09:57:58 2006
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A reminder that we expect (nay demand and insist) that we have all presentations mailed (and that means received not sent) by 12:30 pm today.

We are still a considerable number outstanding.

And earlier is better.

regards

Keith

Keith Drage
Lucent Technologies
drage@lucent.com
tel: +44 1793 776249

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Rohan Mahy wrote:
> 
>
> So we are back to a pure semantic question.  Does the service get to 
> decide how many flows the UA opens (up to an implementation supported 
> maximum), does the UA decide how many flows to open, or is there some 
> negotiation?   I still think the current semantics are fine, because 
> they allow a single flow to a URI that can still load-share and failover 
> to a large number of actual servers.  Of course there are others who 
> don't like that because it relies on DNS for failover.
> 
> thanks,
> -rohan
> 

I thought a bit more about this. I believe the service provider has no 
more say than providing the list of outbound URIs. The UA has to decide 
on how many flows to open, and what is the cost of this decision. I can 
imagine a policy in my laptop that makes the number of flows dependent 
on whether I am plug to the main or running on battery, so that if I am 
running on batteries I keep only one flow. My provider will never know 
what is my battery status: He can only set the maximum number of flows, 
but not the minimum.

/Miguel

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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From sip-bounces@ietf.org Mon Jul 10 10:15:47 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: Scott Lawrence <slawrence@pingtel.com>
Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-hop-limit-diagnostics-02.txt
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Getting back to the Warning issue:

Inline below

Keith

> -----Original Message-----
> From: Scott Lawrence [mailto:slawrence@pingtel.com]
> Sent: 08 June 2006 19:04
> To: Drage, Keith (Keith)
> Cc: sip@ietf.org; ahawrylyshen@ditechnetworks.com; RjS@nostrum.com
> Subject: RE: [Sip] I-D
> ACTION:draft-ietf-sip-hop-limit-diagnostics-02.txt
> 
> 
> On Thu, 2006-06-08 at 18:29 +0100, Drage, Keith (Keith) wrote:
<snipped>
> > > > 2)	Section 2.2 and elsewhere, I am not convinced that the 
> > > use of the
> > > > Warning header is consistent with RFC 3261. The clear 
> implication of
> > > > RFC 3261 is that Warning headers relate only to the session
> > > > description contained in the request.
> > > 
> > > It didn't seem worth the trouble to define a new header, 
> or to require
> > > some multipart body just to identify the last server - I'm open to
> > > suggestions.  It seems a valuable part of the diagnostic 
> information.
> > > 
> > Well if we continue to use the Warning header, this would be another
> > update of RFC 3261 to extend the usage of Warning, at least in my
> > understanding, and that would need to be defined.
> 
> RFC 3261, section 20.43 Warning says:
> 
>            The Warning header field is used to carry 
> additional information
>            about the status of a response.
> and
>            The currently-defined "warn-code"s are listed below, with a
>            recommended warn-text in English and a description 
> of their meaning.
>            These warnings describe failures induced by the 
> session description.
>            The first digit of warning codes beginning with 
> "3" indicates
>            warnings specific to SIP.  Warnings 300 through 
> 329 are reserved for
>            indicating problems with keywords in the session 
> description, 330
>            through 339 are warnings related to basic network 
> services requested
>            in the session description, 370 through 379 are 
> warnings related to
>            quantitative QoS parameters requested in the 
> session description, and
>            390 through 399 are miscellaneous warnings that do 
> not fall into one
>            of the above categories.
> 
> it seems to me that this already explicitly allows for usages 
> other than
> session description errors by allowing "miscellaneous warnings that do
> not fall into one of the above categories". 
> 
> Arguably, this could call for actually registering a code in the 39x
> range; if the WG thinks that's better, I can add an IANA 
> Considerations
> section to do that.
> 
The text you are quoting is I agree vague, but implies that 390 through 399 vcould be so used.

However if continue on in the text of the section you are reading:

   Additional "warn-code"s can be defined through IANA, as defined in
   Section 27.2.

and you go to the IANA registration section in 27.2:

   This specification establishes the Warn-codes sub-registry under
   http://www.iana.org/assignments/sip-parameters and initiates its
   population with the warn-codes listed in Section 20.43.  Additional
   warn-codes are registered by RFC publication.







Rosenberg, et. al.          Standards Track                   [Page 252]

RFC 3261            SIP: Session Initiation Protocol           June 2002


   The descriptive text for the table of warn-codes is:

   Warning codes provide information supplemental to the status code in
   SIP response messages when the failure of the transaction results
   from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.

   The "warn-code" consists of three digits.  A first digit of "3"
   indicates warnings specific to SIP.  Until a future specification
   describes uses of warn-codes other than 3xx, only 3xx warn-codes may
   be registered.

   Warnings 300 through 329 are reserved for indicating problems with
   keywords in the session description, 330 through 339 are warnings
   related to basic network services requested in the session
   description, 370 through 379 are warnings related to quantitative QoS
   parameters requested in the session description, and 390 through 399
   are miscellaneous warnings that do not fall into one of the above
   categories.

I believe this text is clear in requiring IANA to register codes that apply only to SDP errors.

The text is reproduced on the IANA registry page concerned.

Now whether we should change this or not I will leave open to discussion, but that is what it says at the moment.


> > Out of interest, what harm would occur by adding it to the Server
> > header? Alternatively, could the Contact header be allowed in a 483
> > response mean this?
> 
> The Server header is defined as providing information about 
> the software
> of the UAS, and as such the syntax doesn't allow for a field that is
> identifiably the address or host name.
> 
> I'm concerned that including a Contact is more likely to confuse
> implementations that don't expect to see them in an error response
> (especially since it is often going to be coming from a proxy).
> 

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> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> 
> OK, I'll take your word for it. If we then interpret such a
> 'stun-interval=xxx' as an indication of support for STUN (and say 'no STUN
> unless interval is provided'), I'm happy
> 
> Only trouble is: in many cases, the proxy would not know which type of NAT
> there is in between, let alone how long it caches...

If it doesn't know, then it sets the value recommended in the draft.
However, there are ways for the proxy to find out, and there are also
operators who know what the timeout is. (e.g., if the NAT is provided by
them)

-hadriel


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Subject: RE: [Sip] Outbound and multiple connections
Date: Mon, 10 Jul 2006 10:43:18 -0400
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Thread-Topic: [Sip] Outbound and multiple connections
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Joel M. Halpern" <joel@stevecrocker.com>, <sip@ietf.org>
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I wonder if the author is using an IMS orientation where the client
attaches to different edge proxies, but the same registrar proxy, but
that all gets conflated to talking about a "proxy". =20

In any case, good writing technique would suggest scanning a draft and
replacing all instances of "it" with whatever "it" is.  Saving a couple
characters is a bad trade-off for the confusion that often ensues.

Mike


> -----Original Message-----
> From: Joel M. Halpern [mailto:joel@stevecrocker.com]=20
> Sent: Saturday, July 08, 2006 6:10 PM
> To: sip@ietf.org
> Subject: [Sip] Outbound and multiple connections
>=20
> I have been watching this ongoing discussion with some bemusement.
> It appears that there are two very different views of the=20
> purpose for the multiple connections specified in=20
> draft-ietf-sip-outbound-04.txt.  There also seems to be some=20
> discrepancies about where the multiple connections go.
>=20
> With regard to purpose, the question I see is whether the=20
> reliability is for the clients benefit, or the networks.  If=20
> it is for the client's benefit, then as has been suggested=20
> the client should be able to choose whether to take advantage=20
> of it or not.  The multiple connections ought not be MUST=20
> strength.  On the other hand, if it is a mechanism to enable=20
> the network to offer better service (particularly, better=20
> service to incoming requests) then the network would=20
> reasonably expect the multiple connections, and so the usage=20
> should be a MUST.
> --Personally, I think that network techniques for providing=20
> reliability are the network's job / business, and so the=20
> multiple connections are about the clients desire for=20
> reliability, and ought to be SHOULD, or even MAY strength.
>=20
> I believe that the above alternatives lead to confusion about=20
> the draft.  Further, I think that there is related confusion=20
> about the usage of the multiple connections.
>=20
> I note that the wording of the draft is very fuzzy as to=20
> whether the multiple connections are to the same server,=20
> multiple "closely coupled" servers, or distinct servers.  I=20
> had assumed that the connections would be to distinct=20
> servers.  However, re-reading the draft I see wording like=20
> (3.1) "it [proxy] can use any one of the flows on which a=20
> successful registration has been completed."  For a proxy to=20
> have several flows would imply that the client connected to=20
> the same server multiple times.  is in order.  Wording in=20
> section 3.2 also talks about the proxy, which is the=20
> registrar, having multiple connections to the target.
> I suspect that all the wording about a single registrar is to=20
> be interpreted in conjunction with the example case (3.3) of=20
> a single registrar that is actually two "closely coupled"=20
> hosts.  This is a perfectly reasonable deployment.
> However, I think we would be much better off describing this=20
> as if it were two registrars, and indicating that=20
> coordination among the registrars is outside of the scope of=20
> this draft.  And we ought not require that they be closely coupled.
> Whatever the actual intent of the authors of 3.1, 3.2, and=20
> 3.3, we need to decide what we want it to mean, and clarify=20
> the wording so that it says that, clearly.
>=20
> Yours,
> Joel M. Halpern
>=20
> PS: I believe the reference in section 4.4 to "Section 4.2 to=20
> form a new flow" is actually a reference to 4.4.2?
>=20
>=20
>=20
> _______________________________________________
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> Use sipping@ietf.org for new developments on the application of sip
>=20

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Miguel Garcia writes:

 > The UA has to decide 
 > on how many flows to open, and what is the cost of this decision. 

what if the cost is missing half of incoming calls?  if that is not
acceptable, then you exclude configuration where outbound proxies
operate standalone.  and once you do that, outbound does not provide ANY
benefit over what we have today (reliability nor nat traversal).

in fact, outbound would provides LESS than what we have today.  in
current implementations where proxies do nat keepalives, an intelligent
nat can be build that does not actually send ANY keepalives to your
battery operated UA, but still keep nat binding open.

-- juha

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The draft still says that in case of flow failure, the UA should retry
registering every 900...1800 seconds *forever*. For an end user, even
the first 30 minutes of waiting will seem like forever, so the UA should
abort with an error much, much sooner than that. Why say forever in the
draft then?

What is the relationship between outbound, SIP identity and perhaps also
identity-coexistance? Will outgoing requests reuse the outbound flow, is
the edge proxy also the AS, and if so how will it determine the identity
of the sender of the request? The draft could say a few words (and maybe
add to the example) also about outgoing requests.

About the default keepalive timers: the TCP timer is awfully short. A
couple of minutes keepalive interval will make the battery life of any
2G or 3G mobile phone absolutely suck. What is the reason for such an
aggressive interval, if even the worst NAT generally has binding
lifetimes ten times that?

Overall, the draft says in the introduction that this is about SIP
traversing NAT better. However, the draft isn't only about that, but
tries to solve some other n-nines reliability requirement as well. These
could be separate drafts, really.

-- 
Aki Niemi
Nokia Research Center

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Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Mon, 10 Jul 2006 10:59:11 -0400
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Thread-Topic: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>,
	"Rohan Mahy" <rohan@ekabal.com>
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And what is your approach to re-registration avalanches?

Oh, wait that is not a handset problem.

Mike
=20

> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
> Sent: Monday, July 10, 2006 9:58 AM
> To: Rohan Mahy
> Cc: sip@ietf.org; Cullen Jennings (fluffy)
> Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
>=20
> Rohan Mahy wrote:
> >=20
> >
> > So we are back to a pure semantic question.  Does the=20
> service get to=20
> > decide how many flows the UA opens (up to an implementation=20
> supported=20
> > maximum), does the UA decide how many flows to open, or is=20
> there some
> > negotiation?   I still think the current semantics are=20
> fine, because=20
> > they allow a single flow to a URI that can still load-share and=20
> > failover to a large number of actual servers.  Of course there are=20
> > others who don't like that because it relies on DNS for failover.
> >=20
> > thanks,
> > -rohan
> >=20
>=20
> I thought a bit more about this. I believe the service=20
> provider has no more say than providing the list of outbound=20
> URIs. The UA has to decide on how many flows to open, and=20
> what is the cost of this decision. I can imagine a policy in=20
> my laptop that makes the number of flows dependent on whether=20
> I am plug to the main or running on battery, so that if I am=20
> running on batteries I keep only one flow. My provider will=20
> never know what is my battery status: He can only set the=20
> maximum number of flows, but not the minimum.
>=20
> /Miguel
>=20
> --=20
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From sip-bounces@ietf.org Mon Jul 10 11:00:58 2006
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The draft still says that in case of flow failure, the UA should retry
registering every 900...1800 seconds *forever*. For an end user, even
the first 30 minutes of waiting will seem like forever, so the UA should
abort with an error much, much sooner than that. Why say forever in the
draft then?

What is the relationship between outbound, SIP identity and perhaps also
identity-coexistance? Will outgoing requests reuse the outbound flow, is
the edge proxy also the AS, and if so how will it determine the identity
of the sender of the request? The draft could say a few words (and maybe
add to the example) also about outgoing requests.

About the default keepalive timers: the TCP timer is awfully short. A
couple of minutes keepalive interval will make the battery life of any
2G or 3G mobile phone absolutely suck. What is the reason for such an
aggressive interval, if even the worst NAT generally has binding
lifetimes ten times that?

Overall, the draft says in the introduction that this is about SIP
traversing NAT better. However, the draft isn't only about that, but
tries to solve some other n-nines reliability requirement as well. These
could be separate drafts, really.

-- 
Aki Niemi
Nokia Research Center

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From sip-bounces@ietf.org Mon Jul 10 11:00:58 2006
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Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Mon, 10 Jul 2006 10:59:11 -0400
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>,
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And what is your approach to re-registration avalanches?

Oh, wait that is not a handset problem.

Mike
=20

> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
> Sent: Monday, July 10, 2006 9:58 AM
> To: Rohan Mahy
> Cc: sip@ietf.org; Cullen Jennings (fluffy)
> Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
>=20
> Rohan Mahy wrote:
> >=20
> >
> > So we are back to a pure semantic question.  Does the=20
> service get to=20
> > decide how many flows the UA opens (up to an implementation=20
> supported=20
> > maximum), does the UA decide how many flows to open, or is=20
> there some
> > negotiation?   I still think the current semantics are=20
> fine, because=20
> > they allow a single flow to a URI that can still load-share and=20
> > failover to a large number of actual servers.  Of course there are=20
> > others who don't like that because it relies on DNS for failover.
> >=20
> > thanks,
> > -rohan
> >=20
>=20
> I thought a bit more about this. I believe the service=20
> provider has no more say than providing the list of outbound=20
> URIs. The UA has to decide on how many flows to open, and=20
> what is the cost of this decision. I can imagine a policy in=20
> my laptop that makes the number of flows dependent on whether=20
> I am plug to the main or running on battery, so that if I am=20
> running on batteries I keep only one flow. My provider will=20
> never know what is my battery status: He can only set the=20
> maximum number of flows, but not the minimum.
>=20
> /Miguel
>=20
> --=20
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip





From sip-bounces@ietf.org Mon Jul 10 11:05:33 2006
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From: Juha Heinanen <jh@tutpro.com>
To: Aki Niemi <aki.niemi@nokia.com>
Subject: [Sip] Outbound comments
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Aki Niemi writes:

 > About the default keepalive timers: the TCP timer is awfully short. A
 > couple of minutes keepalive interval will make the battery life of any
 > 2G or 3G mobile phone absolutely suck. What is the reason for such an
 > aggressive interval, if even the worst NAT generally has binding
 > lifetimes ten times that?

aki,

uset yleisessa kaytossa olevan natit sidoksen hengissa luokkaa 30
sekunttia.

-- juha

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



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inline.

Miguel Garcia wrote:

> Rohan Mahy wrote:
> 
>>
>>
>> So we are back to a pure semantic question.  Does the service get to 
>> decide how many flows the UA opens (up to an implementation supported 
>> maximum), does the UA decide how many flows to open, or is there some 
>> negotiation?   I still think the current semantics are fine, because 
>> they allow a single flow to a URI that can still load-share and 
>> failover to a large number of actual servers.  Of course there are 
>> others who don't like that because it relies on DNS for failover.
>>
>> thanks,
>> -rohan
>>
> 
> I thought a bit more about this. I believe the service provider has no 
> more say than providing the list of outbound URIs.

No. I think the semantic is clear. If the provider offers N URI, then 
the semantic is, "I, the provider, want you to open N flows".


> The UA has to decide 
> on how many flows to open, and what is the cost of this decision. I can 
> imagine a policy in my laptop that makes the number of flows dependent 
> on whether I am plug to the main or running on battery, so that if I am 
> running on batteries I keep only one flow. My provider will never know 
> what is my battery status: He can only set the maximum number of flows, 
> but not the minimum.

I'm not entirely convinced that, once you're doing one flow with 
sip-outbound, that the incremental cost of an additional flow has a 
major impact on battery life.

That said, I think this is starting to split hairs. The main point, I 
think, is that, under almost all conditions, if a client is offered N 
URI it opens N flows. Only for *exceptional* conditions would it open 
less. I would be OK with a strongly-qualified SHOULD. I.e., the spec 
would say SHOULD and then explain that its only under such extreme 
conditions that it would open fewer.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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On 6/30/06, Cullen Jennings <fluffy@cisco.com> wrote:
>
> On Jun 26, 2006, at 1:00 PM, Jeroen van Bemmel wrote:
>
>
> On the using a GRUU in a cert, would need to consider that any time
> my GRUU changes I need to generate a new cert - if this meant a new
> private/public key pair, this might be expensive.
>

Also consider that GRUUs will change relatively frequently for devices.

Can I get a sense on the list of who thinks storing the GRUU instead
of the AOR in the cert is worth additional consideration in the
context of this document? I would suggest leaving this out of scope
for now.

Jason

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inline.

Erkki.Koivusalo@nokia.com wrote:


> Another reason is embedded into this statement from Jonathan:
> 
> 
>>(3) constructed URIs. 
>>
>>On (3), this is definitely a possibility. However I think the best way
> 
> 
>>to mitigate is to say 'don't do that'. SIP-outbound can have some
> 
> words 
> 
>>which warn implementations that they MUST NOT insert this parameter
> 
> into 
> 
>>URI that they construct from a domain name.
> 
> 
> This suggestion seems to prevent the UA to use DHCP to discover
> and use a local SIP proxy even if such an outbound compliant
> proxy would exist. This does not sound a good idea to me.

Not in my discovery proposal at least. A UA would construct the outbound 
proxy URI from DHCP without the flag. REGISTER is sent through it. 
Resulting service-route URI has the flag. Now keepalives get used.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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Thanks for reading. Responses below.

Christer Holmberg wrote:

> Hi Jonathan,
> 
> There are a couple of issues/deployment restrictions with your
> proposal.
> 
> 1. The home proxy has, as you indicate, to know that an edge proxy is
> needed. If the edge proxy is used only for STUN keepalive that may be
> difficult, because the home proxy would need to know when keepalive
> is needed in the first place. Or, of course it could force all
> registrations to go via the edge proxy.

You lost me on this one. The idea is that, within a particular service 
provider network, the home proxy knows whether the user should be using 
an edge proxy from that provider.

> 
> 2. The home proxy and the edge proxy may be in different
> administrative domains, so the home proxy may not know the address of
> the edge proxy (maybe not even in which domain it's located).

Right. This is a good point. In this particular case, I think you would 
discover the edge proxy through DHCP rather from the AOR. Then, the 
procedure works as I described, with Service-Route providing a URI with 
the keepalive=stun and the Additional-Proxies header with more proxies.

> 
> 3. The request may traverse a numbet of proxies in any case, so the
> home proxy would have to know whether one of those is the edge proxy
> or not.

If the request arrives to the home proxy from a different proxy, it 
wouldn't redirect.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From: "Hadriel Kaplan" <HKaplan@acmepacket.com>
To: "'Michael Hammer \(mhammer\)'" <mhammer@cisco.com>,
	"'Miguel Garcia'" <Miguel.An.Garcia@nokia.com>,
	"'Rohan Mahy'" <rohan@ekabal.com>
Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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I don't put that much value in using multiple concurrent flows for avalanche
avoidance.  There are already ways proxies handle this, though they're
mostly about smoothing out registration rates over time.  And the UAs will
already be doing some smoothing on their own as they detect the failure at
different times (over a 2 min period?).  The worst-case avalanche is
probably right after a power outage, and that's not covered by multiple
flows.  So your proxies need to be able to handle it.

-hadriel


> -----Original Message-----
> From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> Sent: Monday, July 10, 2006 10:59 AM
> To: Miguel Garcia; Rohan Mahy
> Cc: sip@ietf.org; Cullen Jennings (fluffy)
> Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
> 
> And what is your approach to re-registration avalanches?
> 
> Oh, wait that is not a handset problem.
> 
> Mike
> 
> 
> > -----Original Message-----
> > From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
> > Sent: Monday, July 10, 2006 9:58 AM
> > To: Rohan Mahy
> > Cc: sip@ietf.org; Cullen Jennings (fluffy)
> > Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
> >
> > Rohan Mahy wrote:
> > >
> > >
> > > So we are back to a pure semantic question.  Does the
> > service get to
> > > decide how many flows the UA opens (up to an implementation
> > supported
> > > maximum), does the UA decide how many flows to open, or is
> > there some
> > > negotiation?   I still think the current semantics are
> > fine, because
> > > they allow a single flow to a URI that can still load-share and
> > > failover to a large number of actual servers.  Of course there are
> > > others who don't like that because it relies on DNS for failover.
> > >
> > > thanks,
> > > -rohan
> > >
> >
> > I thought a bit more about this. I believe the service
> > provider has no more say than providing the list of outbound
> > URIs. The UA has to decide on how many flows to open, and
> > what is the cost of this decision. I can imagine a policy in
> > my laptop that makes the number of flows dependent on whether
> > I am plug to the main or running on battery, so that if I am
> > running on batteries I keep only one flow. My provider will
> > never know what is my battery status: He can only set the
> > maximum number of flows, but not the minimum.
> >
> > /Miguel
> >
> > --
> > Miguel A. Garcia           tel:+358-50-4804586
> > sip:miguel.an.garcia@openlaboratory.net
> > Nokia Research Center      Helsinki, Finland
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Jason Fischl wrote:

> On 6/30/06, Cullen Jennings <fluffy@cisco.com> wrote:
>
>>
>> On Jun 26, 2006, at 1:00 PM, Jeroen van Bemmel wrote:
>>
>>
>> On the using a GRUU in a cert, would need to consider that any time
>> my GRUU changes I need to generate a new cert - if this meant a new
>> private/public key pair, this might be expensive.
>

Just piggybacking on this comment. First, try doing an openssl genrsa --
it doesn't take very long to create a new key pair for all but really large
keys. Second, I don't understand why one would need to create new keys
for new identities. A certificate is nothing more than a name/key binding
at its heart. There's nothing in the certificate that can sniff out 
whether the
key is fresh or stale, and in the abstract there's nothing wrong with having
a single key signing for a variety of different identities (*).

       Mike

[*] context matters here, of course, but my guess is that for the average UA
      it matters very little.

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Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Mon, 10 Jul 2006 11:43:09 -0400
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>,
	"Miguel Garcia" <Miguel.An.Garcia@nokia.com>,
	"Rohan Mahy" <rohan@ekabal.com>
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That 2-minute period could be smoothed out over a much longer period of
time if there were multiple flows.

Your power outage example assumes a specific topology relative to where
power was lost.  I disagree with assuming a poor network design as
argument for multiple flows not handling the outage.

I do agree that proxies need to handle such overloads, but that leads
right back to the service being out for longer than the acceptable (?) 2
minute period.

I am not adverse to wireless networks choosing to make the tradeoff of
battery time to increased possibility of missing incoming calls, I just
think that there needs to be agreement between the network and client as
to what is acceptable and what those tradeoffs are, such that neither
has negative impacts.

Mike


> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: Monday, July 10, 2006 11:25 AM
> To: Michael Hammer (mhammer); 'Miguel Garcia'; 'Rohan Mahy'
> Cc: sip@ietf.org; Cullen Jennings (fluffy)
> Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
>=20
> I don't put that much value in using multiple concurrent=20
> flows for avalanche avoidance.  There are already ways=20
> proxies handle this, though they're mostly about smoothing=20
> out registration rates over time.  And the UAs will already=20
> be doing some smoothing on their own as they detect the=20
> failure at different times (over a 2 min period?).  The=20
> worst-case avalanche is probably right after a power outage,=20
> and that's not covered by multiple flows.  So your proxies=20
> need to be able to handle it.
>=20
> -hadriel
>=20
>=20
> > -----Original Message-----
> > From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> > Sent: Monday, July 10, 2006 10:59 AM
> > To: Miguel Garcia; Rohan Mahy
> > Cc: sip@ietf.org; Cullen Jennings (fluffy)
> > Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
> >=20
> > And what is your approach to re-registration avalanches?
> >=20
> > Oh, wait that is not a handset problem.
> >=20
> > Mike
> >=20
> >=20
> > > -----Original Message-----
> > > From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
> > > Sent: Monday, July 10, 2006 9:58 AM
> > > To: Rohan Mahy
> > > Cc: sip@ietf.org; Cullen Jennings (fluffy)
> > > Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
> > >
> > > Rohan Mahy wrote:
> > > >
> > > >
> > > > So we are back to a pure semantic question.  Does the
> > > service get to
> > > > decide how many flows the UA opens (up to an implementation
> > > supported
> > > > maximum), does the UA decide how many flows to open, or is
> > > there some
> > > > negotiation?   I still think the current semantics are
> > > fine, because
> > > > they allow a single flow to a URI that can still load-share and=20
> > > > failover to a large number of actual servers.  Of=20
> course there are=20
> > > > others who don't like that because it relies on DNS for=20
> failover.
> > > >
> > > > thanks,
> > > > -rohan
> > > >
> > >
> > > I thought a bit more about this. I believe the service=20
> provider has=20
> > > no more say than providing the list of outbound URIs. The=20
> UA has to=20
> > > decide on how many flows to open, and what is the cost of this=20
> > > decision. I can imagine a policy in my laptop that makes=20
> the number=20
> > > of flows dependent on whether I am plug to the main or running on=20
> > > battery, so that if I am running on batteries I keep only=20
> one flow.=20
> > > My provider will never know what is my battery status: He=20
> can only=20
> > > set the maximum number of flows, but not the minimum.
> > >
> > > /Miguel
> > >
> > > --
> > > Miguel A. Garcia           tel:+358-50-4804586
> > > sip:miguel.an.garcia@openlaboratory.net
> > > Nokia Research Center      Helsinki, Finland
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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Juha Heinanen wrote:
> Miguel Garcia writes:
> 
>  > The UA has to decide 
>  > on how many flows to open, and what is the cost of this decision. 
> 
> what if the cost is missing half of incoming calls?  if that is not
> acceptable, then you exclude configuration where outbound proxies
> operate standalone.  and once you do that, outbound does not provide ANY
> benefit over what we have today (reliability nor nat traversal).

If the user does not want to miss any calls, he can install a policy 
that uses several flows. The draft allows you to do it. What the draft 
does not allow me to do is to install a policy where I use only one flow.

> 
> in fact, outbound would provides LESS than what we have today.  in
> current implementations where proxies do nat keepalives, an intelligent
> nat can be build that does not actually send ANY keepalives to your
> battery operated UA, but still keep nat binding open.

True, in that case you probably don't need STUN or SIP outbound, so no 
need to discuss it either. The same applies if you can get a non-private 
IP address to your UA. We don't discuss the lack of need of the outbound 
draft in those cases.

/Miguel

> 
> -- juha

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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From sip-bounces@ietf.org Mon Jul 10 12:19:49 2006
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Jonathan Rosenberg wrote:
> inline.
> 
> I'm not entirely convinced that, once you're doing one flow with 
> sip-outbound, that the incremental cost of an additional flow has a 
> major impact on battery life.

It does, not because of the flow itself, but because of the keepalives 
every 30 seconds.


> That said, I think this is starting to split hairs. The main point, I 
> think, is that, under almost all conditions, if a client is offered N 
> URI it opens N flows. Only for *exceptional* conditions would it open 
> less. I would be OK with a strongly-qualified SHOULD. I.e., the spec 
> would say SHOULD and then explain that its only under such extreme 
> conditions that it would open fewer.

I am fine with this proposal. And I wouldn't mine to add some paragraph 
describing the conditions under which the UA would establish the N flows.

/Miguel

> Thanks,
> Jonathan R.

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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From sip-bounces@ietf.org Mon Jul 10 13:22:23 2006
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Folks,

per my presentation a minute ago in SIP, if you are interested in 
meeting at some point during the following days to work on
the issue I presented, which relates to SigComp and SIP, please send me 
an email.

Thanks,

Gonzalo

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Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
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So, could one infer from your statement, that if a network uses multiple
flows, but not keepalives, because it knows there are no NATs, then that
would be an acceptable configuration?

Mike


> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
> Sent: Monday, July 10, 2006 12:18 PM
> To: Jonathan Rosenberg (jdrosen)
> Cc: sip@ietf.org; Rohan Mahy; Cullen Jennings (fluffy)
> Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
>=20
> Jonathan Rosenberg wrote:
> > inline.
> >=20
> > I'm not entirely convinced that, once you're doing one flow with=20
> > sip-outbound, that the incremental cost of an additional flow has a=20
> > major impact on battery life.
>=20
> It does, not because of the flow itself, but because of the=20
> keepalives every 30 seconds.
>=20
>=20
> > That said, I think this is starting to split hairs. The=20
> main point, I=20
> > think, is that, under almost all conditions, if a client is=20
> offered N=20
> > URI it opens N flows. Only for *exceptional* conditions=20
> would it open=20
> > less. I would be OK with a strongly-qualified SHOULD. I.e.,=20
> the spec=20
> > would say SHOULD and then explain that its only under such extreme=20
> > conditions that it would open fewer.
>=20
> I am fine with this proposal. And I wouldn't mine to add some=20
> paragraph=20
> describing the conditions under which the UA would establish=20
> the N flows.
>=20
> /Miguel
>=20
> > Thanks,
> > Jonathan R.
>=20
> --=20
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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>=20

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Michael Hammer (mhammer) wrote:
> So, could one infer from your statement, that if a network uses multiple
> flows, but not keepalives, because it knows there are no NATs, then that
> would be an acceptable configuration?
> 

Yes, although I am not sure the benefit outbound will give you in this 
case, but it would be fine.

/Miguel

> Mike
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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From sip-bounces@ietf.org Mon Jul 10 14:25:34 2006
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From: "Hadriel Kaplan" <HKaplan@acmepacket.com>
To: "'Michael Hammer \(mhammer\)'" <mhammer@cisco.com>
Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Mon, 10 Jul 2006 14:23:28 -0400
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My point was just that it's not a panacea for the avalanche issue.  It helps
in various failure modes, doesn't in others.  The power outage was just an
extreme failure mode example; there are others.  

I know everyone's mileage varies, but 2 Minutes lets a heck of a lot of UAs
register, from what I've seen of registrar reg/sec load handling.  Like at
least in the thousands.  If more than thousands of UAs are registering anew,
the odds are it was either because they've all rebooted due to regional
power loss, or their primary proxy died/lost-comms.  In the former case
multiple flows would not help.  In the latter case outbound would have all
the UAs connecting to a backup for that failed primary in 90 seconds (ie,
they use the secondary proxy, but then create a replacement for the failed
flow).  So the avalanche is still there, just for another proxy, no?

-hadriel


> -----Original Message-----
> From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> Sent: Monday, July 10, 2006 11:43 AM
> To: Hadriel Kaplan; Miguel Garcia; Rohan Mahy
> Cc: sip@ietf.org; Cullen Jennings (fluffy)
> Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
> 
> That 2-minute period could be smoothed out over a much longer period of
> time if there were multiple flows.
> 
> Your power outage example assumes a specific topology relative to where
> power was lost.  I disagree with assuming a poor network design as
> argument for multiple flows not handling the outage.
> 
> I do agree that proxies need to handle such overloads, but that leads
> right back to the service being out for longer than the acceptable (?) 2
> minute period.
> 
> I am not adverse to wireless networks choosing to make the tradeoff of
> battery time to increased possibility of missing incoming calls, I just
> think that there needs to be agreement between the network and client as
> to what is acceptable and what those tradeoffs are, such that neither
> has negative impacts.
> 
> Mike


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Subject: RE: [Sip] Review of draft-audet-sip-sips-guidelines-02
Date: Mon, 10 Jul 2006 13:41:27 -0500
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FA> -----Original Message-----
FA> From: Eric Rescorla [mailto:ekr@networkresonance.com]=20
FA> Sent: Friday, July 07, 2006 5:50 PM
FA> To: sip@ietf.org
FA> Subject: [Sip] Review of draft-audet-sip-sips-guidelines-02
FA>=20
FA> This draft attempts to clarify the underspecification of=20
FA> SIPS in RFC 3261. This seems like a good idea in principle,=20
FA> but at least from the security guy perspective, I worry=20
FA> that this draft isn't completely baked yet. I do hasten to=20
FA> add that this is the security guy perspective, so there are=20
FA> probably SIP complexities I'm missing.
FA>=20
FA> That said, ISTM that there are a number of problems that need to be
FA> disentangled:
FA>=20
FA> 1. What a UA which is trying to contact a given URI=20
FA>    can expect in terms of how their messages will be
FA>    handled.
FA> 2. What a UA can safely publish as their AOR and how
FA>    to do it.
FA> 3. What a UA which receives a message can determine
FA>    about how the message was handled.
FA>=20
FA> This draft handles all of these issues together and I think=20
FA> it would be best to handle them separately. Taking a step=20
FA> back, I think we want the following:
FA>=20
FA> 1. A UA which tries to dereference a SIPS URI knows that
FA>    as long as its dealing with compliant proxies its messages
FA>    will be transmitted securely all the way to the peer.
FA>=20
FA> 2. A UA can make itself available in the following ways:
FA>    - Secure only
FA>    - Secure or insecure but hopefully if you're dealing
FA>      with a secure peer then you both get secure.
FA>    - Insecure only.
FA>    So, AOR advertisement should support this.
FA>=20
FA> 3. A UA can determine (again assuming it's dealing with compliant
FA>    proxies) whether the path from the peer was secure.
FA>=20
FA> Do I have these requirements right?

Thanks Eric.

You are putting the finger on exactly where the problem is:
Sips was never really well-defined. The requirements where never
explicit.
We are faced with the "fait accomplit" of RFC 3261.=20

What I am seeing now is the following:
- There is a group of people now that are trying to implement SIP-TLS
and
  can't make any sense of RFC 3261, and need guidance.
- There is another group of people who see flaws with the basic
requirements.

It's going to be tough to reconcile this...

FA>=20
FA> Issue (1) needs to be dealt with by imposing rules about=20
FA> what the proxies do. I.e., that when presented with SIPS=20
FA> they use TLS in the outgoing direction. The edge case--and=20
FA> the one that's creating most of the heartbreak--appears to=20
FA> be what happens inside of trust domains.=20
FA>=20
FA> Based on the discussion in this draft, Issue (2) breaks=20
FA> down into two cases:
FA>=20
FA> (a) What you advertise when you register.
FA> (b) What you put in your own headers (i.e., the To).
FA>=20
FA> Case (a) seems like it's relatively straightforward using=20
FA> the rules described in S 5. It's case (b) that's tricky,=20
FA> due to the asymmetry of "SIPS means you must use TLS to=20
FA> your first hop proxy" and "SIPS means you may or may not=20
FA> use TLS between the last hop proxy and the final=20
FA> destination".  Since that last link is about to become the=20
FA> first hop, this seems highly problematic.
FA>=20
FA> ISTM that there are two natural fixes for this:
FA>=20
FA> (a) Decree that SIPS means TLS all the way (this is a lot less
FA>     problematic in the face of outbound than it otherwise
FA>     would be.)
FA> (b) Decree that it's none of our business what happens inside
FA>     people's trust boundaries and so SIPS only means secure
FA>     in between them.
FA>=20
FA> This draft, however, takes neither approach and instead=20
FA> attempts to thread the needle between these. I'm not sure=20
FA> that this is practical--and it's certainly creating a lot=20
FA> of complexity.

Basically, I'm fetching for opinions on this.

Keep in mind that this is an individual draft still. We need people
to express opinions on this.=20

As it stands, this draft is basically pointing out problems with
RFC 3261.

FA>=20
FA> Issue (3) seems like it ought to be solved the same way=20
FA> that it's ntended to be solved in RFC 3207 (STARTTLS):=20
FA> examine the Via lines or something like them. I would=20
FA> ignore the URI entirely here, especially in view of the=20
FA> confusion noted above.
FA>=20
FA>=20
FA> DETAILED COMMENTS:
FA> S 5.1:
FA> I'm not sure why you need TLS transport to do a=20
FA> registration of a SIPS contact.

My thinking is that if the registration itself is not secure, how can
the
Resources that are binding themselves to a sips URI with that
registration be
secure?

FA> S 6.
FA>=20
FA>    If a UAS does not support SIPS, it MUST reject a request over TLS
FA>    with response code 416.=20
FA>=20
FA> I don't understand how this can happen. Can you explain?

Good point. It can't.


FA>    Upon receiveing a 416 a UAC SHOULD not re-
FA>    attempt the request with a SIP URI by automatically replacing the
FA>    SIPS scheme with a SIP scheme.  If the UAC does=20
FA> re-attempt the call
FA>    with a SIP URI, it SHOULD inform to the user that the=20
FA> security level
FA>    is downgraded.
FA>=20
FA> Shouldn't this last SHOULD be a MUST? It seems like a=20
FA> security condition.

I think that's from RFC 3261 verbatim.

FA>=20
FA>    The presence of a SIPS Request-URI does not necessarily=20
FA> indicate that
FA>    the request was sent end-to-end securely.=20
FA>=20
FA> The presence by the time you receive it, right? CAuse it=20
FA> should mean that at the sender....

Correct.

FA>=20
FA>=20
FA>    26.4.4/RFC 3261 also explains that S/MIME may also be used by the
FA>    originating UAC to ensure that the original form of the To header
FA>    field is carried end-to-end.  While not specifically mentioned in
FA>    26.4.4/RFC 3261, this is meant to imply that [RFC3893]=20
FA> would be used
FA>    to "tunnel" important headers (such as To and From) in=20
FA> an encrypted
FA>    S/MIME body, replicating the information in the SIP message, and
FA>    allowing the UAS to validate the content of those=20
FA> important headers.
FA>    While this approach is certainly legal, another approach=20
FA> is to use
FA>    the SIP Identity mechanism defined in=20
FA> [I-D.ietf-sip-identity].  SIP
FA>    Identity creates a signed identity digest which includes, amongst
FA>    other things, the AOR of the sender (from the From=20
FA> header) and the
FA>    AOR of the original destination (from the To header).  It is
FA>    RECOMMENDED that a UAC use the mechanism in=20
FA> [I-D.ietf-sip-identity]
FA>    instead of the one defined in RFC 3893.
FA>=20
FA> You say "encrypted" here, but you really mean signed, right?

I believe it's all of them for S/MIME.

No encryption for SIP Identity.

FA>=20
FA> S 13.
FA> I'm not sure I find this that helpful.

It's helpful for the author.... The idea would be to remove it later.

FA> S 14.
FA> It seems to me that most of these rules have security implications.

Yes. It should really say TBD....


FA> -Ekr
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>=20
FA>    =20
FA>=20
FA>=20
FA>=20
FA>=20
FA>   =20
FA>=20
FA> _______________________________________________
FA> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
FA> This list is for NEW development of the core SIP Protocol=20
FA> Use sip-implementors@cs.columbia.edu for questions on=20
FA> current sip Use sipping@ietf.org for new developments on=20
FA> the application of sip
FA>=20

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From sip-bounces@ietf.org Mon Jul 10 14:52:42 2006
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Subject: Re: [Sip] Review of draft-audet-sip-sips-guidelines-02
References: <1ECE0EB50388174790F9694F77522CCF0B4E2F42@zrc2hxm0.corp.nortel.com>
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Date: Mon, 10 Jul 2006 11:49:59 -0700
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"Francois Audet" <audet@nortel.com> writes:
> FA> Issue (3) seems like it ought to be solved the same way 
> FA> that it's ntended to be solved in RFC 3207 (STARTTLS): 
> FA> examine the Via lines or something like them. I would 
> FA> ignore the URI entirely here, especially in view of the 
> FA> confusion noted above.
> FA> 
> FA> 
> FA> DETAILED COMMENTS:
> FA> S 5.1:
> FA> I'm not sure why you need TLS transport to do a 
> FA> registration of a SIPS contact.
>
> My thinking is that if the registration itself is not secure, how can
> the
> Resources that are binding themselves to a sips URI with that
> registration be
> secure?

Well, what if it happened on some walled garden network?

> FA> S 13.
> FA> I'm not sure I find this that helpful.
>
> It's helpful for the author.... The idea would be to remove it later.

That makes sense...

-Ekr

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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
From: Aki Niemi <aki.niemi@nokia.com>
To: ext Juha Heinanen <jh@tutpro.com>
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On ma, 2006-07-10 at 17:53 +0300, ext Juha Heinanen wrote:
> in fact, outbound would provides LESS than what we have today. 

>From my point of view, the most important thing missing today is getting
SIP over TCP to work in presence of NATs. This is what outbound
provides, which IMO is more than what's available today.

-- 
Aki Niemi
Nokia Research Center


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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Rohan Mahy" <rohan@ekabal.com>
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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Mon, 10 Jul 2006 21:15:52 +0200
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Rohan,

>> If the system is architected as Juha was describing (2 independent 
>> proxies, UA missing 50% of its calls when only setting up 1 flow), the 
>> chance at missed requests for such a UA is much higher than double that 
>> of a single flow system. I feel that outbound should certainly not 
>> mandate such an architecture.
>
> The UA doesn't miss 50% of its calls in this case.  If only one flow is 
> registered, the registrar/home proxy will forward incoming requests only 
> over that flow.  Its just that the UA is not playing nicely with the 
> proxies, as they would like to load balance over two flows, but they have 
> to forward all the traffic for the UA over one flow instead.
>

As long as things work as you describe here, it's fine with me. The 50% was 
based on Juha's description of the system he envisioned earlier in this 
thread


>> Currently in SIP the assumption is that a single registration suffices to 
>> receive all incoming requests for a given AoR. If we break that, it means 
>> that current UAs won't be able to register with "outbound-enhanced" 
>> registrars and edge proxies. That to me is not acceptable.
>
> there is no requirement in outbound that pre-outbound registrations will 
> not work.
>

True, but this is not the same as a requirement that says that pre-outbound 
registrations MUST still work (and receive 100% of service, not 50%). I 
thought this was implicitly obvious, but recent discussions suggest 
otherwise

>> In other words, a requirement is missing: "Support UAs that do not 
>> implement this extension"
>
> What amount of "support" do you envision?

Just no less - statistically speaking - than a UA would get from a REGISTER 
today.

Regards,

Jeroen

>
> thanks,
> -rohan
> 


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From sip-bounces@ietf.org Mon Jul 10 15:26:15 2006
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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
Date: Mon, 10 Jul 2006 21:24:13 +0200
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Acceptable with a minor change in wording: "if a network OFFERS TO use 
multiple flows".

I compare this with an incoming INVITE that offers both audio and video. An 
UA might accept the audio but refuse the video, for whatever reason. The 
communication experience will be different (some might say "less"), but it's 
the UA's decision.

If there are no keepalives, then that is one compelling reason less to 
establish fewer flows than offered. So phrases like "the UA SHOULD establish 
a flow to each URI in the outbound proxy set" and "it is RECOMMENDED 
that..." are fine.

Regards,

Jeroen

----- Original Message ----- 
From: "Miguel Garcia" <Miguel.An.Garcia@nokia.com>
To: "ext Michael Hammer (mhammer)" <mhammer@cisco.com>
Cc: <sip@ietf.org>; "Rohan Mahy" <rohan@ekabal.com>; "Cullen Jennings 
(fluffy)" <fluffy@cisco.com>
Sent: Monday, July 10, 2006 7:30 PM
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04


> Michael Hammer (mhammer) wrote:
>> So, could one infer from your statement, that if a network uses multiple
>> flows, but not keepalives, because it knows there are no NATs, then that
>> would be an acceptable configuration?
>>
>
> Yes, although I am not sure the benefit outbound will give you in this 
> case, but it would be fine.
>
> /Miguel
>
>> Mike
>>
>
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>
> _______________________________________________
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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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Jeroen van Bemmel writes:

 > True, but this is not the same as a requirement that says that pre-outbound 
 > registrations MUST still work (and receive 100% of service, not 50%). I 
 > thought this was implicitly obvious, but recent discussions suggest 
 > otherwise.

i haven't said that pre-outbound registrations would not work.  what
would not work is that if UA claims outbound support, but establishes
only one flow when two would be the minimum for a particular domain.

-- juha

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From sip-bounces@ietf.org Mon Jul 10 15:37:34 2006
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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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>So, could one infer from your statement, that if a network uses multiple
>flows, but not keepalives, because it knows there are no NATs, then that
>would be an acceptable configuration?

Besides being outside of the scope of outbound, the current text also makes 
it impossible to do optimizations like this (ie telling the UA not to do 
keepalive) because of the text in 7.1:

"When a URI is created that refers to a SIP device that supports STUN
   as described in this section, the 'keepalive' URI parameter, as
   defined in Section 10 MUST be added to the URI, with a value of
   'stun'."

Regards,

jeroen

> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
> Sent: Monday, July 10, 2006 12:18 PM
> To: Jonathan Rosenberg (jdrosen)
> Cc: sip@ietf.org; Rohan Mahy; Cullen Jennings (fluffy)
> Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
>
> Jonathan Rosenberg wrote:
> > inline.
> >
> > I'm not entirely convinced that, once you're doing one flow with
> > sip-outbound, that the incremental cost of an additional flow has a
> > major impact on battery life.
>
> It does, not because of the flow itself, but because of the
> keepalives every 30 seconds.
>
>
> > That said, I think this is starting to split hairs. The
> main point, I
> > think, is that, under almost all conditions, if a client is
> offered N
> > URI it opens N flows. Only for *exceptional* conditions
> would it open
> > less. I would be OK with a strongly-qualified SHOULD. I.e.,
> the spec
> > would say SHOULD and then explain that its only under such extreme
> > conditions that it would open fewer.
>
> I am fine with this proposal. And I wouldn't mine to add some
> paragraph
> describing the conditions under which the UA would establish
> the N flows.
>
> /Miguel
>
> > Thanks,
> > Jonathan R.
>
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.an.garcia@openlaboratory.net
> Nokia Research Center      Helsinki, Finland
>
> _______________________________________________
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> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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I do not think it is the UAs job to provide support for the operators 
architecture.
Nor is it "outbound"s job to mandate specific redundancy mechanisms 
(it currently seems to do so, although I think all of that is 
intended as examples.)

Yours,
Joel

At 03:29 PM 7/10/2006, Juha Heinanen wrote:
>Jeroen van Bemmel writes:
>
>  > True, but this is not the same as a requirement that says that 
> pre-outbound
>  > registrations MUST still work (and receive 100% of service, not 50%). I
>  > thought this was implicitly obvious, but recent discussions suggest
>  > otherwise.
>
>i haven't said that pre-outbound registrations would not work.  what
>would not work is that if UA claims outbound support, but establishes
>only one flow when two would be the minimum for a particular domain.
>
>-- juha
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Mon Jul 10 15:59:57 2006
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To: Juha Heinanen <jh@tutpro.com>
From: "Joel M. Halpern" <joel@stevecrocker.com>
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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My problem is that this couples the network requirements for 
resiliency with the UA's preferences and with the UA's need to get 
through a firewall (often not even under the networks control.)
Outbound is not the place for the network to get the UA to help with 
redundancy.

Yours,
Joel

At 03:47 PM 7/10/2006, Juha Heinanen wrote:
>Joel M. Halpern writes:
>
>  > Nor is it "outbound"s job to mandate specific redundancy mechanisms
>  > (it currently seems to do so, although I think all of that is
>  > intended as examples.)
>
>outbound can support any number of redundancy mechanisms.  if a domain
>chooses to require two flows from a particular UA, then it is up to
>outbound UA to do so.  another domain may have different policy.  as
>long as outbound UA is able to setup at least two flows everything is
>ok.
>
>-- juha


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From: Juha Heinanen <jh@tutpro.com>
To: "Joel M. Halpern" <joel@stevecrocker.com>
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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Joel M. Halpern writes:

 > Outbound is not the place for the network to get the UA to help with 
 > redundancy.

ok, in case i was misled to believe so.  it would have been years of
migration to outbound anyhow and without normative reference to
discovery, its deployment would have been a nightmare in open
environment.  so i'll rest my case and go back to write code.

-- juha

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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: Re: [Sip] Ready for WGLC on Outbound?
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Dean Willis wrote:
> 
> On Jul 8, 2006, at 12:23 PM, Juha Heinanen wrote:
> 
>>
>>> Or just hope it won't melt when you send it a STUN packet, meaning that
>>> you "require the operator to get it right".
>>
>>> The problem as I see it is we have no way of easily detecting an error
>>> if somebody mistakenly lists a non-compliant proxy as an outbound proxy
>>> in DHCP or some other discovery mechanism. I hate leaping off a trapeze
>>> without a safety net . . .
>>
>> no properly implemented proxy melts no matter what you sent to it.  this
>> kind of arguments are completely bogus.
> 
> Dale has noted in a separate thread that STUN breaks TCP framing,

One solution to fix this TCP framing problem would be to add an
attribute in the STUN request, just before the FINGERPRINT attribute,
that contain this octet string:

0x00000D0A6C3A32340D0A0D0A

This embeds the character string "\r\nl:24\r\n\r\n", which should be
enough to have the SIP parser skip the entire STUN message, and return
an error (same than with UDP).

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Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-hop-limit-diagnostics-02.txt
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On Mon, 2006-07-10 at 15:13 +0100, Drage, Keith (Keith) wrote:
> Getting back to the Warning issue:


> and you go to the IANA registration section in 27.2:
> 
>    This specification establishes the Warn-codes sub-registry under
>    http://www.iana.org/assignments/sip-parameters and initiates its
>    population with the warn-codes listed in Section 20.43.  Additional
>    warn-codes are registered by RFC publication.

>    The descriptive text for the table of warn-codes is:

>    Warning codes provide information supplemental to the status code in
>    SIP response messages when the failure of the transaction results
>    from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.
> 
>    The "warn-code" consists of three digits.  A first digit of "3"
>    indicates warnings specific to SIP.  Until a future specification
>    describes uses of warn-codes other than 3xx, only 3xx warn-codes may
>    be registered.
> 
>    Warnings 300 through 329 are reserved for indicating problems with
>    keywords in the session description, 330 through 339 are warnings
>    related to basic network services requested in the session
>    description, 370 through 379 are warnings related to quantitative QoS
>    parameters requested in the session description, and 390 through 399
>    are miscellaneous warnings that do not fall into one of the above
>    categories.
> 
> I believe this text is clear in requiring IANA to register codes that apply only to SDP errors.
> 
> The text is reproduced on the IANA registry page concerned.
> 
> Now whether we should change this or not I will leave open to discussion, but that is what it says at the moment.

Fair enough.

I've explained why I think the Server and Contact headers are not good
for identifying the responding server, and I think that the value of
identifying the responding server is clear.  
 
What does the group think is better:

- define a new mechanism 
- expanding the usage of Warning

 
-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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Agree, it is not a panacea, but a means of mitigation.

I tend to feel things have scaled well when we start talking millions.
But, I know of some systems that think in terms of tens to hundreds of
thousands.

Of course, you would need to detect and re-establish IP connectivity.

As for the avalanche on a replacement proxy in the power-out region
(versus the one outside that region), you could spread the load across a
much longer time period (tens of minutes, if you like), while the backup
proxy could deliver calls following the connectivity re-establishment.

Mike


> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: Monday, July 10, 2006 2:23 PM
> To: Michael Hammer (mhammer)
> Cc: sip@ietf.org
> Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
>=20
> My point was just that it's not a panacea for the avalanche=20
> issue.  It helps in various failure modes, doesn't in others.=20
>  The power outage was just an extreme failure mode example;=20
> there are others. =20
>=20
> I know everyone's mileage varies, but 2 Minutes lets a heck=20
> of a lot of UAs register, from what I've seen of registrar=20
> reg/sec load handling.  Like at least in the thousands.  If=20
> more than thousands of UAs are registering anew, the odds are=20
> it was either because they've all rebooted due to regional=20
> power loss, or their primary proxy died/lost-comms.  In the=20
> former case multiple flows would not help.  In the latter=20
> case outbound would have all the UAs connecting to a backup=20
> for that failed primary in 90 seconds (ie, they use the=20
> secondary proxy, but then create a replacement for the failed=20
> flow).  So the avalanche is still there, just for another proxy, no?
>=20
> -hadriel
>=20
>=20
> > -----Original Message-----
> > From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> > Sent: Monday, July 10, 2006 11:43 AM
> > To: Hadriel Kaplan; Miguel Garcia; Rohan Mahy
> > Cc: sip@ietf.org; Cullen Jennings (fluffy)
> > Subject: RE: [Sip] Re: A coment to draft-ietf-sip-outbound-04
> >=20
> > That 2-minute period could be smoothed out over a much=20
> longer period=20
> > of time if there were multiple flows.
> >=20
> > Your power outage example assumes a specific topology relative to=20
> > where power was lost.  I disagree with assuming a poor=20
> network design=20
> > as argument for multiple flows not handling the outage.
> >=20
> > I do agree that proxies need to handle such overloads, but=20
> that leads=20
> > right back to the service being out for longer than the=20
> acceptable (?)=20
> > 2 minute period.
> >=20
> > I am not adverse to wireless networks choosing to make the=20
> tradeoff of=20
> > battery time to increased possibility of missing incoming calls, I=20
> > just think that there needs to be agreement between the network and=20
> > client as to what is acceptable and what those tradeoffs are, such=20
> > that neither has negative impacts.
> >=20
> > Mike
>=20

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FYI.

-------- Original Message --------
Subject: P2P SIP Ad-hoc on Friday at 12:30 at Room 519A
Date: Tue, 11 Jul 2006 00:50:37 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
To: sipping <sipping@ietf.org>
CC: Mary Barnes <mary.barnes@nortel.com>

Folks,

the ad-hoc meeting on SIP peer-to-peer will take place on Friday, at
12:30, at Room 519A. The room is reserved until 15:00.

Cheers,

Gonzalo



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On Jul 10, 2006, at 5:05 PM, Scott Lawrence wrote:

> On Mon, 2006-07-10 at 15:13 +0100, Drage, Keith (Keith) wrote:
>> Getting back to the Warning issue:
>
> ...
>>
>> I believe this text is clear in requiring IANA to register codes  
>> that apply only to SDP errors.
>>
>> The text is reproduced on the IANA registry page concerned.
>>
>> Now whether we should change this or not I will leave open to  
>> discussion, but that is what it says at the moment.
>
> Fair enough.
>
> I've explained why I think the Server and Contact headers are not good
> for identifying the responding server, and I think that the value of
> identifying the responding server is clear.
>
> What does the group think is better:
>
> - define a new mechanism
> - expanding the usage of Warning

I don't feel strongly, but if we allow the scope to creep to other  
response conditions, then I suspect
we'll be better off with a new mechanism and propose that carrying  
that data (and any other meta-data
we might discover useful) in the body.

RjS

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Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04
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Joel M. Halpern writes:

 > Nor is it "outbound"s job to mandate specific redundancy mechanisms 
 > (it currently seems to do so, although I think all of that is 
 > intended as examples.)

outbound can support any number of redundancy mechanisms.  if a domain
chooses to require two flows from a particular UA, then it is up to
outbound UA to do so.  another domain may have different policy.  as
long as outbound UA is able to setup at least two flows everything is
ok.

-- juha

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By this interpretation, a network could not offer redundant / backup 
connectivity (via outbound) to UAs which want to take advantage of 
that without also mandating that all UAs make connect  to all the entities.
Yours,
Joel

At 03:47 PM 7/10/2006, Juha Heinanen wrote:
>Joel M. Halpern writes:
>
>  > Nor is it "outbound"s job to mandate specific redundancy mechanisms
>  > (it currently seems to do so, although I think all of that is
>  > intended as examples.)
>
>outbound can support any number of redundancy mechanisms.  if a domain
>chooses to require two flows from a particular UA, then it is up to
>outbound UA to do so.  another domain may have different policy.  as
>long as outbound UA is able to setup at least two flows everything is
>ok.
>
>-- juha
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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On Jul 10, 2006, at 1:22 AM, Dean Willis wrote:

>
> On Jul 9, 2006, at 10:48 PM, Ben Campbell wrote:
>
>> Time zones are going to be an issue with any real-time review  
>> process. Maybe it is an issue we can deal with, but I suspect some  
>> number people who would otherwise participate may be locked out,  
>> or at least discouraged, by the idea of doing conference calls in  
>> their normal sleep cycle.
>
> You people sleep? No wonder we're missing milestones! Keith, we  
> need to recalculate all the milestones to accommodate this sleep  
> thing.
>
> Seriously, one would have to schedule an individual review to meet  
> the schedules of the selected review team. We'd have to be careful  
> not to mix people in very bad configurations. This can usually be  
> sorted out for groups of 3-5 people selected from a large pool,  
> assuming they're willing to work between 7:00 AM and midnight in  
> their home time zones. Of course, that does leave out at least one  
> area director I know of . . .

For that group size, I agree. I had somehow assumed larger groups for  
no particular reasons.


>
> --
> Dean
>
>


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Juha Heinanen wrote:
> Adam Roach writes:
>
>  > I'm suggesting a third bullet that reads something like:
>  > 
>  >     * If multiple bindings have the same instance-id, the proxy MUST NOT
>  >       use a binding until all bindings of greater q-value have been tried.
>
> if outbound set has two proxies and UA registers with one using q value
> 0.5 and with the other using q value 0.6, there would be only one
> binding per proxy.  in that case, i hope you are not suggesting that one
> proxy asks the other if it has higher q value for the same instance id.
>   

I'm not adding any new requirements here. The following bullet has 
identical implications:

o  If two bindings have the same instance-id and reg-id, the proxy
      SHOULD prefer the contact that was most recently updated.

/a


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Suggestion for EP indicating it added a token:
=20
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Regards,

jeroen
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From sip-bounces@ietf.org Mon Jul 10 23:18:48 2006
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Subject: [Sip] Outbound: Stunning who?
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I was chatting with folks about this debate on when it is okay to 
send STUN messages.
An interesting point was raised.

In describing the STUN processing (section 7.1) the draft says "When 
a URI is created that refers to a SIP device that supports STUN as 
described in this section, the keepalive URI parameter, as defined in 
Section 10 MUST be added to the URI, with a value of 'stun'.  This 
allows a UA to inspect the URI to decide if it should attempt to send 
STUN requests to this location."
Presumably, the URI under discussion here is the same URI as 
discussed earlier, namely the sip URI of the outbound proxy.  This is 
consistent with the discussion on the list about the use of 
supported: outbound indicating support for STUN.  supported: outbound 
is put in a response by the outbound proxy (registrar.)

However, the document is quite clear that a normal deployment is one 
with an edge proxy between the UAC and the outbound proxy.  If I 
understand right, the outbound proxy URI points to the outbound 
proxy, not the edge proxy.  And the supported header is put on by the 
outbound proxy, not the edge proxy.

The only problem is that if the UAC sends STUN packets, in the 
presence of an edge proxy, those STUN packets will be addressed to, 
and (if they arrive) be consumed by the edge proxy.  In other words, 
the entity which needs to support STUN is not the entity identified 
by the URI with the keepalive=stun parameter, nor is it the entity 
which placed the support: outbound in the header.  Whatever one 
thinks about the safety of sending STUN based on a configured 
parameter, that configuration ought to relate to the server to which 
the STUN is being sent.

As far as I can tell, the desire for and verification of STUN usage 
ought to be a matter between the UAC and the edge proxy, and ought 
not be tied up with the outbound URI or the behavior of the outbound proxy.

Feel free to tell me I have missed the point,
Joel


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Marcia has arranged us a smallish room - seats for about 35 people --  
on Wednesday afternoon at 1520 (after SIP).

On the agenda:

draft-gurbani-sip-sipsec-00.txt
draft-srivastava-sip-e2e-ciphersuites-00
and more on:
draft-audet-sip-sips-guidelines-02

What I'd like to achieve here: a better understanding of the security  
properties we either want or thought we had (but don't actually have)  
with sips:.

--
dean


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Subject: Re: [Sip] Outbound: Stunning who?
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Joel,

see inline

Regards,

Jeroen

----- Original Message ----- 
From: "Joel M. Halpern" <joel@stevecrocker.com>
To: <sip@ietf.org>
Sent: Tuesday, July 11, 2006 5:16 AM
Subject: [Sip] Outbound: Stunning who?


>I was chatting with folks about this debate on when it is okay to send STUN 
>messages.
> An interesting point was raised.
>
> In describing the STUN processing (section 7.1) the draft says "When a URI 
> is created that refers to a SIP device that supports STUN as described in 
> this section, the keepalive URI parameter, as defined in Section 10 MUST 
> be added to the URI, with a value of 'stun'.  This allows a UA to inspect 
> the URI to decide if it should attempt to send STUN requests to this 
> location."
> Presumably, the URI under discussion here is the same URI as discussed 
> earlier, namely the sip URI of the outbound proxy.  This is consistent 
> with the discussion on the list about the use of supported: outbound 
> indicating support for STUN.  supported: outbound is put in a response by 
> the outbound proxy (registrar.)
>
> However, the document is quite clear that a normal deployment is one with 
> an edge proxy between the UAC and the outbound proxy.  If I understand 
> right, the outbound proxy URI points to the outbound proxy, not the edge 
> proxy.  And the supported header is put on by the outbound proxy, not the 
> edge proxy.
>

An "outbound proxy URI" would refer to the outbound proxy, which can be 
either an edge proxy or a "home proxy" (ie the proxy associated with the 
registrar). The URI that is being talked about in section 7.1 refers to a 
"SIP device [...] as describe in this section", which "include a Registrar 
and an Edge proxy". It purposefully leaves both options open.

Interestingly, early drafts of RFC3263 defined an "outbound proxy" as "a 
specific server, independent of the URI", but the term was removed from the 
final version of that RFC.

Section 2.1 of the outbound draft should really include a definition of 
"outbound proxy", but the intention is that it can mean both depending on 
deployment. The supported header is inserted by the *registrar*, which is 
(associated with) the outbound proxy if there are no edge proxies.

I think for clarity, we should also come up with a term for the "proxy 
associated with the registrar" (and add that to section 2.1 too)

> The only problem is that if the UAC sends STUN packets, in the presence of 
> an edge proxy, those STUN packets will be addressed to, and (if they 
> arrive) be consumed by the edge proxy.  In other words, the entity which 
> needs to support STUN is not the entity identified by the URI with the 
> keepalive=stun parameter, nor is it the entity which placed the support: 
> outbound in the header.  Whatever one thinks about the safety of sending 
> STUN based on a configured parameter, that configuration ought to relate 
> to the server to which the STUN is being sent.
>
> As far as I can tell, the desire for and verification of STUN usage ought 
> to be a matter between the UAC and the edge proxy, and ought not be tied 
> up with the outbound URI or the behavior of the outbound proxy.
>

Edge proxies *are* outbound proxies, the "home proxy" is the outbound proxy 
when there are no edge proxies

> Feel free to tell me I have missed the point,
> Joel
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] Outbound: Stunning who?
Date: Tue, 11 Jul 2006 07:21:22 -0400
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On Jul 11, 2006, at 3:28 AM, Jeroen van Bemmel wrote:

> Joel,
>
> see inline
>
[snip]
>> The only problem is that if the UAC sends STUN packets, in the  
>> presence of an edge proxy, those STUN packets will be addressed  
>> to, and (if they arrive) be consumed by the edge proxy.  In other  
>> words, the entity which needs to support STUN is not the entity  
>> identified by the URI with the keepalive=stun parameter, nor is it  
>> the entity which placed the support: outbound in the header.   
>> Whatever one thinks about the safety of sending STUN based on a  
>> configured parameter, that configuration ought to relate to the  
>> server to which the STUN is being sent.
>>
>> As far as I can tell, the desire for and verification of STUN  
>> usage ought to be a matter between the UAC and the edge proxy, and  
>> ought not be tied up with the outbound URI or the behavior of the  
>> outbound proxy.
>>
>
> Edge proxies *are* outbound proxies, the "home proxy" is the  
> outbound proxy when there are no edge proxies
>
Needless to say, I find this incredibly confusing. It's clear Joel  
does too. I wonder how many other people are wandering around in this  
maze as badly as I am.

I thought this whole effort was about managing transport connections  
from UAs to the "SIP cloud" such that the user doesn't lose service  
because SIP routing can't figure out how to get requests destined for  
the UA to the proxy that happens to have the transport connection to  
the UA at this moment.

If I'm misinformed on this, I suspect some other folks are too.

>> Feel free to tell me I have missed the point,
If you missed the point, I missed the point...

I'll quickly comment that when I looked over outbound again last  
night, I got the distinct impression that MOST of the complexity is  
due to trying to solve one particular use case - the UA wants to  
maintain transport connections to proxies in different domains and be  
able to receive calls whichever of the connections (and hence  
whichever of those doimains) happens to be working.

I once worked out a pretty simple scheme (at least one I think is  
much simpler than outbound) for the case where all that's needed is  
redundant transport connectivity to the UA from proxies in the same  
edge proxy farm. (I described this to Dean late last night and I  
think he got it, so maybe there's somebody else who shared my  
hallucination here).

Dave Oran.

>> Joel
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Your description below of the entities involved, and of the outbound 
proxy being an edge proxy or a home proxy is technically correct.  An 
Outbound proxy may be an edge proxy.  But as I read the document, 
there may be an edge proxy which is not an outbound proxy in between.

The draft talks about the outbound set, and about the registration 
being targeted at the elements of the outbound set.
As such, the elements of the outbound set (the outbound proxies, the 
URIs that are known to the UAC) are clearly the registrars.  The 
draft explicitly introduces the notion that the edge proxy may be 
between the UAC and the registrar.  The registrar is the entity who 
fills in supported: outbound.  Not edge proxy.

It is true that the definitions section states that the outbound 
proxy set are the proxies with which the UAC will attempt to maintain 
a direct flow.  But it states that these are "often Edge 
Proxies".  Presumably, they are sometimes not edge proxies.  When 
they are not edge proxies, there presumably is an edge proxy between 
the UAC and the outbound proxy.

The whole wording makes it clear that one registers with the elements 
of the outbound set.

Yours,
Joel M. Halpern

PS: If Outbound is really supposed to be about finding and using the 
first proxy, rather than the registrar, then that proxy ought to be 
the one that has to support outbound.  Alternatively (and more 
usefully), the mechanisms for supporting STUN ought to be tied to 
ones first proxy, rather than to ones registrar, while outbound talks 
about finding and talking to suitable registrars.

At 03:28 AM 7/11/2006, Jeroen van Bemmel wrote:
>Joel,
>
>see inline
>
>Regards,
>
>Jeroen
>
>----- Original Message ----- From: "Joel M. Halpern" <joel@stevecrocker.com>
>To: <sip@ietf.org>
>Sent: Tuesday, July 11, 2006 5:16 AM
>Subject: [Sip] Outbound: Stunning who?
>
>
>>I was chatting with folks about this debate on when it is okay to 
>>send STUN messages.
>>An interesting point was raised.
>>
>>In describing the STUN processing (section 7.1) the draft says 
>>"When a URI is created that refers to a SIP device that supports 
>>STUN as described in this section, the keepalive URI parameter, as 
>>defined in Section 10 MUST be added to the URI, with a value of 
>>'stun'.  This allows a UA to inspect the URI to decide if it should 
>>attempt to send STUN requests to this location."
>>Presumably, the URI under discussion here is the same URI as 
>>discussed earlier, namely the sip URI of the outbound proxy.  This 
>>is consistent with the discussion on the list about the use of 
>>supported: outbound indicating support for STUN.  supported: 
>>outbound is put in a response by the outbound proxy (registrar.)
>>
>>However, the document is quite clear that a normal deployment is 
>>one with an edge proxy between the UAC and the outbound proxy.  If 
>>I understand right, the outbound proxy URI points to the outbound 
>>proxy, not the edge proxy.  And the supported header is put on by 
>>the outbound proxy, not the edge proxy.
>
>An "outbound proxy URI" would refer to the outbound proxy, which can 
>be either an edge proxy or a "home proxy" (ie the proxy associated 
>with the registrar). The URI that is being talked about in section 
>7.1 refers to a "SIP device [...] as describe in this section", 
>which "include a Registrar and an Edge proxy". It purposefully 
>leaves both options open.
>
>Interestingly, early drafts of RFC3263 defined an "outbound proxy" 
>as "a specific server, independent of the URI", but the term was 
>removed from the final version of that RFC.
>
>Section 2.1 of the outbound draft should really include a definition 
>of "outbound proxy", but the intention is that it can mean both 
>depending on deployment. The supported header is inserted by the 
>*registrar*, which is (associated with) the outbound proxy if there 
>are no edge proxies.
>
>I think for clarity, we should also come up with a term for the 
>"proxy associated with the registrar" (and add that to section 2.1 too)
>
>>The only problem is that if the UAC sends STUN packets, in the 
>>presence of an edge proxy, those STUN packets will be addressed to, 
>>and (if they arrive) be consumed by the edge proxy.  In other 
>>words, the entity which needs to support STUN is not the entity 
>>identified by the URI with the keepalive=stun parameter, nor is it 
>>the entity which placed the support: outbound in the 
>>header.  Whatever one thinks about the safety of sending STUN based 
>>on a configured parameter, that configuration ought to relate to 
>>the server to which the STUN is being sent.
>>
>>As far as I can tell, the desire for and verification of STUN usage 
>>ought to be a matter between the UAC and the edge proxy, and ought 
>>not be tied up with the outbound URI or the behavior of the outbound proxy.
>
>Edge proxies *are* outbound proxies, the "home proxy" is the 
>outbound proxy when there are no edge proxies
>
>>Feel free to tell me I have missed the point,
>>Joel
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Wacky-thread sipsec ad-hoc
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I'd like to put in a request for some time at the beginning to frame  
two key things that might help frame the rest of the discussion....

1) Requirements^H^H^H^H^H Thinks we want around having a "Lock Icon"  
on phone

2) A path out of the sips swamp that we are swimming in

Cullen



On Jul 10, 2006, at 11:35 PM, Dean Willis wrote:

>
> Marcia has arranged us a smallish room - seats for about 35 people  
> -- on Wednesday afternoon at 1520 (after SIP).
>
> On the agenda:
>
> draft-gurbani-sip-sipsec-00.txt
> draft-srivastava-sip-e2e-ciphersuites-00
> and more on:
> draft-audet-sip-sips-guidelines-02
>
> What I'd like to achieve here: a better understanding of the  
> security properties we either want or thought we had (but don't  
> actually have) with sips:.
>
> --
> dean
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



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Subject: RE: [Sip] Outbound: Stunning who?
Date: Tue, 11 Jul 2006 11:04:30 -0400
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Thread-Topic: [Sip] Outbound: Stunning who?
Thread-Index: Acak3DunHQkTdu8LQaCYBm+8ESfklwAHajGA
From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Dave Oran \(oran\)" <oran@cisco.com>, "IETF SIP List" <sip@ietf.org>
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Dave,

I think there are 3 possible topologies:
R=3Dregistrar
E=3Dedge
O=3Doutbound
C=3Dclient

+-+  +-+  +-+
|R|  |R|  |R| home
| |  +-+  +-+
| |   |    |
| |  +-+  +-+
|E|  |E|  |E| home
| |  | |  +-+=20
| |  | |   |
| |  | |  +-+
|O|  |O|  |O| visited
+-+  +-+  +-+=20
 |    |    |
+-+  +-+  +-+
|C|  |C|  |C|
+-+  +-+  +-+
 1    2    3

1=3Dmonolithic proxy
2=3Dhome network IMS
3=3Dvisited network IMS

Confusion may depend on which picture people have in their head at the
moment.

All, feel free to correct or modify if I missed something.

Mike
=20

> -----Original Message-----
> From: Dave Oran (oran)=20
> Sent: Tuesday, July 11, 2006 7:21 AM
> To: IETF SIP List
> Subject: Re: [Sip] Outbound: Stunning who?
>=20
>=20
> On Jul 11, 2006, at 3:28 AM, Jeroen van Bemmel wrote:
>=20
> > Joel,
> >
> > see inline
> >
> [snip]
> >> The only problem is that if the UAC sends STUN packets, in the=20
> >> presence of an edge proxy, those STUN packets will be=20
> addressed to,=20
> >> and (if they arrive) be consumed by the edge proxy.  In=20
> other words,=20
> >> the entity which needs to support STUN is not the entity=20
> identified=20
> >> by the URI with the keepalive=3Dstun parameter, nor is it
> >> the entity which placed the support: outbound in the header.  =20
> >> Whatever one thinks about the safety of sending STUN based on a=20
> >> configured parameter, that configuration ought to relate to the=20
> >> server to which the STUN is being sent.
> >>
> >> As far as I can tell, the desire for and verification of=20
> STUN usage=20
> >> ought to be a matter between the UAC and the edge proxy, and ought=20
> >> not be tied up with the outbound URI or the behavior of=20
> the outbound=20
> >> proxy.
> >>
> >
> > Edge proxies *are* outbound proxies, the "home proxy" is=20
> the outbound=20
> > proxy when there are no edge proxies
> >
> Needless to say, I find this incredibly confusing. It's clear=20
> Joel does too. I wonder how many other people are wandering=20
> around in this maze as badly as I am.
>=20
> I thought this whole effort was about managing transport=20
> connections from UAs to the "SIP cloud" such that the user=20
> doesn't lose service because SIP routing can't figure out how=20
> to get requests destined for the UA to the proxy that happens=20
> to have the transport connection to the UA at this moment.
>=20
> If I'm misinformed on this, I suspect some other folks are too.
>=20
> >> Feel free to tell me I have missed the point,
> If you missed the point, I missed the point...
>=20
> I'll quickly comment that when I looked over outbound again=20
> last night, I got the distinct impression that MOST of the=20
> complexity is due to trying to solve one particular use case=20
> - the UA wants to maintain transport connections to proxies=20
> in different domains and be able to receive calls whichever=20
> of the connections (and hence whichever of those doimains)=20
> happens to be working.
>=20
> I once worked out a pretty simple scheme (at least one I=20
> think is much simpler than outbound) for the case where all=20
> that's needed is redundant transport connectivity to the UA=20
> from proxies in the same edge proxy farm. (I described this=20
> to Dean late last night and I think he got it, so maybe=20
> there's somebody else who shared my hallucination here).
>=20
> Dave Oran.
>=20
> >> Joel
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Tue Jul 11 12:46:46 2006
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From: "Joel M. Halpern" <joel@stevecrocker.com>
Subject: RE: [Sip] Outbound: Stunning who?
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That's interesting, but does not align with how I read the outbound draft.
The proxies identified as the "outbound set" (which are presumably 
the outbound proxies) are teh targets of the registration requests 
from the user.  Hence, I believe the outbound proxies are intended to 
be the registrars.
And the Edge Proxy is specifically stated to be the first proxy from the user.

Whatever else is going on, if different readers are getting pictures 
this disparate from the draft, we need to clarify the draft.

Yours,
Joel

PS: There is actually an assumption in all of this that the first 
proxy is outside of ones NAT.  Otherwise, the client can not STUN 
anyone, since the entity he has an IP address for, his first proxy, 
would also be inside the NAT boundary.

At 11:04 AM 7/11/2006, Michael Hammer \(mhammer\) wrote:
>Dave,
>
>I think there are 3 possible topologies:
>R=registrar
>E=edge
>O=outbound
>C=client
>
>+-+  +-+  +-+
>|R|  |R|  |R| home
>| |  +-+  +-+
>| |   |    |
>| |  +-+  +-+
>|E|  |E|  |E| home
>| |  | |  +-+
>| |  | |   |
>| |  | |  +-+
>|O|  |O|  |O| visited
>+-+  +-+  +-+
>  |    |    |
>+-+  +-+  +-+
>|C|  |C|  |C|
>+-+  +-+  +-+
>  1    2    3
>
>1=monolithic proxy
>2=home network IMS
>3=visited network IMS
>
>Confusion may depend on which picture people have in their head at the
>moment.
>
>All, feel free to correct or modify if I missed something.
>
>Mike
>
>
> > -----Original Message-----
> > From: Dave Oran (oran)
> > Sent: Tuesday, July 11, 2006 7:21 AM
> > To: IETF SIP List
> > Subject: Re: [Sip] Outbound: Stunning who?
> >
> >
> > On Jul 11, 2006, at 3:28 AM, Jeroen van Bemmel wrote:
> >
> > > Joel,
> > >
> > > see inline
> > >
> > [snip]
> > >> The only problem is that if the UAC sends STUN packets, in the
> > >> presence of an edge proxy, those STUN packets will be
> > addressed to,
> > >> and (if they arrive) be consumed by the edge proxy.  In
> > other words,
> > >> the entity which needs to support STUN is not the entity
> > identified
> > >> by the URI with the keepalive=stun parameter, nor is it
> > >> the entity which placed the support: outbound in the header.
> > >> Whatever one thinks about the safety of sending STUN based on a
> > >> configured parameter, that configuration ought to relate to the
> > >> server to which the STUN is being sent.
> > >>
> > >> As far as I can tell, the desire for and verification of
> > STUN usage
> > >> ought to be a matter between the UAC and the edge proxy, and ought
> > >> not be tied up with the outbound URI or the behavior of
> > the outbound
> > >> proxy.
> > >>
> > >
> > > Edge proxies *are* outbound proxies, the "home proxy" is
> > the outbound
> > > proxy when there are no edge proxies
> > >
> > Needless to say, I find this incredibly confusing. It's clear
> > Joel does too. I wonder how many other people are wandering
> > around in this maze as badly as I am.
> >
> > I thought this whole effort was about managing transport
> > connections from UAs to the "SIP cloud" such that the user
> > doesn't lose service because SIP routing can't figure out how
> > to get requests destined for the UA to the proxy that happens
> > to have the transport connection to the UA at this moment.
> >
> > If I'm misinformed on this, I suspect some other folks are too.
> >
> > >> Feel free to tell me I have missed the point,
> > If you missed the point, I missed the point...
> >
> > I'll quickly comment that when I looked over outbound again
> > last night, I got the distinct impression that MOST of the
> > complexity is due to trying to solve one particular use case
> > - the UA wants to maintain transport connections to proxies
> > in different domains and be able to receive calls whichever
> > of the connections (and hence whichever of those doimains)
> > happens to be working.
> >
> > I once worked out a pretty simple scheme (at least one I
> > think is much simpler than outbound) for the case where all
> > that's needed is redundant transport connectivity to the UA
> > from proxies in the same edge proxy farm. (I described this
> > to Dean late last night and I think he got it, so maybe
> > there's somebody else who shared my hallucination here).
> >
> > Dave Oran.
> >
> > >> Joel
> > >>
> > >>
> > >> _______________________________________________
> > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >> This list is for NEW development of the core SIP Protocol Use
> > >> sip-implementors@cs.columbia.edu for questions on current sip Use
> > >> sipping@ietf.org for new developments on the application of sip
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > sipping@ietf.org for new developments on the application of sip
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



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From: Dale.Worley@comcast.net
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	(jason@counterpath.com)
Subject: Re: [Sip] draft-ietf-sip-certs-00,
	communicating private keys and self-signed certificates
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	<4497EAE7.9090700@cisco.com> <44985514.4010107@cisco.com>
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   From: "Jason Fischl" <jason@counterpath.com>

   > On the using a GRUU in a cert, would need to consider that any time
   > my GRUU changes I need to generate a new cert - if this meant a new
   > private/public key pair, this might be expensive.

   Also consider that GRUUs will change relatively frequently for devices.

Why would GRUUs change "relatively frequently"?  They can change only
as frequently as the AOR changes, or the instance-id changes.

Dale

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Subject: Re: [Sip] Outbound: Stunning who?
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Hi Joel,


> That's interesting, but does not align with how I read the outbound draft.
> The proxies identified as the "outbound set" (which are presumably the 
> outbound proxies) are teh targets of the registration requests from the 
> user.  Hence, I believe the outbound proxies are intended to be the 
> registrars.

No, the registration requests are targeted at the home domain as usual (ie 
request URI == domain from AoR). Any discovered URI is added as the topmost 
Route header, it is the first hop but not (necessarily) the final 
destination of the REGISTER.

That being said, there seems something wrong with the text in section 4.2. 
When interpreted literally, it reads as if the UA would always have an 
outbound-proxy-set with at least 2 entries, making the example given in 
section 3.2 (direct registration) impossible (because there would 
"typically" be a Route header).

I think the text should say "UAs obtain at configuration time *zero* or more 
SIP URIs representing the default outbound-proxy-set." to allow for this 
case

> And the Edge Proxy is specifically stated to be the first proxy from the 
> user.
>
> Whatever else is going on, if different readers are getting pictures this 
> disparate from the draft, we need to clarify the draft.
>
> Yours,
> Joel
>
> PS: There is actually an assumption in all of this that the first proxy is 
> outside of ones NAT.  Otherwise, the client can not STUN anyone, since the 
> entity he has an IP address for, his first proxy, would also be inside the 
> NAT boundary.
>

That is the assumption / limitation that the outbound draft makes: if there 
is NAT, it is located between the UA and the proxy to which the discovered 
outbound URI resolves.

Regards,

jeroen

> At 11:04 AM 7/11/2006, Michael Hammer \(mhammer\) wrote:
>>Dave,
>>
>>I think there are 3 possible topologies:
>>R=registrar
>>E=edge
>>O=outbound
>>C=client
>>
>>+-+  +-+  +-+
>>|R|  |R|  |R| home
>>| |  +-+  +-+
>>| |   |    |
>>| |  +-+  +-+
>>|E|  |E|  |E| home
>>| |  | |  +-+
>>| |  | |   |
>>| |  | |  +-+
>>|O|  |O|  |O| visited
>>+-+  +-+  +-+
>>  |    |    |
>>+-+  +-+  +-+
>>|C|  |C|  |C|
>>+-+  +-+  +-+
>>  1    2    3
>>
>>1=monolithic proxy
>>2=home network IMS
>>3=visited network IMS
>>
>>Confusion may depend on which picture people have in their head at the
>>moment.
>>
>>All, feel free to correct or modify if I missed something.
>>
>>Mike
>>
>>
>> > -----Original Message-----
>> > From: Dave Oran (oran)
>> > Sent: Tuesday, July 11, 2006 7:21 AM
>> > To: IETF SIP List
>> > Subject: Re: [Sip] Outbound: Stunning who?
>> >
>> >
>> > On Jul 11, 2006, at 3:28 AM, Jeroen van Bemmel wrote:
>> >
>> > > Joel,
>> > >
>> > > see inline
>> > >
>> > [snip]
>> > >> The only problem is that if the UAC sends STUN packets, in the
>> > >> presence of an edge proxy, those STUN packets will be
>> > addressed to,
>> > >> and (if they arrive) be consumed by the edge proxy.  In
>> > other words,
>> > >> the entity which needs to support STUN is not the entity
>> > identified
>> > >> by the URI with the keepalive=stun parameter, nor is it
>> > >> the entity which placed the support: outbound in the header.
>> > >> Whatever one thinks about the safety of sending STUN based on a
>> > >> configured parameter, that configuration ought to relate to the
>> > >> server to which the STUN is being sent.
>> > >>
>> > >> As far as I can tell, the desire for and verification of
>> > STUN usage
>> > >> ought to be a matter between the UAC and the edge proxy, and ought
>> > >> not be tied up with the outbound URI or the behavior of
>> > the outbound
>> > >> proxy.
>> > >>
>> > >
>> > > Edge proxies *are* outbound proxies, the "home proxy" is
>> > the outbound
>> > > proxy when there are no edge proxies
>> > >
>> > Needless to say, I find this incredibly confusing. It's clear
>> > Joel does too. I wonder how many other people are wandering
>> > around in this maze as badly as I am.
>> >
>> > I thought this whole effort was about managing transport
>> > connections from UAs to the "SIP cloud" such that the user
>> > doesn't lose service because SIP routing can't figure out how
>> > to get requests destined for the UA to the proxy that happens
>> > to have the transport connection to the UA at this moment.
>> >
>> > If I'm misinformed on this, I suspect some other folks are too.
>> >
>> > >> Feel free to tell me I have missed the point,
>> > If you missed the point, I missed the point...
>> >
>> > I'll quickly comment that when I looked over outbound again
>> > last night, I got the distinct impression that MOST of the
>> > complexity is due to trying to solve one particular use case
>> > - the UA wants to maintain transport connections to proxies
>> > in different domains and be able to receive calls whichever
>> > of the connections (and hence whichever of those doimains)
>> > happens to be working.
>> >
>> > I once worked out a pretty simple scheme (at least one I
>> > think is much simpler than outbound) for the case where all
>> > that's needed is redundant transport connectivity to the UA
>> > from proxies in the same edge proxy farm. (I described this
>> > to Dean late last night and I think he got it, so maybe
>> > there's somebody else who shared my hallucination here).
>> >
>> > Dave Oran.
>> >
>> > >> Joel
>> > >>
>> > >>
>> > >> _______________________________________________
>> > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > >> This list is for NEW development of the core SIP Protocol Use
>> > >> sip-implementors@cs.columbia.edu for questions on current sip Use
>> > >> sipping@ietf.org for new developments on the application of sip
>> > >
>> > >
>> > > _______________________________________________
>> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > > This list is for NEW development of the core SIP Protocol Use
>> > > sip-implementors@cs.columbia.edu for questions on current sip Use
>> > > sipping@ietf.org for new developments on the application of sip
>> >
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol Use
>> > sip-implementors@cs.columbia.edu for questions on current sip
>> > Use sipping@ietf.org for new developments on the application of sip
>> >
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: RE: [Sip] Outbound: Stunning who?
Date: Tue, 11 Jul 2006 15:22:56 -0400
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Thread-Topic: [Sip] Outbound: Stunning who?
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Joel M. Halpern" <joel@stevecrocker.com>, "IETF SIP List" <sip@ietf.org>
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I probably switched the E's and the O's, but that is why I threw this
out there, because this was the picture that came to mind from reading
the email thread.

Actually, the third possibility may be the one tripping up the words.

Case 1 is monolithic and likely in the home network only

Case 2 may allow for the O/E proxy to be either in a home or visited
network.  Does it matter?

Case 3 may need more explanation in the draft.

Is the following the correct picture?  Maybe my E and O are synonymous
and the O proxy should be labeled something else.

R=3Dregistrar
E=3Dedge
O=3Doutbound
C=3Dclient

+-+  +-+  +-+
|R|  |R|  |R| home
| |  +-+  +-+
| |   |    |
| |  +-+  +-+
|O|  |O|  |O| home
| |  | |  +-+
| |  | |   |
| |  | |  +-+
|E|  |E|  |E| visited
+-+  +-+  +-+
 |    |    |
+-+  +-+  +-+
|C|  |C|  |C|
+-+  +-+  +-+
 1    2    3

1=3Dmonolithic proxy
2=3Dhome network IMS
3=3Dvisited network IMS

If we can't draw the picture, we probably don't know what we mean. =20
If this is wrong, please draw me another picture.

Mike
=20

> -----Original Message-----
> From: Joel M. Halpern [mailto:joel@stevecrocker.com]=20
> Sent: Tuesday, July 11, 2006 12:45 PM
> To: Michael Hammer (mhammer); IETF SIP List
> Subject: RE: [Sip] Outbound: Stunning who?
>=20
> That's interesting, but does not align with how I read the=20
> outbound draft.
> The proxies identified as the "outbound set" (which are=20
> presumably the outbound proxies) are teh targets of the=20
> registration requests from the user.  Hence, I believe the=20
> outbound proxies are intended to be the registrars.
> And the Edge Proxy is specifically stated to be the first=20
> proxy from the user.
>=20
> Whatever else is going on, if different readers are getting=20
> pictures this disparate from the draft, we need to clarify the draft.
>=20
> Yours,
> Joel
>=20
> PS: There is actually an assumption in all of this that the=20
> first proxy is outside of ones NAT.  Otherwise, the client=20
> can not STUN anyone, since the entity he has an IP address=20
> for, his first proxy, would also be inside the NAT boundary.
>=20
> At 11:04 AM 7/11/2006, Michael Hammer \(mhammer\) wrote:
> >Dave,
> >
> >I think there are 3 possible topologies:
> >R=3Dregistrar
> >E=3Dedge
> >O=3Doutbound
> >C=3Dclient
> >
> >+-+  +-+  +-+
> >|R|  |R|  |R| home
> >| |  +-+  +-+
> >| |   |    |
> >| |  +-+  +-+
> >|E|  |E|  |E| home
> >| |  | |  +-+
> >| |  | |   |
> >| |  | |  +-+
> >|O|  |O|  |O| visited
> >+-+  +-+  +-+
> >  |    |    |
> >+-+  +-+  +-+
> >|C|  |C|  |C|
> >+-+  +-+  +-+
> >  1    2    3
> >
> >1=3Dmonolithic proxy
> >2=3Dhome network IMS
> >3=3Dvisited network IMS
> >
> >Confusion may depend on which picture people have in their=20
> head at the=20
> >moment.
> >
> >All, feel free to correct or modify if I missed something.
> >
> >Mike
> >
> >
> > > -----Original Message-----
> > > From: Dave Oran (oran)
> > > Sent: Tuesday, July 11, 2006 7:21 AM
> > > To: IETF SIP List
> > > Subject: Re: [Sip] Outbound: Stunning who?
> > >
> > >
> > > On Jul 11, 2006, at 3:28 AM, Jeroen van Bemmel wrote:
> > >
> > > > Joel,
> > > >
> > > > see inline
> > > >
> > > [snip]
> > > >> The only problem is that if the UAC sends STUN packets, in the=20
> > > >> presence of an edge proxy, those STUN packets will be
> > > addressed to,
> > > >> and (if they arrive) be consumed by the edge proxy.  In
> > > other words,
> > > >> the entity which needs to support STUN is not the entity
> > > identified
> > > >> by the URI with the keepalive=3Dstun parameter, nor is it the=20
> > > >> entity which placed the support: outbound in the header.
> > > >> Whatever one thinks about the safety of sending STUN=20
> based on a=20
> > > >> configured parameter, that configuration ought to=20
> relate to the=20
> > > >> server to which the STUN is being sent.
> > > >>
> > > >> As far as I can tell, the desire for and verification of
> > > STUN usage
> > > >> ought to be a matter between the UAC and the edge proxy, and=20
> > > >> ought not be tied up with the outbound URI or the behavior of
> > > the outbound
> > > >> proxy.
> > > >>
> > > >
> > > > Edge proxies *are* outbound proxies, the "home proxy" is
> > > the outbound
> > > > proxy when there are no edge proxies
> > > >
> > > Needless to say, I find this incredibly confusing. It's=20
> clear Joel=20
> > > does too. I wonder how many other people are wandering around in=20
> > > this maze as badly as I am.
> > >
> > > I thought this whole effort was about managing transport=20
> connections=20
> > > from UAs to the "SIP cloud" such that the user doesn't=20
> lose service=20
> > > because SIP routing can't figure out how to get requests destined=20
> > > for the UA to the proxy that happens to have the transport=20
> > > connection to the UA at this moment.
> > >
> > > If I'm misinformed on this, I suspect some other folks are too.
> > >
> > > >> Feel free to tell me I have missed the point,
> > > If you missed the point, I missed the point...
> > >
> > > I'll quickly comment that when I looked over outbound again last=20
> > > night, I got the distinct impression that MOST of the=20
> complexity is=20
> > > due to trying to solve one particular use case
> > > - the UA wants to maintain transport connections to proxies in=20
> > > different domains and be able to receive calls whichever of the=20
> > > connections (and hence whichever of those doimains) happens to be=20
> > > working.
> > >
> > > I once worked out a pretty simple scheme (at least one I think is=20
> > > much simpler than outbound) for the case where all that's=20
> needed is=20
> > > redundant transport connectivity to the UA from proxies=20
> in the same=20
> > > edge proxy farm. (I described this to Dean late last night and I=20
> > > think he got it, so maybe there's somebody else who shared my=20
> > > hallucination here).
> > >
> > > Dave Oran.
> > >
> > > >> Joel
> > > >>
> > > >>
> > > >> _______________________________________________
> > > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > >> This list is for NEW development of the core SIP Protocol Use=20
> > > >> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > >> sipping@ietf.org for new developments on the application of sip
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol Use=20
> >sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: [Sip] Room# for Wacky-thread sipsec ad-hoc on Wednesday
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We have room: 518C, for Wednesday afternoon after SIP, eta 1520. It's  
a small room.

--
Dean


On Jul 10, 2006, at 10:35 PM, Dean Willis wrote:

>
> Marcia has arranged us a smallish room - seats for about 35 people  
> -- on Wednesday afternoon at 1520 (after SIP).
>
> On the agenda:
>
> draft-gurbani-sip-sipsec-00.txt
> draft-srivastava-sip-e2e-ciphersuites-00
> and more on:
> draft-audet-sip-sips-guidelines-02
>
> What I'd like to achieve here: a better understanding of the  
> security properties we either want or thought we had (but don't  
> actually have) with sips:.
>
> --
> dean
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Outbound: Stunning who?
Date: Tue, 11 Jul 2006 15:55:30 -0500
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On Jul 11, 2006, at 11:44 AM, Joel M. Halpern wrote:

> That's interesting, but does not align with how I read the outbound  
> draft.
> The proxies identified as the "outbound set" (which are presumably  
> the outbound proxies) are teh targets of the registration requests  
> from the user.  Hence, I believe the outbound proxies are intended  
> to be the registrars.
> And the Edge Proxy is specifically stated to be the first proxy  
> from the user.
>
> Whatever else is going on, if different readers are getting  
> pictures this disparate from the draft, we need to clarify the draft.

Yep. I'm pretty sure I invented the term "outbound proxy" (in the sip  
context, at least) some years ago, but even so the way I read the  
draft we were ending up stunning helpless edge proxies that were  
between the UAs and the "outbound" proxies and had no way to complain  
about it. Worse than the Taser-happy Dallas police, I think.

The root of the problem, I think, is that this draft intertwingles  
the proxy behavior (supporting stun) with the registrar behavior  
(supporting "outbound" parallel registrations).

--
Dean



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Subject: Re: [Sip] Outbound: Stunning who?
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> The root of the problem, I think, is that this draft intertwingles  the 
> proxy behavior (supporting stun) with the registrar behavior  (supporting 
> "outbound" parallel registrations).

That, plus the fact that "outbound" is really about receiving inbound 
requests (from the UA perspective). An "outbound" proxy is traditionally 
something to which a UA sends its outbound requests, typically starting with 
REGISTER.

"draft-ietf-sip-inbound-nat-traversal" ?

Jeroen


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From sip-bounces@ietf.org Tue Jul 11 21:19:17 2006
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Due to the overrun in the first session, this is the revised agenda for tomorrow. Apologies for the format but all the information is there.

We will be looking to pull up as much time as possible, so we will be skipping resolution of non-urgent issues. Presenter's if you think something is critical that we may not have thought of, please tell us in advance.

I think we have most of the presentations now, but if for some reason you have not provided your's, please send as soon as possible. 

Note that we also have the security issues discussion in an ad-hoc immediately following the SIP session (after the break). See Dean's separate note to this list.

regards

Keith

1300	Agenda bash	Chairs	This doc

1305	Location conveyance	James Polk	draft-ietf-sip-location-conveyance-03.txt 

1320	SAML for SIP	James Polk, Hannes Tschofenig	draft-ietf-sip-saml-00.txt 

1330	Certificates. Arboreal Rodents
Or Fruitless Canine Vocalization?	Jason Fischl	draft-ietf-sip-certs-01 

1345	Hop Limit Diagnostics
Is my bunny sick?	Scott Lawrence	draft-ietf-sip-hop-limit-diagnostics-03 

1400	Connected identity	John Elwell	draft-ietf-sip-connected-identity-00.txt 
draft-elwell-sip-tispan-connected-identity-01.txt 

1430	Security flows
Fork loop fix
Feature list for advanced specification	Robert Sparks	draft-ietf-sip-sec-flows-01
draft-ietf-sip-fork-loop-fix-02
TBD

1445	Hitchhiker's guide	Jonathan Rosenberg	draft-ietf-sip-hitchhikers-guide-00

1455	SIPS: Guidelines	Francois Audet	draft-audet-sip-sips-guidelines-02

Keith Drage
drage@lucent.com
tel: +44 1793 776249

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From sip-bounces@ietf.org Tue Jul 11 23:43:40 2006
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RFC 4485 (http://www.ietf.org/rfc/rfc4485.txt) section 5 specifies:

   Dialog Usages: SIP allows for requests that normally create their own
      dialog (such as SUBSCRIBE) to be used within a dialog created by
      another method (such as INVITE).  In such a case, there are said
      to be multiple usages of that dialog.  Extensions SHOULD consider
      their interaction with dialog usages.  In particular, extensions
      that define new error response codes SHOULD describe whether that
      response code causes the dialog and all usages to terminate, or
      just a specific usage.

http://www.ietf.org/internet-drafts/draft-ietf-sip-acr-code-01.txt

does this by implication:

   A UAC the does not understand or care about the specific semantics of
   the 433 response will treat it as a 400 response.

But this could be much much more explicit.

http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyance-03.txt

does not seem to have any information on this whatsoever.

Can I ask all editor's of i-ds with new response code values to make sure this is covered.

regards

Keith

Keith Drage
Lucent Technologies
drage@lucent.com
tel: +44 1793 776249

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> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
>
> Ok, I am done't think I am manging to explaim myself. Say A has a
> contract with B and B has a crontract with C. Now a call arrives at
> C, it seem that C wants a assertion to be made by B. Anything A says
> to C is clsoe to useless because C has no clue who A even is. I guese
> I feel like we need to sort out the requirements for what we want in
> a transit case and then we might be able to figure out a solution.

One of the problems with the idea of a b2bua re-signing it is how to trust
the b2bua.  I.e., if the domain the b2bua is in is not the same as the
From's, then you have to somehow trust the different domain.  I agree with
you there's a common model of "I trust the domain connected to me directly".


So if you have an A-B-C chain as you describe, then sure A can sign, then B
can verify A and re-sign with B-cert, and C can verify B.  Now imagine
A-B-C-D; A and B have a b2bua, but C doesn't.  A signs, then B verifies and
signs with B-cert, C does nothing and D gets a From of A with a cert of B
and does what?
In other words, unless they all re-sign, it's limited to a small chain.  But
if they all re-sign, we're back to hop-by-hop trust.  Well... fewer hops
because it's domain-by-domain instead of proxy-by-proxy, so it is better.

But is domain by domain resigning acceptable?  It breaks the end-to-end
model of identity, so I would think not.  And making every domain re-sign
would have a serious flag-day problem. (not to mention quite an impressive
performance/scale issue I suspect)  At the same time, in your example C does
not have a contract with A, so the business model is not end-to-end - so I'm
not sure why the trust would be.

-hadriel



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From sip-bounces@ietf.org Wed Jul 12 07:16:41 2006
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From: "Jesske, R" <R.Jesske@t-com.net>
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Subject: AW: [Sip] concerns on draft-elwell-sip-tispan-connected-identity- 01
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The use of From and P-Asserted Identity for TISPAN Services is =
discussed within IETF since nearly one year.
Discussion at the Paris and Vancouver Meeting and within the SIPPING =
and SIPPING-TISPAN mail list.

The use of the From and P-Asserted header for our OIP-OIR (Originating =
Identification Presentation/Restriction) is inline with the use as =
defined in RFC3261. The From header is used as a not screened identity =
directly send from the user where as the P-Asserted-Identity is send by =
a network entity that can verify the identity of the user. In many =
cases the sent identity is the same.
But also in some cases the user can have more than one identity. A =
public (Asserted) and a private (Not asserted) one but both are =
pointing to the same user. The user could have more than one private =
identity pointing to his UA or several UAs.
Now in the PSTN world we have PBX using also Public and private =
identities (one main line e.G. a Company with many extensions e.G. all =
people working within the company) so we have the same construct as I =
described for the P-Asserted-Identity and From header.
The PBX is sender and receiver of the Requests and Responses so also =
originator of the From header and a P-Preferred-Identity that will be =
verified by a Network element of the trusted Domain. But behind an PBX =
there could be many private identities of users that could be addressed =
directly by the private identity and over the asserted identity =
(switchboard).  =20

So now for the forward direction the use of From and =
P-Asserted-Identity for presenting the originating user is clear. In =
backward direction we can also use the P-Asserted-Identity but a =
equivalent of the Private identity is missing. So the =
draft-elwell-sip-tispan-connected-identity- 01 looks fore some =
possibilities to fulfil this requirements.
These requirements are also documented within =
draft-jesske-sipping-tispan-requirements-03.=20

Best Regards

Roland




> -----Urspr=FCngliche Nachricht-----
> Von: Elwell, John [mailto:john.elwell@siemens.com]=20
> Gesendet: Montag, 10. Juli 2006 04:26
> An: Jonathan Rosenberg; IETF SIP List
> Betreff: RE: [Sip] concerns on=20
> draft-elwell-sip-tispan-connected-identity- 01
>=20
>=20
> Jonathan,
>=20
> Thanks for reading this and commenting. I will leave the=20
> TISPAN guys to
> reply on the TISPAN reasons for needing two identities=20
> (although this was
> discussed at length on an old thread) and their rationale for=20
> using PAI for
> one / Identity for the other.=20
>=20
> All I can say is that the document lays down a concern that=20
> some TISPAN
> people have about the connected-identity draft and possible=20
> ways to overcome
> that concern. One solution, of course, would be to do nothing=20
> - leave the
> connected-identity draft as it is. But in taking that course=20
> we will need to
> show that TISPAN requirements can be met by some other means.
>=20
> John
>=20
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> > Sent: 07 July 2006 20:50
> > To: IETF SIP List
> > Subject: [Sip] concerns on=20
> > draft-elwell-sip-tispan-connected-identity-01
> >=20
> > I wasn't able to find any disucssion on this draft so far in the=20
> > archives; my apologies if I missed it.
> >=20
> > This draft concerns me greatly.
> >=20
> > The first thing that worries me is the general idea that 3gpp=20
> > or tispan=20
> > get to 'choose' which SIP header field contains what type of=20
> > identity.=20
> > They get to do no such thing. These header fields have=20
> > defined meanings.=20
> > Having either organization decide to specify their own=20
> interpretation=20
> > for the type of identity that goes in there is a guarnateed=20
> > receipe for=20
> > huge, massive, and unending interoperability failures.
> >=20
> > On the technical front, I strongly disagree that the From and 
> > P-Asserted-Identity represent different types of identity=20
> in any way.=20
> > They both have always identified the user that is the sender of the =

> > request. In cases where you had an enterprise user Joe making a SIP =

> > call, the From would contain sip:joe@enterprise.com and=20
> this would be=20
> > authenticated by the proxy in the enterprise, followed by=20
> > insertion of a=20
> > P-Asserted-ID which says the same thing. The From, in this=20
> use case,=20
> > should not be sip:enterprise.com.
> >=20
> > Next, I am puzzled by the requirement to deliver separate=20
> > identities to=20
> > the calling party for these two things. I understand the=20
> concept of a=20
> > billing identity being separate from the identity of the=20
> caller. I do=20
> > not, however, understand why the billing ID is of any=20
> interest to the=20
> > called party whatsoever. It seems that this would, at best, be of=20
> > interest only within the domain of the calling party. Since=20
> its also=20
> > directly inferrable from the identity of the caller (presumably the =

> > database contains a linkage between the two), I also don't=20
> see why it=20
> > would need to be carried in the sip message itself. Is there=20
> > a use case=20
> > where there is not a provisioned relationship between the=20
> user making=20
> > the call and the billing entity? Certainly there is a clear direct=20
> > relationship in the enterprise example in the draft.
> >=20
> > Finally, even if I buy the idea that there are two identities=20
> > and that=20
> > they both need to be conveyed, I strongly disagree with the=20
> > notion that=20
> > one of them can be conveyed with SIP-identity, while P-A-ID=20
> is OK for=20
> > the other. There are certain well-known technical limitations with=20
> > P-A-ID which sip-identity addresses. There is nothing about=20
> > the nature=20
> > of the identity which invalidates any of those issues.
> >=20
> > Thanks,
> > Jonathan R.
> > --=20
> > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > Cisco Fellow                                   Parsippany, NJ=20
> > 07054-2711
> > Cisco Systems
> > jdrosen@cisco.com                              FAX:   (973) =
952-5050
> > http://www.jdrosen.net                         PHONE: (973) =
952-5000
> > http://www.cisco.com
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From sip-bounces@ietf.org Wed Jul 12 08:08:39 2006
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Date: Wed, 12 Jul 2006 13:02:21 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] STUN keep-alive: Client vs server generating requests
To: Hadriel Kaplan <HKaplan@acmepacket.com>,
	'Jeroen van Bemmel' <jbemmel@zonnet.nl>, sip@ietf.org
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I think it would be a good idea to have this stun-interval=xxx. The proxy
will have its own upper limit of frequency of STUN requests it can handle,
depending on the number of registered UAs it supports. In general there is
likely to be some administrative action required for choosing the right
compromise between fast recovery from loss of connectivity and the number of
UAs a proxy can support. It is best to do this once at the proxy and have
that indicated to each registering UA. When administering this value,
account has to be taken of the NATs and firewalls involved, and if the value
chosen is too long for some NATs/firewalls, then it will fail. But likewise
if the UA chooses the value, it could be wrong not only for NATs/firewalls
but also for the proxy.

John

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
> Sent: 10 July 2006 15:32
> To: 'Jeroen van Bemmel'; sip@ietf.org
> Subject: RE: [Sip] STUN keep-alive: Client vs server 
> generating requests
> 
> 
> > -----Original Message-----
> > From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> > 
> > OK, I'll take your word for it. If we then interpret such a
> > 'stun-interval=xxx' as an indication of support for STUN 
> (and say 'no STUN
> > unless interval is provided'), I'm happy
> > 
> > Only trouble is: in many cases, the proxy would not know 
> which type of NAT
> > there is in between, let alone how long it caches...
> 
> If it doesn't know, then it sets the value recommended in the draft.
> However, there are ways for the proxy to find out, and there are also
> operators who know what the timeout is. (e.g., if the NAT is 
> provided by
> them)
> 
> -hadriel
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From: "Jason Fischl" <jason@counterpath.com>
To: "Dale.Worley@comcast.net" <Dale.Worley@comcast.net>
Subject: Re: [Sip] draft-ietf-sip-certs-00,
	communicating private keys and self-signed certificates
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On 7/11/06, Dale.Worley@comcast.net <Dale.Worley@comcast.net> wrote:
>    From: "Jason Fischl" <jason@counterpath.com>
>
>    > On the using a GRUU in a cert, would need to consider that any time
>    > my GRUU changes I need to generate a new cert - if this meant a new
>    > private/public key pair, this might be expensive.
>
>    Also consider that GRUUs will change relatively frequently for devices.
>
> Why would GRUUs change "relatively frequently"?  They can change only
> as frequently as the AOR changes, or the instance-id changes.
>

Yes. I was just confused. Sorry about that and thanks for pointing it out.

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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] draft-ietf-sip-certs-00,
	communicating private keys and self-signed certificates
Date: Wed, 12 Jul 2006 10:26:16 -0500
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On Jul 12, 2006, at 7:49 AM, Jason Fischl wrote:

> On 7/11/06, Dale.Worley@comcast.net <Dale.Worley@comcast.net> wrote:
>>    From: "Jason Fischl" <jason@counterpath.com>
>>
>>    > On the using a GRUU in a cert, would need to consider that  
>> any time
>>    > my GRUU changes I need to generate a new cert - if this meant  
>> a new
>>    > private/public key pair, this might be expensive.
>>
>>    Also consider that GRUUs will change relatively frequently for  
>> devices.
>>
>> Why would GRUUs change "relatively frequently"?  They can change only
>> as frequently as the AOR changes, or the instance-id changes.
>>
>
> Yes. I was just confused. Sorry about that and thanks for pointing  
> it out.
>

In practice, I would not be surprised if they actually change more  
frequently -- either with reboot of the assigning server, or perhaps  
on a database rebuild, major upgrade, etc. I'm not aware of anybody  
currently adding assigned GRUUs to their "long term provisioning  
database of record", which is what would be needed to maintain the  
GRUU as long as we might want.

--
Dean


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On Jul 7, 2006, at 12:45 AM, Jonathan Rosenberg wrote:

>
>
> One idea on dealing with the size problem is, instead of including  
> the sip-frag in a response, the resposne just indicates that the  
> hop limit was exceeded. The server which generated the response  
> then sends a request in a totally new transaction, towards the  
> contact of the request. That request contains the sip-frag with all  
> of the information, and could be sent over TCP if needed.
>
> That idea doesn't work for requests which omit contact, and also  
> means that intermediate proxies on the path of the failed request  
> might not see the diagnostic information, since it would come over  
> a different transaction.
>
> Anyway, just a thought.

JDR mentioned during today's WG discussion that he had previously  
suggested this idea on the list. I looked and found it hiding in a  
thread, and thought it was worth putting back up on the list.

So here you go. Maybe this will work.

--
Dean


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From: "Theresa Lovell" <TheresaLovell@mail.ru>
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     In other words, we didn't touch on the subject again. Kirill phoned PPS
     No limits, Jonathan? he thought. Well, then, the time's  not  distant
     "OK, sir. Next time I'll bring  my  dictionary along, sir." Then it was
in to his body, left  only  the  narrow  swept  daggers  of  his  wingtips



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From: "Elwell, John" <john.elwell@siemens.com>
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Cc: "Jesske, R" <R.Jesske@t-com.net>, christer.holmberg@ericsson.com
Subject: [Sip] Outcome of discussion on tispan-connected-identity draft
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During today's SIP meeting there was discussion of
draft-elwell-sip-tispan-connected-identity-01. This was in the context of
open issue 1 in draft-ietf-sip-connected-identity-00. My view of what
emerged was as follows (please correct me if necessary).

There were mixed views on the TISPAN use of the P-Asserted-Identity header
field to indicate the identity used for billing / authorisation purposes, as
opposed to the identity suitable for return call purposes that would be in
the From header field (with the possibility of authentication using Identity
header field). Opinions ranged from agreeing that it is reasonable, on the
one hand, to strong opposition to changing the meaning of the
P-Asserted-Identity header field on the other hand. Although ETSI TISPAN
requirements in this area are stated in
draft-jesske-sipping-tispan-requirements-02 and had been discussed in the
past, discussion with ETSI TISPAN on candidate mechanisms to satisfy these
requirements had not taken place previously. Because the primary topic of
the agenda slot was draft-ietf-sip-connected-identity, and because of the
absence of TISPAN liaison representatives in the meeting, it was not
possible to discuss the issue at length during this meeting. However, I was
encouraged to work with TISPAN representatives and assist them to contribute
further on the topic.

Open issue 1 in draft-ietf-sip-connected-identity-00 will be closed, with no
change. The document will not specify a response identity and will not
specify an option tag to inform a PSTN gateway that an UPDATE request will
follow the 200 response to the INVITE request. If any protocol enhancements
are required to support TISPAN callee identification requirements, these
will be covered in separate documents under a separate charter item.

TISPAN members on this mailing list (IETF SIP) are encouraged to take note
of the above.

John





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From sip-bounces@ietf.org Wed Jul 12 19:21:44 2006
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To follow up on the discussion in the SIP group. I think there are  
roughly three options for the content of Location:

(1) Point to content in the body  --> can't be modified and cannot be  
inserted by proxies (and may not be inspectable by proxies if encrypted)

(2) Point to external location --> can be modified, can be added, but  
can induce loops if changed in mid-stream and incurs another failure  
mode (and possible security risks if the reference is generally  
accessible)

(3) Point to data URL (RFC 2397) as in

Location: data:application/pidf+xml;base64,<base-64 version of PIDF-LO)

Disadvantage: about 30% larger than as body; can't be encrypted  
separately (except as XML-DSIG)

Advantages: can be inserted/modified by proxies (similar to (2)), but  
avoids difficulty of external reference

I think these three methods all have different time-space-flexibility  
trade-offs and thus I suspect they all serve a need.

Henning


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From sip-bounces@ietf.org Wed Jul 12 19:38:14 2006
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Hi Henning,

I like your idea of using the data URL.
It would work nicely for PIDF-LO and SAML assertions.

Ciao
Hannes


Henning Schulzrinne wrote:
> To follow up on the discussion in the SIP group. I think there are  
> roughly three options for the content of Location:
> 
> (1) Point to content in the body  --> can't be modified and cannot be  
> inserted by proxies (and may not be inspectable by proxies if encrypted)
> 
> (2) Point to external location --> can be modified, can be added, but  
> can induce loops if changed in mid-stream and incurs another failure  
> mode (and possible security risks if the reference is generally  
> accessible)
> 
> (3) Point to data URL (RFC 2397) as in
> 
> Location: data:application/pidf+xml;base64,<base-64 version of PIDF-LO)
> 
> Disadvantage: about 30% larger than as body; can't be encrypted  
> separately (except as XML-DSIG)
> 
> Advantages: can be inserted/modified by proxies (similar to (2)), but  
> avoids difficulty of external reference
> 
> I think these three methods all have different time-space-flexibility  
> trade-offs and thus I suspect they all serve a need.
> 
> Henning
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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   From: Dean Willis <dean.willis@softarmor.com>

   In practice, I would not be surprised if they actually change more  
   frequently -- either with reboot of the assigning server, or perhaps  
   on a database rebuild, major upgrade, etc. I'm not aware of anybody  
   currently adding assigned GRUUs to their "long term provisioning  
   database of record", which is what would be needed to maintain the  
   GRUU as long as we might want.

sipX (http://www.sipfoundry.org) constructs GRUUs by hashing the AOR,
instance ID, and a fixed key string.  So its GRUUs really are eternal.

Dale

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From sip-bounces@ietf.org Thu Jul 13 11:11:31 2006
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Subject: [Sip] Size of message/sipfrag in diagnostic responses
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One of the open issues with draft-ietf-hop-limit-diagnostics-0x (for all
x <= 3) is that the response will usually be very large.  If any hop in
the path is UDP, and the large response reaches that point, it may be
fragmented into so many parts that it: 
 - causes congestion problems in the reverse direction
 - fails reassembly (resulting in no response at all and 
   triggering retries)
Since either of these problems is arguably worse than not getting good
diagnostic information, we have concluded that something different needs
to be done.

I've spent some time trying to sort through the various suggestions on
how to cope with this, and I think that the simplest one is the best.  I
propose something like the following for the next revision of the draft:

        When sending an error response that has a message/sipfrag body,
        the responding server MUST examine all the Via header values in
        the request.  If any Via header value specifies UDP, then the
        total size of the response body MUST NOT be greater than 200
        bytes and the server MUST follow the procedures in section X to
        prune the returned message/sipfrag body to fit within this
        limit.  If no request Via header value specifies UDP, then the
        server is free to either prune the returned message/sipfrag body
        as specified in section X or to return the full set of request
        headers.

Even though this falls short of providing as much diagnostic information
as I'd like, I think that this is the best chance of getting _some_
useful information back to both the originator and to the proxies along
the path.  It means that you have a better chance of getting better
diagnostic information when you use TCP, and I think that may actually
provide an incentive that's in a good direction anyway.

The above text does not attempt to solve the general problem of SIP
response size; it is deliberately limited to the class of diagnostic
responses this draft is advocating.  I think that the SIP WG should try
to actually specify general rules for response sizes, but I would rather
decouple that from making this diagnostic information available.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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Robert Sparks wrote:
> I have another open issue on the current algorithm (as it is presented 
> in RFC 3261) which suggests that the "Via" header should be added, 
> without telling that, of course, the branch parameter should NOT be 
> used in Hash calculation (overwise it will always be different and 
> loop will never be detected...)
>  
> the fork-loop-fix draft (at version -02) _REPLACES_ all of that text 
> in 3261. I don't think the replacement text has the problem you
> point to does it?
This is the part which mentions the open issues:

> This second part MUST vary with any field used by the location
>
>    service logic in determining where to retarget or forward this
>    request.  This is necessary to distinguish looped requests from
>    spirals by allowing the proxy to recognize if none of the values
>    affecting the processing of the request have changed.  Hence, The
>    second part MUST depend at least on the received Request-URI and any
>    Route header field values in the received request.  Implementers need
>    to take care to include all fields used by the location service logic
>    in that particular implementation.
>
>    This second part MUST NOT include the request method.  CANCEL and
>    non-200 ACK requests MUST have the same branch parameter value as the
>    corresponding request they cancel or acknowledge.  This branch
>    parameter value is used in correlating those requests at the server
>    handling them (see Sections 17.2.3 and 9.2).
I am confused with this part:
there is no more sent-by part of the Via header? why is not mentionned 
in the open issue part then? (like for Call-Id ,toTag, and fromTag which
should actually be removed...)

> Another quick comment on the section 4 of 
> draft-ietf-sip-fork-loop-fix-02.txt
>>>  4.  Loop Detection Check
>>>
>>>    Proxies required to perform loop-detection by RFC-XXXX (RFC-Editor:
>>>    replace XXXX with the RFC number of this document) MUST perform the
>>>    following loop-detection test before forwarding a request.  Each Via
>>>    header field value in the request whose sent-by value matches a 
>>> value
>>>    placed into previous requests by this proxy MUST be inspected for 
>>> the
>>>    "second part" defined in Section 4.2.1 of RFC-XXXX.  This second 
>>> part
>>>    will not be present if the message was not forked when that Via
>>>    header field value was added.  If the second field is present, the
>>>    proxy MUST perform the second part calculation described in
>>>    Section 4.2.1 of RFC-XXXX on this request and compare the result to
>>>    the value from the Via header field.  If these values are equal, the
>>>    request has looped and the proxy MUST reject the request with a 482
>>>    (Loop Detected) response.  If the values differ, the request is
>>>    spiraling and processing continues to the next step.
>> The question here is, how do I know that the "second part" is present?
>> the common practice is to setup a branch with a "minus" separator, eg:
>> branch=z9hg4bKRANDOMID-LOOPDETECTIONHASH
>> however, it is also common pratice for many implementation to setup 
>> some additionnal information in their branch, also using the "minus" 
>> separator,
>> e.g:
>> branch=z9hG4bKRANDOMID-MYIMPLEMENTATIONSTUFF-LOOPDETECTIONHASH
>>
>> So, since the "presence" of minus (or any other) separator may not be 
>> enough, and in order to be able be sure the "loop detection part" is 
>> here,
>> wouldn't it be useful to setup another kind of "magic cookie" for that?
>
> So the only thing that will look into the branch for the second part 
> is the thing that put that second part there.
> You don't need to know how anyone else forms their branch parameter 
> with respect to this.
>
> Since you put the value in, you know how to recognize whether it's 
> there. I don't see importance in making everyone
> do this the same way.
because if somebody use -XX at the end of the branch parameter (and this 
is common practice), I will have to make unecessary computation on the 
incoming request...
One possible solution to avoid this is to have some kind of magic 
cookie, or a pre-defined separator... If I decide to use '--' as 
separator,  unfortunately, it may 
by used by somebody else in its branch format (and for something ELSE 
than loop detection)...



 


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From sip-bounces@ietf.org Thu Jul 13 11:38:38 2006
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Thu, 13 Jul 2006 17:21:04 +0200
To: Scott Lawrence <slawrence@pingtel.com>
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Another option is to return, in the 4xx response, an http URI pointing 
to a place where the server has stored the request. The requester can 
then go and pull the SIP request.

Might be an issue to interdomain cases, but not sure.

Hisham

On Jul 13, 2006, at 4:53 PM, Scott Lawrence wrote:

> One of the open issues with draft-ietf-hop-limit-diagnostics-0x (for 
> all
> x <= 3) is that the response will usually be very large.  If any hop in
> the path is UDP, and the large response reaches that point, it may be
> fragmented into so many parts that it:
>  - causes congestion problems in the reverse direction
>  - fails reassembly (resulting in no response at all and
>    triggering retries)
> Since either of these problems is arguably worse than not getting good
> diagnostic information, we have concluded that something different 
> needs
> to be done.
>
> I've spent some time trying to sort through the various suggestions on
> how to cope with this, and I think that the simplest one is the best.  
> I
> propose something like the following for the next revision of the 
> draft:
>
>         When sending an error response that has a message/sipfrag body,
>         the responding server MUST examine all the Via header values in
>         the request.  If any Via header value specifies UDP, then the
>         total size of the response body MUST NOT be greater than 200
>         bytes and the server MUST follow the procedures in section X to
>         prune the returned message/sipfrag body to fit within this
>         limit.  If no request Via header value specifies UDP, then the
>         server is free to either prune the returned message/sipfrag 
> body
>         as specified in section X or to return the full set of request
>         headers.
>
> Even though this falls short of providing as much diagnostic 
> information
> as I'd like, I think that this is the best chance of getting _some_
> useful information back to both the originator and to the proxies along
> the path.  It means that you have a better chance of getting better
> diagnostic information when you use TCP, and I think that may actually
> provide an incentive that's in a good direction anyway.
>
> The above text does not attempt to solve the general problem of SIP
> response size; it is deliberately limited to the class of diagnostic
> responses this draft is advocating.  I think that the SIP WG should try
> to actually specify general rules for response sizes, but I would 
> rather
> decouple that from making this diagnostic information available.
>
> -- 
> Scott Lawrence  tel:+1-781-938-5306;ext=162 or 
> sip:slawrence@pingtel.com
>   sipXpbx project coordinator - SIPfoundry    
> http://www.sipfoundry.org/sipX
>   Chief Architect             - Pingtel Corp. http://www.pingtel.com/
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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On Jul 13, 2006, at 11:14 AM, Thomas.Froment@alcatel.fr wrote:

> Robert Sparks wrote:
>> I have another open issue on the current algorithm (as it is  
>> presented in RFC 3261) which suggests that the "Via" header should  
>> be added, without telling that, of course, the branch parameter  
>> should NOT be used in Hash calculation (overwise it will always be  
>> different and loop will never be detected...)
>>  the fork-loop-fix draft (at version -02) _REPLACES_ all of that  
>> text in 3261. I don't think the replacement text has the problem you
>> point to does it?
> This is the part which mentions the open issues:
>
>> This second part MUST vary with any field used by the location
>>
>>    service logic in determining where to retarget or forward this
>>    request.  This is necessary to distinguish looped requests from
>>    spirals by allowing the proxy to recognize if none of the values
>>    affecting the processing of the request have changed.  Hence, The
>>    second part MUST depend at least on the received Request-URI  
>> and any
>>    Route header field values in the received request.   
>> Implementers need
>>    to take care to include all fields used by the location service  
>> logic
>>    in that particular implementation.
>>
>>    This second part MUST NOT include the request method.  CANCEL and
>>    non-200 ACK requests MUST have the same branch parameter value  
>> as the
>>    corresponding request they cancel or acknowledge.  This branch
>>    parameter value is used in correlating those requests at the  
>> server
>>    handling them (see Sections 17.2.3 and 9.2).
> I am confused with this part:
> there is no more sent-by part of the Via header? why is not  
> mentionned in the open issue part then? (like for Call-Id ,toTag,  
> and fromTag which
> should actually be removed...)

I didn't call that out as an open issue because we have clear  
consensus to not include the
topmost via in the calculation. I should have made a note of that in  
the changelog - sorry
for the confusion.

>
>> Another quick comment on the section 4 of draft-ietf-sip-fork-loop- 
>> fix-02.txt
>>>>  4.  Loop Detection Check
>>>>
>>>>    Proxies required to perform loop-detection by RFC-XXXX (RFC- 
>>>> Editor:
>>>>    replace XXXX with the RFC number of this document) MUST  
>>>> perform the
>>>>    following loop-detection test before forwarding a request.   
>>>> Each Via
>>>>    header field value in the request whose sent-by value matches  
>>>> a value
>>>>    placed into previous requests by this proxy MUST be inspected  
>>>> for the
>>>>    "second part" defined in Section 4.2.1 of RFC-XXXX.  This  
>>>> second part
>>>>    will not be present if the message was not forked when that Via
>>>>    header field value was added.  If the second field is  
>>>> present, the
>>>>    proxy MUST perform the second part calculation described in
>>>>    Section 4.2.1 of RFC-XXXX on this request and compare the  
>>>> result to
>>>>    the value from the Via header field.  If these values are  
>>>> equal, the
>>>>    request has looped and the proxy MUST reject the request with  
>>>> a 482
>>>>    (Loop Detected) response.  If the values differ, the request is
>>>>    spiraling and processing continues to the next step.
>>> The question here is, how do I know that the "second part" is  
>>> present?
>>> the common practice is to setup a branch with a "minus"  
>>> separator, eg:
>>> branch=z9hg4bKRANDOMID-LOOPDETECTIONHASH
>>> however, it is also common pratice for many implementation to  
>>> setup some additionnal information in their branch, also using  
>>> the "minus" separator,
>>> e.g:
>>> branch=z9hG4bKRANDOMID-MYIMPLEMENTATIONSTUFF-LOOPDETECTIONHASH
>>>
>>> So, since the "presence" of minus (or any other) separator may  
>>> not be enough, and in order to be able be sure the "loop  
>>> detection part" is here,
>>> wouldn't it be useful to setup another kind of "magic cookie" for  
>>> that?
>>
>> So the only thing that will look into the branch for the second  
>> part is the thing that put that second part there.
>> You don't need to know how anyone else forms their branch  
>> parameter with respect to this.
>>
>> Since you put the value in, you know how to recognize whether it's  
>> there. I don't see importance in making everyone
>> do this the same way.
> because if somebody use -XX at the end of the branch parameter (and  
> this is common practice), I will have to make unecessary  
> computation on the incoming request...
> One possible solution to avoid this is to have some kind of magic  
> cookie, or a pre-defined separator... If I decide to use '--' as  
> separator,  unfortunately, it may by used by somebody else in its  
> branch format (and for something ELSE than loop detection)...
>

What computation are you doing on other people's branch tag? You  
should only be looking inside ones you put there.

I think you're here at IETF - could we take this to an in-person  
conversation?

RjS




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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
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> >> I
> >> propose something like the following for the next revision of the 
> >> draft:
> >>
> >>         When sending an error response that has a message/sipfrag body,
> >>         the responding server MUST examine all the Via header values in
> >>         the request.  If any Via header value specifies UDP, then the
> >>         total size of the response body MUST NOT be greater than 200
> >>         bytes and the server MUST follow the procedures in section X to
> >>         prune the returned message/sipfrag body to fit within this
> >>         limit.  If no request Via header value specifies UDP, then the
> >>         server is free to either prune the returned message/sipfrag 
> >> body
> >>         as specified in section X or to return the full set of request

On Thu, 2006-07-13 at 17:47 +0200, Jeroen van Bemmel wrote:
> gzip on the body might help too

If I'm not mistaken, it's not trivial to figure out how much any given
text will compress with gzip unless you actually try each possibility. 

I'd rather preserve the human-readability and get one fewer headers.

Also, it doesn't do all that well on short inputs.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Thu Jul 13 14:18:21 2006
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
From: Scott Lawrence <slawrence@pingtel.com>
To: Hisham Khartabil <hisham.khartabil@telio.no>
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> > I
> > propose something like the following for the next revision of the 
> > draft:
> >
> >         When sending an error response that has a message/sipfrag body,
> >         the responding server MUST examine all the Via header values in
> >         the request.  If any Via header value specifies UDP, then the
> >         total size of the response body MUST NOT be greater than 200
> >         bytes and the server MUST follow the procedures in section X to
> >         prune the returned message/sipfrag body to fit within this
> >         limit.  If no request Via header value specifies UDP, then the
> >         server is free to either prune the returned message/sipfrag 
> > body
> >         as specified in section X or to return the full set of request
> >         headers.

On Thu, 2006-07-13 at 17:21 +0200, Hisham Khartabil wrote:
> Another option is to return, in the 4xx response, an http URI pointing 
> to a place where the server has stored the request. The requester can 
> then go and pull the SIP request.
> 
> Might be an issue to interdomain cases, but not sure.

I'm fairly sure that it would often be a problem.  

Even if all I get is the warning header, the request line and the
topmost via, I suspect it's more useful than a URL I might not be able
to reach (and I suspect that the URLs would often not be short).  

I also don't want to require that a proxy also be an http server in
order to provide this functionality.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Thu Jul 13 14:22:53 2006
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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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Hi Hisham,
I don't like the http URI concept for that because of the security,
performance, and reachability issues.  I imagine that intermediate nodes in
the return path may well want to log the diags info for certain response
failures, without user intervention.  With a mime attachment no problem
(well, no more than other sip/mime issues).  With an http uri it's an issue
for a bunch of probably obvious reasons.  I'd rather not have to create
diags/logs for the diags problems.  :)
Cheers,
-hadriel


> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no]
> Sent: Thursday, July 13, 2006 11:21 AM
> To: Scott Lawrence
> Cc: SIP WG List
> Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
> 
> Another option is to return, in the 4xx response, an http URI pointing
> to a place where the server has stored the request. The requester can
> then go and pull the SIP request.
> 
> Might be an issue to interdomain cases, but not sure.
> 
> Hisham
> 
> On Jul 13, 2006, at 4:53 PM, Scott Lawrence wrote:
> 
> > One of the open issues with draft-ietf-hop-limit-diagnostics-0x (for
> > all
> > x <= 3) is that the response will usually be very large.  If any hop in
> > the path is UDP, and the large response reaches that point, it may be
> > fragmented into so many parts that it:
> >  - causes congestion problems in the reverse direction
> >  - fails reassembly (resulting in no response at all and
> >    triggering retries)
> > Since either of these problems is arguably worse than not getting good
> > diagnostic information, we have concluded that something different
> > needs
> > to be done.
> >
> > I've spent some time trying to sort through the various suggestions on
> > how to cope with this, and I think that the simplest one is the best.
> > I
> > propose something like the following for the next revision of the
> > draft:
> >
> >         When sending an error response that has a message/sipfrag body,
> >         the responding server MUST examine all the Via header values in
> >         the request.  If any Via header value specifies UDP, then the
> >         total size of the response body MUST NOT be greater than 200
> >         bytes and the server MUST follow the procedures in section X to
> >         prune the returned message/sipfrag body to fit within this
> >         limit.  If no request Via header value specifies UDP, then the
> >         server is free to either prune the returned message/sipfrag
> > body
> >         as specified in section X or to return the full set of request
> >         headers.
> >
> > Even though this falls short of providing as much diagnostic
> > information
> > as I'd like, I think that this is the best chance of getting _some_
> > useful information back to both the originator and to the proxies along
> > the path.  It means that you have a better chance of getting better
> > diagnostic information when you use TCP, and I think that may actually
> > provide an incentive that's in a good direction anyway.
> >
> > The above text does not attempt to solve the general problem of SIP
> > response size; it is deliberately limited to the class of diagnostic
> > responses this draft is advocating.  I think that the SIP WG should try
> > to actually specify general rules for response sizes, but I would
> > rather
> > decouple that from making this diagnostic information available.
> >
> > --
> > Scott Lawrence  tel:+1-781-938-5306;ext=162 or
> > sip:slawrence@pingtel.com
> >   sipXpbx project coordinator - SIPfoundry
> > http://www.sipfoundry.org/sipX
> >   Chief Architect             - Pingtel Corp. http://www.pingtel.com/
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> 
> _______________________________________________
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From sip-bounces@ietf.org Thu Jul 13 15:39:01 2006
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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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Hi,

Inline below.

Thx
Samir

<< Removed Unrelated stuff>>

>=20
> On Thu, 2006-07-13 at 17:47 +0200, Jeroen van Bemmel wrote:
> > gzip on the body might help too
>=20
> If I'm not mistaken, it's not trivial to figure out how much=20
> any given text will compress with gzip unless you actually=20
> try each possibility.=20
>=20
> I'd rather preserve the human-readability and get one fewer headers.
>=20
> Also, it doesn't do all that well on short inputs.

SS>>  Gzip can achieve good compression ratio, if there are loops and
hop (RR, Via ) entries are repeated multiple times. Or should it be for
sigcomp ??

>=20
> --
> Scott Lawrence  tel:+1-781-938-5306;ext=3D162 or=20
> sip:slawrence@pingtel.com
>   sipXpbx project coordinator - SIPfoundry   =20
> http://www.sipfoundry.org/sipX
>   Chief Architect             - Pingtel Corp. http://www.pingtel.com/
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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>=20

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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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From: "Samir Srivastava" <samirsr@nortel.com>
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>=20
> Might be an issue to interdomain cases, but not sure.
>=20
> Hisham
>=20
>

Can SAML assertions take care of this ?

Thx
Samir=20

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This is a function we want to be implemented universally in proxies. So 
we don't want to raise the bar at all high. Incorporating a web server 
into every proxy seems to raise the bar. And retaining the message so it 
can be retrieved, without any clear idea how long that will be is 
another problem. There is a lot of attraction to just sending the 
response and forgetting it.

	Paul

Hisham Khartabil wrote:
> Another option is to return, in the 4xx response, an http URI pointing 
> to a place where the server has stored the request. The requester can 
> then go and pull the SIP request.
> 
> Might be an issue to interdomain cases, but not sure.
> 
> Hisham
> 
> On Jul 13, 2006, at 4:53 PM, Scott Lawrence wrote:
> 
>> One of the open issues with draft-ietf-hop-limit-diagnostics-0x (for all
>> x <= 3) is that the response will usually be very large.  If any hop in
>> the path is UDP, and the large response reaches that point, it may be
>> fragmented into so many parts that it:
>>  - causes congestion problems in the reverse direction
>>  - fails reassembly (resulting in no response at all and
>>    triggering retries)
>> Since either of these problems is arguably worse than not getting good
>> diagnostic information, we have concluded that something different needs
>> to be done.
>>
>> I've spent some time trying to sort through the various suggestions on
>> how to cope with this, and I think that the simplest one is the best.  I
>> propose something like the following for the next revision of the draft:
>>
>>         When sending an error response that has a message/sipfrag body,
>>         the responding server MUST examine all the Via header values in
>>         the request.  If any Via header value specifies UDP, then the
>>         total size of the response body MUST NOT be greater than 200
>>         bytes and the server MUST follow the procedures in section X to
>>         prune the returned message/sipfrag body to fit within this
>>         limit.  If no request Via header value specifies UDP, then the
>>         server is free to either prune the returned message/sipfrag body
>>         as specified in section X or to return the full set of request
>>         headers.
>>
>> Even though this falls short of providing as much diagnostic information
>> as I'd like, I think that this is the best chance of getting _some_
>> useful information back to both the originator and to the proxies along
>> the path.  It means that you have a better chance of getting better
>> diagnostic information when you use TCP, and I think that may actually
>> provide an incentive that's in a good direction anyway.
>>
>> The above text does not attempt to solve the general problem of SIP
>> response size; it is deliberately limited to the class of diagnostic
>> responses this draft is advocating.  I think that the SIP WG should try
>> to actually specify general rules for response sizes, but I would rather
>> decouple that from making this diagnostic information available.
>>
>> -- 
>> Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
>>   sipXpbx project coordinator - SIPfoundry    
>> http://www.sipfoundry.org/sipX
>>   Chief Architect             - Pingtel Corp. http://www.pingtel.com/
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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> 

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Subject: RE:[Sip] Size of message/sipfrag in diagnostic responses
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From: "Samir Srivastava" <samirsr@nortel.com>
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Inline.

> -----Original Message-----=20
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: Thursday, July 13, 2006 2:21 PM
> To: 'Hisham Khartabil'; 'Scott Lawrence'
> Cc: 'SIP WG List'
> Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
>=20
> Hi Hisham,
> I don't like the http URI concept for that because of the=20
> security, performance, and reachability issues.  I imagine=20
> that intermediate nodes in the return path may well want to=20
> log the diags info for certain response failures, without=20
> user intervention.  With a mime attachment no problem (well,=20
> no more than other sip/mime issues).  With an http uri it's=20
> an issue for a bunch of probably obvious reasons.  I'd rather=20
> not have to create diags/logs for the diags problems.  :)=20
> Cheers, -hadriel
>=20

SS>> Nobody wants to handle cascaded faults, diagnostics. If http
operation fails, that's it. Stop.

Apart from security concerns (Which may be asserted with SAML etc, I am
not sure as of now), what other concerns you have.

Thx
Samir

<< Removed unrelated stuff >>=20

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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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> > On Thu, 2006-07-13 at 17:47 +0200, Jeroen van Bemmel wrote:
> > > gzip on the body might help too
> > 
> > If I'm not mistaken, it's not trivial to figure out how much 
> > any given text will compress with gzip unless you actually 
> > try each possibility. 
> > 
> > I'd rather preserve the human-readability and get one fewer headers.
> > 
> > Also, it doesn't do all that well on short inputs.

On Thu, 2006-07-13 at 14:35 -0500, Samir Srivastava wrote:

> SS>>  Gzip can achieve good compression ratio, if there are loops and
> hop (RR, Via ) entries are repeated multiple times. Or should it be for
> sigcomp ??

My point was that the output of the gzip compression would have to be
less than 201 bytes.  You can't just truncate gzip output at any byte
boundary, so you'd have to select some part of the message to compress
and see if you got <201, then try again if you didn't, etc.  Too
complex.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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Fine. So what options are left ? Only compression or prunning.

IMHO, how proxies are configured. They must be having the web interface
for the configuration.

Or they are just plain old vanila text file/cmd line configuration
entities.

Thx
Samir=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Thursday, July 13, 2006 3:25 PM
> To: Hisham Khartabil
> Cc: SIP WG List
> Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
>=20
> This is a function we want to be implemented universally in=20
> proxies. So we don't want to raise the bar at all high.=20
> Incorporating a web server into every proxy seems to raise=20
> the bar. And retaining the message so it can be retrieved,=20
> without any clear idea how long that will be is another=20
> problem. There is a lot of attraction to just sending the=20
> response and forgetting it.
>=20
> 	Paul
>=20

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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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> My point was that the output of the gzip compression would=20
> have to be less than 201 bytes.  You can't just truncate gzip=20
> output at any byte boundary, so you'd have to select some=20
> part of the message to compress and see if you got <201, then=20
> try again if you didn't, etc.  Too complex.

 I was not asking to truncate the compressed output. Options available
are

  1) use sigcomp everywhere. And everything has limits, so it may not
help too in the extreme situations.

  2) http indirection

  3) At any hop, where proxy finds out that response will be increased
further going down, it just rejects the request rightaway asking for a
different transport. This check should be done before MAX-FORWARD check.
When proxy will be doing this, it will not have to attach the response
like in 483 in the body. It is different error code. I am not sure on
this what 3261 says. As we are changing the 3261 behaviour for 483
response, can we do this ? Backward compatibility ??

Thx
Samir

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Subject: RE: [Sip] Re: question on draft-ietf-sip-fork-loop-fix-02.txt
Date: Thu, 13 Jul 2006 16:03:15 -0500
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Hi,

  There is draft available for avoiding loops also.

http://www.ietf.org/internet-drafts/draft-srivastava-sipping-loop-avoida
nce-00.txt

It got comments from the group. Refer following threads for the solution
to them

http://www1.ietf.org/mail-archive/web/sipping/current/msg11084.html

http://www1.ietf.org/mail-archive/web/sipping/current/msg11085.html

http://www1.ietf.org/mail-archive/web/sipping/current/msg11097.html

I will be updating the draft with these. Appreciate your comments also
on this.

It has nice side effect of checking of SIP reachability at the addition
of bindings when they are spanning across the domains i.e. * bindings
are not added in vacum * .

Thx
Samir



> -----Original Message-----
> From: Thomas.Froment@alcatel.fr [mailto:Thomas.Froment@alcatel.fr]=20
> Sent: Thursday, July 13, 2006 11:15 AM
> To: Robert Sparks
> Cc: sip@ietf.org
> Subject: [Sip] Re: question on draft-ietf-sip-fork-loop-fix-02.txt
>=20
> Robert Sparks wrote:
> > I have another open issue on the current algorithm (as it=20
> is presented=20
> > in RFC 3261) which suggests that the "Via" header should be added,=20
> > without telling that, of course, the branch parameter should NOT be=20
> > used in Hash calculation (overwise it will always be different and=20
> > loop will never be detected...)
> > =20
> > the fork-loop-fix draft (at version -02) _REPLACES_ all of=20
> that text=20
> > in 3261. I don't think the replacement text has the problem=20
> you point=20
> > to does it?
> This is the part which mentions the open issues:
>=20
> > This second part MUST vary with any field used by the location
> >
> >    service logic in determining where to retarget or forward this
> >    request.  This is necessary to distinguish looped requests from
> >    spirals by allowing the proxy to recognize if none of the values
> >    affecting the processing of the request have changed.  Hence, The
> >    second part MUST depend at least on the received=20
> Request-URI and any
> >    Route header field values in the received request. =20
> Implementers need
> >    to take care to include all fields used by the location=20
> service logic
> >    in that particular implementation.
> >
> >    This second part MUST NOT include the request method.  CANCEL and
> >    non-200 ACK requests MUST have the same branch parameter=20
> value as the
> >    corresponding request they cancel or acknowledge.  This branch
> >    parameter value is used in correlating those requests at=20
> the server
> >    handling them (see Sections 17.2.3 and 9.2).
> I am confused with this part:
> there is no more sent-by part of the Via header? why is not=20
> mentionned in the open issue part then? (like for Call-Id=20
> ,toTag, and fromTag which should actually be removed...)
>=20
> > Another quick comment on the section 4 of=20
> > draft-ietf-sip-fork-loop-fix-02.txt
> >>>  4.  Loop Detection Check
> >>>
> >>>    Proxies required to perform loop-detection by RFC-XXXX=20
> (RFC-Editor:
> >>>    replace XXXX with the RFC number of this document)=20
> MUST perform the
> >>>    following loop-detection test before forwarding a=20
> request.  Each Via
> >>>    header field value in the request whose sent-by value=20
> matches a=20
> >>> value
> >>>    placed into previous requests by this proxy MUST be=20
> inspected for=20
> >>> the
> >>>    "second part" defined in Section 4.2.1 of RFC-XXXX. =20
> This second=20
> >>> part
> >>>    will not be present if the message was not forked when that Via
> >>>    header field value was added.  If the second field is=20
> present, the
> >>>    proxy MUST perform the second part calculation described in
> >>>    Section 4.2.1 of RFC-XXXX on this request and compare=20
> the result to
> >>>    the value from the Via header field.  If these values=20
> are equal, the
> >>>    request has looped and the proxy MUST reject the=20
> request with a 482
> >>>    (Loop Detected) response.  If the values differ, the request is
> >>>    spiraling and processing continues to the next step.
> >> The question here is, how do I know that the "second part"=20
> is present?
> >> the common practice is to setup a branch with a "minus"=20
> separator, eg:
> >> branch=3Dz9hg4bKRANDOMID-LOOPDETECTIONHASH
> >> however, it is also common pratice for many implementation=20
> to setup=20
> >> some additionnal information in their branch, also using=20
> the "minus"
> >> separator,
> >> e.g:
> >> branch=3Dz9hG4bKRANDOMID-MYIMPLEMENTATIONSTUFF-LOOPDETECTIONHASH
> >>
> >> So, since the "presence" of minus (or any other) separator=20
> may not be=20
> >> enough, and in order to be able be sure the "loop=20
> detection part" is=20
> >> here, wouldn't it be useful to setup another kind of=20
> "magic cookie"=20
> >> for that?
> >
> > So the only thing that will look into the branch for the=20
> second part=20
> > is the thing that put that second part there.
> > You don't need to know how anyone else forms their branch parameter=20
> > with respect to this.
> >
> > Since you put the value in, you know how to recognize whether it's=20
> > there. I don't see importance in making everyone do this=20
> the same way.
> because if somebody use -XX at the end of the branch=20
> parameter (and this is common practice), I will have to make=20
> unecessary computation on the incoming request...
> One possible solution to avoid this is to have some kind of=20
> magic cookie, or a pre-defined separator... If I decide to=20
> use '--' as separator,  unfortunately, it may by used by=20
> somebody else in its branch format (and for something ELSE=20
> than loop detection)...
>=20
>=20
>=20
> =20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20
>=20

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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Thu, 13 Jul 2006 16:14:16 -0500
To: Scott Lawrence <slawrence@pingtel.com>
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On Jul 13, 2006, at 9:53 AM, Scott Lawrence wrote:

>         When sending an error response that has a message/sipfrag  
> body,
>         the responding server MUST examine all the Via header  
> values in
>         the request.  If any Via header value specifies UDP, then the
>         total size of the response body MUST NOT be greater than 200
>         bytes and the server MUST follow the procedures in section  
> X to
>         prune the returned message/sipfrag body to fit within this
>         limit.  If no request Via header value specifies UDP, then the
>         server is free to either prune the returned message/sipfrag  
> body
>         as specified in section X or to return the full set of request
>         headers.

Since UDP is the default, you may need this to say "Does not specify  
TCP, TLS, or SCTP".

Also, this will still explode in environments where evil topology- 
hiding functions (common in SBCs) have suppressed the Via header  
fields. Some people would have me believe that this occurs in most if  
not all IMS deployments, giving us a low probability that large  
responses will ever actually work.

--
Dean



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From sip-bounces@ietf.org Thu Jul 13 18:07:37 2006
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After you presented in SIP meeting the 3 open issues of current SIP loop 
detection algorithm which are identified in sip-fork-loop-fix:
>   Open Issue 1 :  Why do all the Route header field values need to be
>       included?  Only the top two in the received request will normally
>       influence processing (in general, the topmost and any others this
>       proxy removes from the message, plus the topmost in the message
>       the proxy emits).  Can we simplify this to only include those
>       Route header field values that actually affect the processing of
>       the message?
>    Open Issue 2 :  RFC 3261 recommends Call-ID, To-tag, and From-tag and
>       CSeq be included in this second part.  Why?  This are guaranteed
>       to be invariant each time the request passes through this proxy.
>       Further the created value will only be compared with Vias received
>       from previous times the request went through this proxy, so there
>       is no possibility of comparing this value with a hash created from
>       some other request with a different Call-ID, To-tag, or From-tag
>       or CSeq.  I've removed these fields from the recommended set for
>       this version of the draft.
>   
Do we agree on these two?
>    Open Issue 3 :  RFC 3261 recommends including any Proxy-Require and
>       Proxy-Authorization header field values in the second part.  Why
>       it it valuable to call these out as special cases?  What is the
>       use case where they will be the only thing in a message returning
>       to a proxy that varies, making the return a spiral instead of a
>       loop?  I haven't been able to remember the motivation - we need to
>       re-identify that and capture it here or remove the text making
>       these anything other than a case covered by the abstract "fields
>       used by the location service logic".  This draft is taking the
>       second option.
I think Jonathan noticed during the meeting that he remembers a good 
reason to keep Proxy-Require and Proxy-Authorization, but I am not sure 
to know what is the final decision on this open issue.

I have another open issue on the current algorithm (as it is presented 
in RFC 3261) which suggests that the "Via" header should be added,
without telling that, of course, the branch parameter should NOT be used 
in Hash calculation (overwise it will always be different and loop will 
never be detected...)


Another quick comment on the section 4 of 
draft-ietf-sip-fork-loop-fix-02.txt
>  4.  Loop Detection Check
>
>    Proxies required to perform loop-detection by RFC-XXXX (RFC-Editor:
>    replace XXXX with the RFC number of this document) MUST perform the
>    following loop-detection test before forwarding a request.  Each Via
>    header field value in the request whose sent-by value matches a value
>    placed into previous requests by this proxy MUST be inspected for the
>    "second part" defined in Section 4.2.1 of RFC-XXXX.  This second part
>    will not be present if the message was not forked when that Via
>    header field value was added.  If the second field is present, the
>    proxy MUST perform the second part calculation described in
>    Section 4.2.1 of RFC-XXXX on this request and compare the result to
>    the value from the Via header field.  If these values are equal, the
>    request has looped and the proxy MUST reject the request with a 482
>    (Loop Detected) response.  If the values differ, the request is
>    spiraling and processing continues to the next step.
The question here is, how do I know that the "second part" is present?
the common practice is to setup a branch with a "minus" separator, eg:
branch=z9hg4bKRANDOMID-LOOPDETECTIONHASH
however, it is also common pratice for many implementation to setup some 
additionnal information in their branch, also using the "minus" separator,
e.g:
branch=z9hG4bKRANDOMID-MYIMPLEMENTATIONSTUFF-LOOPDETECTIONHASH

So, since the "presence" of minus (or any other) separator may not be 
enough, and in order to be able be sure the "loop detection part" is here,
wouldn't it be useful to setup another kind of "magic cookie" for that?

Thomas



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Right now, the ID lists options 1 and 2 as the only options, and this keeps 
things simple.  There have been a couple of comments here in Montreal that 
adding a data URL isn't really wanted.

One reason given is the inability to encrypt it (easily).  A message  body 
can be encypted using S/MIME, and a URI can have the server posessing the 
PIDF-LO configured to challenge a dereference request - including 
responding with an encrypted PIDF-LO.  Both these senarios have the PIDF-LO 
being able to be encrypted.  I'm not so sure encrypting the data URL is so 
straightforward, or deployable.  Others, please inform me and the list if 
this is untrue.

Cannot the ID proceed as is (in this respect), and once someone wants to 
extend this doc (once it's an RFC), propose that extention as a separate ID?

At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>To follow up on the discussion in the SIP group. I think there are
>roughly three options for the content of Location:
>
>(1) Point to content in the body  --> can't be modified and cannot be
>inserted by proxies (and may not be inspectable by proxies if encrypted)
>
>(2) Point to external location --> can be modified, can be added, but
>can induce loops if changed in mid-stream and incurs another failure
>mode (and possible security risks if the reference is generally
>accessible)
>
>(3) Point to data URL (RFC 2397) as in
>
>Location: data:application/pidf+xml;base64,<base-64 version of PIDF-LO)
>
>Disadvantage: about 30% larger than as body; can't be encrypted
>separately (except as XML-DSIG)
>
>Advantages: can be inserted/modified by proxies (similar to (2)), but
>avoids difficulty of external reference
>
>I think these three methods all have different time-space-flexibility
>trade-offs and thus I suspect they all serve a need.
>
>Henning
>
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] Location header
Date: Thu, 13 Jul 2006 19:08:51 -0400
To: "James M. Polk" <jmpolk@cisco.com>
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I think as long as the location information is used for routing (the  
only reason to have a Location header at all), the encryption  
argument is of dubious utility. I don't see any realistic possibility  
that the UAC can predict the public key of some intermediary router -  
by definition, by making routing dependent on location, you don't  
know where the request will end up. If you did, you'd put that  
address in the request-URI and only the UAS would have to know  
location (in which case we don't need the location header).

Henning


On Jul 13, 2006, at 6:55 PM, James M. Polk wrote:

> Right now, the ID lists options 1 and 2 as the only options, and  
> this keeps things simple.  There have been a couple of comments  
> here in Montreal that adding a data URL isn't really wanted.
>
> One reason given is the inability to encrypt it (easily).  A  
> message  body can be encypted using S/MIME, and a URI can have the  
> server posessing the PIDF-LO configured to challenge a dereference  
> request - including responding with an encrypted PIDF-LO.  Both  
> these senarios have the PIDF-LO being able to be encrypted.  I'm  
> not so sure encrypting the data URL is so straightforward, or  
> deployable.  Others, please inform me and the list if this is untrue.
>
> Cannot the ID proceed as is (in this respect), and once someone  
> wants to extend this doc (once it's an RFC), propose that extention  
> as a separate ID?
>
> At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>> To follow up on the discussion in the SIP group. I think there are
>> roughly three options for the content of Location:
>>
>> (1) Point to content in the body  --> can't be modified and cannot be
>> inserted by proxies (and may not be inspectable by proxies if  
>> encrypted)
>>
>> (2) Point to external location --> can be modified, can be added, but
>> can induce loops if changed in mid-stream and incurs another failure
>> mode (and possible security risks if the reference is generally
>> accessible)
>>
>> (3) Point to data URL (RFC 2397) as in
>>
>> Location: data:application/pidf+xml;base64,<base-64 version of  
>> PIDF-LO)
>>
>> Disadvantage: about 30% larger than as body; can't be encrypted
>> separately (except as XML-DSIG)
>>
>> Advantages: can be inserted/modified by proxies (similar to (2)), but
>> avoids difficulty of external reference
>>
>> I think these three methods all have different time-space-flexibility
>> trade-offs and thus I suspect they all serve a need.
>>
>> Henning
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip


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From sip-bounces@ietf.org Thu Jul 13 19:26:46 2006
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Hi James,

please explain me the difference (from a security point of view) between
* adding a location object into the header that is not encrypted, and
* adding a reference to a location object to the header that can be 
resolved into a location object by everyone that knows the reference?

Ciao
Hannes

James M. Polk wrote:
> Right now, the ID lists options 1 and 2 as the only options, and this 
> keeps things simple.  There have been a couple of comments here in 
> Montreal that adding a data URL isn't really wanted.
> 
> One reason given is the inability to encrypt it (easily).  A message  
> body can be encypted using S/MIME, and a URI can have the server 
> posessing the PIDF-LO configured to challenge a dereference request - 
> including responding with an encrypted PIDF-LO.  Both these senarios 
> have the PIDF-LO being able to be encrypted.  I'm not so sure encrypting 
> the data URL is so straightforward, or deployable.  Others, please 
> inform me and the list if this is untrue.
> 
> Cannot the ID proceed as is (in this respect), and once someone wants to 
> extend this doc (once it's an RFC), propose that extention as a separate 
> ID?
> 
> At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
> 
>> To follow up on the discussion in the SIP group. I think there are
>> roughly three options for the content of Location:
>>
>> (1) Point to content in the body  --> can't be modified and cannot be
>> inserted by proxies (and may not be inspectable by proxies if encrypted)
>>
>> (2) Point to external location --> can be modified, can be added, but
>> can induce loops if changed in mid-stream and incurs another failure
>> mode (and possible security risks if the reference is generally
>> accessible)
>>
>> (3) Point to data URL (RFC 2397) as in
>>
>> Location: data:application/pidf+xml;base64,<base-64 version of PIDF-LO)
>>
>> Disadvantage: about 30% larger than as body; can't be encrypted
>> separately (except as XML-DSIG)
>>
>> Advantages: can be inserted/modified by proxies (similar to (2)), but
>> avoids difficulty of external reference
>>
>> I think these three methods all have different time-space-flexibility
>> trade-offs and thus I suspect they all serve a need.
>>
>> Henning
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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From sip-bounces@ietf.org Thu Jul 13 19:33:23 2006
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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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That's not a problem, in terms of fragmentation.  They'll probably strip the
attachment anyway.  Or at least that's my guess. (I tried to say that in the
meeting)

I don't see much choice - it's too obvious a place to have all sorts of
internal sensitive headers exposed. (for example, generate a request into an
IMS network that you know will fail and trigger this diags response, and
learn about P-headers and trunk-group-ids and other fun things)  It's called
topology hiding for a reason.  :)

One could argue the most likely party capable of actually rectifying the
problem which caused the response is the same IMS network operator who
strips them.  So as long as the stripping device logged them they'll at
least have the info. (hence my previous email on logging them on proxies) 

-hadriel


> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> 
> Also, this will still explode in environments where evil topology-
> hiding functions (common in SBCs) have suppressed the Via header
> fields. Some people would have me believe that this occurs in most if
> not all IMS deployments, giving us a low probability that large
> responses will ever actually work.
> 
> --
> Dean
> 


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Well, the logical conclusion of this is pretty obvious. Why stop at =
PIDF-LO and SAML? Create a Body header, composed of a MIME type and a =
base64 encoded body, into which we can place location information, SAML =
assertions, SDP, instant messages, or anything else that would =
ordinarily be placed into a SIP message body. Deprecate the body of SIP =
messages, and our problems will be solved! Of course, the various =
Content-* headers would all need to be built into the Body header =
eventually, but I'm sure that won't stop us.

Rhetoric and layering concerns aside, I think the confidentiality =
analysis of header vs. body below is unpersuasive. While it is true that =
a body can be encrypted to prevent proxies from inspecting it, that is =
surely also true for the base64 encoded blob that would appear in the =
Body header, and the rational and circumstances associated with =
encrypting either are identical. A pointer to an external location, from =
my perspective, has much more interesting and powerful confidentiality =
properties than an encrypted blob (be it in a body or a header). There =
is no practical way I am aware of that a single body could be =
selectively revealed to multiple elements in a SIP message path other =
than through such a pointer to an external blob. What I mean is that the =
external server could challenge and selectively authorize requests to =
fetch the body which originate from various elements in the message =
path. Compound this with the "unanticipated recipient" problems =
associated with retargeting and you honestly begin to wonder if any =
other approach for confidentiality works in SIP.

I am also unpersuaded that an external reference is any more likely to =
induce loops than proxies using the Body header. After all, if two =
proxies keep changing the location information in such a way that it =
causes a loop, it hardly matters if that location information is =
provided by-reference or by-value - that is a problem with allowing =
proxies to modify location information at all.

I suspect it is only the integrity differences between header and body =
that are material here. Headers can be modified, bodies cannot. For some =
bodies, people argue this is a feature, for others a bug. From time to =
time people propose a flag signifying "this body can be modified". The =
current proposed flag is encoding the body as a header. The general =
problem with such flags, however they are cast, is that they are not =
themselves integrity-protected, and the recipient of a request has no =
way to know who applied the flags. So far, despite numerous challenges =
on numerous occasions, this WG has not exhibited a consensus to overturn =
the RFC3261 normative language that prevents intermediaries from =
changing bodies.

In short, I am not convinced that recasting bodies as headers =
accomplishes anything further than changing their integrity properties, =
and I remain quite skeptical that changing the integrity properties of =
bodies is warranted or wise.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> Sent: Wednesday, July 12, 2006 7:36 PM
> To: Henning Schulzrinne
> Cc: sip@ietf.org
> Subject: Re: [Sip] Location header
>=20
>=20
> Hi Henning,
>=20
> I like your idea of using the data URL.
> It would work nicely for PIDF-LO and SAML assertions.
>=20
> Ciao
> Hannes
>=20
>=20
> Henning Schulzrinne wrote:
> > To follow up on the discussion in the SIP group. I think there are =20
> > roughly three options for the content of Location:
> >=20
> > (1) Point to content in the body  --> can't be modified and=20
> cannot be =20
> > inserted by proxies (and may not be inspectable by proxies=20
> if encrypted)
> >=20
> > (2) Point to external location --> can be modified, can be=20
> added, but =20
> > can induce loops if changed in mid-stream and incurs=20
> another failure =20
> > mode (and possible security risks if the reference is generally =20
> > accessible)
> >=20
> > (3) Point to data URL (RFC 2397) as in
> >=20
> > Location: data:application/pidf+xml;base64,<base-64 version=20
> of PIDF-LO)
> >=20
> > Disadvantage: about 30% larger than as body; can't be encrypted =20
> > separately (except as XML-DSIG)
> >=20
> > Advantages: can be inserted/modified by proxies (similar to=20
> (2)), but =20
> > avoids difficulty of external reference
> >=20
> > I think these three methods all have different=20
> time-space-flexibility =20
> > trade-offs and thus I suspect they all serve a need.
> >=20
> > Henning
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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I'll avoid the temptation to pursue the general reductio-ad-absurdio  
arguments and focus on the proposal at hand. At least from my SIP  
background, SIP headers are appropriate for material meant for  
request routing and that are thus meant to be inspected and/or  
modified by proxies rather than (just) end systems. SDP presumably  
does not fall into that category, even though SBCs seem to think so.

Length or base-64 encoding doesn't seem to be determining factor of  
the use of a header vs. body since I suspect that the base-64 encoded  
version of the "Identity" header, to pick one example, will often be  
larger than the Location header proposal.

Thus, the only remaining difference between

Location: some pointer to a location object
and
Location: some location object

is the value vs. pointer distinction, i.e., one additional layer of  
dereferencing. I fail to see the deep architectural implications of  
this distinction, but maybe these can be explored.

The modifiability of the header vs. body is not a major issue since  
either can be signed (or not). Even in the current conveyance draft,  
a proxy can simply replace one http pointer with another if it  
doesn't like the original one, so I again fail to see the major  
difference.

We've discussed pointer vs. value issues in various places such as  
GEOPRIV, so I don't want to repeat the discussion, particularly since  
I'm likely to miss some of the arguments. My basic concern about  
changeability is not the proxies changing the location but the end  
system having no idea when and how long they need to keep the  
location information up-to-date and available. In some cases, it's a  
good idea for a moving object to update its location store, in others  
(such as routing) it is potentially a bad idea since you could  
suddenly find yourself at a proxy that doesn't know how to route the  
request.  (Picture a car crossing the county boundary where the  
internal county PSAP routing is done by a separate proxy.)

As a side issue, the draft currently provides no guidance on what  
kind of properties the referenced location information needs to have  
(how long to keep, whether it can change or not, etc.). This is more  
than just specifying HTTP GET.

Note that I did not advocate that end systems inserting location  
information use the data: mechanism, as they don't need to. The by- 
value case is mostly of interest in the case that we need a proxy to  
insert location information. Without this case, we wouldn't need the  
location header at all since proxies could just inspect the message  
body and find the location information by disposition information.  
For the case of proxy insertion, the proxy needs to maintain a  
location store that is accessible by all downstream proxies, by  
unknown entities and for unknown periods of time, including external  
(outside-in) traversal of firewalls. It would be nice to have a  
simpler option for that case.

Or are you opposed to proxies inserting location information? If so,  
I apologize for mis-understanding your basic argument.

Henning


On Jul 13, 2006, at 11:34 AM, Peterson, Jon wrote:

>
> Well, the logical conclusion of this is pretty obvious. Why stop at  
> PIDF-LO and SAML? Create a Body header, composed of a MIME type and  
> a base64 encoded body, into which we can place location  
> information, SAML assertions, SDP, instant messages, or anything  
> else that would ordinarily be placed into a SIP message body.  
> Deprecate the body of SIP messages, and our problems will be  
> solved! Of course, the various Content-* headers would all need to  
> be built into the Body header eventually, but I'm sure that won't  
> stop us.
>
> Rhetoric and layering concerns aside, I think the confidentiality  
> analysis of header vs. body below is unpersuasive. While it is true  
> that a body can be encrypted to prevent proxies from inspecting it,  
> that is surely also true for the base64 encoded blob that would  
> appear in the Body header, and the rational and circumstances  
> associated with encrypting either are identical. A pointer to an  
> external location, from my perspective, has much more interesting  
> and powerful confidentiality properties than an encrypted blob (be  
> it in a body or a header). There is no practical way I am aware of  
> that a single body could be selectively revealed to multiple  
> elements in a SIP message path other than through such a pointer to  
> an external blob. What I mean is that the external server could  
> challenge and selectively authorize requests to fetch the body  
> which originate from various elements in the message path. Compound  
> this with the "unanticipated recipient" problems associated with  
> retargeting and you honestly begin to wonder if any other approach  
> for confidentiality works in SIP.
>
> I am also unpersuaded that an external reference is any more likely  
> to induce loops than proxies using the Body header. After all, if  
> two proxies keep changing the location information in such a way  
> that it causes a loop, it hardly matters if that location  
> information is provided by-reference or by-value - that is a  
> problem with allowing proxies to modify location information at all.
>
> I suspect it is only the integrity differences between header and  
> body that are material here. Headers can be modified, bodies  
> cannot. For some bodies, people argue this is a feature, for others  
> a bug. From time to time people propose a flag signifying "this  
> body can be modified". The current proposed flag is encoding the  
> body as a header. The general problem with such flags, however they  
> are cast, is that they are not themselves integrity-protected, and  
> the recipient of a request has no way to know who applied the  
> flags. So far, despite numerous challenges on numerous occasions,  
> this WG has not exhibited a consensus to overturn the RFC3261  
> normative language that prevents intermediaries from changing bodies.
>
> In short, I am not convinced that recasting bodies as headers  
> accomplishes anything further than changing their integrity  
> properties, and I remain quite skeptical that changing the  
> integrity properties of bodies is warranted or wise.
>
> Jon Peterson
> NeuStar, Inc.
>
>> -----Original Message-----
>> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
>> Sent: Wednesday, July 12, 2006 7:36 PM
>> To: Henning Schulzrinne
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Location header
>>
>>
>> Hi Henning,
>>
>> I like your idea of using the data URL.
>> It would work nicely for PIDF-LO and SAML assertions.
>>
>> Ciao
>> Hannes
>>
>>
>> Henning Schulzrinne wrote:
>>> To follow up on the discussion in the SIP group. I think there are
>>> roughly three options for the content of Location:
>>>
>>> (1) Point to content in the body  --> can't be modified and
>> cannot be
>>> inserted by proxies (and may not be inspectable by proxies
>> if encrypted)
>>>
>>> (2) Point to external location --> can be modified, can be
>> added, but
>>> can induce loops if changed in mid-stream and incurs
>> another failure
>>> mode (and possible security risks if the reference is generally
>>> accessible)
>>>
>>> (3) Point to data URL (RFC 2397) as in
>>>
>>> Location: data:application/pidf+xml;base64,<base-64 version
>> of PIDF-LO)
>>>
>>> Disadvantage: about 30% larger than as body; can't be encrypted
>>> separately (except as XML-DSIG)
>>>
>>> Advantages: can be inserted/modified by proxies (similar to
>> (2)), but
>>> avoids difficulty of external reference
>>>
>>> I think these three methods all have different
>> time-space-flexibility
>>> trade-offs and thus I suspect they all serve a need.
>>>
>>> Henning
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
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>> Use sipping@ietf.org for new developments on the application of sip
>>


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From sip-bounces@ietf.org Thu Jul 13 23:36:58 2006
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To: "Hisham Khartabil" <hisham.khartabil@telio.no>,
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Thu, 13 Jul 2006 17:47:20 +0200
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gzip on the body might help too

Regards,

Jeroen

----- Original Message ----- 
From: "Hisham Khartabil" <hisham.khartabil@telio.no>
To: "Scott Lawrence" <slawrence@pingtel.com>
Cc: "SIP WG List" <sip@ietf.org>
Sent: Thursday, July 13, 2006 5:21 PM
Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses


> Another option is to return, in the 4xx response, an http URI pointing 
> to a place where the server has stored the request. The requester can 
> then go and pull the SIP request.
> 
> Might be an issue to interdomain cases, but not sure.
> 
> Hisham
> 
> On Jul 13, 2006, at 4:53 PM, Scott Lawrence wrote:
> 
>> One of the open issues with draft-ietf-hop-limit-diagnostics-0x (for 
>> all
>> x <= 3) is that the response will usually be very large.  If any hop in
>> the path is UDP, and the large response reaches that point, it may be
>> fragmented into so many parts that it:
>>  - causes congestion problems in the reverse direction
>>  - fails reassembly (resulting in no response at all and
>>    triggering retries)
>> Since either of these problems is arguably worse than not getting good
>> diagnostic information, we have concluded that something different 
>> needs
>> to be done.
>>
>> I've spent some time trying to sort through the various suggestions on
>> how to cope with this, and I think that the simplest one is the best.  
>> I
>> propose something like the following for the next revision of the 
>> draft:
>>
>>         When sending an error response that has a message/sipfrag body,
>>         the responding server MUST examine all the Via header values in
>>         the request.  If any Via header value specifies UDP, then the
>>         total size of the response body MUST NOT be greater than 200
>>         bytes and the server MUST follow the procedures in section X to
>>         prune the returned message/sipfrag body to fit within this
>>         limit.  If no request Via header value specifies UDP, then the
>>         server is free to either prune the returned message/sipfrag 
>> body
>>         as specified in section X or to return the full set of request
>>         headers.
>>
>> Even though this falls short of providing as much diagnostic 
>> information
>> as I'd like, I think that this is the best chance of getting _some_
>> useful information back to both the originator and to the proxies along
>> the path.  It means that you have a better chance of getting better
>> diagnostic information when you use TCP, and I think that may actually
>> provide an incentive that's in a good direction anyway.
>>
>> The above text does not attempt to solve the general problem of SIP
>> response size; it is deliberately limited to the class of diagnostic
>> responses this draft is advocating.  I think that the SIP WG should try
>> to actually specify general rules for response sizes, but I would 
>> rather
>> decouple that from making this diagnostic information available.
>>
>> -- 
>> Scott Lawrence  tel:+1-781-938-5306;ext=162 or 
>> sip:slawrence@pingtel.com
>>   sipXpbx project coordinator - SIPfoundry    
>> http://www.sipfoundry.org/sipX
>>   Chief Architect             - Pingtel Corp. http://www.pingtel.com/
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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On Jul 13, 2006, at 9:47 AM, Thomas.Froment@alcatel.fr wrote:

> After you presented in SIP meeting the 3 open issues of current SIP  
> loop detection algorithm which are identified in sip-fork-loop-fix:
>>   Open Issue 1 :  Why do all the Route header field values need to be
>>       included?  Only the top two in the received request will  
>> normally
>>       influence processing (in general, the topmost and any others  
>> this
>>       proxy removes from the message, plus the topmost in the message
>>       the proxy emits).  Can we simplify this to only include those
>>       Route header field values that actually affect the  
>> processing of
>>       the message?
>>    Open Issue 2 :  RFC 3261 recommends Call-ID, To-tag, and From- 
>> tag and
>>       CSeq be included in this second part.  Why?  This are  
>> guaranteed
>>       to be invariant each time the request passes through this  
>> proxy.
>>       Further the created value will only be compared with Vias  
>> received
>>       from previous times the request went through this proxy, so  
>> there
>>       is no possibility of comparing this value with a hash  
>> created from
>>       some other request with a different Call-ID, To-tag, or From- 
>> tag
>>       or CSeq.  I've removed these fields from the recommended set  
>> for
>>       this version of the draft.
>>
> Do we agree on these two?
>>    Open Issue 3 :  RFC 3261 recommends including any Proxy-Require  
>> and
>>       Proxy-Authorization header field values in the second part.   
>> Why
>>       it it valuable to call these out as special cases?  What is the
>>       use case where they will be the only thing in a message  
>> returning
>>       to a proxy that varies, making the return a spiral instead of a
>>       loop?  I haven't been able to remember the motivation - we  
>> need to
>>       re-identify that and capture it here or remove the text making
>>       these anything other than a case covered by the abstract  
>> "fields
>>       used by the location service logic".  This draft is taking the
>>       second option.
> I think Jonathan noticed during the meeting that he remembers a  
> good reason to keep Proxy-Require and Proxy-Authorization, but I am  
> not sure to know what is the final decision on this open issue.
>
> I have another open issue on the current algorithm (as it is  
> presented in RFC 3261) which suggests that the "Via" header should  
> be added,
> without telling that, of course, the branch parameter should NOT be  
> used in Hash calculation (overwise it will always be different and  
> loop will never be detected...)

the fork-loop-fix draft (at version -02) _REPLACES_ all of that text  
in 3261. I don't think the replacement text has the problem you
point to does it?

>
>
> Another quick comment on the section 4 of draft-ietf-sip-fork-loop- 
> fix-02.txt
>>  4.  Loop Detection Check
>>
>>    Proxies required to perform loop-detection by RFC-XXXX (RFC- 
>> Editor:
>>    replace XXXX with the RFC number of this document) MUST perform  
>> the
>>    following loop-detection test before forwarding a request.   
>> Each Via
>>    header field value in the request whose sent-by value matches a  
>> value
>>    placed into previous requests by this proxy MUST be inspected  
>> for the
>>    "second part" defined in Section 4.2.1 of RFC-XXXX.  This  
>> second part
>>    will not be present if the message was not forked when that Via
>>    header field value was added.  If the second field is present, the
>>    proxy MUST perform the second part calculation described in
>>    Section 4.2.1 of RFC-XXXX on this request and compare the  
>> result to
>>    the value from the Via header field.  If these values are  
>> equal, the
>>    request has looped and the proxy MUST reject the request with a  
>> 482
>>    (Loop Detected) response.  If the values differ, the request is
>>    spiraling and processing continues to the next step.
> The question here is, how do I know that the "second part" is present?
> the common practice is to setup a branch with a "minus" separator, eg:
> branch=z9hg4bKRANDOMID-LOOPDETECTIONHASH
> however, it is also common pratice for many implementation to setup  
> some additionnal information in their branch, also using the  
> "minus" separator,
> e.g:
> branch=z9hG4bKRANDOMID-MYIMPLEMENTATIONSTUFF-LOOPDETECTIONHASH
>
> So, since the "presence" of minus (or any other) separator may not  
> be enough, and in order to be able be sure the "loop detection  
> part" is here,
> wouldn't it be useful to setup another kind of "magic cookie" for  
> that?

So the only thing that will look into the branch for the second part  
is the thing that put that second part there.
You don't need to know how anyone else forms their branch parameter  
with respect to this.

Since you put the value in, you know how to recognize whether it's  
there. I don't see importance in making everyone
do this the same way.

RjS

>
> Thomas
>
>


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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
Date: Fri, 14 Jul 2006 00:01:00 -0500
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From: "Samir Srivastava" <samirsr@nortel.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
	"Hisham Khartabil" <hisham.khartabil@telio.no>,
	"Scott Lawrence" <slawrence@pingtel.com>
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Joren,

If it is good case where all hops are different, what will be the
compression ratio ?

If it is looping/spiraling, then it will be having good compression
ratio for 483.=20

What about in general (No hop limit use case, no body in the response)
where response size is big ?

Thx
Samir  =20

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]=20
> Sent: Thursday, July 13, 2006 11:47 AM
> To: Hisham Khartabil; Scott Lawrence
> Cc: SIP WG List
> Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
>=20
> gzip on the body might help too
>=20
> Regards,
>=20
> Jeroen
>=20
>=20

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To: Hannes Tschofenig <Hannes.Tschofenig@gmx.net>
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Subject: Re: [Sip] Location header
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At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
>Hi James,
>
>please explain me the difference (from a security point of view) between
>* adding a location object into the header that is not encrypted, and
>* adding a reference to a location object to the header that can be 
>resolved into a location object by everyone that knows the reference?

The server which has the by-referenced PIDF-LO can challenge whoever asks 
for the PIDF-LO.  The header URI allows for each intermediary to ask for 
the dereference, but only those that have permission, surviving the 
challenge(s), get the PIDF-LO.  This means not all dereference attempts 
learn the UAC's location.  This cannot be accomplished in a header.


>Ciao
>Hannes
>
>James M. Polk wrote:
>>Right now, the ID lists options 1 and 2 as the only options, and this 
>>keeps things simple.  There have been a couple of comments here in 
>>Montreal that adding a data URL isn't really wanted.
>>One reason given is the inability to encrypt it (easily).  A message
>>body can be encypted using S/MIME, and a URI can have the server 
>>posessing the PIDF-LO configured to challenge a dereference request - 
>>including responding with an encrypted PIDF-LO.  Both these senarios have 
>>the PIDF-LO being able to be encrypted.  I'm not so sure encrypting the 
>>data URL is so straightforward, or deployable.  Others, please inform me 
>>and the list if this is untrue.
>>Cannot the ID proceed as is (in this respect), and once someone wants to 
>>extend this doc (once it's an RFC), propose that extention as a separate ID?
>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>>
>>>To follow up on the discussion in the SIP group. I think there are
>>>roughly three options for the content of Location:
>>>
>>>(1) Point to content in the body  --> can't be modified and cannot be
>>>inserted by proxies (and may not be inspectable by proxies if encrypted)
>>>
>>>(2) Point to external location --> can be modified, can be added, but
>>>can induce loops if changed in mid-stream and incurs another failure
>>>mode (and possible security risks if the reference is generally
>>>accessible)
>>>
>>>(3) Point to data URL (RFC 2397) as in
>>>
>>>Location: data:application/pidf+xml;base64,<base-64 version of PIDF-LO)
>>>
>>>Disadvantage: about 30% larger than as body; can't be encrypted
>>>separately (except as XML-DSIG)
>>>
>>>Advantages: can be inserted/modified by proxies (similar to (2)), but
>>>avoids difficulty of external reference
>>>
>>>I think these three methods all have different time-space-flexibility
>>>trade-offs and thus I suspect they all serve a need.
>>>
>>>Henning
>>>
>>>
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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>>
>>_______________________________________________
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>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Fri Jul 14 02:58:25 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Fri, 14 Jul 2006 08:56:23 +0200
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Samir,

I was thinking specifically for the 483 case, reporting the via sequence.

You have {200 + the size of the original body + request headers that you can 
omit + savings from using short header forms} to respond with, perhaps a 
simple text-based run-length encoding scheme could work as well.

E.g.

111.222.111.222:1234/UDP,111.222.111.223:1234/UDP,17(111.222.111.224:1234/UDP,111.222.111.225:1234/UDP),17(111.222.111.224:1234/TCP,111.222.111.225:1234/TCP)

FQDNs could be slightly longer, but assuming about 40 bytes per entry, you 
could report at least 5 (omitting the last one (UA itself) and the 
one-but-last one, since the UA will know those). Perhaps omitting the 
transport (port too?) (UA will know initial transport, if required only the 
moment of transistion could be encoded). Then you could report 10 entries 
easily. Add to that a string host encoding:

1=111.222.111.222:1234
2=111.222.111.223:1234
3=111.222.111.224:1234
4=111.222.111.225:1234

1,2,34(3,4)

Regards,

Jeroen


----- Original Message ----- 
From: "Samir Srivastava" <samirsr@nortel.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>; "Hisham Khartabil" 
<hisham.khartabil@telio.no>; "Scott Lawrence" <slawrence@pingtel.com>
Cc: "SIP WG List" <sip@ietf.org>
Sent: Friday, July 14, 2006 7:01 AM
Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses


Joren,

If it is good case where all hops are different, what will be the
compression ratio ?

If it is looping/spiraling, then it will be having good compression
ratio for 483.

What about in general (No hop limit use case, no body in the response)
where response size is big ?

Thx
Samir

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Thursday, July 13, 2006 11:47 AM
> To: Hisham Khartabil; Scott Lawrence
> Cc: SIP WG List
> Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
>
> gzip on the body might help too
>
> Regards,
>
> Jeroen
>
> 


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From sip-bounces@ietf.org Fri Jul 14 03:15:34 2006
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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
Date: Fri, 14 Jul 2006 02:13:20 -0500
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From: "Samir Srivastava" <samirsr@nortel.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
	"Hisham Khartabil" <hisham.khartabil@telio.no>,
	"Scott Lawrence" <slawrence@pingtel.com>
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Hi,

There was discussion in the meeting, whether response message exceeding
should be fixed in general or 483 only.

Chairs (Dean) said that Hop limit diagnostics cannot move without that
fix.

Thx
Samir=20

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]=20
> Sent: Friday, July 14, 2006 2:56 AM
> To: Srivastava, Samir (SC100:8826); Hisham Khartabil; Scott Lawrence
> Cc: SIP WG List
> Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
>=20
> Samir,
>=20
> I was thinking specifically for the 483 case, reporting the=20
> via sequence.
>=20
> You have {200 + the size of the original body + request=20
> headers that you can omit + savings from using short header=20
> forms} to respond with, perhaps a simple text-based=20
> run-length encoding scheme could work as well.
>=20
> E.g.
>=20
> 111.222.111.222:1234/UDP,111.222.111.223:1234/UDP,17(111.222.1
> 11.224:1234/UDP,111.222.111.225:1234/UDP),17(111.222.111.224:1
> 234/TCP,111.222.111.225:1234/TCP)
>=20
> FQDNs could be slightly longer, but assuming about 40 bytes=20
> per entry, you could report at least 5 (omitting the last one=20
> (UA itself) and the one-but-last one, since the UA will know=20
> those). Perhaps omitting the transport (port too?) (UA will=20
> know initial transport, if required only the moment of=20
> transistion could be encoded). Then you could report 10=20
> entries easily. Add to that a string host encoding:
>=20
> 1=3D111.222.111.222:1234
> 2=3D111.222.111.223:1234
> 3=3D111.222.111.224:1234
> 4=3D111.222.111.225:1234
>=20
> 1,2,34(3,4)
>=20
> Regards,
>=20
> Jeroen
>=20
>=20
> ----- Original Message -----
> From: "Samir Srivastava" <samirsr@nortel.com>
> To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>; "Hisham Khartabil"=20
> <hisham.khartabil@telio.no>; "Scott Lawrence" <slawrence@pingtel.com>
> Cc: "SIP WG List" <sip@ietf.org>
> Sent: Friday, July 14, 2006 7:01 AM
> Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
>=20
>=20
> Joren,
>=20
> If it is good case where all hops are different, what will be the
> compression ratio ?
>=20
> If it is looping/spiraling, then it will be having good compression
> ratio for 483.
>=20
> What about in general (No hop limit use case, no body in the response)
> where response size is big ?
>=20
> Thx
> Samir
>=20
> > -----Original Message-----
> > From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> > Sent: Thursday, July 13, 2006 11:47 AM
> > To: Hisham Khartabil; Scott Lawrence
> > Cc: SIP WG List
> > Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
> >
> > gzip on the body might help too
> >
> > Regards,
> >
> > Jeroen
> >
> >=20
>=20
>=20

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This list is for NEW development of the core SIP Protocol
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Dear All,

I don't know if this is the correct mailing list to put my question and =
if not my apologies for that.

Consider the following IMS scenario


(IMS core Network) -> S-CSCF -> AS


If the connection between AS and S-CSCF fails what is the expected =
behavior of S-CSCF? It must release all active calls sending BYE =
messages to caller and called clients in order to terminate the calls?

Thanks in advance

Regards

AA=20



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From sip-bounces@ietf.org Fri Jul 14 09:47:42 2006
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Hi James,

James M. Polk wrote:
> At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
> 
>> Hi James,
>>
>> please explain me the difference (from a security point of view) between
>> * adding a location object into the header that is not encrypted, and
>> * adding a reference to a location object to the header that can be 
>> resolved into a location object by everyone that knows the reference?
> 
> 
> The server which has the by-referenced PIDF-LO can challenge whoever 
> asks for the PIDF-LO.  The header URI allows for each intermediary to 
> ask for the dereference, but only those that have permission, surviving 
> the challenge(s), get the PIDF-LO.  This means not all dereference 
> attempts learn the UAC's location.  This cannot be accomplished in a 
> header.
This would assume that the entity that created the location object has a 
  relationship with the final location recipient (and to share some 
credential with them).

I think that this is a totally unrealistic assumption.


Ciao
Hannes

> 
> 
>> Ciao
>> Hannes
>>
>> James M. Polk wrote:
>>
>>> Right now, the ID lists options 1 and 2 as the only options, and this 
>>> keeps things simple.  There have been a couple of comments here in 
>>> Montreal that adding a data URL isn't really wanted.
>>> One reason given is the inability to encrypt it (easily).  A message
>>> body can be encypted using S/MIME, and a URI can have the server 
>>> posessing the PIDF-LO configured to challenge a dereference request - 
>>> including responding with an encrypted PIDF-LO.  Both these senarios 
>>> have the PIDF-LO being able to be encrypted.  I'm not so sure 
>>> encrypting the data URL is so straightforward, or deployable.  
>>> Others, please inform me and the list if this is untrue.
>>> Cannot the ID proceed as is (in this respect), and once someone wants 
>>> to extend this doc (once it's an RFC), propose that extention as a 
>>> separate ID?
>>> At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>>>
>>>> To follow up on the discussion in the SIP group. I think there are
>>>> roughly three options for the content of Location:
>>>>
>>>> (1) Point to content in the body  --> can't be modified and cannot be
>>>> inserted by proxies (and may not be inspectable by proxies if 
>>>> encrypted)
>>>>
>>>> (2) Point to external location --> can be modified, can be added, but
>>>> can induce loops if changed in mid-stream and incurs another failure
>>>> mode (and possible security risks if the reference is generally
>>>> accessible)
>>>>
>>>> (3) Point to data URL (RFC 2397) as in
>>>>
>>>> Location: data:application/pidf+xml;base64,<base-64 version of PIDF-LO)
>>>>
>>>> Disadvantage: about 30% larger than as body; can't be encrypted
>>>> separately (except as XML-DSIG)
>>>>
>>>> Advantages: can be inserted/modified by proxies (similar to (2)), but
>>>> avoids difficulty of external reference
>>>>
>>>> I think these three methods all have different time-space-flexibility
>>>> trade-offs and thus I suspect they all serve a need.
>>>>
>>>> Henning
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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> 
> 
> 


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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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On Fri, 2006-07-14 at 02:13 -0500, Samir Srivastava wrote:
> Hi,
> 
> There was discussion in the meeting, whether response message exceeding
> should be fixed in general or 483 only.
> 
> Chairs (Dean) said that Hop limit diagnostics cannot move without that
> fix.

The guidance I have received is that we cannot proceed with specifying a
new diagnostic response behavior without at least limiting the size of
the responses it generates for paths that include UDP.  

We are not required (and have not attempted in the proposed text) to
solve the problem of response sizes in general.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
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On Thu, 2006-07-13 at 15:18 -0500, Samir Srivastava wrote:
> > My point was that the output of the gzip compression would 
> > have to be less than 201 bytes.  You can't just truncate gzip 
> > output at any byte boundary, so you'd have to select some 
> > part of the message to compress and see if you got <201, then 
> > try again if you didn't, etc.  Too complex.
> 
>  I was not asking to truncate the compressed output. Options available
> are
> 
>   1) use sigcomp everywhere. And everything has limits, so it may not
> help too in the extreme situations.

I believe that requiring sigcomp support in a proxy in order to provide
these responses would be too large a barrier to deployment.  I want this
to be so easy that lots of implementations will do it quickly.

>   2) http indirection

I think this has been dealt with.

>   3) At any hop, where proxy finds out that response will be increased
> further going down, it just rejects the request rightaway asking for a
> different transport. This check should be done before MAX-FORWARD check.
> When proxy will be doing this, it will not have to attach the response
> like in 483 in the body. It is different error code. I am not sure on
> this what 3261 says. As we are changing the 3261 behaviour for 483
> response, can we do this ? Backward compatibility ??

There is no way for a proxy to detect that a request would fail further
along the path.

Yes, we are updating what 3261 says.  I have been doing interoperability
testing of a variant of these responses for 2 years and (other than the
UDP size problem), have not encountered any UA that had a problem with
these responses (even ones that had _lots_ of other problems with
perfectly valid SIP).


-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
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From sip-bounces@ietf.org Fri Jul 14 10:21:04 2006
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
From: Scott Lawrence <slawrence@pingtel.com>
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On Fri, 2006-07-14 at 08:56 +0200, Jeroen van Bemmel wrote:
> Samir,
> 
> I was thinking specifically for the 483 case, reporting the via sequence.

This is another issue that I'll start a thread on soon, but I'd like to
be able to send this kind of diagnostic response for some other errors
as well, and any solution to the size question should, I think, be
independent of the error type.

> You have {200 + the size of the original body + request headers that you can 
> omit + savings from using short header forms} to respond with, perhaps a 
> simple text-based run-length encoding scheme could work as well.

I believe that it is important to minimize the manipulation of the
returned diagnostic information because that introduces opportunities
for the rejecting server to misrepresent the request, which adds to the
problem.

Pruning the response is (alas) unavoidable given UDP.  The
message/sipfrag specification (as I understand it) allow for modifying
the message only by removing entire headers, and I think that's the most
we should do.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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Hi Mike,

Michael Hammer (mhammer) wrote:
> Hannes,
> 
> If you are right, then you are saying that symmetric secret keys are out
> of the question.  That leaves some sort of public/private key type of
> operation.

The problem is not only the authentication but the authorization.

Virtually everyone could be location recipient.

> 
> Contrary to what Jon suggested earlier, there are ways of passing an
> encrypted object to multiple recipients, namely by using a secret key to
> encrypt the LO, then adding the secret key enrypted by the public key of
> the intended recipient as an object for each recipient.

We haven't even come to the problem of multiple recipient with a single 
message.

... and: S/MIME does not provide these capabilities.

   This also has
> the characteristic that down-stream nodes upon learning of additional
> recipients, can add new Pu(secret) objects for those new recipients as
> needed.  That does assume a chain of trust among those given access to
> the secret.
> 
> Is that practical?  I think so, but others may have their opinion.
> 


This is just an optimization of encrypting location objects to 
individual recipients. I am not sure we are even closen to optimizations.


> BTW, can someone re-summarize what the requirements are?

Requirements for what exactly? Most of the requirements are listed in 
the document provided by James & Brian.

> It must be modifiable -> must be a header not body
> It must provide confidentiality -> encrypted or selectively handed out
> (by ref)
> Downstream recipient unknown to UAC must have access -> ???
> ...
What is 'it'?


Ciao
Hannes

> 
> Mike
> 
> 
> 
>>-----Original Message-----
>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>>Sent: Friday, July 14, 2006 9:39 AM
>>To: James Polk (jmpolk)
>>Cc: sip@ietf.org; Henning Schulzrinne
>>Subject: Re: [Sip] Location header
>>
>>Hi James,
>>
>>James M. Polk wrote:
>>
>>>At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
>>>
>>>
>>>>Hi James,
>>>>
>>>>please explain me the difference (from a security point of view) 
>>>>between
>>>>* adding a location object into the header that is not 
>>
>>encrypted, and
>>
>>>>* adding a reference to a location object to the header 
>>
>>that can be 
>>
>>>>resolved into a location object by everyone that knows the 
>>
>>reference?
>>
>>>
>>>The server which has the by-referenced PIDF-LO can 
>>
>>challenge whoever 
>>
>>>asks for the PIDF-LO.  The header URI allows for each 
>>
>>intermediary to 
>>
>>>ask for the dereference, but only those that have permission, 
>>>surviving the challenge(s), get the PIDF-LO.  This means not all 
>>>dereference attempts learn the UAC's location.  This cannot be 
>>>accomplished in a header.
>>
>>This would assume that the entity that created the location 
>>object has a
>>  relationship with the final location recipient (and to 
>>share some credential with them).
>>
>>I think that this is a totally unrealistic assumption.
>>
>>
>>Ciao
>>Hannes
>>
>>
>>>
>>>>Ciao
>>>>Hannes
>>>>
>>>>James M. Polk wrote:
>>>>
>>>>
>>>>>Right now, the ID lists options 1 and 2 as the only options, and 
>>>>>this keeps things simple.  There have been a couple of 
>>
>>comments here 
>>
>>>>>in Montreal that adding a data URL isn't really wanted.
>>>>>One reason given is the inability to encrypt it (easily). 
>>
>> A message 
>>
>>>>>body can be encypted using S/MIME, and a URI can have the server 
>>>>>posessing the PIDF-LO configured to challenge a 
>>
>>dereference request 
>>
>>>>>- including responding with an encrypted PIDF-LO.  Both these 
>>>>>senarios have the PIDF-LO being able to be encrypted.  I'm not so 
>>>>>sure encrypting the data URL is so straightforward, or deployable.
>>>>>Others, please inform me and the list if this is untrue.
>>>>>Cannot the ID proceed as is (in this respect), and once someone 
>>>>>wants to extend this doc (once it's an RFC), propose that 
>>
>>extention 
>>
>>>>>as a separate ID?
>>>>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>>>>>
>>>>>
>>>>>>To follow up on the discussion in the SIP group. I think 
>>
>>there are 
>>
>>>>>>roughly three options for the content of Location:
>>>>>>
>>>>>>(1) Point to content in the body  --> can't be modified 
>>
>>and cannot 
>>
>>>>>>be inserted by proxies (and may not be inspectable by proxies if
>>>>>>encrypted)
>>>>>>
>>>>>>(2) Point to external location --> can be modified, can 
>>
>>be added, 
>>
>>>>>>but can induce loops if changed in mid-stream and incurs another 
>>>>>>failure mode (and possible security risks if the reference is 
>>>>>>generally
>>>>>>accessible)
>>>>>>
>>>>>>(3) Point to data URL (RFC 2397) as in
>>>>>>
>>>>>>Location: data:application/pidf+xml;base64,<base-64 version of 
>>>>>>PIDF-LO)
>>>>>>
>>>>>>Disadvantage: about 30% larger than as body; can't be encrypted 
>>>>>>separately (except as XML-DSIG)
>>>>>>
>>>>>>Advantages: can be inserted/modified by proxies (similar 
>>
>>to (2)), 
>>
>>>>>>but avoids difficulty of external reference
>>>>>>
>>>>>>I think these three methods all have different 
>>>>>>time-space-flexibility trade-offs and thus I suspect 
>>
>>they all serve a need.
>>
>>>>>>Henning
>>>>>>
>>>>>>
>>>>>>_______________________________________________
>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>sip-implementors@cs.columbia.edu for questions on 
>>
>>current sip Use 
>>
>>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>>>sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol Use 
>>sip-implementors@cs.columbia.edu for questions on current sip 
>>Use sipping@ietf.org for new developments on the application of sip
>>
> 
> 
> 


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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] Location header
Date: Fri, 14 Jul 2006 10:19:56 -0400
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On Jul 14, 2006, at 10:05 AM, Michael Hammer ((mhammer)) wrote:

> Hannes,
>
> Contrary to what Jon suggested earlier, there are ways of passing an
> encrypted object to multiple recipients, namely by using a secret  
> key to
> encrypt the LO, then adding the secret key enrypted by the public  
> key of
> the intended recipient as an object for each recipient.  This also has
> the characteristic that down-stream nodes upon learning of additional
> recipients, can add new Pu(secret) objects for those new recipients as
> needed.  That does assume a chain of trust among those given access to
> the secret.
>
> Is that practical?  I think so, but others may have their opinion.
>

I suspect there are multiple deployment scenarios, so it is probably  
impossible to say that it is (or not), in general. However, we might  
take two location-based routing scenarios that we have discussed:

(1) emergency calling

(2) calling for towing service

For either, I'm having a really hard time picturing how the sender  
(or the originating VSP) can know the credentials or even identity of  
the proxies that are on the signaling path. My VSP is located in  
Germany; it is unlikely to have any relationship with the Montreal  
fire department or, more importantly, the routing model of the  
emergency calling network in Canada. I think the model that has been  
discussed is that there might be a province/state proxy, which routes  
to a county proxy which routes to a city proxy. Only the first one  
would be visible in LoST, so the caller has no way to determine the  
identity or public key of the other two.

Clearly, there are going to be cases where such multiple-destination  
encryption is feasible, but this has no bearing on the value vs.  
reference discussion, since you can do the kind of thing you  
mentioned either way.


> BTW, can someone re-summarize what the requirements are?
> It must be modifiable -> must be a header not body

Right.

> It must provide confidentiality -> encrypted or selectively handed out
> (by ref)

I would qualify this as saying that it must be able to do this in  
some cases. Like other routing-related information (To, caller  
preferences, etc.), it's unlikely to be common, given your last  
requirement.

Assuming you use TLS, the extra encryption of the information offers  
almost no value, since all those that are destinations for a TLS  
session also need to inspect the Location information. This is why we  
don't encrypt the From/To header, Route header and other similar  
routing/policy information, either.

By the way, an interesting model is to split the routing from the end  
system information, if they need different degrees of precision: The  
routing-related information may only need city-level precision, for  
example, and the end-to-end information would have house-level  
precision. This doesn't help much with emergency calling since the  
caller won't know the destination public key ahead of time, as far as  
I can tell. (Unless all PSAPs in a state, say, share the same public/ 
private key pair.)


> Downstream recipient unknown to UAC must have access -> ???

Indeed, with the minor qualification of "recipient*s*", i.e., there  
might be several.

Henning



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Subject: RE: [Sip] Location header
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Hannes,

I went back and re-read those requirements.  My problem here is that the
arguments for one approach versus another are not always covered by
those requirements.  In other words, a requirement is being inferred
from the general requirement "do privacy".  So, if someone argues that X
is not good enough because it doesn't meet Y, and Y is not a
requirement, then the argument is baseless.

Much of the debate surrounds assumptions about topology and trust
relationships (or not) between various nodes.

Disagreement on assumptions/requirements -> Disagreement on solutions.

Mike


> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: Friday, July 14, 2006 10:20 AM
> To: Michael Hammer (mhammer)
> Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
> Subject: Re: [Sip] Location header
>=20
> Hi Mike,
>=20
> Michael Hammer (mhammer) wrote:
> > Hannes,
> >=20
> > If you are right, then you are saying that symmetric secret=20
> keys are=20
> > out of the question.  That leaves some sort of=20
> public/private key type=20
> > of operation.
>=20
> The problem is not only the authentication but the authorization.
>=20
> Virtually everyone could be location recipient.
>=20
> >=20
> > Contrary to what Jon suggested earlier, there are ways of=20
> passing an=20
> > encrypted object to multiple recipients, namely by using a=20
> secret key=20
> > to encrypt the LO, then adding the secret key enrypted by=20
> the public=20
> > key of the intended recipient as an object for each recipient.
>=20
> We haven't even come to the problem of multiple recipient=20
> with a single message.
>=20
> ... and: S/MIME does not provide these capabilities.
>=20
>    This also has
> > the characteristic that down-stream nodes upon learning of=20
> additional=20
> > recipients, can add new Pu(secret) objects for those new=20
> recipients as=20
> > needed.  That does assume a chain of trust among those=20
> given access to=20
> > the secret.
> >=20
> > Is that practical?  I think so, but others may have their opinion.
> >=20
>=20
>=20
> This is just an optimization of encrypting location objects=20
> to individual recipients. I am not sure we are even closen to=20
> optimizations.
>=20
>=20
> > BTW, can someone re-summarize what the requirements are?
>=20
> Requirements for what exactly? Most of the requirements are listed in=20
> the document provided by James & Brian.
>=20
> > It must be modifiable -> must be a header not body
> > It must provide confidentiality -> encrypted or selectively=20
> handed out
> > (by ref)
> > Downstream recipient unknown to UAC must have access -> ???
> > ...
> What is 'it'?
>=20
>=20
> Ciao
> Hannes
>=20
> >=20
> > Mike
> >=20
> >=20
> >=20
> >>-----Original Message-----
> >>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> >>Sent: Friday, July 14, 2006 9:39 AM
> >>To: James Polk (jmpolk)
> >>Cc: sip@ietf.org; Henning Schulzrinne
> >>Subject: Re: [Sip] Location header
> >>
> >>Hi James,
> >>
> >>James M. Polk wrote:
> >>
> >>>At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
> >>>
> >>>
> >>>>Hi James,
> >>>>
> >>>>please explain me the difference (from a security point of view)=20
> >>>>between
> >>>>* adding a location object into the header that is not=20
> >>
> >>encrypted, and
> >>
> >>>>* adding a reference to a location object to the header=20
> >>
> >>that can be=20
> >>
> >>>>resolved into a location object by everyone that knows the=20
> >>
> >>reference?
> >>
> >>>
> >>>The server which has the by-referenced PIDF-LO can=20
> >>
> >>challenge whoever=20
> >>
> >>>asks for the PIDF-LO.  The header URI allows for each=20
> >>
> >>intermediary to=20
> >>
> >>>ask for the dereference, but only those that have permission,=20
> >>>surviving the challenge(s), get the PIDF-LO.  This means not all=20
> >>>dereference attempts learn the UAC's location.  This cannot be=20
> >>>accomplished in a header.
> >>
> >>This would assume that the entity that created the location=20
> >>object has a
> >>  relationship with the final location recipient (and to=20
> >>share some credential with them).
> >>
> >>I think that this is a totally unrealistic assumption.
> >>
> >>
> >>Ciao
> >>Hannes
> >>
> >>
> >>>
> >>>>Ciao
> >>>>Hannes
> >>>>
> >>>>James M. Polk wrote:
> >>>>
> >>>>
> >>>>>Right now, the ID lists options 1 and 2 as the only options, and=20
> >>>>>this keeps things simple.  There have been a couple of=20
> >>
> >>comments here=20
> >>
> >>>>>in Montreal that adding a data URL isn't really wanted.
> >>>>>One reason given is the inability to encrypt it (easily).=20
> >>
> >> A message=20
> >>
> >>>>>body can be encypted using S/MIME, and a URI can have the server=20
> >>>>>posessing the PIDF-LO configured to challenge a=20
> >>
> >>dereference request=20
> >>
> >>>>>- including responding with an encrypted PIDF-LO.  Both these=20
> >>>>>senarios have the PIDF-LO being able to be encrypted. =20
> I'm not so=20
> >>>>>sure encrypting the data URL is so straightforward, or=20
> deployable.
> >>>>>Others, please inform me and the list if this is untrue.
> >>>>>Cannot the ID proceed as is (in this respect), and once someone=20
> >>>>>wants to extend this doc (once it's an RFC), propose that=20
> >>
> >>extention=20
> >>
> >>>>>as a separate ID?
> >>>>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
> >>>>>
> >>>>>
> >>>>>>To follow up on the discussion in the SIP group. I think=20
> >>
> >>there are=20
> >>
> >>>>>>roughly three options for the content of Location:
> >>>>>>
> >>>>>>(1) Point to content in the body  --> can't be modified=20
> >>
> >>and cannot=20
> >>
> >>>>>>be inserted by proxies (and may not be inspectable by proxies if
> >>>>>>encrypted)
> >>>>>>
> >>>>>>(2) Point to external location --> can be modified, can=20
> >>
> >>be added,=20
> >>
> >>>>>>but can induce loops if changed in mid-stream and=20
> incurs another=20
> >>>>>>failure mode (and possible security risks if the reference is=20
> >>>>>>generally
> >>>>>>accessible)
> >>>>>>
> >>>>>>(3) Point to data URL (RFC 2397) as in
> >>>>>>
> >>>>>>Location: data:application/pidf+xml;base64,<base-64 version of=20
> >>>>>>PIDF-LO)
> >>>>>>
> >>>>>>Disadvantage: about 30% larger than as body; can't be encrypted=20
> >>>>>>separately (except as XML-DSIG)
> >>>>>>
> >>>>>>Advantages: can be inserted/modified by proxies (similar=20
> >>
> >>to (2)),=20
> >>
> >>>>>>but avoids difficulty of external reference
> >>>>>>
> >>>>>>I think these three methods all have different=20
> >>>>>>time-space-flexibility trade-offs and thus I suspect=20
> >>
> >>they all serve a need.
> >>
> >>>>>>Henning
> >>>>>>
> >>>>>>
> >>>>>>_______________________________________________
> >>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>>This list is for NEW development of the core SIP Protocol Use=20
> >>>>>>sip-implementors@cs.columbia.edu for questions on=20
> >>
> >>current sip Use=20
> >>
> >>>>>>sipping@ietf.org for new developments on the application of sip
> >>>>>
> >>>>>
> >>>>>_______________________________________________
> >>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>This list is for NEW development of the core SIP Protocol Use=20
> >>>>>sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>>>sipping@ietf.org for new developments on the application of sip
> >>>
> >>>
> >>>
> >>
> >>_______________________________________________
> >>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>This list is for NEW development of the core SIP Protocol Use=20
> >>sip-implementors@cs.columbia.edu for questions on current sip=20
> >>Use sipping@ietf.org for new developments on the application of sip
> >>
> >=20
> >=20
> >=20
>=20

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Subject: RE: [Sip] Location header
Date: Fri, 14 Jul 2006 11:15:42 -0400
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Thread-Topic: [Sip] Location header
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>
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I wouldn't call them my requirements per se, I was just trying to read
between the lines of the discussion and extract why some folks were
arguing for or against based on a statement or requirement/assumption.

The inclusion v. reference and header v. body arguments aside, it seemed
as though there were two dimensions here:

Set of nodes with which the Client has trust relationship versus not
Set of nodes that have permitted "need to know" location versus not

1) Relationship + need to know
2) No Relation + need to know
3) Relationship + no need to know
4) No Relation + need to know

I thought each of the above needed to be addressed.  But, my suggestion
for 4) based on overlapping trust groups seemed to be deemed jumping
ahead of the immediate issues.
=20
Mike


> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: Friday, July 14, 2006 10:50 AM
> To: Michael Hammer (mhammer)
> Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
> Subject: Re: [Sip] Location header
>=20
> Hi Mike,
>=20
> I agree that the current document does not cover all the=20
> requirements and does not go into a deep discussion of the=20
> security/privacy aspects.=20
> I am happy to discuss these.
>=20
> Maybe you could restate your requirements again in more details.
>=20
> Ciao
> Hannes
>=20
> Michael Hammer (mhammer) wrote:
> > Hannes,
> >=20
> > I went back and re-read those requirements.  My problem=20
> here is that=20
> > the arguments for one approach versus another are not=20
> always covered=20
> > by those requirements.  In other words, a requirement is being=20
> > inferred from the general requirement "do privacy".  So, if someone=20
> > argues that X is not good enough because it doesn't meet Y,=20
> and Y is=20
> > not a requirement, then the argument is baseless.
> >=20
> > Much of the debate surrounds assumptions about topology and trust=20
> > relationships (or not) between various nodes.
> >=20
> > Disagreement on assumptions/requirements -> Disagreement on=20
> solutions.
> >=20
> > Mike
> >=20
> >=20
> >=20
> >>-----Original Message-----
> >>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> >>Sent: Friday, July 14, 2006 10:20 AM
> >>To: Michael Hammer (mhammer)
> >>Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
> >>Subject: Re: [Sip] Location header
> >>
> >>Hi Mike,
> >>
> >>Michael Hammer (mhammer) wrote:
> >>
> >>>Hannes,
> >>>
> >>>If you are right, then you are saying that symmetric secret
> >>
> >>keys are
> >>
> >>>out of the question.  That leaves some sort of
> >>
> >>public/private key type
> >>
> >>>of operation.
> >>
> >>The problem is not only the authentication but the authorization.
> >>
> >>Virtually everyone could be location recipient.
> >>
> >>
> >>>Contrary to what Jon suggested earlier, there are ways of
> >>
> >>passing an
> >>
> >>>encrypted object to multiple recipients, namely by using a
> >>
> >>secret key
> >>
> >>>to encrypt the LO, then adding the secret key enrypted by
> >>
> >>the public
> >>
> >>>key of the intended recipient as an object for each recipient.
> >>
> >>We haven't even come to the problem of multiple recipient with a=20
> >>single message.
> >>
> >>... and: S/MIME does not provide these capabilities.
> >>
> >>   This also has
> >>
> >>>the characteristic that down-stream nodes upon learning of
> >>
> >>additional
> >>
> >>>recipients, can add new Pu(secret) objects for those new
> >>
> >>recipients as
> >>
> >>>needed.  That does assume a chain of trust among those
> >>
> >>given access to
> >>
> >>>the secret.
> >>>
> >>>Is that practical?  I think so, but others may have their opinion.
> >>>
> >>
> >>
> >>This is just an optimization of encrypting location objects to=20
> >>individual recipients. I am not sure we are even closen to=20
> >>optimizations.
> >>
> >>
> >>
> >>>BTW, can someone re-summarize what the requirements are?
> >>
> >>Requirements for what exactly? Most of the requirements are=20
> listed in=20
> >>the document provided by James & Brian.
> >>
> >>
> >>>It must be modifiable -> must be a header not body It must provide=20
> >>>confidentiality -> encrypted or selectively
> >>
> >>handed out
> >>
> >>>(by ref)
> >>>Downstream recipient unknown to UAC must have access -> ???
> >>>...
> >>
> >>What is 'it'?
> >>
> >>
> >>Ciao
> >>Hannes
> >>
> >>
> >>>Mike
> >>>
> >>>
> >>>
> >>>
> >>>>-----Original Message-----
> >>>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> >>>>Sent: Friday, July 14, 2006 9:39 AM
> >>>>To: James Polk (jmpolk)
> >>>>Cc: sip@ietf.org; Henning Schulzrinne
> >>>>Subject: Re: [Sip] Location header
> >>>>
> >>>>Hi James,
> >>>>
> >>>>James M. Polk wrote:
> >>>>
> >>>>
> >>>>>At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
> >>>>>
> >>>>>
> >>>>>
> >>>>>>Hi James,
> >>>>>>
> >>>>>>please explain me the difference (from a security point=20
> of view)=20
> >>>>>>between
> >>>>>>* adding a location object into the header that is not
> >>>>
> >>>>encrypted, and
> >>>>
> >>>>
> >>>>>>* adding a reference to a location object to the header
> >>>>
> >>>>that can be
> >>>>
> >>>>
> >>>>>>resolved into a location object by everyone that knows the
> >>>>
> >>>>reference?
> >>>>
> >>>>
> >>>>>The server which has the by-referenced PIDF-LO can
> >>>>
> >>>>challenge whoever
> >>>>
> >>>>
> >>>>>asks for the PIDF-LO.  The header URI allows for each
> >>>>
> >>>>intermediary to
> >>>>
> >>>>
> >>>>>ask for the dereference, but only those that have permission,=20
> >>>>>surviving the challenge(s), get the PIDF-LO.  This means not all=20
> >>>>>dereference attempts learn the UAC's location.  This cannot be=20
> >>>>>accomplished in a header.
> >>>>
> >>>>This would assume that the entity that created the=20
> location object=20
> >>>>has a  relationship with the final location recipient=20
> (and to share=20
> >>>>some credential with them).
> >>>>
> >>>>I think that this is a totally unrealistic assumption.
> >>>>
> >>>>
> >>>>Ciao
> >>>>Hannes
> >>>>
> >>>>
> >>>>
> >>>>>>Ciao
> >>>>>>Hannes
> >>>>>>
> >>>>>>James M. Polk wrote:
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>Right now, the ID lists options 1 and 2 as the only=20
> options, and=20
> >>>>>>>this keeps things simple.  There have been a couple of
> >>>>
> >>>>comments here
> >>>>
> >>>>
> >>>>>>>in Montreal that adding a data URL isn't really wanted.
> >>>>>>>One reason given is the inability to encrypt it (easily).=20
> >>>>
> >>>>A message
> >>>>
> >>>>
> >>>>>>>body can be encypted using S/MIME, and a URI can have=20
> the server=20
> >>>>>>>posessing the PIDF-LO configured to challenge a
> >>>>
> >>>>dereference request
> >>>>
> >>>>
> >>>>>>>- including responding with an encrypted PIDF-LO.  Both these=20
> >>>>>>>senarios have the PIDF-LO being able to be encrypted.
> >>
> >>I'm not so
> >>
> >>>>>>>sure encrypting the data URL is so straightforward, or
> >>
> >>deployable.
> >>
> >>>>>>>Others, please inform me and the list if this is untrue.
> >>>>>>>Cannot the ID proceed as is (in this respect), and=20
> once someone=20
> >>>>>>>wants to extend this doc (once it's an RFC), propose that
> >>>>
> >>>>extention
> >>>>
> >>>>
> >>>>>>>as a separate ID?
> >>>>>>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>>To follow up on the discussion in the SIP group. I think
> >>>>
> >>>>there are
> >>>>
> >>>>
> >>>>>>>>roughly three options for the content of Location:
> >>>>>>>>
> >>>>>>>>(1) Point to content in the body  --> can't be modified
> >>>>
> >>>>and cannot
> >>>>
> >>>>
> >>>>>>>>be inserted by proxies (and may not be inspectable by=20
> proxies if
> >>>>>>>>encrypted)
> >>>>>>>>
> >>>>>>>>(2) Point to external location --> can be modified, can
> >>>>
> >>>>be added,
> >>>>
> >>>>
> >>>>>>>>but can induce loops if changed in mid-stream and
> >>
> >>incurs another
> >>
> >>>>>>>>failure mode (and possible security risks if the reference is=20
> >>>>>>>>generally
> >>>>>>>>accessible)
> >>>>>>>>
> >>>>>>>>(3) Point to data URL (RFC 2397) as in
> >>>>>>>>
> >>>>>>>>Location: data:application/pidf+xml;base64,<base-64 version of
> >>>>>>>>PIDF-LO)
> >>>>>>>>
> >>>>>>>>Disadvantage: about 30% larger than as body; can't be=20
> encrypted=20
> >>>>>>>>separately (except as XML-DSIG)
> >>>>>>>>
> >>>>>>>>Advantages: can be inserted/modified by proxies (similar
> >>>>
> >>>>to (2)),
> >>>>
> >>>>
> >>>>>>>>but avoids difficulty of external reference
> >>>>>>>>
> >>>>>>>>I think these three methods all have different=20
> >>>>>>>>time-space-flexibility trade-offs and thus I suspect
> >>>>
> >>>>they all serve a need.
> >>>>
> >>>>
> >>>>>>>>Henning
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>_______________________________________________
> >>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>>>>This list is for NEW development of the core SIP Protocol Use=20
> >>>>>>>>sip-implementors@cs.columbia.edu for questions on
> >>>>
> >>>>current sip Use
> >>>>
> >>>>
> >>>>>>>>sipping@ietf.org for new developments on the=20
> application of sip
> >>>>>>>
> >>>>>>>
> >>>>>>>_______________________________________________
> >>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>>>This list is for NEW development of the core SIP Protocol Use=20
> >>>>>>>sip-implementors@cs.columbia.edu for questions on
> >>
> >>current sip Use
> >>
> >>>>>>>sipping@ietf.org for new developments on the application of sip
> >>>>>
> >>>>>
> >>>>>
> >>>>_______________________________________________
> >>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>This list is for NEW development of the core SIP Protocol Use=20
> >>>>sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>>>sipping@ietf.org for new developments on the application of sip
> >>>>
> >>>
> >>>
> >>>
> >=20
> >=20
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Fri Jul 14 11:18:09 2006
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Hi Mike,

I agree that the current document does not cover all the requirements 
and does not go into a deep discussion of the security/privacy aspects. 
I am happy to discuss these.

Maybe you could restate your requirements again in more details.

Ciao
Hannes

Michael Hammer (mhammer) wrote:
> Hannes,
> 
> I went back and re-read those requirements.  My problem here is that the
> arguments for one approach versus another are not always covered by
> those requirements.  In other words, a requirement is being inferred
> from the general requirement "do privacy".  So, if someone argues that X
> is not good enough because it doesn't meet Y, and Y is not a
> requirement, then the argument is baseless.
> 
> Much of the debate surrounds assumptions about topology and trust
> relationships (or not) between various nodes.
> 
> Disagreement on assumptions/requirements -> Disagreement on solutions.
> 
> Mike
> 
> 
> 
>>-----Original Message-----
>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>>Sent: Friday, July 14, 2006 10:20 AM
>>To: Michael Hammer (mhammer)
>>Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
>>Subject: Re: [Sip] Location header
>>
>>Hi Mike,
>>
>>Michael Hammer (mhammer) wrote:
>>
>>>Hannes,
>>>
>>>If you are right, then you are saying that symmetric secret 
>>
>>keys are 
>>
>>>out of the question.  That leaves some sort of 
>>
>>public/private key type 
>>
>>>of operation.
>>
>>The problem is not only the authentication but the authorization.
>>
>>Virtually everyone could be location recipient.
>>
>>
>>>Contrary to what Jon suggested earlier, there are ways of 
>>
>>passing an 
>>
>>>encrypted object to multiple recipients, namely by using a 
>>
>>secret key 
>>
>>>to encrypt the LO, then adding the secret key enrypted by 
>>
>>the public 
>>
>>>key of the intended recipient as an object for each recipient.
>>
>>We haven't even come to the problem of multiple recipient 
>>with a single message.
>>
>>... and: S/MIME does not provide these capabilities.
>>
>>   This also has
>>
>>>the characteristic that down-stream nodes upon learning of 
>>
>>additional 
>>
>>>recipients, can add new Pu(secret) objects for those new 
>>
>>recipients as 
>>
>>>needed.  That does assume a chain of trust among those 
>>
>>given access to 
>>
>>>the secret.
>>>
>>>Is that practical?  I think so, but others may have their opinion.
>>>
>>
>>
>>This is just an optimization of encrypting location objects 
>>to individual recipients. I am not sure we are even closen to 
>>optimizations.
>>
>>
>>
>>>BTW, can someone re-summarize what the requirements are?
>>
>>Requirements for what exactly? Most of the requirements are listed in 
>>the document provided by James & Brian.
>>
>>
>>>It must be modifiable -> must be a header not body
>>>It must provide confidentiality -> encrypted or selectively 
>>
>>handed out
>>
>>>(by ref)
>>>Downstream recipient unknown to UAC must have access -> ???
>>>...
>>
>>What is 'it'?
>>
>>
>>Ciao
>>Hannes
>>
>>
>>>Mike
>>>
>>>
>>>
>>>
>>>>-----Original Message-----
>>>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>>>>Sent: Friday, July 14, 2006 9:39 AM
>>>>To: James Polk (jmpolk)
>>>>Cc: sip@ietf.org; Henning Schulzrinne
>>>>Subject: Re: [Sip] Location header
>>>>
>>>>Hi James,
>>>>
>>>>James M. Polk wrote:
>>>>
>>>>
>>>>>At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
>>>>>
>>>>>
>>>>>
>>>>>>Hi James,
>>>>>>
>>>>>>please explain me the difference (from a security point of view) 
>>>>>>between
>>>>>>* adding a location object into the header that is not 
>>>>
>>>>encrypted, and
>>>>
>>>>
>>>>>>* adding a reference to a location object to the header 
>>>>
>>>>that can be 
>>>>
>>>>
>>>>>>resolved into a location object by everyone that knows the 
>>>>
>>>>reference?
>>>>
>>>>
>>>>>The server which has the by-referenced PIDF-LO can 
>>>>
>>>>challenge whoever 
>>>>
>>>>
>>>>>asks for the PIDF-LO.  The header URI allows for each 
>>>>
>>>>intermediary to 
>>>>
>>>>
>>>>>ask for the dereference, but only those that have permission, 
>>>>>surviving the challenge(s), get the PIDF-LO.  This means not all 
>>>>>dereference attempts learn the UAC's location.  This cannot be 
>>>>>accomplished in a header.
>>>>
>>>>This would assume that the entity that created the location 
>>>>object has a
>>>> relationship with the final location recipient (and to 
>>>>share some credential with them).
>>>>
>>>>I think that this is a totally unrealistic assumption.
>>>>
>>>>
>>>>Ciao
>>>>Hannes
>>>>
>>>>
>>>>
>>>>>>Ciao
>>>>>>Hannes
>>>>>>
>>>>>>James M. Polk wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>>Right now, the ID lists options 1 and 2 as the only options, and 
>>>>>>>this keeps things simple.  There have been a couple of 
>>>>
>>>>comments here 
>>>>
>>>>
>>>>>>>in Montreal that adding a data URL isn't really wanted.
>>>>>>>One reason given is the inability to encrypt it (easily). 
>>>>
>>>>A message 
>>>>
>>>>
>>>>>>>body can be encypted using S/MIME, and a URI can have the server 
>>>>>>>posessing the PIDF-LO configured to challenge a 
>>>>
>>>>dereference request 
>>>>
>>>>
>>>>>>>- including responding with an encrypted PIDF-LO.  Both these 
>>>>>>>senarios have the PIDF-LO being able to be encrypted.  
>>
>>I'm not so 
>>
>>>>>>>sure encrypting the data URL is so straightforward, or 
>>
>>deployable.
>>
>>>>>>>Others, please inform me and the list if this is untrue.
>>>>>>>Cannot the ID proceed as is (in this respect), and once someone 
>>>>>>>wants to extend this doc (once it's an RFC), propose that 
>>>>
>>>>extention 
>>>>
>>>>
>>>>>>>as a separate ID?
>>>>>>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>To follow up on the discussion in the SIP group. I think 
>>>>
>>>>there are 
>>>>
>>>>
>>>>>>>>roughly three options for the content of Location:
>>>>>>>>
>>>>>>>>(1) Point to content in the body  --> can't be modified 
>>>>
>>>>and cannot 
>>>>
>>>>
>>>>>>>>be inserted by proxies (and may not be inspectable by proxies if
>>>>>>>>encrypted)
>>>>>>>>
>>>>>>>>(2) Point to external location --> can be modified, can 
>>>>
>>>>be added, 
>>>>
>>>>
>>>>>>>>but can inFrom sip-bounces@ietf.org Fri Jul 14 11:18:09 2006
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To: "Michael Hammer (mhammer)" <mhammer@cisco.com>
Subject: Re: [Sip] Location header
References: <072C5B76F7CEAB488172C6F64B30B5E301B36584@xmb-rtp-20b.amer.cisco.com>
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Hi Mike,

I agree that the current document does not cover all the requirements 
and does not go into a deep discussion of the security/privacy aspects. 
I am happy to discuss these.

Maybe you could restate your requirements again in more details.

Ciao
Hannes

Michael Hammer (mhammer) wrote:
> Hannes,
> 
> I went back and re-read those requirements.  My problem here is that the
> arguments for one approach versus another are not always covered by
> those requirements.  In other words, a requirement is being inferred
> from the general requirement "do privacy".  So, if someone argues that X
> is not good enough because it doesn't meet Y, and Y is not a
> requirement, then the argument is baseless.
> 
> Much of the debate surrounds assumptions about topology and trust
> relationships (or not) between various nodes.
> 
> Disagreement on assumptions/requirements -> Disagreement on solutions.
> 
> Mike
> 
> 
> 
>>-----Original Message-----
>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>>Sent: Friday, July 14, 2006 10:20 AM
>>To: Michael Hammer (mhammer)
>>Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
>>Subject: Re: [Sip] Location header
>>
>>Hi Mike,
>>
>>Michael Hammer (mhammer) wrote:
>>
>>>Hannes,
>>>
>>>If you are right, then you are saying that symmetric secret 
>>
>>keys are 
>>
>>>out of the question.  That leaves some sort of 
>>
>>public/private key type 
>>
>>>of operation.
>>
>>The problem is not only the authentication but the authorization.
>>
>>Virtually everyone could be location recipient.
>>
>>
>>>Contrary to what Jon suggested earlier, there are ways of 
>>
>>passing an 
>>
>>>encrypted object to multiple recipients, namely by using a 
>>
>>secret key 
>>
>>>to encrypt the LO, then adding the secret key enrypted by 
>>
>>the public 
>>
>>>key of the intended recipient as an object for each recipient.
>>
>>We haven't even come to the problem of multiple recipient 
>>with a single message.
>>
>>... and: S/MIME does not provide these capabilities.
>>
>>   This also has
>>
>>>the charactduce loops if changed in mid-stream and 
>>
>>incurs another 
>>
>>>>>>>>failure mode (and possible security risks if the reference is 
>>>>>>>>generally
>>>>>>>>accessible)
>>>>>>>>
>>>>>>>>(3) Point to data URL (RFC 2397) as in
>>>>>>>>
>>>>>>>>Location: data:application/pidf+xml;base64,<base-64 version of 
>>>>>>>>PIDF-LO)
>>>>>>>>
>>>>>>>>Disadvantage: about 30% larger than as body; can't be encrypted 
>>>>>>>>separately (except as XML-DSIG)
>>>>>>>>
>>>>>>>>Advantages: can be inserted/modified by proxies (similar 
>>>>
>>>>to (2)), 
>>>>
>>>>
>>>>>>>>but avoids difficulty of external reference
>>>>>>>>
>>>>>>>>I think these three methods all have different 
>>>>>>>>time-space-flexibility trade-offs and thus I suspect 
>>>>
>>>>they all serve a need.
>>>>
>>>>
>>>>>>>>Henning
>>>>>>>>
>>>>>>>>
>>>>>>>>_______________________________________________
>>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>>>sip-implementors@cs.columbia.edu for questions on 
>>>>
>>>>current sip Use 
>>>>
>>>>
>>>>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>>>
>>>>>>>
>>>>>>>_______________________________________________
>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>>sip-implementors@cs.columbia.edu for questions on 
>>
>>current sip Use 
>>
>>>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>
>>>>>
>>>>>
>>>>_______________________________________________
>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>sip-implementors@cs.columbia.edu for questions on current sip 
>>>>Use sipping@ietf.org for new developments on the application of sip
>>>>
>>>
>>>
>>>
> 
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From sip-bounces@ietf.org Fri Jul 14 11:18:09 2006
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From: Markus Hofmann <mhofmann@nero.com>
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To: =?ISO-8859-1?Q?Ant=F3nio_Manuel_Amaral?= <antonio-mn-amaral@ptinovacao.pt>
Subject: Re: [Sip] S-CSCF expected behavior
References: <75D349FBF7C131408F5061B7168072C30329DB49@INOAVREX06.ptin.corpPT.com>
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>>
>>additional 
>>
>>>recipients, can add new Pu(secret) objects for those new 
>>
>>recipients as 
>>
>>>needed.  That does assume a chain of trust among those 
>>
>>given access to 
>>
>>>the secret.
>>>
>>>Is that practical?  I think so, but others may have their opinion.
>>>
>>
>>
>>This is just an optimization of encrypting location objects 
>>to individual recipients. I am not sure we are even closen to 
>>optimizations.
>>
>>
>>
>>>BTW, can someone re-summarize what the requirements are?
>>
>>Requirements for what exactly? Most of the requirements are listed in 
>>the document provided by James & Brian.
>>
>>
>>>It must be modifiable -> must be a header not body
>>>It must provide confidentiality -> encrypted or selectively 
>>
>>handed out
>>
>>>(by ref)
>>>Downstream recipient unknown to UAC must have access -> ???
>>>...
>>
>>What is 'it'?
>>
>>
>>Ciao
>>Hannes
>>
>>
>>>Mike
>>>
>>>
>>>
>>>
>>>>-----Original Message-----
>>>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>>>>Sent: Friday, July 14, 2006 9:39 AM
>>>>To: James Polk (jmpolk)
>>>>Cc: sip@ietf.org; Henning Schulzrinne
>>>>Subject: Re: [Sip] Location header
>>>>
>>>>Hi James,
>>>>
>>>>James M. Polk wrote:
>>>>
>>>>
>>>>>At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
>>>>>
>>>>>
>>>>>
>>>>>>Hi James,
>>>>>>
>>>>>>please explain me the difference (from a security point of view) 
>>>>>>between
>>>>>>* adding a location object into the header that is not 
>>>>
>>>>encrypted, and
>>>>
>>>>
>>>>>>* adding a reference to a location object to the header 
>>>>
>>>>that can be 
>>>>
>>>>
>>>>>>resolved into a location object by everyone that knows the 
>>>>
>>>>reference?
>>>>
>>>>
>>>>>The server which has the by-referenced PIDF-LO can 
>>>>
>>>>challenge whoever 
>>>>
>>>>
>>>>>asks for the PIDF-LO.  The header URI allows for each 
>>>>
>>>>intermediary to 
>>>>
>>>>
>>>>>ask for the dereference, but only those that have permission, 
>>>>>surviving the challenge(s), get the PIDF-LO.  This means not all 
>>>>>dereference attempts learn the UAC's location.  This cannot be 
>>>>>accomplished in a header.
>>>>
>>>>This would assume that the entity that created the location 
>>>>object has a
>>>> relationship with the final location recipient (and to 
>>>>share some credential with them).
>>>>
>>>>I think that this is a totally unrealistic assumption.
>>>>
>>>>
>>>>Ciao
>>>>Hannes
>>>>
>>>>
>>>>
>>>>>>Ciao
>>>>>>Hannes
>>>>>>
>>>>>>James M. Polk wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>>Right now, the ID lists options 1 and 2 as the only options, and 
>>>>>>>this keeps things simple.  There have been a couple of 
>>>>
>>>>comments here 
>>>>
>>>>
>>>>>>>in Montreal that adding a data URL isn't really wanted.
>>>>>>>One reason given is the inability to encrypt it (easily). 
>>>>
>>>>A message 
>>>>
>>>>
>>>>>>>body can be encypted using S/MIME, and a URI can have the server 
>>>>>>>posessing the PIDF-LO configured to challenge a 
>>>>
>>>>dereference request 
>>>>
>>>>
>>>>>>>- including responding with an encrypted PIDF-LO.  Both these 
>>>>>>>senarios have the PIDF-LO being able to be encrypted.  
>>
>>I'm not so 
>>
>>>>>>>sure encrypting the data URL is so straightforward, or 
>>
>>deployable.
>>
>>>>>>>Others, please inform me and the list if this is untrue.
>>>>>>>Cannot the ID proceed as is (in this respect), and once someone 
>>>>>>>wants to extend this doc (once it's an RFC), propose that 
>>>>
>>>>extention 
>>>>
>>>>
>>>>>>>as a separate ID?
>>>>>>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>To follow up on the discussion in the SIP group. I think 
>>>>
>>>>there are 
>>>>
>>>>
>>>>>>>>roughly three options for the content of Location:
>>>>>>>>
>>>>>>>>(1) Point to content in the body  --> can't be modified 
>>>>
>>>>and cannot 
>>>>
>>>>
>>>>>>>>be inserted by proxies (and may not be inspectable by proxies if
>>>>>>>>encrypted)
>>>>>>>>
>>>>>>>>(2) Point to external location --> can be modified, can 
>>>>
>>>>be added, 
>>>>
>>>>
>>>>>>>>but can in69a74e02bbee44ab4f8eafdbcedd94a1
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Hi Ant=F3nio,

I can't answer your question. I think you should contact the 3G which is
responsible for IMS.

Regards

Markus

Ant=F3nio Manuel Amaral wrote:
> Dear All,
>=20
> I don't know if this is the correct mailing list to put my question and=
 if not my apologies for that.
>=20
> Consider the following IMS scenario
>=20
>=20
> (IMS core Network) -> S-CSCF -> AS
>=20
>=20
> If the connection between AS and S-CSCF fails what is the expected beha=
vior of S-CSCF? It must release all active calls sending BYE messages to =
caller and called clients in order to terminate the calls?
>=20
> Thanks in advance
>=20
> Regards
>=20
> AA=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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duce loops if changed in mid-stream and 
>>
>>incurs another 
>>
>>>>>>>>failure mode (and possible security risks if the reference is 
>>>>>>>>generally
>>>>>>>>accessible)
>>>>>>>>
>>>>>>>>(3) Point to data URL (RFC 2397) as in
>>>>>>>>
>>>>>>>>Location: data:application/pidf+xml;base64,<base-64 version of 
>>>>>>>>PIDF-LO)
>>>>>>>>
>>>>>>>>Disadvantage: about 30% larger than as body; can't be encrypted 
>>>>>>>>separately (except as XML-DSIG)
>>>>>>>>
>>>>>>>>Advantages: can be inserted/modified by proxies (similar 
>>>>
>>>>to (2)), 
>>>>
>>>>
>>>>>>>>but avoids difficulty of external reference
>>>>>>>>
>>>>>>>>I think these three methods all have different 
>>>>>>>>time-space-flexibility trade-offs and thus I suspect 
>>>>
>>>>they all serve a need.
>>>>
>>>>
>>>>>>>>Henning
>>>>>>>>
>>>>>>>>
>>>>>>>>_______________________________________________
>>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>>>sip-implementors@cs.columbia.edu for questions on 
>>>>
>>>>current sip Use 
>>>>
>>>>
>>>>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>>>
>>>>>>>
>>>>>>>_______________________________________________
>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>>sip-implementors@cs.columbia.edu for questions on 
>>
>>current sip Use 
>>
>>>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>
>>>>>
>>>>>
>>>>_______________________________________________
>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>sip-implementors@cs.columbia.edu for questions on current sip 
>>>>Use sipping@ietf.org for new developments on the application of sip
>>>>
>>>
>>>
>>>
> 
> 


_______________________________________________
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From sip-bounces@ietf.org Fri Jul 14 11:18:09 2006
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Hi Ant=F3nio,

I can't answer your question. I think you should contact the 3G which is
responsible for IMS.

Regards

Markus

Ant=F3nio Manuel Amaral wrote:
> Dear All,
>=20
> I don't know if this is the correct mailing list to put my question and=
 if not my apologies for that.
>=20
> Consider the following IMS scenario
>=20
>=20
> (IMS core Network) -> S-CSCF -> AS
>=20
>=20
> If the connection between AS and S-CSCF fails what is the expected beha=
vior of S-CSCF? It must release all active calls sending BYE messages to =
caller and called clients in order to terminate the calls?
>=20
> Thanks in advance
>=20
> Regards
>=20
> AA=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip





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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] Location header
Date: Fri, 14 Jul 2006 09:24:54 -0400
To: "James M. Polk" <jmpolk@cisco.com>
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For the purposes of routing, such challenges are unlikely to be  
feasible or even plausible. Are you going to give a special secret to  
all possible carriers, PSAPs, E-CSCFs in the world, for every single  
individual? Good luck...

On Jul 14, 2006, at 1:28 AM, James M. Polk wrote:

> At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
>> Hi James,
>>
>> please explain me the difference (from a security point of view)  
>> between
>> * adding a location object into the header that is not encrypted, and
>> * adding a reference to a location object to the header that can  
>> be resolved into a location object by everyone that knows the  
>> reference?
>
> The server which has the by-referenced PIDF-LO can challenge  
> whoever asks for the PIDF-LO.  The header URI allows for each  
> intermediary to ask for the dereference, but only those that have  
> permission, surviving the challenge(s), get the PIDF-LO.  This  
> means not all dereference attempts learn the UAC's location.  This  
> cannot be accomplished in a header.
>
>
>> Ciao
>> Hannes
>>
>> James M. Polk wrote:
>>> Right now, the ID lists options 1 and 2 as the only options, and  
>>> this keeps things simple.  There have been a couple of comments  
>>> here in Montreal that adding a data URL isn't really wanted.
>>> One reason given is the inability to encrypt it (easily).  A message
>>> body can be encypted using S/MIME, and a URI can have the server  
>>> posessing the PIDF-LO configured to challenge a dereference  
>>> request - including responding with an encrypted PIDF-LO.  Both  
>>> these senarios have the PIDF-LO being able to be encrypted.  I'm  
>>> not so sure encrypting the data URL is so straightforward, or  
>>> deployable.  Others, please inform me and the list if this is  
>>> untrue.
>>> Cannot the ID proceed as is (in this respect), and once someone  
>>> wants to extend this doc (once it's an RFC), propose that  
>>> extention as a separate ID?
>>> At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>>>
>>>> To follow up on the discussion in the SIP group. I think there are
>>>> roughly three options for the content of Location:
>>>>
>>>> (1) Point to content in the body  --> can't be modified and  
>>>> cannot be
>>>> inserted by proxies (and may not be inspectable by proxies if  
>>>> encrypted)
>>>>
>>>> (2) Point to external location --> can be modified, can be  
>>>> added, but
>>>> can induce loops if changed in mid-stream and incurs another  
>>>> failure
>>>> mode (and possible security risks if the reference is generally
>>>> accessible)
>>>>
>>>> (3) Point to data URL (RFC 2397) as in
>>>>
>>>> Location: data:application/pidf+xml;base64,<base-64 version of  
>>>> PIDF-LO)
>>>>
>>>> Disadvantage: about 30% larger than as body; can't be encrypted
>>>> separately (except as XML-DSIG)
>>>>
>>>> Advantages: can be inserted/modified by proxies (similar to  
>>>> (2)), but
>>>> avoids difficulty of external reference
>>>>
>>>> I think these three methods all have different time-space- 
>>>> flexibility
>>>> trade-offs and thus I suspect they all serve a need.
>>>>
>>>> Henning
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip


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Subject: RE: [Sip] Location header
Date: Fri, 14 Jul 2006 10:05:36 -0400
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Thread-Topic: [Sip] Location header
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>,
	"James Polk \(jmpolk\)" <jmpolk@cisco.com>
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Hannes,

If you are right, then you are saying that symmetric secret keys are out
of the question.  That leaves some sort of public/private key type of
operation.

Contrary to what Jon suggested earlier, there are ways of passing an
encrypted object to multiple recipients, namely by using a secret key to
encrypt the LO, then adding the secret key enrypted by the public key of
the intended recipient as an object for each recipient.  This also has
the characteristic that down-stream nodes upon learning of additional
recipients, can add new Pu(secret) objects for those new recipients as
needed.  That does assume a chain of trust among those given access to
the secret.

Is that practical?  I think so, but others may have their opinion.

BTW, can someone re-summarize what the requirements are?
It must be modifiable -> must be a header not body
It must provide confidentiality -> encrypted or selectively handed out
(by ref)
Downstream recipient unknown to UAC must have access -> ???
...

Mike


> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: Friday, July 14, 2006 9:39 AM
> To: James Polk (jmpolk)
> Cc: sip@ietf.org; Henning Schulzrinne
> Subject: Re: [Sip] Location header
>=20
> Hi James,
>=20
> James M. Polk wrote:
> > At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
> >=20
> >> Hi James,
> >>
> >> please explain me the difference (from a security point of view)=20
> >> between
> >> * adding a location object into the header that is not=20
> encrypted, and
> >> * adding a reference to a location object to the header=20
> that can be=20
> >> resolved into a location object by everyone that knows the=20
> reference?
> >=20
> >=20
> > The server which has the by-referenced PIDF-LO can=20
> challenge whoever=20
> > asks for the PIDF-LO.  The header URI allows for each=20
> intermediary to=20
> > ask for the dereference, but only those that have permission,=20
> > surviving the challenge(s), get the PIDF-LO.  This means not all=20
> > dereference attempts learn the UAC's location.  This cannot be=20
> > accomplished in a header.
> This would assume that the entity that created the location=20
> object has a
>   relationship with the final location recipient (and to=20
> share some credential with them).
>=20
> I think that this is a totally unrealistic assumption.
>=20
>=20
> Ciao
> Hannes
>=20
> >=20
> >=20
> >> Ciao
> >> Hannes
> >>
> >> James M. Polk wrote:
> >>
> >>> Right now, the ID lists options 1 and 2 as the only options, and=20
> >>> this keeps things simple.  There have been a couple of=20
> comments here=20
> >>> in Montreal that adding a data URL isn't really wanted.
> >>> One reason given is the inability to encrypt it (easily).=20
>  A message=20
> >>> body can be encypted using S/MIME, and a URI can have the server=20
> >>> posessing the PIDF-LO configured to challenge a=20
> dereference request=20
> >>> - including responding with an encrypted PIDF-LO.  Both these=20
> >>> senarios have the PIDF-LO being able to be encrypted.  I'm not so=20
> >>> sure encrypting the data URL is so straightforward, or deployable.
> >>> Others, please inform me and the list if this is untrue.
> >>> Cannot the ID proceed as is (in this respect), and once someone=20
> >>> wants to extend this doc (once it's an RFC), propose that=20
> extention=20
> >>> as a separate ID?
> >>> At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
> >>>
> >>>> To follow up on the discussion in the SIP group. I think=20
> there are=20
> >>>> roughly three options for the content of Location:
> >>>>
> >>>> (1) Point to content in the body  --> can't be modified=20
> and cannot=20
> >>>> be inserted by proxies (and may not be inspectable by proxies if
> >>>> encrypted)
> >>>>
> >>>> (2) Point to external location --> can be modified, can=20
> be added,=20
> >>>> but can induce loops if changed in mid-stream and incurs another=20
> >>>> failure mode (and possible security risks if the reference is=20
> >>>> generally
> >>>> accessible)
> >>>>
> >>>> (3) Point to data URL (RFC 2397) as in
> >>>>
> >>>> Location: data:application/pidf+xml;base64,<base-64 version of=20
> >>>> PIDF-LO)
> >>>>
> >>>> Disadvantage: about 30% larger than as body; can't be encrypted=20
> >>>> separately (except as XML-DSIG)
> >>>>
> >>>> Advantages: can be inserted/modified by proxies (similar=20
> to (2)),=20
> >>>> but avoids difficulty of external reference
> >>>>
> >>>> I think these three methods all have different=20
> >>>> time-space-flexibility trade-offs and thus I suspect=20
> they all serve a need.
> >>>>
> >>>> Henning
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sipping@ietf.org for new developments on the application of sip
> >>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
> >=20
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
From: Scott Lawrence <slawrence@pingtel.com>
To: Dean Willis <dean.willis@softarmor.com>
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On Thu, 2006-07-13 at 16:14 -0500, Dean Willis wrote:
> On Jul 13, 2006, at 9:53 AM, Scott Lawrence wrote:
> 
> >         When sending an error response that has a message/sipfrag  
> > body,
> >         the responding server MUST examine all the Via header  
> > values in
> >         the request.  If any Via header value specifies UDP, then the
> >         total size of the response body MUST NOT be greater than 200
> >         bytes and the server MUST follow the procedures in section  
> > X to
> >         prune the returned message/sipfrag body to fit within this
> >         limit.  If no request Via header value specifies UDP, then the
> >         server is free to either prune the returned message/sipfrag  
> > body
> >         as specified in section X or to return the full set of request
> >         headers.
> 
> Since UDP is the default, you may need this to say "Does not specify  
> TCP, TLS, or SCTP".

I don't believe that we're in danger of the SIP standard adding any
other transports that share the unreliable and network-hostile
characteristics of UDP, but it seems likely that there will in the
future be other reliable transports that can cope with large responses,
so I'd prefer to express this in a way that doesn't need to be updated
when that happens.

Would it be more clear to replace 'specifies UDP' with 'indicates that
UDP is used' ?

> Also, this will still explode in environments where evil topology- 
> hiding functions (common in SBCs) have suppressed the Via header  
> fields. Some people would have me believe that this occurs in most if  
> not all IMS deployments, giving us a low probability that large  
> responses will ever actually work.

People who deploy such things and don't keep them up to date when the
protocol changes deserve what happens to them.  I don't think that we
need to maintain backward compatibility or otherwise protect the
clueless from implementations that have chosen (for, I am sure, good
and/or revenue-generating reasons) to violate MUST-level rules.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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Hi Mike,

Michael Hammer (mhammer) wrote:
> I wouldn't call them my requirements per se, I was just trying to read
> between the lines of the discussion and extract why some folks were
> arguing for or against based on a statement or requirement/assumption.
> 
> The inclusion v. reference and header v. body arguments aside, it seemed
> as though there were two dimensions here:
> 
> Set of nodes with which the Client has trust relationship versus not

I don't like the term trust since it is too fuzzy.
Furthermore, we have multiple entities playing a role here: Target, SIP 
Proxy, Location Recipient

> Set of nodes that have permitted "need to know" location versus not

I agree with this one.

> 
> 1) Relationship + need to know
> 2) No Relation + need to know
> 3) Relationship + no need to know
> 4) No Relation + need to know
> 
> I thought each of the above needed to be addressed.  But, my suggestion
> for 4) based on overlapping trust groups seemed to be deemed jumping
> ahead of the immediate issues.
I will write a longer mail that describes the scenario I can see.

Ciao
Hannes

>  
> Mike
> 
> 
> 
>>-----Original Message-----
>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>>Sent: Friday, July 14, 2006 10:50 AM
>>To: Michael Hammer (mhammer)
>>Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
>>Subject: Re: [Sip] Location header
>>
>>Hi Mike,
>>
>>I agree that the current document does not cover all the 
>>requirements and does not go into a deep discussion of the 
>>security/privacy aspects. 
>>I am happy to discuss these.
>>
>>Maybe you could restate your requirements again in more details.
>>
>>Ciao
>>Hannes
>>
>>Michael Hammer (mhammer) wrote:
>>
>>>Hannes,
>>>
>>>I went back and re-read those requirements.  My problem 
>>
>>here is that 
>>
>>>the arguments for one approach versus another are not 
>>
>>always covered 
>>
>>>by those requirements.  In other words, a requirement is being 
>>>inferred from the general requirement "do privacy".  So, if someone 
>>>argues that X is not good enough because it doesn't meet Y, 
>>
>>and Y is 
>>
>>>not a requirement, then the argument is baseless.
>>>
>>>Much of the debate surrounds assumptions about topology and trust 
>>>relationships (or not) between various nodes.
>>>
>>>Disagreement on assumptions/requirements -> Disagreement on 
>>
>>solutions.
>>
>>>Mike
>>>
>>>
>>>
>>>
>>>>-----Original Message-----
>>>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
>>>>Sent: Friday, July 14, 2006 10:20 AM
>>>>To: Michael Hammer (mhammer)
>>>>Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
>>>>Subject: Re: [Sip] Location header
>>>>
>>>>Hi Mike,
>>>>
>>>>Michael Hammer (mhammer) wrote:
>>>>
>>>>
>>>>>Hannes,
>>>>>
>>>>>If you are right, then you are saying that symmetric secret
>>>>
>>>>keys are
>>>>
>>>>
>>>>>out of the question.  That leaves some sort of
>>>>
>>>>public/private key type
>>>>
>>>>
>>>>>of operation.
>>>>
>>>>The problem is not only the authentication but the authorization.
>>>>
>>>>Virtually everyone could be location recipient.
>>>>
>>>>
>>>>
>>>>>Contrary to what Jon suggested earlier, there are ways of
>>>>
>>>>passing an
>>>>
>>>>
>>>>>encrypted object to multiple recipients, namely by using a
>>>>
>>>>secret key
>>>>
>>>>
>>>>>to encrypt the LO, then adding the secret key enrypted by
>>>>
>>>>the public
>>>>
>>>>
>>>>>key of the intended recipient as an object for each recipient.
>>>>
>>>>We haven't even come to the problem of multiple recipient with a 
>>>>single message.
>>>>
>>>>... and: S/MIME does not provide these capabilities.
>>>>
>>>>  This also has
>>>>
>>>>
>>>>>the characteristic that down-stream nodes upon learning of
>>>>
>>>>additional
>>>>
>>>>
>>>>>recipients, can add new Pu(secret) objects for those new
>>>>
>>>>recipients as
>>>>
>>>>
>>>>>needed.  That does assume a chain of trust among those
>>>>
>>>>given access to
>>>>
>>>>
>>>>>the secret.
>>>>>
>>>>>Is that practical?  I think so, but others may have their opinion.
>>>>>
>>>>
>>>>
>>>>This is just an optimization of encrypting location objects to 
>>>>individual recipients. I am not sure we are even closen to 
>>>>optimizations.
>>>>
>>>>
>>>>
>>>>
>>>>>BTW, can someone re-summarize what the requirements are?
>>>>
>>>>Requirements for what exactly? Most of the requirements are 
>>
>>listed in 
>>
>>>>the document provided by James & Brian.
>>>>
>>>>
>>>>
>>>>>It must be modifiable -> must be a header not body It must provide 
>>>>>confidentiality -> encrypted or selectively
>>>>
>>>>handed out
>>>>
>>>>
>>>>>(by ref)
>>>>>Downstream recipient unknown to UAC must have access -> ???
>>>>>...
>>>>
>>>>What is 'it'?
>>>>
>>>>
>>>>Ciao
>>>>Hannes
>>>>
>>>>
>>>>
>>>>>Mike
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>-----Original Message-----
>>>>>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
>>>>>>Sent: Friday, July 14, 2006 9:39 AM
>>>>>>To: James Polk (jmpolk)
>>>>>>Cc: sip@ietf.org; Henning Schulzrinne
>>>>>>Subject: Re: [Sip] Location header
>>>>>>
>>>>>>Hi James,
>>>>>>
>>>>>>James M. Polk wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>>At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>Hi James,
>>>>>>>>
>>>>>>>>please explain me the difference (from a security point 
>>
>>of view) 
>>
>>>>>>>>between
>>>>>>>>* adding a location object into the header that is not
>>>>>>
>>>>>>encrypted, and
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>* adding a reference to a location object to the header
>>>>>>
>>>>>>that can be
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>resolved into a location object by everyone that knows the
>>>>>>
>>>>>>reference?
>>>>>>
>>>>>>
>>>>>>
>>>>>>>The server which has the by-referenced PIDF-LO can
>>>>>>
>>>>>>challenge whoever
>>>>>>
>>>>>>
>>>>>>
>>>>>>>asks for the PIDF-LO.  The header URI allows for each
>>>>>>
>>>>>>intermediary to
>>>>>>
>>>>>>
>>>>>>
>>>>>>>ask for the dereference, but only those that have permission, 
>>>>>>>surviving the challenge(s), get the PIDF-LO.  This means not all 
>>>>>>>dereference attempts learn the UAC's location.  This cannot be 
>>>>>>>accomplished in a header.
>>>>>>
>>>>>>This would assume that the entity that created the 
>>
>>location object 
>>
>>>>>>has a  relationship with the final location recipient 
>>
>>(and to share 
>>
>>>>>>some credential with them).
>>>>>>
>>>>>>I think that this is a totally unrealistic assumption.
>>>>>>
>>>>>>
>>>>>>Ciao
>>>>>>Hannes
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>Ciao
>>>>>>>>Hannes
>>>>>>>>
>>>>>>>>James M. Polk wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>Right now, the ID lists options 1 and 2 as the only 
>>
>>options, and 
>>
>>>>>>>>>this keeps things simple.  There have been a couple of
>>>>>>
>>>>>>comments here
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>in Montreal that adding a data URL isn't really wanted.
>>>>>>>>>One reason given is the inability to encrypt it (easily). 
>>>>>>
>>>>>>A message
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>body can be encypted using S/MIME, and a URI can have 
>>
>>the server 
>>
>>>>>>>>>posessing the PIDF-LO configured to challenge a
>>>>>>
>>>>>>dereference request
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>- including responding with an encrypted PIDF-LO.  Both these 
>>>>>>>>>senarios have the PIDF-LO being able to be encrypted.
>>>>
>>>>I'm not so
>>>>
>>>>
>>>>>>>>>sure encrypting the data URL is so straightforward, or
>>>>
>>>>deployable.
>>>>
>>>>
>>>>>>>>>Others, please inform me and the list if this is untrue.
>>>>>>>>>Cannot the ID proceed as is (in this respect), and 
>>
>>once someone 
>>
>>>>>>>>>wants to extend this doc (once it's an RFC), propose that
>>>>>>
>>>>>>extention
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>as a separate ID?
>>>>>>>>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>To follow up on the discussion in the SIP group. I think
>>>>>>
>>>>>>there are
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>>roughly three options for the content of Location:
>>>>>>>>>>
>>>>>>>>>>(1) Point to content in the body  --> can't be modified
>>>>>>
>>>>>>and cannot
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>>be inserted by proxies (and may not be inspectable by 
>>
>>proxies if
>>
>>>>>>>>>>encrypted)
>>>>>>>>>>
>>>>>>>>>>(2) Point to external location --> can be modified, can
>>>>>>
>>>>>>be added,
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>>but can induce loops if changed in mid-stream and
>>>>
>>>>incurs another
>>>>
>>>>
>>>>>>>>>>failure mode (and possible security risks if the reference is 
>>>>>>>>>>generally
>>>>>>>>>>accessible)
>>>>>>>>>>
>>>>>>>>>>(3) Point to data URL (RFC 2397) as in
>>>>>>>>>>
>>>>>>>>>>Location: data:application/pidf+xml;base64,<base-64 version of
>>>>>>>>>>PIDF-LO)
>>>>>>>>>>
>>>>>>>>>>Disadvantage: about 30% larger than as body; can't be 
>>
>>encrypted 
>>
>>>>>>>>>>separately (except as XML-DSIG)
>>>>>>>>>>
>>>>>>>>>>Advantages: can be inserted/modified by proxies (similar
>>>>>>
>>>>>>to (2)),
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>>but avoids difficulty of external reference
>>>>>>>>>>
>>>>>>>>>>I think these three methods all have different 
>>>>>>>>>>time-space-flexibility trade-offs and thus I suspect
>>>>>>
>>>>>>they all serve a need.
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>>Henning
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>_______________________________________________
>>>>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>>>>>sip-implementors@cs.columbia.edu for questions on
>>>>>>
>>>>>>current sip Use
>>>>>>
>>>>>>
>>>>>>
>>>>>>>>>>sipping@ietf.org for new developments on the 
>>
>>application of sip
>>
>>>>>>>>>
>>>>>>>>>_______________________________________________
>>>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>>>>sip-implementors@cs.columbia.edu for questions on
>>>>
>>>>current sip Use
>>>>
>>>>
>>>>>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>_______________________________________________
>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>>sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>>
>>>>>
>>>>>
>>>>>
>>>
> 
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Yeah, trust is fuzzy. =20

But it all boils down to whether two parties have a means to pass
information securely to each other, while excluding others. =20

The trust part is that party A gives it to party B "knowing" that B can
see it. =20
A has to "trust" the B will do the right thing with the information.

Mike


> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: Friday, July 14, 2006 2:03 PM
> To: Michael Hammer (mhammer)
> Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
> Subject: Re: [Sip] Location header
>=20
> Hi Mike,
>=20
> Michael Hammer (mhammer) wrote:
> > I wouldn't call them my requirements per se, I was just=20
> trying to read=20
> > between the lines of the discussion and extract why some folks were=20
> > arguing for or against based on a statement or=20
> requirement/assumption.
> >=20
> > The inclusion v. reference and header v. body arguments aside, it=20
> > seemed as though there were two dimensions here:
> >=20
> > Set of nodes with which the Client has trust relationship versus not
>=20
> I don't like the term trust since it is too fuzzy.
> Furthermore, we have multiple entities playing a role here:=20
> Target, SIP Proxy, Location Recipient
>=20
> > Set of nodes that have permitted "need to know" location versus not
>=20
> I agree with this one.
>=20
> >=20
> > 1) Relationship + need to know
> > 2) No Relation + need to know
> > 3) Relationship + no need to know
> > 4) No Relation + need to know
> >=20
> > I thought each of the above needed to be addressed.  But, my=20
> > suggestion for 4) based on overlapping trust groups seemed to be=20
> > deemed jumping ahead of the immediate issues.
> I will write a longer mail that describes the scenario I can see.
>=20
> Ciao
> Hannes
>=20
> > =20
> > Mike
> >=20
> >=20
> >=20
> >>-----Original Message-----
> >>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> >>Sent: Friday, July 14, 2006 10:50 AM
> >>To: Michael Hammer (mhammer)
> >>Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
> >>Subject: Re: [Sip] Location header
> >>
> >>Hi Mike,
> >>
> >>I agree that the current document does not cover all the=20
> requirements=20
> >>and does not go into a deep discussion of the security/privacy=20
> >>aspects.
> >>I am happy to discuss these.
> >>
> >>Maybe you could restate your requirements again in more details.
> >>
> >>Ciao
> >>Hannes
> >>
> >>Michael Hammer (mhammer) wrote:
> >>
> >>>Hannes,
> >>>
> >>>I went back and re-read those requirements.  My problem
> >>
> >>here is that
> >>
> >>>the arguments for one approach versus another are not
> >>
> >>always covered
> >>
> >>>by those requirements.  In other words, a requirement is being=20
> >>>inferred from the general requirement "do privacy".  So,=20
> if someone=20
> >>>argues that X is not good enough because it doesn't meet Y,
> >>
> >>and Y is
> >>
> >>>not a requirement, then the argument is baseless.
> >>>
> >>>Much of the debate surrounds assumptions about topology and trust=20
> >>>relationships (or not) between various nodes.
> >>>
> >>>Disagreement on assumptions/requirements -> Disagreement on
> >>
> >>solutions.
> >>
> >>>Mike
> >>>
> >>>
> >>>
> >>>
> >>>>-----Original Message-----
> >>>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> >>>>Sent: Friday, July 14, 2006 10:20 AM
> >>>>To: Michael Hammer (mhammer)
> >>>>Cc: James Polk (jmpolk); sip@ietf.org; Henning Schulzrinne
> >>>>Subject: Re: [Sip] Location header
> >>>>
> >>>>Hi Mike,
> >>>>
> >>>>Michael Hammer (mhammer) wrote:
> >>>>
> >>>>
> >>>>>Hannes,
> >>>>>
> >>>>>If you are right, then you are saying that symmetric secret
> >>>>
> >>>>keys are
> >>>>
> >>>>
> >>>>>out of the question.  That leaves some sort of
> >>>>
> >>>>public/private key type
> >>>>
> >>>>
> >>>>>of operation.
> >>>>
> >>>>The problem is not only the authentication but the authorization.
> >>>>
> >>>>Virtually everyone could be location recipient.
> >>>>
> >>>>
> >>>>
> >>>>>Contrary to what Jon suggested earlier, there are ways of
> >>>>
> >>>>passing an
> >>>>
> >>>>
> >>>>>encrypted object to multiple recipients, namely by using a
> >>>>
> >>>>secret key
> >>>>
> >>>>
> >>>>>to encrypt the LO, then adding the secret key enrypted by
> >>>>
> >>>>the public
> >>>>
> >>>>
> >>>>>key of the intended recipient as an object for each recipient.
> >>>>
> >>>>We haven't even come to the problem of multiple recipient with a=20
> >>>>single message.
> >>>>
> >>>>... and: S/MIME does not provide these capabilities.
> >>>>
> >>>>  This also has
> >>>>
> >>>>
> >>>>>the characteristic that down-stream nodes upon learning of
> >>>>
> >>>>additional
> >>>>
> >>>>
> >>>>>recipients, can add new Pu(secret) objects for those new
> >>>>
> >>>>recipients as
> >>>>
> >>>>
> >>>>>needed.  That does assume a chain of trust among those
> >>>>
> >>>>given access to
> >>>>
> >>>>
> >>>>>the secret.
> >>>>>
> >>>>>Is that practical?  I think so, but others may have=20
> their opinion.
> >>>>>
> >>>>
> >>>>
> >>>>This is just an optimization of encrypting location objects to=20
> >>>>individual recipients. I am not sure we are even closen to=20
> >>>>optimizations.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>>BTW, can someone re-summarize what the requirements are?
> >>>>
> >>>>Requirements for what exactly? Most of the requirements are
> >>
> >>listed in
> >>
> >>>>the document provided by James & Brian.
> >>>>
> >>>>
> >>>>
> >>>>>It must be modifiable -> must be a header not body It=20
> must provide=20
> >>>>>confidentiality -> encrypted or selectively
> >>>>
> >>>>handed out
> >>>>
> >>>>
> >>>>>(by ref)
> >>>>>Downstream recipient unknown to UAC must have access -> ???
> >>>>>...
> >>>>
> >>>>What is 'it'?
> >>>>
> >>>>
> >>>>Ciao
> >>>>Hannes
> >>>>
> >>>>
> >>>>
> >>>>>Mike
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>>-----Original Message-----
> >>>>>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> >>>>>>Sent: Friday, July 14, 2006 9:39 AM
> >>>>>>To: James Polk (jmpolk)
> >>>>>>Cc: sip@ietf.org; Henning Schulzrinne
> >>>>>>Subject: Re: [Sip] Location header
> >>>>>>
> >>>>>>Hi James,
> >>>>>>
> >>>>>>James M. Polk wrote:
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>At 07:25 PM 7/13/2006 -0400, Hannes Tschofenig wrote:
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>>Hi James,
> >>>>>>>>
> >>>>>>>>please explain me the difference (from a security point
> >>
> >>of view)
> >>
> >>>>>>>>between
> >>>>>>>>* adding a location object into the header that is not
> >>>>>>
> >>>>>>encrypted, and
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>* adding a reference to a location object to the header
> >>>>>>
> >>>>>>that can be
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>resolved into a location object by everyone that knows the
> >>>>>>
> >>>>>>reference?
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>The server which has the by-referenced PIDF-LO can
> >>>>>>
> >>>>>>challenge whoever
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>asks for the PIDF-LO.  The header URI allows for each
> >>>>>>
> >>>>>>intermediary to
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>ask for the dereference, but only those that have permission,=20
> >>>>>>>surviving the challenge(s), get the PIDF-LO.  This=20
> means not all=20
> >>>>>>>dereference attempts learn the UAC's location.  This cannot be=20
> >>>>>>>accomplished in a header.
> >>>>>>
> >>>>>>This would assume that the entity that created the
> >>
> >>location object
> >>
> >>>>>>has a  relationship with the final location recipient
> >>
> >>(and to share
> >>
> >>>>>>some credential with them).
> >>>>>>
> >>>>>>I think that this is a totally unrealistic assumption.
> >>>>>>
> >>>>>>
> >>>>>>Ciao
> >>>>>>Hannes
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>Ciao
> >>>>>>>>Hannes
> >>>>>>>>
> >>>>>>>>James M. Polk wrote:
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>>Right now, the ID lists options 1 and 2 as the only
> >>
> >>options, and
> >>
> >>>>>>>>>this keeps things simple.  There have been a couple of
> >>>>>>
> >>>>>>comments here
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>in Montreal that adding a data URL isn't really wanted.
> >>>>>>>>>One reason given is the inability to encrypt it (easily).=20
> >>>>>>
> >>>>>>A message
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>body can be encypted using S/MIME, and a URI can have
> >>
> >>the server
> >>
> >>>>>>>>>posessing the PIDF-LO configured to challenge a
> >>>>>>
> >>>>>>dereference request
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>- including responding with an encrypted PIDF-LO. =20
> Both these=20
> >>>>>>>>>senarios have the PIDF-LO being able to be encrypted.
> >>>>
> >>>>I'm not so
> >>>>
> >>>>
> >>>>>>>>>sure encrypting the data URL is so straightforward, or
> >>>>
> >>>>deployable.
> >>>>
> >>>>
> >>>>>>>>>Others, please inform me and the list if this is untrue.
> >>>>>>>>>Cannot the ID proceed as is (in this respect), and
> >>
> >>once someone
> >>
> >>>>>>>>>wants to extend this doc (once it's an RFC), propose that
> >>>>>>
> >>>>>>extention
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>as a separate ID?
> >>>>>>>>>At 07:19 PM 7/12/2006 -0400, Henning Schulzrinne wrote:
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>>To follow up on the discussion in the SIP group. I think
> >>>>>>
> >>>>>>there are
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>>roughly three options for the content of Location:
> >>>>>>>>>>
> >>>>>>>>>>(1) Point to content in the body  --> can't be modified
> >>>>>>
> >>>>>>and cannot
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>>be inserted by proxies (and may not be inspectable by
> >>
> >>proxies if
> >>
> >>>>>>>>>>encrypted)
> >>>>>>>>>>
> >>>>>>>>>>(2) Point to external location --> can be modified, can
> >>>>>>
> >>>>>>be added,
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>>but can induce loops if changed in mid-stream and
> >>>>
> >>>>incurs another
> >>>>
> >>>>
> >>>>>>>>>>failure mode (and possible security risks if the=20
> reference is=20
> >>>>>>>>>>generally
> >>>>>>>>>>accessible)
> >>>>>>>>>>
> >>>>>>>>>>(3) Point to data URL (RFC 2397) as in
> >>>>>>>>>>
> >>>>>>>>>>Location: data:application/pidf+xml;base64,<base-64=20
> version of
> >>>>>>>>>>PIDF-LO)
> >>>>>>>>>>
> >>>>>>>>>>Disadvantage: about 30% larger than as body; can't be
> >>
> >>encrypted
> >>
> >>>>>>>>>>separately (except as XML-DSIG)
> >>>>>>>>>>
> >>>>>>>>>>Advantages: can be inserted/modified by proxies (similar
> >>>>>>
> >>>>>>to (2)),
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>>but avoids difficulty of external reference
> >>>>>>>>>>
> >>>>>>>>>>I think these three methods all have different=20
> >>>>>>>>>>time-space-flexibility trade-offs and thus I suspect
> >>>>>>
> >>>>>>they all serve a need.
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>>Henning
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>_______________________________________________
> >>>>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>>>>>>This list is for NEW development of the core SIP=20
> Protocol Use=20
> >>>>>>>>>>sip-implementors@cs.columbia.edu for questions on
> >>>>>>
> >>>>>>current sip Use
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>>>>sipping@ietf.org for new developments on the
> >>
> >>application of sip
> >>
> >>>>>>>>>
> >>>>>>>>>_______________________________________________
> >>>>>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>>>>>This list is for NEW development of the core SIP=20
> Protocol Use=20
> >>>>>>>>>sip-implementors@cs.columbia.edu for questions on
> >>>>
> >>>>current sip Use
> >>>>
> >>>>
> >>>>>>>>>sipping@ietf.org for new developments on the=20
> application of sip
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>_______________________________________________
> >>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>>>This list is for NEW development of the core SIP Protocol Use=20
> >>>>>>sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>>>>sipping@ietf.org for new developments on the application of sip
> >>>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>
> >=20
> >=20
>=20

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> -----Original Message-----
> From: Scott Lawrence [mailto:slawrence@pingtel.com]
> 
> On Thu, 2006-07-13 at 16:14 -0500, Dean Willis wrote:
> > Also, this will still explode in environments where evil topology-
> > hiding functions (common in SBCs) have suppressed the Via header
> > fields. Some people would have me believe that this occurs in most if
> > not all IMS deployments, giving us a low probability that large
> > responses will ever actually work.
> 
> People who deploy such things and don't keep them up to date when the
> protocol changes deserve what happens to them.  I don't think that we
> need to maintain backward compatibility or otherwise protect the
> clueless from implementations that have chosen (for, I am sure, good
> and/or revenue-generating reasons) to violate MUST-level rules.

Actually, I think his point was that such evil practices will impact _your_
non-evil providers/UAs.  And I take back what I said about it not being a
problem, because I see Dean's point now and can think of how it can be a
problem.  The problem is if you're a good guy calling _through_ an evil
provider (unbeknownst to you) transiting to other good providers.

If the loop is in the other good provider, and they implement this draft,
but the evil one doesn't strip the attachment, then the extra-large response
may not get back to you because of UDP issues.  So you'll not get the final
response, and have a bad calling experience, and create issues for the good
provider as well.  That's not good, for the good guys.  While I know at
least some Via-hiding boxes can strip the attachment out, that doesn't mean
all of them will/can.  

So perhaps you should size-limit all responses.  The odds of some hop or
other not being UDP isn't that good anyway today, and not many will want to
implement a draft which negatively impacts them for even a subset of cases.
Though I would like TCP-usage to get the benefit of TCP to promote its use
in the future, if you want rapid ubiquitous adoption of this draft mechanism
then you have to co-exist with such use cases, IMHO.  

-hadriel 


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inline.

Henning Schulzrinne wrote:


> I suspect there are multiple deployment scenarios, so it is probably  
> impossible to say that it is (or not), in general. However, we might  
> take two location-based routing scenarios that we have discussed:
> 
> (1) emergency calling
> 
> (2) calling for towing service
> 
> For either, I'm having a really hard time picturing how the sender  (or 
> the originating VSP) can know the credentials or even identity of  the 
> proxies that are on the signaling path. My VSP is located in  Germany; 
> it is unlikely to have any relationship with the Montreal  fire 
> department or, more importantly, the routing model of the  emergency 
> calling network in Canada. 

True. Whats interesting is that, as Jon and others have said, 
location-by-reference allows the owner of the location to apply many 
different types of authorization policies.

One type of authorization policy could simply be 'anyone on the path of 
this request is allowed to access the data for the duration of the 
dialog'. To provide this, as long as each hop was encrypted ala sips, if 
  a proxy inserted an HTTP URI with a crypto-random URI, the proxy would 
grant access to the location information as long as the dialog is still 
active.

Of course, you can always use traditional white lists if you happen to 
be in an environment where there is a pre-arranged relationship between 
parties. Location by reference allows both parties.

I do think, though, that Henning raises a good point, which is that, if 
you do specify location by reference, you need to be very crisp on how 
long the entity is expected to cache the data, and whether it can change 
it.

Thanks,
Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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   From: Scott Lawrence <slawrence@pingtel.com>

   On Thu, 2006-07-13 at 17:47 +0200, Jeroen van Bemmel wrote:
   > gzip on the body might help too

   If I'm not mistaken, it's not trivial to figure out how much any given
   text will compress with gzip unless you actually try each possibility. 

   Also, it doesn't do all that well on short inputs.

I just googled up a few web pages and e-mail archives containing a
total of 56 of SIP messages, extracted them, and ran gzip on them.  It
turns out that gzip compresses them poorly -- the median compression
is 70% (compressed size / original size); 67% is at about the 25th
percentile.  My guess is that many of the longer fields are
authentication, call-ID, and branch-parameter fields, all of which are
pseudo-random.

Dale

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   From: Scott Lawrence <slawrence@pingtel.com>

   > Another option is to return, in the 4xx response, an http URI pointing 
   > to a place where the server has stored the request. The requester can 
   > then go and pull the SIP request.
   > 
   > Might be an issue to interdomain cases, but not sure.

   I'm fairly sure that it would often be a problem.  

If we're going to this level of complexity, the right mechanism would
likely be a SUBSCRIBE that gets a return NOTIFY containing the failed
message -- using SIP to retrieve the message means that the
interdomain issues are likely to be solved already.

But I think that raises the bar for implementation too high -- people
are unlikely to invest in installing that amount of machinery until
the utility of the meachanism is proven.  Let's implement the
straightforward response body now.

Later, if we decide a "remote retrival" mechanism would be useful,
define how to put the branch-parameter from the top Via of the failing
request in a SUBSCRIBE to retrieve the whole message...

Dale

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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
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   From: "Samir Srivastava" <samirsr@nortel.com>

   If it is looping/spiraling, then it will be having good compression
   ratio for 483. 

No, it won't, because most of the bytes in Via headers are in the
branch parameters, which are all different, even when the message
loops.

Fortunately, a rather good job of pruning can be done by a simple
algorithm:  Sort all the headers by a weighting that says how valuable
they are for diagnostics.  The sort needs to be stable, in that all
headers of the same type have their order preserved.  Then retain an
initial subset of the headers without going over the desired body
length.

The ordering of the headers will be something like this:

Most valuable:	Via

		Accept-*

		Content-*

Least valuable:	To
		From
		CSeq
		Call-ID
		Max-Forwards
		Contact
		Allow

The last group of headers can always be omitted, even if you have no
length restriction, because the UAC can reconstruct them, either (1)
it is identical to the same header in the response, (2) it is
generated with the same value in every request by the UAC, or (3) its
value is constant in this situation (e.g., Max-Forwards is 0).

Dale

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   > > Also, this will still explode in environments where evil topology-
   > > hiding functions (common in SBCs) have suppressed the Via header

A representative of a company which makes evil devices mentioned to me
that the Evil Device (TM) will most likely entirely consume such a 483
response (after logging it for internal use), but then generate a
fresh 483 as if it came from the Evil Device at the point of entry to
the Walled Garden (TM).

This actually handles the separation of concerns sensibly -- If the
routing error is inside of the Walled Garden, the Evil Device's log
will capture the information that the Walled Garden's management will
need to fix the problem.  If due to some obscurity, the error happened
before the request entered into the Walled Garden and generated a
bogus request to the Walled Garden domain which caused the Walled
Garden to fail, then a 483 from the Evil Device will show everything
needed to diagnose the problem (including the request-URI at the point
of entry to the Walled Garden).

The messy case is when the loop goes through the Walled Garden but is
not entirely contained by it, though correlating the logs of all the
Even Devices should reveal the problem.

Dale

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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Sat, 15 Jul 2006 00:23:45 -0500
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On Jul 14, 2006, at 8:48 AM, Scott Lawrence wrote:

>
> I don't believe that we're in danger of the SIP standard adding any
> other transports that share the unreliable and network-hostile
> characteristics of UDP, but it seems likely that there will in the
> future be other reliable transports that can cope with large  
> responses,
> so I'd prefer to express this in a way that doesn't need to be updated
> when that happens.
>

I'll buy that.

> Would it be more clear to replace 'specifies UDP' with 'indicates that
> UDP is used' ?
>

There's probably some variation of language that would work here  
without saying TCP, etc.

--
Dean




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Subject: Re: [Sip] Location header
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> One type of authorization policy could simply be 'anyone on the  
> path of this request is allowed to access the data for the duration  
> of the dialog'. To provide this, as long as each hop was encrypted  
> ala sips, if  a proxy inserted an HTTP URI with a crypto-random  
> URI, the proxy would grant access to the location information as  
> long as the dialog is still active.

That's essentially the same security model offered by the location-by- 
value approach, as far as I can tell.

I agree that location-by-reference offers other security models; I  
never claimed that this is an either-or model; we can easily support  
both without changes in the header definition. After all, they are  
all URLs.

The trade-off is that for the simple on-path-can-read, the by- 
reference approach incurs significant overhead, as the location  
provider has to be able to maintain an HTTP server that is accessible  
by any potential destination of the call. For the case of end-system  
inserted location, this now means another configuration option since  
the UA or proxy has to have write access to an HTTP server, which  
isn't even standardized right now. (HTTP PUT? XCAP? WebDAV? ftp? some  
other special protocol?) For proxy-inserted, that server has to be  
outside a NAT and globally reachable.

In other words, for the common case you stipulate, by-reference only  
adds complexity and failure modes, with no increase in security or  
efficiency. (Actually, by reference is likely to be less efficient  
for the outbound-proxy inserts case, as multiple downstream proxies  
all have to open (TLS) connections to the server holding the  
location, each at several hundred or thousand bytes of traffic to get  
a 60-byte location object.)


>
> Of course, you can always use traditional white lists if you happen  
> to be in an environment where there is a pre-arranged relationship  
> between parties. Location by reference allows both parties.

I'm not disagreeing and don't think I've ever said that location-by- 
reference is not useful in some circumstances. There's just no point  
paying the complexity and reliability cost when there is no  
functional difference.

I think the basic design principle holds: simple things should be  
simple to do.

Henning


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Subject: RE: [Sip] S-CSCF expected behavior
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Antonio

I suggest you subscribe to the 3GPP CT1 email list and ask this question =
there.

http://list.3gpp.org/scripts/wa.exe?SUBED1=3D3gpp_tsg_ct_wg1&A=3D1

Andrew

-----Original Message-----
From: Ant=F3nio Manuel Amaral [mailto:antonio-mn-amaral@ptinovacao.pt]=20
Sent: Friday, July 14, 2006 6:17 AM
To: Sip@ietf.org
Subject: [Sip] S-CSCF expected behavior


Dear All,

I don't know if this is the correct mailing list to put my question and =
if not my apologies for that.

Consider the following IMS scenario


(IMS core Network) -> S-CSCF -> AS


If the connection between AS and S-CSCF fails what is the expected =
behavior of S-CSCF? It must release all active calls sending BYE =
messages to caller and called clients in order to terminate the calls?

Thanks in advance

Regards

AA=20



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To: <sip@ietf.org>,
	<Dale.Worley@comcast.net>
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	<200607150329.k6F3Tkul032204@dragon.ariadne.com>
Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Sat, 15 Jul 2006 17:56:22 +0200
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Dale,

see inline

Regards,

jeroen

>   If it is looping/spiraling, then it will be having good compression
>   ratio for 483.
>
> No, it won't, because most of the bytes in Via headers are in the
> branch parameters, which are all different, even when the message
> loops.
>

I would say: prune those Via branch parameters. The only useful information 
they possibly provide, is whether the element is RFC3261 compliant or not. 
So take the first 7 chars of each via branch, and strip the rest.

Then compression will be better too, the only problem is that the sender 
needs to support it (or at least be capable to log binary messages)

> Fortunately, a rather good job of pruning can be done by a simple
> algorithm:  Sort all the headers by a weighting that says how valuable
> they are for diagnostics.  The sort needs to be stable, in that all
> headers of the same type have their order preserved.  Then retain an
> initial subset of the headers without going over the desired body
> length.
>
> The ordering of the headers will be something like this:
>
> Most valuable: Via
>

Add to this: the resulting request URI, Record-Route headers (many proxies 
append state in those, which might help) and Route headers. And as above, 
limit via branches to 7 chars (or even less, eg replace them with '1' or '0' 
depending on whether the magic cookie was there. Retain other parameters, 
such as 'alias' which might indicate that connection reuse is happening)

> Accept-*

including Accept-Contact and Reject-Contact, although in general all these 
UAC-inserted headers are less valuable; many UACs will either still have the 
original request (for retransmissions, in particular for UDP) or be able to 
reconstruct it.

>
> Content-*
>
> Least valuable: To
> From
> CSeq
> Call-ID
> Max-Forwards
> Contact

You'd perhaps also want any Contact header (though the UAC might reconstruct 
that). It could contain callee capabilities that explain why the request was 
routed in a particular way.

> Allow
>
> The last group of headers can always be omitted, even if you have no
> length restriction, because the UAC can reconstruct them, either (1)
> it is identical to the same header in the response, (2) it is
> generated with the same value in every request by the UAC, or (3) its
> value is constant in this situation (e.g., Max-Forwards is 0).
>
> Dale
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


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Ok, let's bubble back up to use cases.

What's the use case for a data URL giving Location in a header here, and 
does that function have to be solved/addressed now, in this document?

I might add that just because something is possible, doesn't mean it needs 
to be specified every time.  Reducing complexity means not taking on more 
than is necessary for the expected uses of a new protocol or extension to 
that protocol.

If, in the future, that additional capability is needed, a new document can 
be written as an extension to this one.  This allows, at least, for an 
implementation to be in 100% compliance with the earlier spec and choose 
not to do the additional function, which that implementation chooses not to 
offer.  Again, this reduces complexity to where it is needed, yet allows 
any implementation to add functionality by incorporating additional spec(s).

At 09:50 AM 7/15/2006 -0400, Henning Schulzrinne wrote:
>>One type of authorization policy could simply be 'anyone on the
>>path of this request is allowed to access the data for the duration
>>of the dialog'. To provide this, as long as each hop was encrypted
>>ala sips, if  a proxy inserted an HTTP URI with a crypto-random
>>URI, the proxy would grant access to the location information as
>>long as the dialog is still active.
>
>That's essentially the same security model offered by the location-by- 
>value approach, as far as I can tell.
>
>I agree that location-by-reference offers other security models; I
>never claimed that this is an either-or model; we can easily support
>both without changes in the header definition. After all, they are
>all URLs.
>
>The trade-off is that for the simple on-path-can-read, the by- reference 
>approach incurs significant overhead, as the location
>provider has to be able to maintain an HTTP server that is accessible
>by any potential destination of the call. For the case of end-system
>inserted location, this now means another configuration option since
>the UA or proxy has to have write access to an HTTP server, which
>isn't even standardized right now. (HTTP PUT? XCAP? WebDAV? ftp? some
>other special protocol?) For proxy-inserted, that server has to be
>outside a NAT and globally reachable.
>
>In other words, for the common case you stipulate, by-reference only
>adds complexity and failure modes, with no increase in security or
>efficiency. (Actually, by reference is likely to be less efficient
>for the outbound-proxy inserts case, as multiple downstream proxies
>all have to open (TLS) connections to the server holding the
>location, each at several hundred or thousand bytes of traffic to get
>a 60-byte location object.)
>
>
>>
>>Of course, you can always use traditional white lists if you happen
>>to be in an environment where there is a pre-arranged relationship
>>between parties. Location by reference allows both parties.
>
>I'm not disagreeing and don't think I've ever said that location-by- 
>reference is not useful in some circumstances. There's just no point
>paying the complexity and reliability cost when there is no
>functional difference.
>
>I think the basic design principle holds: simple things should be
>simple to do.
>
>Henning
>
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Sun Jul 16 09:57:39 2006
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Subject: RE: [Sip] Update to RFC3329, aka Security Agreement
Date: Sun, 16 Jul 2006 16:54:36 +0300
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Thread-Topic: [Sip] Update to RFC3329, aka Security Agreement
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Hi Aki and all...


This email follows my conversation with Gonzalo at last IETF 66th -
regarding RFC3329 issues.

Here are some issues we thought of. I wonder if it can assist your
draft. Anyway I will be glad to constantly contribute to this draft.

Here are the comments/additions to the issues:

1.	
RFC3329 describes how the same security-agreement is used in several
transactions, but does not specify the relations between those
transactions. For example, a security-agreement can be obtained via
OPTIONS transaction and then used by a call-leg. Since there is no
SIP-wise relation between those two SIP entities, a key for comparison
must be defined. This is extremely problematic on server side, because
any incoming transaction must be verified according to the
security-agreement used in previous (unrelated and even nonexistent)
transactions.
One possible solution that is not good for all cases: match of
security-agreements to SIP dialogs using peer transport address.

2.	
A client that obtained a security-agreement with the server is
responsible to select and initiate a security mechanism to use in
further requests. The security-agreement negotiation process, however,
does not necessarily indicate to the server what the client is about to
choose (for example, if the server is the initiator of the negotiation).

Possible Solution: The server will leave all resources open (for
example, IPsec ports) waiting for the first protected request from the
client. Once a request is received from the client, the server will
conclude the choice made by the client and will release other resources.


3.	
The security-agreement process indicates to the client which security
mechanism to use in its requests to the server. However, there are no
explicit instructions for the server on how to send requests to the
client. RFC3329 indicates the possibility to use the security that the
client obtained for requests from the server, but it does not specify
the correct behavior. Moreover, the server might want to send a request
to the client before the client initiated the security mechanism (see
2).
Possible solution: Once the client sends a protected request to the
server via TLS or IPSec connection, the server will store this
connection and use it in its requests to the client (connection reuse
alike). Before that, any request from the server will be sent using
regular address resolution.

4.	
RFC3329 does not discuss the possibility of several suggestions of the
same security mechanism in one negotiation (for example, suggesting
ipsec-3gpp with two different algorithms).
draft-niemi-rfc3329-issues-00.txt states that future standardization
will describe how it shell be done.
Possible solution: In the mean time, till this issue is fixed; do not
allow negotiating the same security mechanism twice.

5.	
RFC3329 requires security-agreement to be held upon 494 responses. 3GPP
TS.33.203 however, allows the security-agreement to be held on other
responses, specifically 401 responses that are generated by the second
hop. 
Possible solution: Allow in RFC 3329 negotiating upon any 4xx response,
i.e., can add Security-Sever list to 4xx responses on server side, and
look for Security-Server list in 4xx responses on client side.

6.	
Another problem arouse from point 4, where the 401 response may contain
Authentication info originating at a second hop, but (theoretically) may
also contain Authentication info that is part of the security-agreement
negotiation. It may not be possible to differ between the two.
Possible solution: Consider authentication information as part of a
security-agreement only if it arrived in a 494 response.

I will be glad to know what you think of these issues...

Thanks 

Sharon Laivand.




-----Original Message-----
From: Aki Niemi [mailto:aki.niemi@nokia.com] 
Sent: Tuesday, March 21, 2006 2:26 AM
To: sip@ietf.org
Subject: [Sip] Update to RFC3329, aka Security Agreement

All,

During the past couple of years, there have been a few reported issues
on RFC 3329, based on implementation experience.

I've gathered the set of issues in an I-D:
http://www.ietf.org/internet-drafts/draft-niemi-rfc3329-issues-00.txt

We've already talked about rfc3329bis among the authors, and we feel
this would be appropriate in order to straighten out the issues and
improve interop. Any comments are welcome.

Cheers,
Aki

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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
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On Sat, 2006-07-15 at 17:56 +0200, Jeroen van Bemmel wrote:
> Dale,
> 
> see inline
> 
> Regards,
> 
> jeroen
> 
> >   If it is looping/spiraling, then it will be having good compression
> >   ratio for 483.
> >
> > No, it won't, because most of the bytes in Via headers are in the
> > branch parameters, which are all different, even when the message
> > loops.
> >
> 
> I would say: prune those Via branch parameters. The only useful information 
> they possibly provide, is whether the element is RFC3261 compliant or not. 
> So take the first 7 chars of each via branch, and strip the rest.

There are two reasons I don't like this approach:

1) The message/sipfrag definition does not allow header fields to be
modified - just removed or included as is.

2) The branch parameters may well be useful to correlate information in
log files (in fact, they may also have information coded into them that
can be read by someone who knows how they are constructed).

> Then compression will be better too, the only problem is that the sender 
> needs to support it (or at least be capable to log binary messages)

None of which addresses the fact that you cannot easily predict how much
of the message to include so that you get <201 bytes out of the
compressor.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
From: Scott Lawrence <slawrence@pingtel.com>
To: Dale.Worley@comcast.net
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On Fri, 2006-07-14 at 23:29 -0400, Dale.Worley@comcast.net wrote:
>    From: "Samir Srivastava" <samirsr@nortel.com>
> 
>    If it is looping/spiraling, then it will be having good compression
>    ratio for 483. 
> 
> No, it won't, because most of the bytes in Via headers are in the
> branch parameters, which are all different, even when the message
> loops.
> 
> Fortunately, a rather good job of pruning can be done by a simple
> algorithm:  Sort all the headers by a weighting that says how valuable
> they are for diagnostics.  The sort needs to be stable, in that all
> headers of the same type have their order preserved.  Then retain an
> initial subset of the headers without going over the desired body
> length.

The more complicated the algorithm, the more chance of introducing
errors that confuse rather than illuminate.  I'd rather send back less
but preserve all ordering than risk corrupting the information that does
get back.  Bad diagnostic information is worse than no diagnostic
information.   The key word is 'simple'.

I will propose some additional text for the pruning section to recommend
removing those headers that are identical to those in the returned
response (call-id, cseq, etc).

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Sun Jul 16 12:07:52 2006
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Subject: RE: [Sip] Size of message/sipfrag in diagnostic responses
From: Scott Lawrence <slawrence@pingtel.com>
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On Fri, 2006-07-14 at 15:16 -0400, Hadriel Kaplan wrote:
> 
> > -----Original Message-----
> > From: Scott Lawrence [mailto:slawrence@pingtel.com]
> > 
> > On Thu, 2006-07-13 at 16:14 -0500, Dean Willis wrote:
> > > Also, this will still explode in environments where evil topology-
> > > hiding functions (common in SBCs) have suppressed the Via header
> > > fields. Some people would have me believe that this occurs in most if
> > > not all IMS deployments, giving us a low probability that large
> > > responses will ever actually work.
> > 
> > People who deploy such things and don't keep them up to date when the
> > protocol changes deserve what happens to them.  I don't think that we
> > need to maintain backward compatibility or otherwise protect the
> > clueless from implementations that have chosen (for, I am sure, good
> > and/or revenue-generating reasons) to violate MUST-level rules.
> 
> Actually, I think his point was that such evil practices will impact _your_
> non-evil providers/UAs.  And I take back what I said about it not being a
> problem, because I see Dean's point now and can think of how it can be a
> problem.  The problem is if you're a good guy calling _through_ an evil
> provider (unbeknownst to you) transiting to other good providers.
> 
> If the loop is in the other good provider, and they implement this draft,
> but the evil one doesn't strip the attachment, then the extra-large response
> may not get back to you because of UDP issues.  So you'll not get the final
> response, and have a bad calling experience, and create issues for the good
> provider as well.  That's not good, for the good guys.  While I know at
> least some Via-hiding boxes can strip the attachment out, that doesn't mean
> all of them will/can.  
> 
> So perhaps you should size-limit all responses.  The odds of some hop or
> other not being UDP isn't that good anyway today, and not many will want to
> implement a draft which negatively impacts them for even a subset of cases.
> Though I would like TCP-usage to get the benefit of TCP to promote its use
> in the future, if you want rapid ubiquitous adoption of this draft mechanism
> then you have to co-exist with such use cases, IMHO.  

I think that crippling a useful feature to allow for elements that
blatantly violate the rules is going too far.  

In the scenario you describe, it will be pretty easy to detect that the
problem occurs on the other size of the evilbox and that the evilbox has
prevented the response from getting back, because if you do the
progressive hop-limit trick it will show that you get responses from
everywhere along the path until you get above the size limit.

In any event, the proposed text says:

        If no request Via header value specifies UDP, then the
        server is free to either prune the returned message/sipfrag body
        as specified in section X or to return the full set of request
        headers.

so if an implementation wishes to always prune to under 200, it is free
to do so and avoid even this potential problem.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Sun Jul 16 12:11:41 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Scott Lawrence" <slawrence@pingtel.com>
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Sun, 16 Jul 2006 18:09:39 +0200
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Hi Scott,

>> see inline
>>
>> Regards,
>>
>> jeroen
>>
>> >   If it is looping/spiraling, then it will be having good compression
>> >   ratio for 483.
>> >
>> > No, it won't, because most of the bytes in Via headers are in the
>> > branch parameters, which are all different, even when the message
>> > loops.
>> >
>>
>> I would say: prune those Via branch parameters. The only useful 
>> information
>> they possibly provide, is whether the element is RFC3261 compliant or 
>> not.
>> So take the first 7 chars of each via branch, and strip the rest.
>
> There are two reasons I don't like this approach:
>
> 1) The message/sipfrag definition does not allow header fields to be
> modified - just removed or included as is.
>

OK, so then perhaps either the message/sipfrag definition needs to be fixed 
as well, or message/sipfrag is not fit for the purpose we have here. 
"message/sipfrag-pruned" ?

> 2) The branch parameters may well be useful to correlate information in
> log files (in fact, they may also have information coded into them that
> can be read by someone who knows how they are constructed).
>

Yes, all information may potentially be helpful. The problem is (as I'm sure 
you're aware of): "Somethin' 's gotta give". In my opinion, if the choice is 
between removing via's starting from the bottom (as the current draft 
suggests, such that the UA can detect that something is missing) or removing 
branch parameters until it fits, I'd rather have all via headers without 
branches. That for sure is something the UA can interpret, as opposed to a 
proprietary branch-encoded scheme which is much less likely to be 
understood. If desired, you could start with all vias including branches, 
and start pruning branch parameters first (so you'd still have some branch 
parameters left, preferrably at the bottom I would say)

>> Then compression will be better too, the only problem is that the sender
>> needs to support it (or at least be capable to log binary messages)
>
> None of which addresses the fact that you cannot easily predict how much
> of the message to include so that you get <201 bytes out of the
> compressor.

Perhaps a streaming compressor like deflate (RFC1950) would then be better. 
I'm no expert, but I believe you could cut that off when you reach the 
maximum size?

Regards,

Jeroen

>
> -- 
> Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
>  sipXpbx project coordinator - SIPfoundry 
> http://www.sipfoundry.org/sipX
>  Chief Architect             - Pingtel Corp. http://www.pingtel.com/
>
> 


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From sip-bounces@ietf.org Sun Jul 16 13:05:33 2006
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
From: Scott Lawrence <slawrence@pingtel.com>
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On Sun, 2006-07-16 at 18:09 +0200, Jeroen van Bemmel wrote:
> Hi Scott,
> 
> >> see inline
> >>
> >> Regards,
> >>
> >> jeroen
> >>
> >> >   If it is looping/spiraling, then it will be having good compression
> >> >   ratio for 483.
> >> >
> >> > No, it won't, because most of the bytes in Via headers are in the
> >> > branch parameters, which are all different, even when the message
> >> > loops.
> >> >
> >>
> >> I would say: prune those Via branch parameters. The only useful 
> >> information
> >> they possibly provide, is whether the element is RFC3261 compliant or 
> >> not.
> >> So take the first 7 chars of each via branch, and strip the rest.
> >
> > There are two reasons I don't like this approach:
> >
> > 1) The message/sipfrag definition does not allow header fields to be
> > modified - just removed or included as is.
> >
> 
> OK, so then perhaps either the message/sipfrag definition needs to be fixed 
> as well, or message/sipfrag is not fit for the purpose we have here. 
> "message/sipfrag-pruned" ?
> 
> > 2) The branch parameters may well be useful to correlate information in
> > log files (in fact, they may also have information coded into them that
> > can be read by someone who knows how they are constructed).
> >
> 
> Yes, all information may potentially be helpful. The problem is (as I'm sure 
> you're aware of): "Somethin' 's gotta give".

My choice would be UDP (only half :-) )

>  In my opinion, if the choice is 
> between removing via's starting from the bottom (as the current draft 
> suggests, such that the UA can detect that something is missing) or removing 
> branch parameters until it fits, I'd rather have all via headers without 
> branches. 

On long paths, you won't have room for that in 200 bytes anyway.

> That for sure is something the UA can interpret, as opposed to a 
> proprietary branch-encoded scheme which is much less likely to be 
> understood. 

I don't actually hope for UAs to interpret this - just human beings.


-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Mon Jul 17 04:58:23 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: =?iso-8859-1?Q?=27Ant=F3nio_Manuel_Amaral=27?=
	<antonio-mn-amaral@ptinovacao.pt>, "'Sip@ietf.org'" <Sip@ietf.org>
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To make this directly relevant to SIP, the IMS S-CSCF behaves exactly =
as specified in RFC 3261. If it has a pending transaction, then section =
16.8 and section 16.9 of RFC 3261 apply.=20

The session timer (RFC 4028) may optionally be used by any of the IMS =
entities that behave as a SIP proxy.

regards

Keith

> -----Original Message-----
> From: Ant=F3nio Manuel Amaral =
[mailto:antonio-mn-amaral@ptinovacao.pt]
> Sent: 14 July 2006 11:17
> To: Sip@ietf.org
> Subject: [Sip] S-CSCF expected behavior
>=20
>=20
>=20
> Dear All,
>=20
> I don't know if this is the correct mailing list to put my=20
> question and if not my apologies for that.
>=20
> Consider the following IMS scenario
>=20
>=20
> (IMS core Network) -> S-CSCF -> AS
>=20
>=20
> If the connection between AS and S-CSCF fails what is the=20
> expected behavior of S-CSCF? It must release all active calls=20
> sending BYE messages to caller and called clients in order to=20
> terminate the calls?
>=20
> Thanks in advance
>=20
> Regards
>=20
> AA=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From sip-bounces@ietf.org Mon Jul 17 07:13:05 2006
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Subject: RE: [Sip] Update to RFC3329, aka Security Agreement
From: Aki Niemi <aki.niemi@nokia.com>
To: ext Sharon Laivand <sharonl@radvision.com>
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On su, 2006-07-16 at 16:54 +0300, ext Sharon Laivand wrote:
> Hi Aki and all...
> 
> 
> This email follows my conversation with Gonzalo at last IETF 66th -
> regarding RFC3329 issues.
> 
> Here are some issues we thought of. I wonder if it can assist your
> draft. Anyway I will be glad to constantly contribute to this draft.

Thanks for your feedback. Some comments inline.

> Here are the comments/additions to the issues:
> 
> 1.	
> RFC3329 describes how the same security-agreement is used in several
> transactions, but does not specify the relations between those
> transactions. For example, a security-agreement can be obtained via
> OPTIONS transaction and then used by a call-leg. Since there is no
> SIP-wise relation between those two SIP entities, a key for comparison
> must be defined. This is extremely problematic on server side, because
> any incoming transaction must be verified according to the
> security-agreement used in previous (unrelated and even nonexistent)
> transactions.
> One possible solution that is not good for all cases: match of
> security-agreements to SIP dialogs using peer transport address.

The primary reason this is underspecified is that back when RFC 3329 was
in the works, we didn't have a clear architecture for this sort of
thing. 

We do now, as outbound and SIP identity both rely on a UA having a
"sentinel" of sorts in the network to whom it has a persistent security
association and can send and receive traffic in a secure, identity-safe
way. 

I would say the most common case is that this association is
bootstrapped using a REGISTER, which was, in fact, the assumption of RFC
3329 as well. Perhaps not that clearly stated, though.

> 2.	
> A client that obtained a security-agreement with the server is
> responsible to select and initiate a security mechanism to use in
> further requests. The security-agreement negotiation process, however,
> does not necessarily indicate to the server what the client is about to
> choose (for example, if the server is the initiator of the negotiation).
> 
> Possible Solution: The server will leave all resources open (for
> example, IPsec ports) waiting for the first protected request from the
> client. Once a request is received from the client, the server will
> conclude the choice made by the client and will release other resources.

That is a possible solution, although I've since come to believe that
choice in these sorts of situations is a bug and not a feature. This
becomes especially hairy if there would be need to indicate preference
accross different security mechanisms. Example: I am willing to do TLS,
but only with the PSK cipher suites; otherwise, I want to do transport
mode IPsec with IKE; but if it fails then perhaps TLS with RSA one-way.

My suggestion in this case would be to *not* add TLS cipher suite
identifiers to sec-agree, but e.g., to only ever do negotiation in the
TLS handshake, effectively deprecating sec-agree...

> 3.	
> The security-agreement process indicates to the client which security
> mechanism to use in its requests to the server. However, there are no
> explicit instructions for the server on how to send requests to the
> client. RFC3329 indicates the possibility to use the security that the
> client obtained for requests from the server, but it does not specify
> the correct behavior. Moreover, the server might want to send a request
> to the client before the client initiated the security mechanism (see
> 2).
> Possible solution: Once the client sends a protected request to the
> server via TLS or IPSec connection, the server will store this
> connection and use it in its requests to the client (connection reuse
> alike). Before that, any request from the server will be sent using
> regular address resolution.

I agree. This is effectively what sip-outbound defines; the only missing
thing is to clarify the security agreement procedure and say that before
the setup of the secure channel is finished, the server would actually
not send any requests to the UA. This would actually be a natural
consequence of using a REGISTER to bootstrap the negotiation.

> 4.	
> RFC3329 does not discuss the possibility of several suggestions of the
> same security mechanism in one negotiation (for example, suggesting
> ipsec-3gpp with two different algorithms).
> draft-niemi-rfc3329-issues-00.txt states that future standardization
> will describe how it shell be done.
> Possible solution: In the mean time, till this issue is fixed; do not
> allow negotiating the same security mechanism twice.

I think RFC 3329 does allow several ipsec-3gpp mechanisms with different
algorithms to be offered, even giving each one a different q-value for
preference. Of course, in addition to not being well documented, as you
add more supported algorithms, there is a combinatorial explosion
resulting in more and more (redundant) ipsec-3gpp mechanisms offered.

There's been a suggestion to change the 3GPP grammar to allow the hash
and cipher algorithms be represented as a comma-separated list, which
seems like a reasonable idea.

> 5.	
> RFC3329 requires security-agreement to be held upon 494 responses. 3GPP
> TS.33.203 however, allows the security-agreement to be held on other
> responses, specifically 401 responses that are generated by the second
> hop. 
> Possible solution: Allow in RFC 3329 negotiating upon any 4xx response,
> i.e., can add Security-Sever list to 4xx responses on server side, and
> look for Security-Server list in 4xx responses on client side.

I'm not sure how to reconcile this in specification -- I *think* anyone
actually implementing RFC 3329 is doing so to support the 3GPP TS. So in
reality, this may not be such a big issue, but rather a place where 3GPP
and/or we could clarify and document existing implementation practice.

That said, allowing Security-Server to appear in a 401 would better
accomodate using RFC 3329 with sip-outbound.

> 6.	
> Another problem arouse from point 4, where the 401 response may contain
> Authentication info originating at a second hop, but (theoretically) may
> also contain Authentication info that is part of the security-agreement
> negotiation. It may not be possible to differ between the two.
> Possible solution: Consider authentication information as part of a
> security-agreement only if it arrived in a 494 response.

This is an interesting point. It might actually happen that a
REGISTER/407 be challenged again with a 401 and a Security-Server. In
which case the original (outbound) proxy would somehow need to be
accomodated.

I'm not sure how to address this... So let me ask a meta-question
instead. ;-) 

How do we want to approach addressing these issues? The rfc3329-issues
draft basically proposes three alternatives:

1) Document all of the found issues with suggestions of how to deal with
them in an RFC errata entry.

2) Do #1 and fix the d-ver calculation as an update to RFC3329.

3) Revise RFC 3329.

Personally, I'm inclined to go with #1, maybe #2, but definitely not #3.
Reason being something I mentioned already above; I think keeping IPsec
around as a potential way to secure SIP is not viable. I think TLS is a
much better suited for securing SIP hops, and especially here, limiting
the amount of choices makes interop better.

Also, one of the key benefits of RFC 3329 is that it protects against
bid-down; however, given most UAs in the wild today don't implement it
pre-empts that benefit pretty well at this point. We sort of missed the
boat on that one, unfortunately.



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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Mon Jul 17 11:22:26 2006
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From: Fatih "Eyüp" NAR <fenar@yahoo.com>
Subject: RE: [Sip] S-CSCF expected behavior
To: "Drage, Keith \(Keith\)" <drage@lucent.com>,
  "'António" Manuel Amaral' <antonio-mn-amaral@ptinovacao.pt>,
  "'Sip@ietf.org'" <Sip@ietf.org>
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what about active sessions (INVITE transaction
completed successfully with 200OK-ACK previously) on
S-CSCF?

ex: For the prepaid customer type once the connection
dropped between AS and S-CSCF only way to prevent -
"minus" credit is to end the session with BYE by S-.



--- "Drage, Keith (Keith)" <drage@lucent.com> wrote:

> To make this directly relevant to SIP, the IMS
> S-CSCF behaves exactly as specified in RFC 3261. If
> it has a pending transaction, then section 16.8 and
> section 16.9 of RFC 3261 apply. 
> 
> The session timer (RFC 4028) may optionally be used
> by any of the IMS entities that behave as a SIP
> proxy.
> 
> regards
> 
> Keith
> 
> > -----Original Message-----
> > From: António Manuel Amaral
> [mailto:antonio-mn-amaral@ptinovacao.pt]
> > Sent: 14 July 2006 11:17
> > To: Sip@ietf.org
> > Subject: [Sip] S-CSCF expected behavior
> > 
> > 
> > 
> > Dear All,
> > 
> > I don't know if this is the correct mailing list
> to put my 
> > question and if not my apologies for that.
> > 
> > Consider the following IMS scenario
> > 
> > 
> > (IMS core Network) -> S-CSCF -> AS
> > 
> > 
> > If the connection between AS and S-CSCF fails what
> is the 
> > expected behavior of S-CSCF? It must release all
> active calls 
> > sending BYE messages to caller and called clients
> in order to 
> > terminate the calls?
> > 
> > Thanks in advance
> > 
> > Regards
> > 
> > AA 
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP
> Protocol
> > Use sip-implementors@cs.columbia.edu for questions
> on current sip
> > Use sipping@ietf.org for new developments on the
> application of sip
> > 
> 
> _______________________________________________
> Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP
> Protocol
> Use sip-implementors@cs.columbia.edu for questions
> on current sip
> Use sipping@ietf.org for new developments on the
> application of sip
> 


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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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=20
Hi,
=20
If the UA chooses to only establish a single flow there is really =
nothing the network can do about it, even if the network allows the UA =
to establish multiple flows.
=20
But, as we have discussed before, it would be useful if the network =
could indicate the max number of flows it allows the UA to establish. =
And, if thet network indicates max-flow=3D1 the UA can assume that the =
network can provide reliability even with a single flow.
=20
Regards,
=20
Christer
=20

________________________________

From: Joel M. Halpern [mailto:joel@stevecrocker.com]
Sent: Tue 11/07/2006 01:25
To: Juha Heinanen
Cc: Miguel Garcia; Rohan Mahy; sip@ietf.org; Cullen Jennings
Subject: Re: [Sip] Re: A coment to draft-ietf-sip-outbound-04



By this interpretation, a network could not offer redundant / backup
connectivity (via outbound) to UAs which want to take advantage of
that without also mandating that all UAs make connect  to all the =
entities.
Yours,
Joel

At 03:47 PM 7/10/2006, Juha Heinanen wrote:
>Joel M. Halpern writes:
>
>  > Nor is it "outbound"s job to mandate specific redundancy mechanisms
>  > (it currently seems to do so, although I think all of that is
>  > intended as examples.)
>
>outbound can support any number of redundancy mechanisms.  if a domain
>chooses to require two flows from a particular UA, then it is up to
>outbound UA to do so.  another domain may have different policy.  as
>long as outbound UA is able to setup at least two flows everything is
>ok.
>
>-- juha
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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<DIV dir=3Dltr><FONT face=3DArial size=3D2>If the UA chooses to only =
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<DIV dir=3Dltr><FONT face=3DArial size=3D2>But, as we have discussed =
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be useful if the network could indicate the max number of flows it =
allows the UA =0A=
to establish. And, if thet network indicates max-flow=3D1 the UA can =
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<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Regards,</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Christer</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>From:</B> Joel M. Halpern =0A=
[mailto:joel@stevecrocker.com]<BR><B>Sent:</B> Tue 11/07/2006 =0A=
01:25<BR><B>To:</B> Juha Heinanen<BR><B>Cc:</B> Miguel Garcia; Rohan =
Mahy; =0A=
sip@ietf.org; Cullen Jennings<BR><B>Subject:</B> Re: [Sip] Re: A coment =
to =0A=
draft-ietf-sip-outbound-04<BR></FONT><BR></DIV>=0A=
<DIV>=0A=
<P><FONT size=3D2>By this interpretation, a network could not offer =
redundant / =0A=
backup<BR>connectivity (via outbound) to UAs which want to take =
advantage =0A=
of<BR>that without also mandating that all UAs make connect&nbsp; to all =
the =0A=
entities.<BR>Yours,<BR>Joel<BR><BR>At 03:47 PM 7/10/2006, Juha Heinanen =0A=
wrote:<BR>&gt;Joel M. Halpern writes:<BR>&gt;<BR>&gt;&nbsp; &gt; Nor is =
it =0A=
"outbound"s job to mandate specific redundancy mechanisms<BR>&gt;&nbsp; =
&gt; (it =0A=
currently seems to do so, although I think all of that is<BR>&gt;&nbsp; =
&gt; =0A=
intended as examples.)<BR>&gt;<BR>&gt;outbound can support any number of =0A=
redundancy mechanisms.&nbsp; if a domain<BR>&gt;chooses to require two =
flows =0A=
from a particular UA, then it is up to<BR>&gt;outbound UA to do =
so.&nbsp; =0A=
another domain may have different policy.&nbsp; as<BR>&gt;long as =
outbound UA is =0A=
able to setup at least two flows everything =
is<BR>&gt;ok.<BR>&gt;<BR>&gt;-- =0A=
juha<BR>&gt;<BR>&gt;_______________________________________________<BR>&g=
t;Sip =0A=
mailing list&nbsp; <A =0A=
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
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list is for NEW development of the core SIP Protocol<BR>&gt;Use =0A=
sip-implementors@cs.columbia.edu for questions on current sip<BR>&gt;Use =0A=
sipping@ietf.org for new developments on the application of =0A=
sip<BR><BR><BR>_______________________________________________<BR>Sip =
mailing =0A=
list&nbsp; <A =0A=
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
/mailman/listinfo/sip</A><BR>This =0A=
list is for NEW development of the core SIP Protocol<BR>Use =0A=
sip-implementors@cs.columbia.edu for questions on current sip<BR>Use =0A=
sipping@ietf.org for new developments on the application of =0A=
sip<BR></FONT></P></DIV>=0A=
=0A=
</BODY>=0A=
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Registration for SIPit 19.

This SIPit will take place at the University of New Hampshire
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participants, so register early.

Registration will close September 29.

For more information, and to register, see http://www.sipit.net

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From sip-bounces@ietf.org Mon Jul 17 19:38:59 2006
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From: Dale.Worley@comcast.net
In-reply-to: <001e01c6a827$38b8bda0$31713b51@BEMBUSTER> (jbemmel@zonnet.nl)
Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
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   From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>

   I would say: prune those Via branch parameters. The only useful information 
   they possibly provide, is whether the element is RFC3261 compliant or not. 
   So take the first 7 chars of each via branch, and strip the rest.

   Then compression will be better too, the only problem is that the sender 
   needs to support it (or at least be capable to log binary messages)

This idea has advantages.  But it branches into the territory of
revising the definition of message/sipfrag, and possibly dinking with
parsers for it.

As for compression, in theory one could add a Content-Encoding header
-- but again, that's burdening implementations.  Most compression
algorithms are stream-oriented, and one could order the headers
according to importance, then start compressing, and chop the output
at 200 bytes.  But again, you'd force implementations to have a gzip
decompresser that returns "the output so far" rather than just an
error indication.

IMHO, the primary design goal is to make it *easy* to deploy this
mechanism widely.  After it has proven its worth, and attracted
significant commitment from implementors -- and we've discovered what
the problems with it are *in practice* -- then redesign for better
performance.

"First make it work.  Then make it fast."

   You'd perhaps also want any Contact header (though the UAC might reconstruct 
   that). It could contain callee capabilities that explain why the request was 
   routed in a particular way.

Presumably the Contact header in the failed request has the same value
as was put there when the UAC sent the request.  The response should
contain a Contact, of course, but that's not what we're talking about.

Dale

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From sip-bounces@ietf.org Mon Jul 17 19:52:07 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
Date: Tue, 18 Jul 2006 01:49:47 +0200
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Simplest and fastest is probably: for UDP, return the warning and as much as 
will fit in ( 200 + Content-Length ) bytes
To get more diagnostics, send the request again using TCP (similar to 
authentication challenge but different transport)

Trade-off is that there is no guarantee that the same thing will happen (but 
likely).

Of course you can still add things like "if sender supports deflate, it is 
recommended to use that"

Regards,

jeroen

----- Original Message ----- 
From: <Dale.Worley@comcast.net>
To: <sip@ietf.org>
Sent: Tuesday, July 18, 2006 1:37 AM
Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses


>   From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
>
>   I would say: prune those Via branch parameters. The only useful 
> information
>   they possibly provide, is whether the element is RFC3261 compliant or 
> not.
>   So take the first 7 chars of each via branch, and strip the rest.
>
>   Then compression will be better too, the only problem is that the sender
>   needs to support it (or at least be capable to log binary messages)
>
> This idea has advantages.  But it branches into the territory of
> revising the definition of message/sipfrag, and possibly dinking with
> parsers for it.
>
> As for compression, in theory one could add a Content-Encoding header
> -- but again, that's burdening implementations.  Most compression
> algorithms are stream-oriented, and one could order the headers
> according to importance, then start compressing, and chop the output
> at 200 bytes.  But again, you'd force implementations to have a gzip
> decompresser that returns "the output so far" rather than just an
> error indication.
>
> IMHO, the primary design goal is to make it *easy* to deploy this
> mechanism widely.  After it has proven its worth, and attracted
> significant commitment from implementors -- and we've discovered what
> the problems with it are *in practice* -- then redesign for better
> performance.
>
> "First make it work.  Then make it fast."
>
>   You'd perhaps also want any Contact header (though the UAC might 
> reconstruct
>   that). It could contain callee capabilities that explain why the request 
> was
>   routed in a particular way.
>
> Presumably the Contact header in the failed request has the same value
> as was put there when the UAC sent the request.  The response should
> contain a Contact, of course, but that's not what we're talking about.
>
> Dale
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


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From sip-bounces@ietf.org Mon Jul 17 21:01:28 2006
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Subject: Re: [Sip] Size of message/sipfrag in diagnostic responses
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   From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>

   Of course you can still add things like "if sender supports
   deflate, it is recommended to use that"

Ah, yes.  And conveniently, there is an "Accept-Encoding" header for
the UAC to indicate just that.

Dale

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From sip-bounces@ietf.org Tue Jul 18 08:55:36 2006
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From: "Brian Rosen" <br@brianrosen.net>
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In sip-location-conveyance, we allow a URI in the Location header to provide
for location-by-reference.  We must have some kind of text that says what
the recipient does with this URI.

Our proposal is to change the ABNF, which currently allows this URI to be
AbsoluteURI (meaning any scheme) and replace it with HTTP/HTTPS only.  Of
course, there would still be the cid: for message body identification.  We
would then define a dereference process as doing an HTTP/HTTPS GET, with no
parameters, on the URI, and expect to receive in return a PIDF-LO, exactly
as it would appear in a body.  The text would state MUST implement TLS,
except for the emergency case.  The text would further state that additional
dereferencing procedures could be defined by a standards track RFC.

Brian and James


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From sip-bounces@ietf.org Tue Jul 18 08:58:36 2006
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It has been pointed out that the new 424 Bad Location Information error code
defined in location-conveyance does not provide sufficient information for
the recipient to know what was wrong with the location that prompted the
error.

We propose to resolve this by allowing a Reason header in the 424(Bad
Location Information) response.  We note that RFC3326 Which defines the
Reason header states that it normally isn't used in a response, but a
standards track RFC could allow this functionality.  We propose that this
document do so.  Our expectation would be that geopriv would process an RFC
that establishes a sub-registry for the Reason codes.  We'll write a draft
on this to get the ball started if this seems like a reasonable approach.
We would not make location-conveyance normatively dependent on the
subregistry.

Brian and James


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From sip-bounces@ietf.org Tue Jul 18 09:01:10 2006
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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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Brian, James:

this is simply false. If desired, XML-DSIG is a well-known and  
accepted mechanism to achieve integrity. Such specification is no  
longer than the MUST NOT that you propose. You are also willfully  
ignoring the other technical arguments made as part of this discussion.

Henning

On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:

> There was a desire expressed by Hannes to provide a way for a proxy to
> insert location-by-value.  He had proposed a SIP Location header
> specific way.  In IETF66, Henning suggested just using a data URI.
>
> Doing this may run afoul of the geopriv location privacy concerns,
> because there is no way for the user to sign the header to provide
> assurance that the PIDF, and specifically the retention and other  
> policy
> bits are preserved.  Note that S/MIME provides sufficient integrity
> protection for a body.  It's probably okay to use TLS per hop to  
> provide
> privacy, but not integrity protection.  The authors believe that there
> is no really compelling use case for this feature, and the effort to
> define acceptable security mechanisms for location data in a header
> would be a significant effort, further delaying the draft.
>
> We propose, therefore, to state in sip-location-conveyance, that a  
> Data
> URI MUST NOT be used in the Location header until a standards track  
> RFC
> defines a suitable security mechanism to protect the PIDF in the  
> header.
>
> Brian and James
>
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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From sip-bounces@ietf.org Tue Jul 18 09:20:22 2006
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Subject: Re: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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I have to agree with Henning here.

You can sign the PIDF-LO, encode it and then put it into the data URI.

Even if you do not sign the PIDF-LO in the header it makes no difference 
since that's what you accomplish by sending a location reference (in the 
header) as well.

The argument that you can use authorization policies to decide who is 
allowed to resolve the reference to the location object is a bit 
misleading. Where do these policies come from?

Ciao
Hannes

Henning Schulzrinne wrote:
> Brian, James:
> 
> this is simply false. If desired, XML-DSIG is a well-known and  accepted 
> mechanism to achieve integrity. Such specification is no  longer than 
> the MUST NOT that you propose. You are also willfully  ignoring the 
> other technical arguments made as part of this discussion.
> 
> Henning
> 
> On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:
> 
>> There was a desire expressed by Hannes to provide a way for a proxy to
>> insert location-by-value.  He had proposed a SIP Location header
>> specific way.  In IETF66, Henning suggested just using a data URI.
>>
>> Doing this may run afoul of the geopriv location privacy concerns,
>> because there is no way for the user to sign the header to provide
>> assurance that the PIDF, and specifically the retention and other  policy
>> bits are preserved.  Note that S/MIME provides sufficient integrity
>> protection for a body.  It's probably okay to use TLS per hop to  provide
>> privacy, but not integrity protection.  The authors believe that there
>> is no really compelling use case for this feature, and the effort to
>> define acceptable security mechanisms for location data in a header
>> would be a significant effort, further delaying the draft.
>>
>> We propose, therefore, to state in sip-location-conveyance, that a  Data
>> URI MUST NOT be used in the Location header until a standards track  RFC
>> defines a suitable security mechanism to protect the PIDF in the  header.
>>
>> Brian and James
>>
>> _______________________________________________
>> Geopriv mailing list
>> Geopriv@ietf.org
>> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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From sip-bounces@ietf.org Tue Jul 18 09:28:13 2006
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On Mon, 2006-07-17 at 20:59 -0400, Dale.Worley@comcast.net wrote:
>    From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
> 
>    Of course you can still add things like "if sender supports
>    deflate, it is recommended to use that"
> 
> Ah, yes.  And conveniently, there is an "Accept-Encoding" header for
> the UAC to indicate just that.

But since that applies generically to any response, I don't think there
is any need to call it out specifically here.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Architect             - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Tue Jul 18 09:40:24 2006
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
Date: Tue, 18 Jul 2006 09:38:15 -0400
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I find this discussion somewhat strange, given that draft-ietf-sip- 
location-conveyance-03 expends significant wording to detail all the  
cases where security (of any kind) is *not* to be used.  To pick just  
one

UAC-9  A UAC MUST be capable of transmitting a SIP request without
           protecting the PIDF-LO message body.  It is RECOMMENDED this
           not be the default configuration of any UA.  This requirement
           is orthogonal to the use of TLS or IPSec hop-by-hop between
           SIP elements.


I probably missed it, but I didn't see a 'MUST use S/MIME' or 'MUST  
use TLS or S/MIME for by-reference' in the document. Indeed, another  
message regarding this draft explicitly allowed HTTP as a retrieval  
mechanism, with no mention of HTTPS, S/MIME or other security wrapper.

The closest I see is

SIP endpoints SHOULD implement S/MIME to encrypt
    the PIDF-LO message body (part) end-to-end.

It then continues

    The SIPS-URI from [RFC3261] MUST be implemented for message  
protection (message
    integrity and confidentiality) and SHOULD be used when S/MIME is not
    used.

Once SIPS is used, there is no technical difference between by-value  
(without XML-DSIG), by-attachment and by-reference that I can find.

The draft leaves unexplained such details as to what key or even  
which crypto mechanisms the UAs should use to give proxies a chance  
to get at the data. It is also rather vague on whether  
"protecting" (in req. UAC-9) means encryption or integrity.

Thus, we should at least be consistent: If we don't want unsigned and/ 
or unencrypted location information, ever, the draft should state  
that. We can then discuss which mechanisms are appropriate in each of  
the delivery mechanisms (attachment, by value, by reference) and how  
we can get intermediaries access to this information. Just waving the  
S/MIME flag is no more useful than pulling out the "just do IPsec"  
security section template.

In summary: All three delivery mechanisms allow exactly equivalent  
security properties; the current draft does not mandate the use of  
such mechanisms for the two it specifies.

Henning


On Jul 18, 2006, at 9:18 AM, Hannes Tschofenig wrote:

> I have to agree with Henning here.
>
> You can sign the PIDF-LO, encode it and then put it into the data URI.
>
> Even if you do not sign the PIDF-LO in the header it makes no  
> difference since that's what you accomplish by sending a location  
> reference (in the header) as well.
>
> The argument that you can use authorization policies to decide who  
> is allowed to resolve the reference to the location object is a bit  
> misleading. Where do these policies come from?
>
> Ciao
> Hannes
>
> Henning Schulzrinne wrote:
>> Brian, James:
>> this is simply false. If desired, XML-DSIG is a well-known and   
>> accepted mechanism to achieve integrity. Such specification is no   
>> longer than the MUST NOT that you propose. You are also willfully   
>> ignoring the other technical arguments made as part of this  
>> discussion.
>> Henning
>> On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:
>>> There was a desire expressed by Hannes to provide a way for a  
>>> proxy to
>>> insert location-by-value.  He had proposed a SIP Location header
>>> specific way.  In IETF66, Henning suggested just using a data URI.
>>>
>>> Doing this may run afoul of the geopriv location privacy concerns,
>>> because there is no way for the user to sign the header to provide
>>> assurance that the PIDF, and specifically the retention and  
>>> other  policy
>>> bits are preserved.  Note that S/MIME provides sufficient integrity
>>> protection for a body.  It's probably okay to use TLS per hop to   
>>> provide
>>> privacy, but not integrity protection.  The authors believe that  
>>> there
>>> is no really compelling use case for this feature, and the effort to
>>> define acceptable security mechanisms for location data in a header
>>> would be a significant effort, further delaying the draft.
>>>
>>> We propose, therefore, to state in sip-location-conveyance, that  
>>> a  Data
>>> URI MUST NOT be used in the Location header until a standards  
>>> track  RFC
>>> defines a suitable security mechanism to protect the PIDF in the   
>>> header.
>>>
>>> Brian and James
>>>
>>> _______________________________________________
>>> Geopriv mailing list
>>> Geopriv@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/geopriv
>> _______________________________________________
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From sip-bounces@ietf.org Tue Jul 18 09:59:05 2006
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To: "Rosen, Brian" <Brian.Rosen@neustar.biz>
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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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Brian,

a four-step process:

- take any PIDF-LO
- wrap in XML-DSIG as specified by the relevant W3C recommendation
- convert to base-64
- make into a data URL, as in the example I gave in an earlier message

   data:application/pidf+xml;base64,<base64-data>

Or did I miss the point of your question?

See http://en.wikipedia.org/wiki/Data:_URL for additional data URL  
examples, btw.

Henning

On Jul 18, 2006, at 9:06 AM, Rosen, Brian wrote:

> Perhaps some enlightenment is in order.  It's not clear to me how you
> dsig a Data uri in a Location header (or maybe you mean dsig the  
> pidf-lo
> and put both the pidf-lo and the signature in the Data URI) somehow?
> Are there some examples of this kind of thing elsewhere?
>
> If we are ignoring technical arguments (as opposed to use cases), it's
> out of forgetfulness, and not willfulness.  What were you referring  
> to?
>
> Brian
>


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From sip-bounces@ietf.org Tue Jul 18 10:24:56 2006
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Subject: [Sip] Location Conveyance: Scenarios
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Hi all,=20

I have compiled a few message flows that would make it easier to discuss
the different scenarios. I have listed some challenges along with each
of the scenarios.=20


1) Location by End Host / Per Value/ in Body

    UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C

       |  SIP Message                |          |       |       |
       |---------------------------->|          |       |       |
       |  Location (value/body)      |                  |
       |                             |----------------->|
       |                             |                  |
       |                SIP Message                     |
       |<-----------------------------------------------|
       |                                                |

Problems:=20
* Likelyhood of the UA using S/MIME; nice on paper but does it reflect
reality?
* UA Alice needs to know that (a) intermediaries do not need to process
the PIDF-LO=20
(b) recipient has a certificate
* Unclear what happens if there are multiple recipients.=20



2) Location by End Host / Per Reference / in Body


   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C

      |  SIP Message                |          |       |       |
      |---------------------------->|          |       |       |
      |  Location (ref./body)       |                  |
      |                             |----------------->|
      |                             |                  |
      |              Dereference                       |
      =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
      |                                                |
      |                                                |
      |                SIP Message                     |
      |<-----------------------------------------------|
      |                                                |

Problems:=20
* Can be protected using S/MIME (see issue above)
* If not protected then the considerations of who is allowed to
dereference to a PIDF-LO are applicable. There is certainly a difference
between a reference pointing to a PIDF-LO at the UA itself and a
reference obtained using one of the discussed Geopriv L7 protocols.=20
It might be difficult for the end host to make the PIDF-LO available. It
might be difficult for him to have a server-side certificate but it
might be easier to control access to it.=20


3) Location by SIP Proxy / Per Reference / in Header

   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C

      |  SIP Message                |          |       |       |
      |---------------------------->|          |       |       |
      |                             | SIP Message      |
      |                             |----------------->|
      |                             | Location         |
      |                             | (ref./header)    |
      |                             |                  |
      |                             |                  |
      |                             |   Dereference    |
      |                             =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
      |                             |                  |
      |                SIP Message                     |
      |<-----------------------------------------------|
      |                                                |
      |                                                |
      |                                                |

Problems:

* What should the SIP proxy but into the PIDF-LO when it is created
(policies, identifiers, etc.)?
* Authorization aspects: see above (Why cannot just everyone that sees
the reference in the header passing by dereference it?)
* Signing of the reference theoretically possible. SIP Proxy, however,
needs to know the intended recipient.=20


4) Location by SIP Proxy / Per Value / in Header

  UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C

     |  SIP Message                |          |       |       |
     |---------------------------->|          |       |       |
     |                             | SIP Message      |
     |                             |----------------->|
     |                             | Location         |
     |                             | (value/header)   |
     |                             |                  |
     |                SIP Message                     |
     |<-----------------------------------------------|
     |                                                |
     |                                                |
     |                                                |

Problems:

* What should the SIP proxy but into the PIDF-LO when it is created
(policies, identifiers, etc.)?
* Data URI approach possible but with increased message size.=20
* Signing of the PIDF-LO possible but approach similar. SIP Proxy,
however, needs to know the intended recipient.


5) Location by SBC / Per Value / in Body

  UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
                                 (SBC)
     |  SIP Message                |          |       |       |
     |---------------------------->|          |       |       |
     |  Location (value/body)      | SIP Message      |
     |                             |----------------->|
     |                             | Location         |
     |                             | (value/body)     |
     |                             |                  |
     |                SIP Message                     |
     |<-----------------------------------------------|
     |                                                |
     |                                                |
     |                                                |

Problems:

* If there is no solution for the cases available where the SIP proxy
can add a PIDF-LO and a reference is not acceptable (because of the
latency) it is likely that boxes will just do it.=20
* Does not play "nicely" with SIP.=20
* Problems with E2E security
=20

6) Location by End Host / Per Referene / in Header

  UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C

     |  SIP Message                |          |       |       |
     |---------------------------->|          |       |       |
     |  Location (ref./header)     |                  |
     |                             |----------------->|
     |                             |                  |
     |              Dereference                       |
     =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
     |                                                |
     |                                                |
     |                SIP Message                     |
     |<-----------------------------------------------|
     |                                                |

Problems:

* Not sure about S/MIME usage
* Simlar considerations as in (2)

Finally, I think it would be useful to differentiate between location
based services in general and emergency services since the requirements
are, in most cases, different.=20

Ciao
Hannes

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From sip-bounces@ietf.org Tue Jul 18 10:59:31 2006
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Thanks, Hannes; these are helpful to focus the discussion.

I think, however, that the issue of security (encryption or integrity) 
and conveyance are essentially orthogonal. In other words, you can apply 
the same techniques, with similar problems, in all three conveyance 
mechanisms (by-value, by-attachment, by-reference). The procedural 
details differ (such as XML-DSIG and XML encryption vs. S/MIME vs. TLS), 
but the strength and the certificate problems are similar, albeit not 
identical. (For example, it is probably somewhat harder to deal with 
multi-recipient cases in S/MIME than in other mechanisms.)

It might help streamline the discussion if we can agree on this general 
statement or show that it is mistaken.

Henning

Tschofenig, Hannes wrote:
> Hi all, 
> 
> I have compiled a few message flows that would make it easier to discuss
> the different scenarios. I have listed some challenges along with each
> of the scenarios. 
> 
> 
> 1) Location by End Host / Per Value/ in Body
> 
>     UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> 
>        |  SIP Message                |          |       |       |
>        |---------------------------->|          |       |       |
>        |  Location (value/body)      |                  |
>        |                             |----------------->|
>        |                             |                  |
>        |                SIP Message                     |
>        |<-----------------------------------------------|
>        |                                                |
> 
> Problems: 
> * Likelyhood of the UA using S/MIME; nice on paper but does it reflect
> reality?
> * UA Alice needs to know that (a) intermediaries do not need to process
> the PIDF-LO 
> (b) recipient has a certificate
> * Unclear what happens if there are multiple recipients. 
> 
> 
> 
> 2) Location by End Host / Per Reference / in Body
> 
> 
>    UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> 
>       |  SIP Message                |          |       |       |
>       |---------------------------->|          |       |       |
>       |  Location (ref./body)       |                  |
>       |                             |----------------->|
>       |                             |                  |
>       |              Dereference                       |
>       |<==============================================>|
>       |                                                |
>       |                                                |
>       |                SIP Message                     |
>       |<-----------------------------------------------|
>       |                                                |
> 
> Problems: 
> * Can be protected using S/MIME (see issue above)
> * If not protected then the considerations of who is allowed to
> dereference to a PIDF-LO are applicable. There is certainly a difference
> between a reference pointing to a PIDF-LO at the UA itself and a
> reference obtained using one of the discussed Geopriv L7 protocols. 
> It might be difficult for the end host to make the PIDF-LO available. It
> might be difficult for him to have a server-side certificate but it
> might be easier to control access to it. 
> 
> 
> 3) Location by SIP Proxy / Per Reference / in Header
> 
>    UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> 
>       |  SIP Message                |          |       |       |
>       |---------------------------->|          |       |       |
>       |                             | SIP Message      |
>       |                             |----------------->|
>       |                             | Location         |
>       |                             | (ref./header)    |
>       |                             |                  |
>       |                             |                  |
>       |                             |   Dereference    |
>       |                             |<================>|
>       |                             |                  |
>       |                SIP Message                     |
>       |<-----------------------------------------------|
>       |                                                |
>       |                                                |
>       |                                                |
> 
> Problems:
> 
> * What should the SIP proxy but into the PIDF-LO when it is created
> (policies, identifiers, etc.)?
> * Authorization aspects: see above (Why cannot just everyone that sees
> the reference in the header passing by dereference it?)
> * Signing of the reference theoretically possible. SIP Proxy, however,
> needs to know the intended recipient. 
> 
> 
> 4) Location by SIP Proxy / Per Value / in Header
> 
>   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> 
>      |  SIP Message                |          |       |       |
>      |---------------------------->|          |       |       |
>      |                             | SIP Message      |
>      |                             |----------------->|
>      |                             | Location         |
>      |                             | (value/header)   |
>      |                             |                  |
>      |                SIP Message                     |
>      |<-----------------------------------------------|
>      |                                                |
>      |                                                |
>      |                                                |
> 
> Problems:
> 
> * What should the SIP proxy but into the PIDF-LO when it is created
> (policies, identifiers, etc.)?
> * Data URI approach possible but with increased message size. 
> * Signing of the PIDF-LO possible but approach similar. SIP Proxy,
> however, needs to know the intended recipient.
> 
> 
> 5) Location by SBC / Per Value / in Body
> 
>   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
>                                  (SBC)
>      |  SIP Message                |          |       |       |
>      |---------------------------->|          |       |       |
>      |  Location (value/body)      | SIP Message      |
>      |                             |----------------->|
>      |                             | Location         |
>      |                             | (value/body)     |
>      |                             |                  |
>      |                SIP Message                     |
>      |<-----------------------------------------------|
>      |                                                |
>      |                                                |
>      |                                                |
> 
> Problems:
> 
> * If there is no solution for the cases available where the SIP proxy
> can add a PIDF-LO and a reference is not acceptable (because of the
> latency) it is likely that boxes will just do it. 
> * Does not play "nicely" with SIP. 
> * Problems with E2E security
>  
> 
> 6) Location by End Host / Per Referene / in Header
> 
>   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> 
>      |  SIP Message                |          |       |       |
>      |---------------------------->|          |       |       |
>      |  Location (ref./header)     |                  |
>      |                             |----------------->|
>      |                             |                  |
>      |              Dereference                       |
>      |<==============================================>|
>      |                                                |
>      |                                                |
>      |                SIP Message                     |
>      |<-----------------------------------------------|
>      |                                                |
> 
> Problems:
> 
> * Not sure about S/MIME usage
> * Simlar considerations as in (2)
> 
> Finally, I think it would be useful to differentiate between location
> based services in general and emergency services since the requirements
> are, in most cases, different. 
> 
> Ciao
> Hannes
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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Thread-Topic: [Geopriv] Location Conveyance: Scenarios
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From: "Tschofenig, Hannes" <hannes.tschofenig@siemens.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>
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Subject: [Sip] AW: [Geopriv] Location Conveyance: Scenarios
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Hi Henning,


> Thanks, Hannes; these are helpful to focus the discussion.
>=20
> I think, however, that the issue of security (encryption or=20
> integrity)=20
> and conveyance are essentially orthogonal. In other words,=20
> you can apply=20
> the same techniques, with similar problems, in all three conveyance=20
> mechanisms (by-value, by-attachment, by-reference).

I fully agree with you.=20
I don't recall that the document ever discussed the possibility of
protecting a location-by-reference.=20

Hence, I was puzzled why the encryption is a big deal in one case and
not an issue at all in the second one.=20

> The procedural=20
> details differ (such as XML-DSIG and XML encryption vs.=20
> S/MIME vs. TLS),=20
> but the strength and the certificate problems are similar, albeit not=20
> identical. (For example, it is probably somewhat harder to deal with=20
> multi-recipient cases in S/MIME than in other mechanisms.)
I agree with you.=20


>=20
> It might help streamline the discussion if we can agree on=20
> this general=20
> statement or show that it is mistaken.
>=20


Ciao
Hannes

> Henning
>=20
> Tschofenig, Hannes wrote:
> > Hi all,=20
> >=20
> > I have compiled a few message flows that would make it=20
> easier to discuss
> > the different scenarios. I have listed some challenges=20
> along with each
> > of the scenarios.=20
> >=20
> >=20
> > 1) Location by End Host / Per Value/ in Body
> >=20
> >     UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> >=20
> >        |  SIP Message                |          |       |       |
> >        |---------------------------->|          |       |       |
> >        |  Location (value/body)      |                  |
> >        |                             |----------------->|
> >        |                             |                  |
> >        |                SIP Message                     |
> >        |<-----------------------------------------------|
> >        |                                                |
> >=20
> > Problems:=20
> > * Likelyhood of the UA using S/MIME; nice on paper but does=20
> it reflect
> > reality?
> > * UA Alice needs to know that (a) intermediaries do not=20
> need to process
> > the PIDF-LO=20
> > (b) recipient has a certificate
> > * Unclear what happens if there are multiple recipients.=20
> >=20
> >=20
> >=20
> > 2) Location by End Host / Per Reference / in Body
> >=20
> >=20
> >    UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> >=20
> >       |  SIP Message                |          |       |       |
> >       |---------------------------->|          |       |       |
> >       |  Location (ref./body)       |                  |
> >       |                             |----------------->|
> >       |                             |                  |
> >       |              Dereference                       |
> >       =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
> >       |                                                |
> >       |                                                |
> >       |                SIP Message                     |
> >       |<-----------------------------------------------|
> >       |                                                |
> >=20
> > Problems:=20
> > * Can be protected using S/MIME (see issue above)
> > * If not protected then the considerations of who is allowed to
> > dereference to a PIDF-LO are applicable. There is certainly=20
> a difference
> > between a reference pointing to a PIDF-LO at the UA itself and a
> > reference obtained using one of the discussed Geopriv L7 protocols.=20
> > It might be difficult for the end host to make the PIDF-LO=20
> available. It
> > might be difficult for him to have a server-side certificate but it
> > might be easier to control access to it.=20
> >=20
> >=20
> > 3) Location by SIP Proxy / Per Reference / in Header
> >=20
> >    UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> >=20
> >       |  SIP Message                |          |       |       |
> >       |---------------------------->|          |       |       |
> >       |                             | SIP Message      |
> >       |                             |----------------->|
> >       |                             | Location         |
> >       |                             | (ref./header)    |
> >       |                             |                  |
> >       |                             |                  |
> >       |                             |   Dereference    |
> >       |                             =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
> >       |                             |                  |
> >       |                SIP Message                     |
> >       |<-----------------------------------------------|
> >       |                                                |
> >       |                                                |
> >       |                                                |
> >=20
> > Problems:
> >=20
> > * What should the SIP proxy but into the PIDF-LO when it is created
> > (policies, identifiers, etc.)?
> > * Authorization aspects: see above (Why cannot just=20
> everyone that sees
> > the reference in the header passing by dereference it?)
> > * Signing of the reference theoretically possible. SIP=20
> Proxy, however,
> > needs to know the intended recipient.=20
> >=20
> >=20
> > 4) Location by SIP Proxy / Per Value / in Header
> >=20
> >   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> >=20
> >      |  SIP Message                |          |       |       |
> >      |---------------------------->|          |       |       |
> >      |                             | SIP Message      |
> >      |                             |----------------->|
> >      |                             | Location         |
> >      |                             | (value/header)   |
> >      |                             |                  |
> >      |                SIP Message                     |
> >      |<-----------------------------------------------|
> >      |                                                |
> >      |                                                |
> >      |                                                |
> >=20
> > Problems:
> >=20
> > * What should the SIP proxy but into the PIDF-LO when it is created
> > (policies, identifiers, etc.)?
> > * Data URI approach possible but with increased message size.=20
> > * Signing of the PIDF-LO possible but approach similar. SIP Proxy,
> > however, needs to know the intended recipient.
> >=20
> >=20
> > 5) Location by SBC / Per Value / in Body
> >=20
> >   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> >                                  (SBC)
> >      |  SIP Message                |          |       |       |
> >      |---------------------------->|          |       |       |
> >      |  Location (value/body)      | SIP Message      |
> >      |                             |----------------->|
> >      |                             | Location         |
> >      |                             | (value/body)     |
> >      |                             |                  |
> >      |                SIP Message                     |
> >      |<-----------------------------------------------|
> >      |                                                |
> >      |                                                |
> >      |                                                |
> >=20
> > Problems:
> >=20
> > * If there is no solution for the cases available where the=20
> SIP proxy
> > can add a PIDF-LO and a reference is not acceptable (because of the
> > latency) it is likely that boxes will just do it.=20
> > * Does not play "nicely" with SIP.=20
> > * Problems with E2E security
> > =20
> >=20
> > 6) Location by End Host / Per Referene / in Header
> >=20
> >   UA Alice                     SIP Proxy   UAS-A   UAS-B   UAS-C
> >=20
> >      |  SIP Message                |          |       |       |
> >      |---------------------------->|          |       |       |
> >      |  Location (ref./header)     |                  |
> >      |                             |----------------->|
> >      |                             |                  |
> >      |              Dereference                       |
> >      =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
> >      |                                                |
> >      |                                                |
> >      |                SIP Message                     |
> >      |<-----------------------------------------------|
> >      |                                                |
> >=20
> > Problems:
> >=20
> > * Not sure about S/MIME usage
> > * Simlar considerations as in (2)
> >=20
> > Finally, I think it would be useful to differentiate=20
> between location
> > based services in general and emergency services since the=20
> requirements
> > are, in most cases, different.=20
> >=20
> > Ciao
> > Hannes
> >=20
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20

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Aki Niemi wrote:
> On su, 2006-07-16 at 16:54 +0300, ext Sharon Laivand wrote:
>> Hi Aki and all...
>>
>>
>> This email follows my conversation with Gonzalo at last IETF 66th -
>> regarding RFC3329 issues.
>>
>> Here are some issues we thought of. I wonder if it can assist your
>> draft. Anyway I will be glad to constantly contribute to this draft.
> 
> Thanks for your feedback. Some comments inline.
> 
>> Here are the comments/additions to the issues:
>>
>> 1.	
>> RFC3329 describes how the same security-agreement is used in several
>> transactions, but does not specify the relations between those
>> transactions. For example, a security-agreement can be obtained via
>> OPTIONS transaction and then used by a call-leg. Since there is no
>> SIP-wise relation between those two SIP entities, a key for comparison
>> must be defined. This is extremely problematic on server side, because
>> any incoming transaction must be verified according to the
>> security-agreement used in previous (unrelated and even nonexistent)
>> transactions.
>> One possible solution that is not good for all cases: match of
>> security-agreements to SIP dialogs using peer transport address.
> 
> The primary reason this is underspecified is that back when RFC 3329 was
> in the works, we didn't have a clear architecture for this sort of
> thing. 
> 
> We do now, as outbound and SIP identity both rely on a UA having a
> "sentinel" of sorts in the network to whom it has a persistent security
> association and can send and receive traffic in a secure, identity-safe
> way. 
> 
> I would say the most common case is that this association is
> bootstrapped using a REGISTER, which was, in fact, the assumption of RFC
> 3329 as well. Perhaps not that clearly stated, though.

When using Digest, I think it is *only* in conjunction with REGISTER 
that you can bootstrap a security association. Furthermore, it has to be 
a first-party REGISTER, where To and From (and/or P-Asserted-Identity) 
are the same. The reason is because Digest authenticates the *source* of 
the request, not the destination. For some request where source and 
destination are different, you can't in general know whether the success 
of the request is dependent on the source being able to authenticate.

For instance, suppose there was a REGISTER where From is Alice, and To 
is Bob. Bob might send a 401 challenge, and Alice might successfully 
respond to it. But that doesn't mean that any entity responsible for 
Alice believes this is Alice, it only means that Bob believes this is 
Alice. It could be that 'Bob' is just a hack server in the network that 
believes any answer to its challenges.

The end result is that 3rd party registrations are not allowed in IMS, 
and because of the design of the system support for them can't readily 
be added.

	Paul

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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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James M. Polk wrote:
> Henning
> 
> Remind us what the unique use-case is for a data URL, where neither a 
> by-reference URI nor a by-value PIDF-LO can be used?

The use case is proxy-inserted location, where the proxy does not have 
access to a globally-accessible HTTP server where it can deposit this 
information. This could be a proxy behind a (corporate) firewall or NAT, 
for example.


> 
> In other words, why is this needed, and what would it be used for that 
> every element necessary couldn't use one of the two methods of Location 
> Conveyance already specified in the doc?

See above. The incremental complexity is essentially zero.

> 
> Complexity is an issue in location conveyance - especially during 
> emergency calling, but I've heard you state this isn't an issue with a 
> data URL.  Are you sure?  Are others sure?

I don't get your complexity point. The complexity of a data URL is 
likely to be significantly lower than

- setting up a custom web server outside the security or NAT perimeter
- that allows remote writes by the proxy server
- using a proxy-to-webserver protocol that isn't standardized
- using a webserver-to-proxy protocol (to get the object identifier 
back) that isn't standardized, either
- automatically cleaning up location state just at the right time (not 
too early to avoid resolution problems and not too late to avoid 
exposure and denial-of-service problems where the attacker uses bogus 
SIP INVITEs to create lots of location entries)
- and hoping that the server is always accessible, so that emergency 
calls don't fail

The data URL solution is self-contained; the reference solution is 
basically of the "and then magic happens" kind, where the proposed 
solution only provides a tiny part of what's needed to actually build a 
real system.

Henning

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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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Henning

Remind us what the unique use-case is for a data URL, where neither a 
by-reference URI nor a by-value PIDF-LO can be used?

In other words, why is this needed, and what would it be used for that 
every element necessary couldn't use one of the two methods of Location 
Conveyance already specified in the doc?

Complexity is an issue in location conveyance - especially during emergency 
calling, but I've heard you state this isn't an issue with a data URL.  Are 
you sure?  Are others sure?

At 09:57 AM 7/18/2006 -0400, Henning Schulzrinne wrote:
>Brian,
>
>a four-step process:
>
>- take any PIDF-LO
>- wrap in XML-DSIG as specified by the relevant W3C recommendation
>- convert to base-64
>- make into a data URL, as in the example I gave in an earlier message
>
>   data:application/pidf+xml;base64,<base64-data>
>
>Or did I miss the point of your question?
>
>See http://en.wikipedia.org/wiki/Data:_URL for additional data URL
>examples, btw.
>
>Henning
>
>On Jul 18, 2006, at 9:06 AM, Rosen, Brian wrote:
>
>>Perhaps some enlightenment is in order.  It's not clear to me how you
>>dsig a Data uri in a Location header (or maybe you mean dsig the
>>pidf-lo
>>and put both the pidf-lo and the signature in the Data URI) somehow?
>>Are there some examples of this kind of thing elsewhere?
>>
>>If we are ignoring technical arguments (as opposed to use cases), it's
>>out of forgetfulness, and not willfulness.  What were you referring
>>to?
>>
>>Brian
>
>
>_______________________________________________
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At 04:27 PM 7/18/2006 -0400, Henning Schulzrinne wrote:
>James M. Polk wrote:
>>Henning
>>Remind us what the unique use-case is for a data URL, where neither a 
>>by-reference URI nor a by-value PIDF-LO can be used?
>
>The use case is proxy-inserted location, where the proxy does not have 
>access to a globally-accessible HTTP server where it can deposit this 
>information. This could be a proxy behind a (corporate) firewall or NAT, 
>for example.

ok, but how would a UA know it's behind a corporate NAT/FW, instead of at a 
home using the SP's LIS beyond the NAT/FW, to know to use a data URL 
instead of by-value or by-reference PIDF-LO?

>>In other words, why is this needed, and what would it be used for that 
>>every element necessary couldn't use one of the two methods of Location 
>>Conveyance already specified in the doc?
>
>See above. The incremental complexity is essentially zero.
>
>>Complexity is an issue in location conveyance - especially during 
>>emergency calling, but I've heard you state this isn't an issue with a 
>>data URL.  Are you sure?  Are others sure?
>
>I don't get your complexity point.

We're talking about all SIP elements that understand this extension now, if 
we do as you and Hannes are suggesting, include a third 
representation.  That's more code for every UA to include, and every 
location aware SIP server.

I hear your arguments below essentialling meaning location by-reference is 
a woefully underspecified user-case, and that it is the one that ought to 
be the one that may be dropped from the list within this extension proposal 
(because there is not enough there).  Are you going down that path here?

>The complexity of a data URL is likely to be significantly lower than
>
>- setting up a custom web server outside the security or NAT perimeter
>- that allows remote writes by the proxy server
>- using a proxy-to-webserver protocol that isn't standardized
>- using a webserver-to-proxy protocol (to get the object identifier back) 
>that isn't standardized, either
>- automatically cleaning up location state just at the right time (not too 
>early to avoid resolution problems and not too late to avoid exposure and 
>denial-of-service problems where the attacker uses bogus SIP INVITEs to 
>create lots of location entries)
>- and hoping that the server is always accessible, so that emergency calls 
>don't fail
>
>The data URL solution is self-contained; the reference solution is 
>basically of the "and then magic happens" kind, where the proposed 
>solution only provides a tiny part of what's needed to actually build a 
>real system.

should by-reference be taken out as an incomplete piece to an 
underspecified solution, IYO?


>Henning

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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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>> The use case is proxy-inserted location, where the proxy does not have 
                    ^^^^^^^^^^^^^^

I'm assuming that the proxy knows whether it has access to a web server 
or not. The UA really wouldn't know or care in that case.

> ok, but how would a UA know it's behind a corporate NAT/FW, instead of 
> at a home using the SP's LIS beyond the NAT/FW, to know to use a data 
> URL instead of by-value or by-reference PIDF-LO?

I would expect a UA to always use by-attachment unless it used an L7 
location mechanism that provided a reference to an object on a LIS 
somewhere. (This, by the way, raises some additional issues; how can the 
  UA know that the pointer it got from the LIS is accessible, by policy, 
to a random third party?)


> We're talking about all SIP elements that understand this extension now, 
> if we do as you and Hannes are suggesting, include a third 
> representation.  That's more code for every UA to include, and every 
> location aware SIP server.

I generally share concerns about code complexity, but we're talking 
pretty trivial code here. In a PHP-like language, it would be something like

if (url['scheme'] == 'http') {
   $fd = fopen($url);
   $lo = fread($fd);
} elseif (url['scheme'] == 'data') {
   $u = split(',', $url);
   $lo = base64_decode($u[1]);
}

Only the last elseif is data-specific. In other words, not exactly a big 
deal.

> 
> I hear your arguments below essentialling meaning location by-reference 
> is a woefully underspecified user-case, and that it is the one that 
> ought to be the one that may be dropped from the list within this 
> extension proposal (because there is not enough there).  Are you going 
> down that path here?

There may well be deployments where by-reference is valuable (such as 
where the LIS already provides by-reference location information to the 
proxy), but I fear that if we only do S/MIME attachments, the practical 
outcome is going to be lots of SBCs modifying SIP bodies.


> should by-reference be taken out as an incomplete piece to an 
> underspecified solution, IYO?

I believe it is currently underspecified; I won't go as far as 
suggesting it be taken out as noted above.


> 
> 
>> Henning

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1317283984==--



From sip-bounces@ietf.org Tue Jul 18 19:16:20 2006
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============2087304446==--



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Hi,

If registrar receives a REGISTER request with two Contact headers, and 
registrar wants to register only one binding,
how it can indicate this in the response message (should it omit the 
Contact header not supported in 200 ok response)

One scenario happening is UAC sending two Contact headers with same IP and 
port but protocol as TCP and UDP, but server does not support TCP, so it 
want to only create binding for UDP Contact header, so it wants to 
indicate to UAC that it has created only one binding.

Thanks,
Udit

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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">If registrar receives a REGISTER request
with two Contact headers, and registrar wants to register only one binding,</font>
<br><font size=2 face="sans-serif">how it can indicate this in the response
message (should it omit the Contact header not supported in 200 ok response)</font>
<br>
<br><font size=2 face="sans-serif">One scenario happening is UAC sending
two Contact headers with same IP and port but protocol as TCP and UDP,
but server does not support TCP, so it want to only create binding for
UDP Contact header, so it wants to indicate to UAC that it has created
only one binding.</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br><font size=2 face="sans-serif">Udit</font>
<br>
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Use sipping@ietf.org for new developments on the application of sip
--===============1912661103==--




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Subject: Re: [Sip] How to reject a binding for REGISTER
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Hi
I didnt find this mentioned in RFC 3261 and so this means that its upto 
the particular Registrar implementation.
But I think it should send 200 ok with the current bindings (ie omitting 
the un-acceptable contact). 200 ok is the only appropriate response here, 
because REGISTER request is valid and atleast one contact is accepted by 
the Registrar.

Regards
Retesh




Udit_Goyal@3com.com 
07/19/2006 10:38 AM


To
sip@ietf.org
cc
sip-implementors@cs.columbia.edu
Subject
[Sip] How to reject a binding for REGISTER







Hi, 

If registrar receives a REGISTER request with two Contact headers, and 
registrar wants to register only one binding, 
how it can indicate this in the response message (should it omit the 
Contact header not supported in 200 ok response) 

One scenario happening is UAC sending two Contact headers with same IP and 
port but protocol as TCP and UDP, but server does not support TCP, so it 
want to only create binding for UDP Contact header, so it wants to 
indicate to UAC that it has created only one binding. 

Thanks, 
Udit _______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


***********************  FSS-Unclassified   ***********************
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<br><font size=2 face="sans-serif">Hi</font>
<br><font size=2 face="sans-serif">I didnt find this mentioned in RFC 3261
and so this means that its upto the particular Registrar implementation.</font>
<br><font size=2 face="sans-serif">But I think it should send 200 ok with
the current bindings (ie omitting the un-acceptable contact). 200 ok is
the only appropriate response here, because REGISTER request is valid and
atleast one contact is accepted by the Registrar.</font>
<br>
<br><font size=2 face="sans-serif">Regards</font>
<br><font size=2 face="sans-serif">Retesh</font>
<br>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td width=40%><font size=1 face="sans-serif"><b>Udit_Goyal@3com.com</b>
</font>
<p><font size=1 face="sans-serif">07/19/2006 10:38 AM</font>
<br>
<td width=59%>
<table width=100%>
<tr>
<td>
<div align=right><font size=1 face="sans-serif">To</font></div>
<td valign=top><font size=1 face="sans-serif">sip@ietf.org</font>
<tr>
<td>
<div align=right><font size=1 face="sans-serif">cc</font></div>
<td valign=top><font size=1 face="sans-serif">sip-implementors@cs.columbia.edu</font>
<tr>
<td>
<div align=right><font size=1 face="sans-serif">Subject</font></div>
<td valign=top><font size=1 face="sans-serif">[Sip] How to reject a binding
for REGISTER</font></table>
<br>
<table>
<tr valign=top>
<td>
<td></table>
<br></table>
<br>
<br>
<br><font size=2 face="sans-serif"><br>
Hi,</font><font size=3> <br>
</font><font size=2 face="sans-serif"><br>
If registrar receives a REGISTER request with two Contact headers, and
registrar wants to register only one binding,</font><font size=3> </font><font size=2 face="sans-serif"><br>
how it can indicate this in the response message (should it omit the Contact
header not supported in 200 ok response)</font><font size=3> <br>
</font><font size=2 face="sans-serif"><br>
One scenario happening is UAC sending two Contact headers with same IP
and port but protocol as TCP and UDP, but server does not support TCP,
so it want to only create binding for UDP Contact header, so it wants to
indicate to UAC that it has created only one binding.</font><font size=3>
<br>
</font><font size=2 face="sans-serif"><br>
Thanks,</font><font size=3> </font><font size=2 face="sans-serif"><br>
Udit</font><font size=3> </font><font size=2><tt>_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip</tt></font>
<br><font size=2 face="sans-serif"><br>
<br>
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From sip-bounces@ietf.org Wed Jul 19 08:38:04 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Thomson, Martin'" <Martin.Thomson@andrew.com>, "Rosen, Brian"
	<Brian.Rosen@neustar.biz>, Henning Schulzrinne <hgs@cs.columbia.edu>
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Cc: sip@ietf.org, geopriv@ietf.org
Subject: [Sip] RE: [Geopriv] Location-by-value in a SIP Location Header
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(As SIP WG chair)

Just to set some limits to this discussion, let me state where we are in the progression of this work item. Remember it is now being classed as urgent and we need to get an RFC out of the door, and the SIP chairs are working with the GEOPRIV chairs as to the quickest way to do this:

1)	The requirements in the document have been approved by the SIPPING group (a long time ago) and therefore we should not be inventing new requirements without a clear use case that has clear agreement.

2)	The SIP group have already identified that two mechanisms to support these requirements should be documented (e.g. the location by value in a body, and the location by reference in a header. These two mechanisms appear to support all the current requirements. In other words, SIP have agreed the technical solution, and we are in the stage of trying to document it.

3)	Using a data URL appears to add a third mechanism, if only because it will at least need to have a different set of security considerations documented for it. 

We should only be adding a third mechanism if:

a)	the existing mechanism can be shown not to meet all the requirements; or
b)	the use case where it offers advantages is clearly identified; and
c)	the compatibility issues are addressed (i.e. does the UAS need to support all of what is now three mechanisms or is there some other support requirement); and
d)	this can be shown as impossible to be discussed as a future extension.

Could I also suggest (in addition to the above) that if people want to proceed with this discussion, it would be appropriate to submit an i-d targeted at the SIP WG identifying the technical solution proposed (with as much completeness as is possible). And remember if you think this discussion is needed, it needs to occur quickly. 

regards

Keith

> -----Original Message-----
> From: Thomson, Martin [mailto:Martin.Thomson@andrew.com]
> Sent: 19 July 2006 00:15
> To: Rosen, Brian; Henning Schulzrinne
> Cc: sip@ietf.org; geopriv@ietf.org
> Subject: RE: [Geopriv] Location-by-value in a SIP Location Header
> 
> 
> If you want technically feasible, I can post a URI to the 
> list that includes a complete PIDF-LO with DSig.  It's a 
> little long though.
> 
> I'm not suggesting that this a good idea, but I see no reason 
> to prevent it.
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Tuesday, 18 July 2006 11:07 PM
> > To: Henning Schulzrinne
> > Cc: sip@ietf.org; geopriv@ietf.org
> > Subject: RE: [Geopriv] Location-by-value in a SIP Location Header
> > 
> > Perhaps some enlightenment is in order.  It's not clear to 
> me how you
> > dsig a Data uri in a Location header (or maybe you mean 
> dsig the pidf-lo
> > and put both the pidf-lo and the signature in the Data URI) somehow?
> > Are there some examples of this kind of thing elsewhere?
> > 
> > If we are ignoring technical arguments (as opposed to use 
> cases), it's
> > out of forgetfulness, and not willfulness.  What were you 
> referring to?
> > 
> > Brian
> > 
> > 
> > 
> > -----Original Message-----
> > From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> > Sent: Tuesday, July 18, 2006 8:59 AM
> > To: Rosen, Brian
> > Cc: sip@ietf.org; geopriv@ietf.org
> > Subject: Re: [Geopriv] Location-by-value in a SIP Location Header
> > 
> > Brian, James:
> > 
> > this is simply false. If desired, XML-DSIG is a well-known and
> > accepted mechanism to achieve integrity. Such specification is no
> > longer than the MUST NOT that you propose. You are also willfully
> > ignoring the other technical arguments made as part of this 
> discussion.
> > 
> > Henning
> > 
> > On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:
> > 
> > > There was a desire expressed by Hannes to provide a way 
> for a proxy to
> > > insert location-by-value.  He had proposed a SIP Location header
> > > specific way.  In IETF66, Henning suggested just using a data URI.
> > >
> > > Doing this may run afoul of the geopriv location privacy concerns,
> > > because there is no way for the user to sign the header to provide
> > > assurance that the PIDF, and specifically the retention and other
> > > policy
> > > bits are preserved.  Note that S/MIME provides sufficient 
> integrity
> > > protection for a body.  It's probably okay to use TLS per hop to
> > > provide
> > > privacy, but not integrity protection.  The authors 
> believe that there
> > > is no really compelling use case for this feature, and 
> the effort to
> > > define acceptable security mechanisms for location data 
> in a header
> > > would be a significant effort, further delaying the draft.
> > >
> > > We propose, therefore, to state in sip-location-conveyance, that a
> > > Data
> > > URI MUST NOT be used in the Location header until a 
> standards track
> > > RFC
> > > defines a suitable security mechanism to protect the PIDF in the
> > > header.
> > >
> > > Brian and James
> > >
> > > _______________________________________________
> > > Geopriv mailing list
> > > Geopriv@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/geopriv
> > 
> > 
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> 
> --------------------------------------------------------------
> ----------------------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.  
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
> --------------------------------------------------------------
> ----------------------------------
> [mf2]
> 

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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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Hi Keith,

Drage, Keith (Keith) wrote:
> (As SIP WG chair)
> 
> Just to set some limits to this discussion, let me state where we are
> in the progression of this work item. Remember it is now being
> classed as urgent and we need to get an RFC out of the door, and the
> SIP chairs are working with the GEOPRIV chairs as to the quickest way
> to do this:
> 
> 1)	The requirements in the document have been approved by the SIPPING
> group (a long time ago) and therefore we should not be inventing new
> requirements without a clear use case that has clear agreement.

At that time when the requirements were written we weren't even close to 
understand how the VoIP emergency service solutions would look like.

> 
> 2)	The SIP group have already identified that two mechanisms to
> support these requirements should be documented (e.g. the location by
> value in a body, and the location by reference in a header. These two
> mechanisms appear to support all the current requirements. In other
> words, SIP have agreed the technical solution, and we are in the
> stage of trying to document it.

Actually, as James pointed out the document supports implicitly three 
mechanisms:
* Location by Value in the Body (by end host)
* Location by Reference in the Header (by end host and by the proxy)
* Location by Reference in the Body (by end host)

In my scenario mail I wanted to point out that it is necessary to 
describe these 3 mechanisms and to discuss the difference between an end 
host and a proxy doing the processing.

> 
> 3)	Using a data URL appears to add a third mechanism, if only because
> it will at least need to have a different set of security
> considerations documented for it.

As I argued in my long list of scenarios I don't believe that the 
security properties are different compared to the approach where 
Location by Reference (with the reference being added by the proxy) is used.

> We should only be adding a third mechanism if:
> 
> a)	the existing mechanism can be shown not to meet all the
> requirements; or b)	the use case where it offers advantages is
> clearly identified; and c)	the compatibility issues are addressed
> (i.e. does the UAS need to support all of what is now three
> mechanisms or is there some other support requirement); and d)	this
> can be shown as impossible to be discussed as a future extension.

Within the emergency work we got the feedback that a number of folks are 
interested in attaching location information per value by a SIP proxy.
I don't want them to do it without having us to look at the implications.

The 3GPP architecture wasn't quite detailed enough on this subject to 
judge whether they are one of the potential consumers. Are they? You 
know it better than I do. We still need to have a discussion on some of 
these issues in ECRIT. Ted provided excellent meeting minutes.


> 
> Could I also suggest (in addition to the above) that if people want
> to proceed with this discussion, it would be appropriate to submit an
> i-d targeted at the SIP WG identifying the technical solution
> proposed (with as much completeness as is possible). And remember if
> you think this discussion is needed, it needs to occur quickly.

Sure, we can do this.

Still, I believe it also helps the solutions offered in the current 
document if we discuss the security aspects. I am interested to have a 
good location conveyance document that covers a discussion about these 
issues.

In GEOPRIV we had a long discussion about the security aspects of 
retrieving location information from the access network using a layer 7 
protocol. If we ignore the details then there is a lot of similarity.

> 
> regards

Ciao
Hannes

> 
> Keith
> 
> 
>> -----Original Message----- From: Thomson, Martin
>> [mailto:Martin.Thomson@andrew.com] Sent: 19 July 2006 00:15 To:
>> Rosen, Brian; Henning Schulzrinne Cc: sip@ietf.org;
>> geopriv@ietf.org Subject: RE: [Geopriv] Location-by-value in a SIP
>> Location Header
>> 
>> 
>> If you want technically feasible, I can post a URI to the list that
>> includes a complete PIDF-LO with DSig.  It's a little long though.
>> 
>> I'm not suggesting that this a good idea, but I see no reason to
>> prevent it.
>> 
>> 
>>> -----Original Message----- From: Rosen, Brian
>>> [mailto:Brian.Rosen@neustar.biz] Sent: Tuesday, 18 July 2006
>>> 11:07 PM To: Henning Schulzrinne Cc: sip@ietf.org;
>>> geopriv@ietf.org Subject: RE: [Geopriv] Location-by-value in a
>>> SIP Location Header
>>> 
>>> Perhaps some enlightenment is in order.  It's not clear to
>> 
>> me how you
>> 
>>> dsig a Data uri in a Location header (or maybe you mean
>> 
>> dsig the pidf-lo
>> 
>>> and put both the pidf-lo and the signature in the Data URI)
>>> somehow? Are there some examples of this kind of thing elsewhere?
>>> 
>>> 
>>> If we are ignoring technical arguments (as opposed to use
>> 
>> cases), it's
>> 
>>> out of forgetfulness, and not willfulness.  What were you
>> 
>> referring to?
>> 
>>> Brian
>>> 
>>> 
>>> 
>>> -----Original Message----- From: Henning Schulzrinne
>>> [mailto:hgs@cs.columbia.edu] Sent: Tuesday, July 18, 2006 8:59 AM
>>>  To: Rosen, Brian Cc: sip@ietf.org; geopriv@ietf.org Subject: Re:
>>> [Geopriv] Location-by-value in a SIP Location Header
>>> 
>>> Brian, James:
>>> 
>>> this is simply false. If desired, XML-DSIG is a well-known and 
>>> accepted mechanism to achieve integrity. Such specification is no
>>>  longer than the MUST NOT that you propose. You are also
>>> willfully ignoring the other technical arguments made as part of
>>> this
>> 
>> discussion.
>> 
>>> Henning
>>> 
>>> On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:
>>> 
>>> 
>>>> There was a desire expressed by Hannes to provide a way
>> 
>> for a proxy to
>> 
>>>> insert location-by-value.  He had proposed a SIP Location
>>>> header specific way.  In IETF66, Henning suggested just using a
>>>> data URI.
>>>> 
>>>> Doing this may run afoul of the geopriv location privacy
>>>> concerns, because there is no way for the user to sign the
>>>> header to provide assurance that the PIDF, and specifically the
>>>> retention and other policy bits are preserved.  Note that
>>>> S/MIME provides sufficient
>> 
>> integrity
>> 
>>>> protection for a body.  It's probably okay to use TLS per hop
>>>> to provide privacy, but not integrity protection.  The authors
>>>> 
>> 
>> believe that there
>> 
>>>> is no really compelling use case for this feature, and
>> 
>> the effort to
>> 
>>>> define acceptable security mechanisms for location data
>> 
>> in a header
>> 
>>>> would be a significant effort, further delaying the draft.
>>>> 
>>>> We propose, therefore, to state in sip-location-conveyance,
>>>> that a Data URI MUST NOT be used in the Location header until a
>>>> 
>> 
>> standards track
>> 
>>>> RFC defines a suitable security mechanism to protect the PIDF
>>>> in the header.
>>>> 
>>>> Brian and James
>>>> 
>>>> _______________________________________________ Geopriv mailing
>>>> list Geopriv@ietf.org 
>>>> https://www1.ietf.org/mailman/listinfo/geopriv
>>> 
>>> 
>>> _______________________________________________ Geopriv mailing
>>> list Geopriv@ietf.org 
>>> https://www1.ietf.org/mailman/listinfo/geopriv
>> 
>> -------------------------------------------------------------- 
>> ---------------------------------- This message is for the
>> designated recipient only and may contain privileged, proprietary,
>> or otherwise private information. If you have received it in error,
>> please notify the sender immediately and delete the original.  Any
>> unauthorized use of this email is prohibited. 
>> -------------------------------------------------------------- 
>> ---------------------------------- [mf2]
>> 
> 
> 
> _______________________________________________ Geopriv mailing list 
> Geopriv@ietf.org https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From sip-bounces@ietf.org Wed Jul 19 09:33:21 2006
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Subject: [Sip] Re: [Simple] UDP - what if response is bigger than 1500 bytes
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Silvestr.Peknik@tietoenator.com wrote:
> Hello, 
>
> Could someone help me with following problem? 
>
> I am not sure what should be done when the response to SIP request sent
> over UDP is bigger than 1500 bytes? It can happen e.g. when PUBBLISH
> request is responded by 415 Unsupported Media Type and Accept,
> Accept-Encoding and Accept-Language headers are added.
>   

This is a good question: I think it was mentionned in last SIP WG IETF 
meeting that having responses greater than path MTU  (-200 bytes)
in SIP was not currently well specified. It was an open issue on 
draft-ietf-sip-hop-limit-diagnostics-03.txt which raises this problem
since it leads naturally to send big SIP responses.
SIP WG people may probably answer you if something is planned to handle 
this problem...
Thomas
> Thank you, 
> Silvestr Peknik
>
> Extract from RFC3261:
> 18.1.1 Sending requests:
> ...
> 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. If this causes a change in the transport protocol from the
>    one indicated in the top Via, the value in the top Via MUST be
>    changed.  This prevents fragmentation of messages over UDP and
>    provides congestion control for larger messages.  However,
>    implementations MUST be able to handle messages up to the maximum
>    datagram packet size.  For UDP, this size is 65,535 bytes, including
>    IP and UDP headers.
>
>       The 200 byte "buffer" between the message size and the MTU
>       accommodates the fact that the response in SIP can be larger than
>       the request.  This happens due to the addition of Record-Route
>       header field values to the responses to INVITE, for example.  With
>       the extra buffer, the response can be about 170 bytes larger than
>       the request, and still not be fragmented on IPv4 (about 30 bytes
>       is consumed by IP/UDP, assuming no IPSec).  1300 is chosen when
>       path MTU is not known, based on the assumption of a 1500 byte
>       Ethernet MTU.
> ... 
>
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>
>   


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From sip-bounces@ietf.org Wed Jul 19 09:57:00 2006
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To: Retesh Chadha <retesh.chadha@flextronicssoftware.com>
Subject: Re: [Sip] How to reject a binding for REGISTER
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I think the Registrar should just accept the binding.   Registars and 
proxies are not always the same application.   The decision what  
transport to be used to contact the registered user is between the Proxy 
and the caller, not between the Registrar and the registered user.


Retesh Chadha wrote:
>
> Hi
> I didnt find this mentioned in RFC 3261 and so this means that its 
> upto the particular Registrar implementation.
> But I think it should send 200 ok with the current bindings (ie 
> omitting the un-acceptable contact). 200 ok is the only appropriate 
> response here, because REGISTER request is valid and atleast one 
> contact is accepted by the Registrar.
>
> Regards
> Retesh
>
>
>
> *Udit_Goyal@3com.com*
>
> 07/19/2006 10:38 AM
>
> 	
> To
> 	sip@ietf.org
> cc
> 	sip-implementors@cs.columbia.edu
> Subject
> 	[Sip] How to reject a binding for REGISTER
>
>
>
> 	
>
>
>
>
>
>
> Hi,
>
> If registrar receives a REGISTER request with two Contact headers, and 
> registrar wants to register only one binding,
> how it can indicate this in the response message (should it omit the 
> Contact header not supported in 200 ok response)
>
> One scenario happening is UAC sending two Contact headers with same IP 
> and port but protocol as TCP and UDP, but server does not support TCP, 
> so it want to only create binding for UDP Contact header, so it wants 
> to indicate to UAC that it has created only one binding.
>
> Thanks,
> Udit _______________________________________________
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>
>
> ***********************  FSS-Unclassified   ***********************
> ------------------------------------------------------------------------
>
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From sip-bounces@ietf.org Wed Jul 19 11:39:30 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Fatih \"Ey|p\" NAR'" <fenar@yahoo.com>, "Drage, Keith (Keith)"
	<drage@lucent.com>, "\"'Antsnio\" Manuel Amaral'"
	<antonio-mn-amaral@ptinovacao.pt>, "'Sip@ietf.org'" <Sip@ietf.org>
Subject: RE: [Sip] S-CSCF expected behavior
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Unless I have misread your question, then RFC 4028, as previously =
indicated, is your only mechanism on active calls.

regards

Keith

> -----Original Message-----
> From: Fatih "Ey|p" NAR [mailto:fenar@yahoo.com]
> Sent: 17 July 2006 16:20
> To: Drage, Keith (Keith); "'Antsnio" Manuel Amaral'; 'Sip@ietf.org'
> Subject: RE: [Sip] S-CSCF expected behavior
>=20
>=20
> what about active sessions (INVITE transaction
> completed successfully with 200OK-ACK previously) on
> S-CSCF?
>=20
> ex: For the prepaid customer type once the connection
> dropped between AS and S-CSCF only way to prevent -
> "minus" credit is to end the session with BYE by S-.
>=20
>=20
>=20
> --- "Drage, Keith (Keith)" <drage@lucent.com> wrote:
>=20
> > To make this directly relevant to SIP, the IMS
> > S-CSCF behaves exactly as specified in RFC 3261. If
> > it has a pending transaction, then section 16.8 and
> > section 16.9 of RFC 3261 apply.=20
> >=20
> > The session timer (RFC 4028) may optionally be used
> > by any of the IMS entities that behave as a SIP
> > proxy.
> >=20
> > regards
> >=20
> > Keith
> >=20
> > > -----Original Message-----
> > > From: Ant=F3nio Manuel Amaral
> > [mailto:antonio-mn-amaral@ptinovacao.pt]
> > > Sent: 14 July 2006 11:17
> > > To: Sip@ietf.org
> > > Subject: [Sip] S-CSCF expected behavior
> > >=20
> > >=20
> > >=20
> > > Dear All,
> > >=20
> > > I don't know if this is the correct mailing list
> > to put my=20
> > > question and if not my apologies for that.
> > >=20
> > > Consider the following IMS scenario
> > >=20
> > >=20
> > > (IMS core Network) -> S-CSCF -> AS
> > >=20
> > >=20
> > > If the connection between AS and S-CSCF fails what
> > is the=20
> > > expected behavior of S-CSCF? It must release all
> > active calls=20
> > > sending BYE messages to caller and called clients
> > in order to=20
> > > terminate the calls?
> > >=20
> > > Thanks in advance
> > >=20
> > > Regards
> > >=20
> > > AA=20
> > >=20
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list=20
> > https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP
> > Protocol
> > > Use sip-implementors@cs.columbia.edu for questions
> > on current sip
> > > Use sipping@ietf.org for new developments on the
> > application of sip
> > >=20
> >=20
> > _______________________________________________
> > Sip mailing list=20
> > https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP
> > Protocol
> > Use sip-implementors@cs.columbia.edu for questions
> > on current sip
> > Use sipping@ietf.org for new developments on the
> > application of sip
> >=20
>=20
>=20
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around=20
> http://mail.yahoo.com=20
>=20

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Brian Rosen wrote:
> In sip-location-conveyance, we allow a URI in the Location header to provide
> for location-by-reference.  We must have some kind of text that says what
> the recipient does with this URI.
> 
> Our proposal is to change the ABNF, which currently allows this URI to be
> AbsoluteURI (meaning any scheme) and replace it with HTTP/HTTPS only.  Of
> course, there would still be the cid: for message body identification.  We
> would then define a dereference process as doing an HTTP/HTTPS GET, with no
> parameters, on the URI, and expect to receive in return a PIDF-LO, exactly
> as it would appear in a body.  The text would state MUST implement TLS,
> except for the emergency case.  The text would further state that additional
> dereferencing procedures could be defined by a standards track RFC.

I'm not sure it is as simple as this.

What happens when a client gets a subscribable location reference URI 
via the L7-LCP, and puts that into a Location header in an INVITE?  How 
is the subscription done with HTTP?

-andy


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From sip-bounces@ietf.org Wed Jul 19 11:53:03 2006
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Thread-Topic: [Geopriv] Location-by-value in a SIP Location Header
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>
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Perhaps some enlightenment is in order.  It's not clear to me how you
dsig a Data uri in a Location header (or maybe you mean dsig the pidf-lo
and put both the pidf-lo and the signature in the Data URI) somehow?
Are there some examples of this kind of thing elsewhere?

If we are ignoring technical arguments (as opposed to use cases), it's
out of forgetfulness, and not willfulness.  What were you referring to?

Brian



-----Original Message-----
From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]=20
Sent: Tuesday, July 18, 2006 8:59 AM
To: Rosen, Brian
Cc: sip@ietf.org; geopriv@ietf.org
Subject: Re: [Geopriv] Location-by-value in a SIP Location Header

Brian, James:

this is simply false. If desired, XML-DSIG is a well-known and =20
accepted mechanism to achieve integrity. Such specification is no =20
longer than the MUST NOT that you propose. You are also willfully =20
ignoring the other technical arguments made as part of this discussion.

Henning

On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:

> There was a desire expressed by Hannes to provide a way for a proxy to
> insert location-by-value.  He had proposed a SIP Location header
> specific way.  In IETF66, Henning suggested just using a data URI.
>
> Doing this may run afoul of the geopriv location privacy concerns,
> because there is no way for the user to sign the header to provide
> assurance that the PIDF, and specifically the retention and other =20
> policy
> bits are preserved.  Note that S/MIME provides sufficient integrity
> protection for a body.  It's probably okay to use TLS per hop to =20
> provide
> privacy, but not integrity protection.  The authors believe that there
> is no really compelling use case for this feature, and the effort to
> define acceptable security mechanisms for location data in a header
> would be a significant effort, further delaying the draft.
>
> We propose, therefore, to state in sip-location-conveyance, that a =20
> Data
> URI MUST NOT be used in the Location header until a standards track =20
> RFC
> defines a suitable security mechanism to protect the PIDF in the =20
> header.
>
> Brian and James
>
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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From sip-bounces@ietf.org Wed Jul 19 11:53:31 2006
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From: "Gaurav Kheterpal" <gkheterpal@ismartpanache.com>
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Subject: [Sip] RE: [Sip-implementors] How to reject a binding for REGISTER
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Refer to Section 10.2.1 in RFC 3261:

"The 2xx response to the REGISTER request will contain, in a Contact header
field, a complete list of bindings that have been registered for this
address-of-record at this registrar."

The registrar will only add those URIs in the Contact headers for 200 OK for
which it has been successfully been able to create a AOR binding.

As for the differentiation between TCP & UDP, I think it can be done in a
way similar to the original INVITE request:

Contact:..@<...>;transport=udp

If the registrar does not support TCP, it will not add the corresponding
CONTACT header in 200 OK.

A similar thread for INVITE might help: 
https://lists.cs.columbia.edu/pipermail/sip-implementors/2004-July/006694.ht
ml

I hope this helps.

Regards,
Gaurav

-----Original Message-----
From: sip-implementors-bounces@cs.columbia.edu
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of
Udit_Goyal@3com.com
Sent: Wednesday, July 19, 2006 10:38 AM
To: sip@ietf.org
Cc: sip-implementors@cs.columbia.edu
Subject: [Sip-implementors] How to reject a binding for REGISTER

Hi,

If registrar receives a REGISTER request with two Contact headers, and 
registrar wants to register only one binding,
how it can indicate this in the response message (should it omit the 
Contact header not supported in 200 ok response)

One scenario happening is UAC sending two Contact headers with same IP and 
port but protocol as TCP and UDP, but server does not support TCP, so it 
want to only create binding for UDP Contact header, so it wants to 
indicate to UAC that it has created only one binding.

Thanks,
Udit
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From: "Marc Linsner" <mlinsner@cisco.com>
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Subject: [Sip] RE: [Geopriv] Location-by-value in a SIP Location Header
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Keith,

A couple of issues with your statements below:

- As you point out in #2, the SIP group has identified two mechanisms,
location by value in the body and location by reference in the header.  The
issue stems from the fact that location by reference is yet to be a
mechanism.  This has been mired down with technical discussions in GeoPriv
for a long time.

- Your list of reasons needs to be expanded to include 'the identified
mechanism has no technical solution'.

As you express, there are many reasons to complete this work quickly, but to
have a total solution it is important to examine these alternatives
mechanisms.  Like you, I certainly hope this comes to a resolution very
soon.

Thanks,

-Marc-



> 
> (As SIP WG chair)
> 
> Just to set some limits to this discussion, let me state 
> where we are in the progression of this work item. Remember 
> it is now being classed as urgent and we need to get an RFC 
> out of the door, and the SIP chairs are working with the 
> GEOPRIV chairs as to the quickest way to do this:
> 
> 1)	The requirements in the document have been approved by 
> the SIPPING group (a long time ago) and therefore we should 
> not be inventing new requirements without a clear use case 
> that has clear agreement.
> 
> 2)	The SIP group have already identified that two 
> mechanisms to support these requirements should be documented 
> (e.g. the location by value in a body, and the location by 
> reference in a header. These two mechanisms appear to support 
> all the current requirements. In other words, SIP have agreed 
> the technical solution, and we are in the stage of trying to 
> document it.
> 
> 3)	Using a data URL appears to add a third mechanism, if 
> only because it will at least need to have a different set of 
> security considerations documented for it. 
> 
> We should only be adding a third mechanism if:
> 
> a)	the existing mechanism can be shown not to meet all the 
> requirements; or
> b)	the use case where it offers advantages is clearly 
> identified; and
> c)	the compatibility issues are addressed (i.e. does the 
> UAS need to support all of what is now three mechanisms or is 
> there some other support requirement); and
> d)	this can be shown as impossible to be discussed as a 
> future extension.
> 
> Could I also suggest (in addition to the above) that if 
> people want to proceed with this discussion, it would be 
> appropriate to submit an i-d targeted at the SIP WG 
> identifying the technical solution proposed (with as much 
> completeness as is possible). And remember if you think this 
> discussion is needed, it needs to occur quickly. 
> 
> regards
> 
> Keith
> 
> > -----Original Message-----
> > From: Thomson, Martin [mailto:Martin.Thomson@andrew.com]
> > Sent: 19 July 2006 00:15
> > To: Rosen, Brian; Henning Schulzrinne
> > Cc: sip@ietf.org; geopriv@ietf.org
> > Subject: RE: [Geopriv] Location-by-value in a SIP Location Header
> > 
> > 
> > If you want technically feasible, I can post a URI to the list that 
> > includes a complete PIDF-LO with DSig.  It's a little long though.
> > 
> > I'm not suggesting that this a good idea, but I see no reason to 
> > prevent it.
> > 
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > Sent: Tuesday, 18 July 2006 11:07 PM
> > > To: Henning Schulzrinne
> > > Cc: sip@ietf.org; geopriv@ietf.org
> > > Subject: RE: [Geopriv] Location-by-value in a SIP Location Header
> > > 
> > > Perhaps some enlightenment is in order.  It's not clear to
> > me how you
> > > dsig a Data uri in a Location header (or maybe you mean
> > dsig the pidf-lo
> > > and put both the pidf-lo and the signature in the Data 
> URI) somehow?
> > > Are there some examples of this kind of thing elsewhere?
> > > 
> > > If we are ignoring technical arguments (as opposed to use
> > cases), it's
> > > out of forgetfulness, and not willfulness.  What were you
> > referring to?
> > > 
> > > Brian
> > > 
> > > 
> > > 
> > > -----Original Message-----
> > > From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> > > Sent: Tuesday, July 18, 2006 8:59 AM
> > > To: Rosen, Brian
> > > Cc: sip@ietf.org; geopriv@ietf.org
> > > Subject: Re: [Geopriv] Location-by-value in a SIP Location Header
> > > 
> > > Brian, James:
> > > 
> > > this is simply false. If desired, XML-DSIG is a well-known and 
> > > accepted mechanism to achieve integrity. Such specification is no 
> > > longer than the MUST NOT that you propose. You are also willfully 
> > > ignoring the other technical arguments made as part of this
> > discussion.
> > > 
> > > Henning
> > > 
> > > On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:
> > > 
> > > > There was a desire expressed by Hannes to provide a way
> > for a proxy to
> > > > insert location-by-value.  He had proposed a SIP 
> Location header 
> > > > specific way.  In IETF66, Henning suggested just using 
> a data URI.
> > > >
> > > > Doing this may run afoul of the geopriv location 
> privacy concerns, 
> > > > because there is no way for the user to sign the header 
> to provide 
> > > > assurance that the PIDF, and specifically the retention 
> and other 
> > > > policy bits are preserved.  Note that S/MIME provides sufficient
> > integrity
> > > > protection for a body.  It's probably okay to use TLS 
> per hop to 
> > > > provide privacy, but not integrity protection.  The authors
> > believe that there
> > > > is no really compelling use case for this feature, and
> > the effort to
> > > > define acceptable security mechanisms for location data
> > in a header
> > > > would be a significant effort, further delaying the draft.
> > > >
> > > > We propose, therefore, to state in 
> sip-location-conveyance, that a 
> > > > Data URI MUST NOT be used in the Location header until a
> > standards track
> > > > RFC
> > > > defines a suitable security mechanism to protect the 
> PIDF in the 
> > > > header.
> > > >
> > > > Brian and James
> > > >
> > > > _______________________________________________
> > > > Geopriv mailing list
> > > > Geopriv@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > 
> > > 
> > > _______________________________________________
> > > Geopriv mailing list
> > > Geopriv@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/geopriv
> > 
> > --------------------------------------------------------------
> > ----------------------------------
> > This message is for the designated recipient only and may contain 
> > privileged, proprietary, or otherwise private information.
> > If you have received it in error, please notify the sender 
> immediately 
> > and delete the original.  Any unauthorized use of this email is 
> > prohibited.
> > --------------------------------------------------------------
> > ----------------------------------
> > [mf2]
> > 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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To: Andrew Newton <andy@hxr.us>, Brian Rosen <br@brianrosen.net>
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Subject: Re: [Sip] Re: [Geopriv] SIP Location Conveyance:
	dereferencing	location-by-reference
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At 05:25 PM 7/18/2006 -0400, Andrew Newton wrote:


>Brian Rosen wrote:
>>In sip-location-conveyance, we allow a URI in the Location header to provide
>>for location-by-reference.  We must have some kind of text that says what
>>the recipient does with this URI.
>>Our proposal is to change the ABNF, which currently allows this URI to be
>>AbsoluteURI (meaning any scheme) and replace it with HTTP/HTTPS only.  Of
>>course, there would still be the cid: for message body identification.  We
>>would then define a dereference process as doing an HTTP/HTTPS GET, with no
>>parameters, on the URI, and expect to receive in return a PIDF-LO, exactly
>>as it would appear in a body.  The text would state MUST implement TLS,
>>except for the emergency case.  The text would further state that additional
>>dereferencing procedures could be defined by a standards track RFC.
>
>I'm not sure it is as simple as this.
>
>What happens when a client gets a subscribable location reference URI via 
>the L7-LCP,

you mean a SIP or SIPS URI?

You expect SIP to be a dereference capable protocol?  Why?

>and puts that into a Location header in an INVITE?

INVITEs don't do subscriptions, SUBSCRIBEs do, so this L7 value put into a 
Location header that's in an INVITE wouldn't create a subscription in the 
first place.

The above offering is to make sure dereferencing is done with HTTP/HTTPS, 
not SIP.

>  How is the subscription done with HTTP?

It isn't done with an INVITE either. sorry...


>-andy
>
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Wed Jul 19 13:00:20 2006
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I've posted the raw notes from the recorders of SIP meetings at IETF  
66 on our supplemental site:

http://www.softarmor.com/sipwg/meets/ietf66/notes/index.html

We'll work these up into minutes.

--
Dean


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From sip-bounces@ietf.org Wed Jul 19 13:02:40 2006
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Subject: Re: [Sip] Re: [Geopriv] SIP Location
	Conveyance:	dereferencing	location-by-reference
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I suspect we're talking about two different use cases:

- proxies will find subscriptions rather tedious when doing call routing 
(do a SUBSCRIBE, wait for authorization, NOTIFY just to get a location 
to route a call?), and don't get any benefit from the subscription model;

- UASs will find subscriptions helpful, in some limited circumstances 
(e.g., if I call OnStar for directions, they might want to see where I'm 
going in real time, rather than where I was five minutes ago while all 
their agents were busy assisting other customers)

I don't see particular security issues as long as the UAC realizes that 
any subscription URI that it hands out might be used by anybody that is 
on the call path. In some cases, that's an acceptable model, 
particularly if the caller identity is obscured, so that you only know 
that "random caller abc17 is at location X waiting for tow truck". (In 
that case, subscriptions are probably somewhat pointless.)

In short, I think we should allow all of these and simply make sensible 
recommendations, such as that location information meant for call 
routing should NOT just use subscriptions and also provide location 
by-value/by-attachment, simply because location subscriptions are too 
painful and useless for proxies in practice.

Henning

Andrew Newton wrote:
> James M. Polk wrote:
>> At 05:25 PM 7/18/2006 -0400, Andrew Newton wrote:
>>> What happens when a client gets a subscribable location reference URI 
>>> via the L7-LCP,
>>
>> you mean a SIP or SIPS URI?
>>
>> You expect SIP to be a dereference capable protocol?  Why?
> 
> I actually didn't mention SIP as the dereference protocol.  What I am 
> asking is which dereference protocol with a subscription mechanism are 
> you going to specify?  Or should implementers assume XMPP for this 
> feature? :)
> 
>>> and puts that into a Location header in an INVITE?
>>
>> INVITEs don't do subscriptions, SUBSCRIBEs do, so this L7 value put 
>> into a Location header that's in an INVITE wouldn't create a 
>> subscription in the first place.
> 
> Thanks for the clarification, but I was talking about the dereference 
> being orchestrated via subscription rather than polling.  I wasn't 
> really talking about the conveyance itself, other than to ask how a 
> conveyee is to know what to do with the reference.
> 
>> The above offering is to make sure dereferencing is done with 
>> HTTP/HTTPS, not SIP.
> 
> I know, which is why I asked...
> 
>>>  How is the subscription done with HTTP?
> 
> -andy
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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From sip-bounces@ietf.org Wed Jul 19 14:02:48 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Andrew Newton'" <andy@hxr.us>,
	"'James M. Polk'" <jmpolk@cisco.com>
Subject: RE: [Sip] Re: [Geopriv] SIP Location Conveyance:
	dereferencing	location-by-reference
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The original message said:
  The text would further state that additional dereferencing procedures
  could be defined by a standards track RFC.

Therefore, the proposed L7 protocol could specify its own dereference
procedure, including a subscription process.  We already have the mechanisms
needed to get a presence subscription out of an AoR.

Given the security concerns about location, it does seem wise to restrict
what kind of URI can be a location-by-reference to those that have reviews
of the geopriv privacy concerns.  I don't think we need to make the
mechanism completely general at the outset.  We propose to have the basic
mechanism be readily extensible by standards track RFCs.  That would seem to
balance security concerns against utility.

Brian

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Wednesday, July 19, 2006 12:51 PM
> To: James M. Polk
> Cc: Brian Rosen; sip@ietf.org; geopriv@ietf.org
> Subject: Re: [Sip] Re: [Geopriv] SIP Location Conveyance: dereferencing
> location-by-reference
> 
> James M. Polk wrote:
> > At 05:25 PM 7/18/2006 -0400, Andrew Newton wrote:
> >> What happens when a client gets a subscribable location reference URI
> >> via the L7-LCP,
> >
> > you mean a SIP or SIPS URI?
> >
> > You expect SIP to be a dereference capable protocol?  Why?
> 
> I actually didn't mention SIP as the dereference protocol.  What I am
> asking is which dereference protocol with a subscription mechanism are
> you going to specify?  Or should implementers assume XMPP for this
> feature? :)
> 
> >> and puts that into a Location header in an INVITE?
> >
> > INVITEs don't do subscriptions, SUBSCRIBEs do, so this L7 value put into
> > a Location header that's in an INVITE wouldn't create a subscription in
> > the first place.
> 
> Thanks for the clarification, but I was talking about the dereference
> being orchestrated via subscription rather than polling.  I wasn't
> really talking about the conveyance itself, other than to ask how a
> conveyee is to know what to do with the reference.
> 
> > The above offering is to make sure dereferencing is done with
> > HTTP/HTTPS, not SIP.
> 
> I know, which is why I asked...
> 
> >>  How is the subscription done with HTTP?
> 
> -andy


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From sip-bounces@ietf.org Wed Jul 19 15:14:21 2006
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Scott:

I went through draft-ietf-sip-hop-limit-diagnostics-03; here
are my comments on the draft.  I went through the existing
WG list discussions regarding this draft to make sure that I
was not duplicating any items that have already been discussed.
To the best of my knowledge, the following have not yet been
discussed.

1) In Montreal, I had mentioned to you that the ICMP sends back
  more than 64 bits of original datagram, and that maybe 64 "bits"
  was transcribed in error.  As it turns out, rfc792 does indeed
  say 64 bits of orgininal datagram.

  However, in practice, we have observed far more data is returned
  from the original datagram than 64-bits.  RFC1122 allows
  implementations to send more than 8 octets (see Section 3.2.2 of
  rfc1122).  So... what we have observed is that Linux returns 520
  bytes of the original datagram, and Solaris returns 64 bytes;
  but for Solaris, this is configurable in the kernel.

  And ICMPv6 (rfc2463) tries to fit as much of the original
  datagram without exceeding the minimum IPv6 MTU in the ICMP
  message.

  So, what does this all mean to hop-limit-diagnostics-03?
  Probably no changes, but interesting and relevant background
  information if you will like to maintain it in the text.

2) Section 3.1, third paragraph talks about removal of "Via
  and Route headers".  I think you meant Record-Route headers
  instead of (or in addition to?) Route headers.  In a
  dialog-establishing INVITE, there'll be greater chances of
  Record-Route headers being present than Route headers (which,
  if they are present consists of a pre-loaded Route set).
  Moreover, the Route headers would be consumed at every hop
  thereby reducing the size of the message, whereas the R-R
  headers are inserted, thus increasing the size.

  It seems that we ought to preserve the Via headers as much
  as possible, at the expense of R-R or Route headers.  The
  Vias give diagnostic information on which intermediaries
  have handled the request, which is of interest.  The Route header,
  on the other hand, portends which intermediary will handle
  the request beyond the element that generated a 483.

3) Nit: Section 3.1, third paragraph, first sentence is
  incomplete.  It states, "...acceptable; the only exception to
  this rule is when ."

4) In the SIP messages, headers that continue should be
  indented (LWS).  The Via headers, for instance, contain a
  branch parameter that begins on a new line without the
  continuation LWS.

5) Section 5 appears to lay the legal groundwork for why a
  UAC should accept a response that contains a body for which
  the UAC did not express an interest in receiving.  I think
  this is good to have.  I would change the first sentence
  of the second paragraph to read as follows:

    Note that a UAC MUST be prepared to receive message bodies
    that it does not understand and did not request in a
    response; this is ...

  That is, move the object of what the UAC must be prepared to
  do closer to the UAC.

6) In Section 6, it appears that the 483 response has been
  pruned, although the text does not indicate so.  By my
  count, if the UAC started with a M-F of 9, then it should
  have 9 Via headers, whereas the example shows the number
  of Via headers to be 7.  You may just want to simply mention
  that the request has been pruned in the response.

Other than that, I am glad that the draft is out -- need
this stuff.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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I've been reading about the ZRTP lately.  I read Henning's comment at the IETF meeting about how S/MIME relies on the receiver's public key to be available to allow for encryption/decryption to work.

Why couldnt we put in place a mechanism that did a key exchange for SIP signaling similar to whats done in ZRTP (i.e. modified Diffie-Hellman) prior to any signaling traffic, then use the exchanged keys as the basis for S/MIME encryption of the bodies?  Or we could even get really carried away and use it to do encryption of the entire SIP message if the signaling did not traverse proxies?  This could be useful for encrypting UDP signaling, again if no proxy were involved.

Thoughts?
FM


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Thomson, Martin wrote:
> Discussions to date on a subscription URI indicate that it is for LIS -> Target purposes only.  We have not yet reached the point where we discuss the creation of a reference.  There are obvious similarities, but I think that we would need to resolve some privacy and security concerns first.

Excuse me for not understanding, but LIS -> Target seems to miss the 
need outlined in the use case given on the L7-LCP DT list... which is a 
mobile device that has a link that very much looks like a fixed wired 
connection (and LI is derived from the network, i.e. not GPS).

-andy


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From sip-bounces@ietf.org Wed Jul 19 15:31:32 2006
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James M. Polk wrote:
> At 05:25 PM 7/18/2006 -0400, Andrew Newton wrote:
>> What happens when a client gets a subscribable location reference URI 
>> via the L7-LCP,
> 
> you mean a SIP or SIPS URI?
> 
> You expect SIP to be a dereference capable protocol?  Why?

I actually didn't mention SIP as the dereference protocol.  What I am 
asking is which dereference protocol with a subscription mechanism are 
you going to specify?  Or should implementers assume XMPP for this 
feature? :)

>> and puts that into a Location header in an INVITE?
> 
> INVITEs don't do subscriptions, SUBSCRIBEs do, so this L7 value put into 
> a Location header that's in an INVITE wouldn't create a subscription in 
> the first place.

Thanks for the clarification, but I was talking about the dereference 
being orchestrated via subscription rather than polling.  I wasn't 
really talking about the conveyance itself, other than to ask how a 
conveyee is to know what to do with the reference.

> The above offering is to make sure dereferencing is done with 
> HTTP/HTTPS, not SIP.

I know, which is why I asked...

>>  How is the subscription done with HTTP?

-andy


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Henning Schulzrinne wrote:
> 
> In short, I think we should allow all of these and simply make sensible 
> recommendations, such as that location information meant for call 
> routing should NOT just use subscriptions and also provide location 
> by-value/by-attachment, simply because location subscriptions are too 
> painful and useless for proxies in practice.

Agreed.  And I'm not advocating that subscription is the only method, it 
was just my understanding that there was a desire to have it.

What I would like is a small list of things I need to implement. 
AbsoluteURI leaves the number of possibilities wide open.

-andy


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From sip-bounces@ietf.org Wed Jul 19 15:39:41 2006
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If you run a DH then the outcome is a symmetric key. S/MIME does not 
work with symmetric keys.

Ciao
Hannes

fwmiller@cornfed.com wrote:
> I've been reading about the ZRTP lately.  I read Henning's comment at
> the IETF meeting about how S/MIME relies on the receiver's public key
> to be available to allow for encryption/decryption to work.
> 
> Why couldnt we put in place a mechanism that did a key exchange for
> SIP signaling similar to whats done in ZRTP (i.e. modified
> Diffie-Hellman) prior to any signaling traffic, then use the
> exchanged keys as the basis for S/MIME encryption of the bodies?  Or
> we could even get really carried away and use it to do encryption of
> the entire SIP message if the signaling did not traverse proxies?
> This could be useful for encrypting UDP signaling, again if no proxy
> were involved.
> 
> Thoughts? FM
> 
> 
> _______________________________________________ Sip mailing list
> https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
> development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
> 
> 


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The idea is the general concept of doing some kind of key exchange between the UAs prior to doing signaling.  This would address the problem of maintaining the public key infrastructure.

If that particular algorithm and key set does not work, perhaps some algoritm that had the "UAS" generate an asymmetric public key dynamically would work.  The key exchange could be for each endpoint to have the other endpoint generate an asymmetric key?  I'm not a crypto person but this seems like it might be doable, does any crypto person here know of how that could be done?


FM



------------------------------------------------
On Wed, 19 Jul 2006 21:37:47 +0200, Hannes Tschofenig <Hannes.Tschofenig@gmx.net> wrote:

> If you run a DH then the outcome is a symmetric key. S/MIME does not 
> work with symmetric keys.
> 
> Ciao
> Hannes
> 
> fwmiller@cornfed.com wrote:
> > I've been reading about the ZRTP lately.  I read Henning's comment at
> > the IETF meeting about how S/MIME relies on the receiver's public key
> > to be available to allow for encryption/decryption to work.
> > 
> > Why couldnt we put in place a mechanism that did a key exchange for
> > SIP signaling similar to whats done in ZRTP (i.e. modified
> > Diffie-Hellman) prior to any signaling traffic, then use the
> > exchanged keys as the basis for S/MIME encryption of the bodies?  Or
> > we could even get really carried away and use it to do encryption of
> > the entire SIP message if the signaling did not traverse proxies?
> > This could be useful for encrypting UDP signaling, again if no proxy
> > were involved.
> > 
> > Thoughts? FM
> > 
> > 
> > _______________________________________________ Sip mailing list
> > https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
> > development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip Use
> > sipping@ietf.org for new developments on the application of sip
> > 
> > 
> 

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Date: Wed, 19 Jul 2006 14:46:51 -0500
To: Andrew Newton <andy@hxr.us>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] Re: [Geopriv] SIP Location Conveyance:
	dereferencing	location-by-reference
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At 12:51 PM 7/19/2006 -0400, Andrew Newton wrote:
>James M. Polk wrote:
>>At 05:25 PM 7/18/2006 -0400, Andrew Newton wrote:
>>>What happens when a client gets a subscribable location reference URI 
>>>via the L7-LCP,
>>you mean a SIP or SIPS URI?
>>You expect SIP to be a dereference capable protocol?  Why?
>
>I actually didn't mention SIP as the dereference protocol.

you mentioned subscriptions, and sip-location-conveyance doesn't, but SIP 
does with SUBSCRIBE - so I made that leap.

Why do you want a subscription, and where's the requirement for that? The 
ID currently doesn't have one.

>What I am asking is which dereference protocol with a subscription 
>mechanism are you going to specify?  Or should implementers assume XMPP 
>for this feature? :)
>
>>>and puts that into a Location header in an INVITE?
>>INVITEs don't do subscriptions, SUBSCRIBEs do, so this L7 value put into 
>>a Location header that's in an INVITE wouldn't create a subscription in 
>>the first place.
>
>Thanks for the clarification, but I was talking about the dereference 
>being orchestrated via subscription rather than polling.  I wasn't really 
>talking about the conveyance itself, other than to ask how a conveyee is 
>to know what to do with the reference.
>
>>The above offering is to make sure dereferencing is done with HTTP/HTTPS, 
>>not SIP.
>
>I know, which is why I asked...
>
>>>  How is the subscription done with HTTP?
>
>-andy
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From sip-bounces@ietf.org Wed Jul 19 17:05:16 2006
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To: fwmiller@cornfed.com
Subject: Re: [Sip] SIP Security mechanism?
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Hi,

it is true that the recent security work in SIP focused on media security.

Some of the proposed mechanisms could also provide security for SIP 
signaling if signaling messages are exchanged between the end points 
(after the initial exchange via proxies that carries, for example, a 
finger print attribute).

Ciao
Hannes


fwmiller@cornfed.com wrote:
> The idea is the general concept of doing some kind of key exchange
> between the UAs prior to doing signaling.  This would address the
> problem of maintaining the public key infrastructure.
> 
> If that particular algorithm and key set does not work, perhaps some
> algoritm that had the "UAS" generate an asymmetric public key
> dynamically would work.  The key exchange could be for each endpoint
> to have the other endpoint generate an asymmetric key?  I'm not a
> crypto person but this seems like it might be doable, does any crypto
> person here know of how that could be done?
> 
> 
> FM
> 
> 
> 
> ------------------------------------------------ On Wed, 19 Jul 2006
> 21:37:47 +0200, Hannes Tschofenig <Hannes.Tschofenig@gmx.net> wrote:
> 
> 
>> If you run a DH then the outcome is a symmetric key. S/MIME does
>> not work with symmetric keys.
>> 
>> Ciao Hannes
>> 
>> fwmiller@cornfed.com wrote:
>> 
>>> I've been reading about the ZRTP lately.  I read Henning's
>>> comment at the IETF meeting about how S/MIME relies on the
>>> receiver's public key to be available to allow for
>>> encryption/decryption to work.
>>> 
>>> Why couldnt we put in place a mechanism that did a key exchange
>>> for SIP signaling similar to whats done in ZRTP (i.e. modified 
>>> Diffie-Hellman) prior to any signaling traffic, then use the 
>>> exchanged keys as the basis for S/MIME encryption of the bodies?
>>> Or we could even get really carried away and use it to do
>>> encryption of the entire SIP message if the signaling did not
>>> traverse proxies? This could be useful for encrypting UDP
>>> signaling, again if no proxy were involved.
>>> 
>>> Thoughts? FM
>>> 
>>> 
>>> _______________________________________________ Sip mailing list 
>>> https://www1.ietf.org/mailman/listinfo/sip This list is for NEW 
>>> development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>  sipping@ietf.org for new developments on the application of sip
>>> 
>>> 
>> 
> 
> 


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From sip-bounces@ietf.org Wed Jul 19 17:28:18 2006
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From: "Brian Rosen" <br@brianrosen.net>
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Subject: [Sip] Routing of calls based on location
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NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU REPLY, SEND IT ONLY TO
GEOPRIV

The SIP Location Conveyance document anticipates the ability to route a call
based on the location of the caller.  The original impetus for this is
emergency calling.  In ecrit, a mechanism has been defined (LoST) that takes
a service request (in the form of a newly defined service urn) and a
location and returns a URI.  This mechanism is not limited to emergency
calling, and could be used for arbitrary location based routing by defining
new service urns.

However, this opens up the issue of geopriv privacy concerns because the
sender does not know the identity of the element (proxy) that will route the
call using LoST.  It may be that the UAC itself does this, but it may be
some other intermediary.

If the identity of the intermediary is not known, then protecting the
location information becomes more difficult, because we cannot encrypt using
keying material from the intended recipient.  In fact, we would need to use
hop-by-hop security, with transitive trust.  This is what the emergency case
does, but it also allows no channel security if necessary (usually meaning,
try it with TLS and if that fails, try again without TLS).

We propose to explicitly allow hop-by-hop security when sending location
using SIP for the purpose of location based routing.  We will specify you
MUST use TLS if sending location by reference (and there will be some words
about the choice of URI construction which is raised in another email I am
about to send).

An important issue that we need to discuss is what "Do Not Distribute" means
in this case.  We would like it to allow the data to be passed onward.
Indeed, SIP proxies shouldn't be dropping headers.  We propose that it means
that it cannot be distributed except that the location header (and body of
course) is passed as it normally would.

Note that although this is an issue that concerns the SIP working group,
because the location conveyance document is a sip document, it really is a
geopriv issue.  It also affects ecrit, since we are using its protocol
beyond its initial purpose.  We ask that this discussion be held on the
geopriv list, and not copied to either sip or ecrit.

Brian


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From sip-bounces@ietf.org Wed Jul 19 17:41:34 2006
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From: "Brian Rosen" <br@brianrosen.net>
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Subject: [Sip] More on the contents of the Location header URI - http
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I started a discussion of what can be in the location header besides a cid:.
We remain convinced that AbsoluteURI is a poor choice; we must be more
specific, including privacy concerns. 

It is now clear that simply saying "https:" is not enough.  There could be
multiple location resolution protocols that use http transport.  We need
something else that would differentiate which one.  The simple GET mechanism
we proposed would need at least a parameter of some sort.  

After thinking about it for a while, I think the best resolution is Andy's
suggestion of a registry, used in conjunction with a parameter.  I'm
ambivalent on whether the parameter is a URI parameter or a parameter on the
Location header.  We would define the registry (and the parameter) in
location-conveyance and create one entry (the simple GET with PIDF-LO
return).  It may be, for example, that the proposed L7 location acquisition
protocol defines another http based resolution protocol.  That document
could define another entry in the registry.

Brian


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From sip-bounces@ietf.org Wed Jul 19 18:48:05 2006
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Subject: [Sip] Re: [Simple] UDP - what if response is bigger than 1500
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Thomas Froment wrote:
> This is a good question: I think it was mentionned in last SIP WG IETF 
> meeting that having responses greater than path MTU  (-200 bytes)
> in SIP was not currently well specified. It was an open issue on 
> draft-ietf-sip-hop-limit-diagnostics-03.txt which raises this problem
> since it leads naturally to send big SIP responses.
> SIP WG people may probably answer you if something is planned to handle 
> this problem...

This remains a problem; I have had a quick chat with Scott
Lawrence on it and we'll be submitting a draft latter part
of next week on one way to deal with this.

Thanks.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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From sip-bounces@ietf.org Wed Jul 19 18:48:48 2006
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From: "Brian Rosen" <br@brianrosen.net>
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Date: Wed, 19 Jul 2006 18:45:38 -0400
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Subject: [Sip] Allowing sip: (or prez:) in the location header
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Suppose we allowed sip: or prez: as the location header URI.  The resolution
of this scheme would be "subscribe to presence".

Note that if you asked for immediate expiration (0), you will get the
immediate NOTIFY.  This constitutes one reasonable way to provide
location-by-reference.  It is a form of "pull" (you send a SUBSCRIBE, you
get one NOTIFY in return with the PIDF-LO).  If your subscription has a
longer expiration, it is a form of "push", and you might want to have a
filter to limit how many pushes you get.

Note that normally, presence provides identity as well as location.  It may
be that in some uses for the Location header, such as a location based
routing, the identity information is not appropriate.  The policy of the
presence server for that use may be to only return the LO part of the PIDF.
Also note that the form of the URI itself could inadvertently provide
identity information (prez:brian.rosen@neustar.biz), which might not be
desirable.  If the idea to allow sip: and prez: is acceptable in the
Location header URI, the text would contain a discussion of this issue.

Brian


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From sip-bounces@ietf.org Wed Jul 19 19:28:23 2006
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Brian,

I am not sure that I agree with you on this one.
The problem is not in the request protocol at all, the GET works fine
providing the return type is always a PIDF-LO.

Cheers
James

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Thursday, 20 July 2006 7:33 AM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: [Geopriv] More on the contents of the Location header URI -
http
>=20
> I started a discussion of what can be in the location header besides a
> cid:.
> We remain convinced that AbsoluteURI is a poor choice; we must be more
> specific, including privacy concerns.
>=20
> It is now clear that simply saying "https:" is not enough.  There
could
> be multiple location resolution protocols that use http transport.  We
> need something else that would differentiate which one.  The simple
GET
> mechanism we proposed would need at least a parameter of some sort.
>=20
> After thinking about it for a while, I think the best resolution is
> Andy's suggestion of a registry, used in conjunction with a parameter.
> I'm ambivalent on whether the parameter is a URI parameter or a
> parameter on the Location header.  We would define the registry (and
the
> parameter) in location-conveyance and create one entry (the simple GET
> with PIDF-LO return).  It may be, for example, that the proposed L7
> location acquisition protocol defines another http based resolution
> protocol.  That document could define another entry in the registry.
>=20
> Brian
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1958694652==--



From sip-bounces@ietf.org Wed Jul 19 19:58:00 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Winterbottom, James'" <James.Winterbottom@andrew.com>,
	"Rosen, Brian" <Brian.Rosen@neustar.biz>, <sip@ietf.org>
Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location header
	URI- http
Date: Wed, 19 Jul 2006 19:51:58 -0400
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The problem is that HTTP is used as a transport protocol by several
mechanisms, and you need to know from the contents of the location header
which mechanism is used.  Is it SOAP, raw GET, LCP, RELO, HELD or something
else, all of which would say "https:blahblah"?

Brian

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Wednesday, July 19, 2006 7:26 PM
> To: Rosen, Brian; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: [Sip] RE: [Geopriv] More on the contents of the Location header
> URI- http
> 
> Brian,
> 
> I am not sure that I agree with you on this one.
> The problem is not in the request protocol at all, the GET works fine
> providing the return type is always a PIDF-LO.
> 
> Cheers
> James
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Thursday, 20 July 2006 7:33 AM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: [Geopriv] More on the contents of the Location header URI -
> http
> >
> > I started a discussion of what can be in the location header besides a
> > cid:.
> > We remain convinced that AbsoluteURI is a poor choice; we must be more
> > specific, including privacy concerns.
> >
> > It is now clear that simply saying "https:" is not enough.  There
> could
> > be multiple location resolution protocols that use http transport.  We
> > need something else that would differentiate which one.  The simple
> GET
> > mechanism we proposed would need at least a parameter of some sort.
> >
> > After thinking about it for a while, I think the best resolution is
> > Andy's suggestion of a registry, used in conjunction with a parameter.
> > I'm ambivalent on whether the parameter is a URI parameter or a
> > parameter on the Location header.  We would define the registry (and
> the
> > parameter) in location-conveyance and create one entry (the simple GET
> > with PIDF-LO return).  It may be, for example, that the proposed L7
> > location acquisition protocol defines another http based resolution
> > protocol.  That document could define another entry in the registry.
> >
> > Brian
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> 
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> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Wed Jul 19 19:58:22 2006
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Subject: [Sip] Protocol Shepherding Writeup for SIP MIB -- draft-ietf-sip-mib
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As the shepherding WG chair for sip-mib, I've submitted the following  
writeup to the IESG.


---- forwarded text follows ----

The SIP working group would like to request publication of the  
standards-track document draft-ietf-sip-mib-11, which has already  
completed MIB Doctor review. A Protocol Shepherd writeup follows,  
including a draft Document Announcement Writeup.



(1.a)  Who is the Document Shepherd for this document?  Has the
           Document Shepherd personally reviewed this version of the
           document and, in particular, does he or she believe this
           version is ready for forwarding to the IESG for publication?

The Document Shepherd for draft-ietf-sip-mib-11.txt is Dean Willis,  
co-chair of the SIP Working Group. He has reviewed this document and  
believes it is ready for forwarding to the iESG for publication.


    (1.b)  Has the document had adequate review both from key WG members
           and from key non-WG members?  Does the Document Shepherd have
           any concerns about the depth or breadth of the reviews that
           have been performed?

The document has been widely reviewed within the working group and  
within the wider community, including discussion at several standards  
organizations.  It has also been reviewed by Dan Romascanu as the  
specified MIB doctor. The document shepherd has no concerns about the  
depth or breadth of the reviews that have been performed.


    (1.c)  Does the Document Shepherd have concerns that the document
           needs more review from a particular or broader perspective,
           e.g., security, operational complexity, someone familiar with
           AAA, internationalization or XML?

The document shepherd has no concerns that the draft needs broader  
review.



    (1.d)  Does the Document Shepherd have any specific concerns or
           issues with this document that the Responsible Area Director
           and/or the IESG should be aware of?  For example, perhaps he
           or she is uncomfortable with certain parts of the  
document, or
           has concerns whether there really is a need for it.  In any
           event, if those issues have been discussed in the WG and the
           WG has indicated that it still wishes to advance the  
document,
           detail those concerns here.


The document shepherd has personal concerns about whether there is  
any need for a SIP MIB given ongoing evolution in the systems  
management environment, but there is a reasonable consensus in the  
working group and broader community to advance the document despite  
these concerns.


    (1.e)  How solid is the WG consensus behind this document?  Does it
           represent the strong concurrence of a few individuals, with
           others being silent, or does the WG as a whole understand and
           agree with it?

The document enjoys a strong consensus of all working group  
participants known to the document shepherd who care about a MIB at all.



    (1.f)  Has anyone threatened an appeal or otherwise indicated  
extreme
           discontent?  If so, please summarise the areas of conflict in
           separate email messages to the Responsible Area Director.   
(It
           should be in a separate email because this questionnaire will
           be entered into the ID Tracker.)

There are no known indications of extreme discontent.


    (1.g)  Has the Document Shepherd verified that the document  
satisfies
           all ID nits?  (See http://www.ietf.org/ID-Checklist.html and
           http://tools.ietf.org/tools/idnits/).  Boilerplate checks are
           not enough; this check needs to be thorough.

The document shepherd read through the document,  executed the  
current idnits against it, and worked through the checklist at http:// 
www.ietf.org/ID-Checklist.html and found no issues. The MIB data in  
the document was verified by a third party using the usual tools.



    (1.h)  Has the document split its references into normative and
           informative?  Are there normative references to documents  
that
           are not ready for advancement or are otherwise in an unclear
           state?  If such normative references exist, what is the
           strategy for their completion?  Are there normative  
references
           that are downward references, as described in [RFC3967]?  If
           so, list these downward references to support the Area
           Director in the Last Call procedure for them [RFC3967].

The references are correctly divided, and all normative references  
are to completed standards-track documents.


    (1.i)  The IESG approval announcement includes a Document
           Announcement Write-Up.  Please provide such a Document
           Announcement Write-Up.  Recent examples can be found in the
           "Action" announcements for approved documents.  The approval
           announcement contains the following sections:

           Technical Summary
              Relevant content can frequently be found in the abstract
              and/or introduction of the document.  If not, this may be
              an indication that there are deficiencies in the abstract
              or introduction.

           Working Group Summary
              Was there anything in WG process that is worth noting?   
For
              example, was there controversy about particular points or
              were there decisions where the consensus was particularly
              rough?

           Document Quality
              Are there existing implementations of the protocol?   
Have a
              significant number of vendors indicated their plan to
              implement the specification?  Are there any reviewers that
              merit special mention as having done a thorough review,
              e.g., one that resulted in important changes or a
              conclusion that the document had no substantive issues?



Document Announcement Writeup for SIP MIB, draft-ietf-sip-mib

Technical Summary:

This document defines a portion of the Management Information Base (a  
MIB) including a set of objects used to manage  Session Initiation  
Protocol (SIP) entities, including SIP User Agents, Proxies, Redirect  
(aka Location) Servers, and Registrars.  Note that this MIB is  
restricted to management and monitoring activities. Following  
extended discussion, the working group chose not to pursue device  
configuration using the SIP MIB, and will rely instead on the SIP  
Configuration Framework being developed in the SIPPING working group.

Working Group Summary:

The SIP Working Group developed this document over a greater than six- 
year period, with the first draft appearing in March of 2000. One of  
the key decisions in its development was the choice to exclude device  
configuration functions from the MIB. This decision was reached after  
the SIP community decided to pursue a SIP-specific configuration  
model as outlined in "A Framework for Session Initiation Protocol  
User Agent Profile Delivery", draft-ietf-sipping-config- 
framework-08.txt. This decision significantly reduced the scope and  
complexity of the SIP MIB. While not all participants in the working  
group are convinced of the need for a SIP MIB, there seems to be a  
very strong consensus on the technical detail of the draft and a  
general agreement on its applicability.

Document Quality:

The document shepherd is aware of at least one pre-specification  
implementation of the SIP MIB developed by a major equipment  
provider. The SIPIT interoperability events have not included SIP MIB  
testing in their events, and the coordinator of those events reports  
that he stopped asking about SIP MIB implementations some years ago  
because of the general lack of interest in the development community.  
However, the industry is beginning to see requirements for SIP MIB  
compliance appear in large-scale Requests For Information and  
Requests For Proposal. The document has also received considerable  
attention from related standards development organizations, including  
PacketCable. Development of this specification was assisted greatly  
by the support of Dan Romascanu acting as the designated "MIB  
Doctor", with additional commentary received from David Harrington..   
The document's "Contributors" section also cites review by  Tom  
Taylor, Kavitha Patchayappan, Dan Romascanu, Cullen Jennings, Orit  
Levin, AC Mahendran, Mary Barnes, Rohan Mahy, Bob Penfield, Charles  
Eckel and Dean Willis. Both Allison Mankin and Cullen Jennings  
oversaw the development as Area Directors, with Cullen Jennings  
serving as the Responsible Area Director at the time of publication  
request. Dean Willis served as the WG Chair Shepherd.





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From sip-bounces@ietf.org Wed Jul 19 20:02:09 2006
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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Brian Rosen" <br@brianrosen.net>,
	"Rosen, Brian" <Brian.Rosen@neustar.biz>, <sip@ietf.org>
Date: Wed, 19 Jul 2006 19:00:14 -0500
Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location header
	URI- http
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Providing all mechanisms support a base GET AND they return a PIDF-LO
there is no problem. That is all that needs to be said.



> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]
> Sent: Thursday, 20 July 2006 9:52 AM
> To: Winterbottom, James; Rosen, Brian; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location
> header URI- http
>=20
> The problem is that HTTP is used as a transport protocol by several
> mechanisms, and you need to know from the contents of the location
header
> which mechanism is used.  Is it SOAP, raw GET, LCP, RELO, HELD or
> something
> else, all of which would say "https:blahblah"?
>=20
> Brian
>=20
> > -----Original Message-----
> > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > Sent: Wednesday, July 19, 2006 7:26 PM
> > To: Rosen, Brian; sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: [Sip] RE: [Geopriv] More on the contents of the Location
header
> > URI- http
> >
> > Brian,
> >
> > I am not sure that I agree with you on this one.
> > The problem is not in the request protocol at all, the GET works
fine
> > providing the return type is always a PIDF-LO.
> >
> > Cheers
> > James
> >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > Sent: Thursday, 20 July 2006 7:33 AM
> > > To: sip@ietf.org
> > > Cc: geopriv@ietf.org
> > > Subject: [Geopriv] More on the contents of the Location header URI
-
> > http
> > >
> > > I started a discussion of what can be in the location header
besides a
> > > cid:.
> > > We remain convinced that AbsoluteURI is a poor choice; we must be
more
> > > specific, including privacy concerns.
> > >
> > > It is now clear that simply saying "https:" is not enough.  There
> > could
> > > be multiple location resolution protocols that use http transport.
We
> > > need something else that would differentiate which one.  The
simple
> > GET
> > > mechanism we proposed would need at least a parameter of some
sort.
> > >
> > > After thinking about it for a while, I think the best resolution
is
> > > Andy's suggestion of a registry, used in conjunction with a
parameter.
> > > I'm ambivalent on whether the parameter is a URI parameter or a
> > > parameter on the Location header.  We would define the registry
(and
> > the
> > > parameter) in location-conveyance and create one entry (the simple
GET
> > > with PIDF-LO return).  It may be, for example, that the proposed
L7
> > > location acquisition protocol defines another http based
resolution
> > > protocol.  That document could define another entry in the
registry.
> > >
> > > Brian
> > >
> > > _______________________________________________
> > > Geopriv mailing list
> > > Geopriv@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/geopriv
> >
> >
------------------------------------------------------------------------
> --
> > ----------------------
> > This message is for the designated recipient only and may
> > contain privileged, proprietary, or otherwise private information.
> > If you have received it in error, please notify the sender
> > immediately and delete the original.  Any unauthorized use of
> > this email is prohibited.
> >
------------------------------------------------------------------------
> --
> > ----------------------
> > [mf2]
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>=20

---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
contain privileged, proprietary, or otherwise private information. =20
If you have received it in error, please notify the sender
immediately and delete the original.  Any unauthorized use of
this email is prohibited.
---------------------------------------------------------------------------=
---------------------
[mf2]

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Wed Jul 19 20:04:28 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Thomson, Martin'" <Martin.Thomson@andrew.com>,
	<sip@ietf.org>
Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
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I think we are going to be explicit about what is allowed to be in the
Location header, and not use AbsoluteURI.  We think something DOES have to
document every possible retrieval mechanism and discuss the geopriv privacy
issues and how they are dealt with.  That is precisely what we propose.
We'll define some of them in location-conveyance, and specify how a
standards track document adds more possibilities.

Please note that we don't want a lot of possibilities here; every
dereferencer has to implement all choices.

Incidentally, I think we will specify in the language that an extension to
the allowed URIs in a Location header is NOT a SIP extension, and does not
come under the 3427 process.   This would allow geopriv, for example, to
define the L7 protocol, and if it has a location-by-reference construction
with a dereferencing procedure, it would be allowed to add to the methods
listed in location-conveyance without running through 3427.  That language,
since it is part of sip-location, would have to be acceptable to
sip/sipping.

Agree that the identity issues of URL construction are generic and can be
treated in the text as generic, not tied to any particular scheme.  It may
be more likely in prez:, simply because the default type of uri is an AoR,
which almost always has identity data.

Brian

> -----Original Message-----
> From: Thomson, Martin [mailto:Martin.Thomson@andrew.com]
> Sent: Wednesday, July 19, 2006 7:36 PM
> To: Brian Rosen; sip@ietf.org
> Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
> 
> Brian,
> 
> Yes, you can use a sip: or pres: URI to retrieve location by-reference.
> It works.  Yes, you can use a pull method to retrieve the information;
> that's a feature of SUBSCRIBE/NOTIFY.  I see no reason to prevent this,
> but you don't actually have to document every possible retrieval method.
> 
> Identity might be a problem, but that can be addressed by the use of
> unlinked pseudonyms.  This is not specific to the dereference method.  The
> same problem exists no matter how you resolve location - or are you
> suggesting that the sip: or pres: URIs dereference to a PIDF-LO, but other
> forms would not?  Clearly, that problem already exists with the cid: URL
> and should exist with the https: URL.
> 
> What I take from this is that you need to have text that explains the
> impact of the association between identity and location.  Those concerns
> are largely covered by other GEOPRIV documents, but there is no harm in
> being explicit.
> 
> Cheers,
> Martin
> 
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Thursday, 20 July 2006 8:46 AM
> > To: sip@ietf.org
> > Subject: [Sip] Allowing sip: (or prez:) in the location header
> >
> > Suppose we allowed sip: or prez: as the location header URI.  The
> > resolution
> > of this scheme would be "subscribe to presence".
> >
> > Note that if you asked for immediate expiration (0), you will get the
> > immediate NOTIFY.  This constitutes one reasonable way to provide
> > location-by-reference.  It is a form of "pull" (you send a SUBSCRIBE,
> you
> > get one NOTIFY in return with the PIDF-LO).  If your subscription has a
> > longer expiration, it is a form of "push", and you might want to have a
> > filter to limit how many pushes you get.
> >
> > Note that normally, presence provides identity as well as location.  It
> > may
> > be that in some uses for the Location header, such as a location based
> > routing, the identity information is not appropriate.  The policy of the
> > presence server for that use may be to only return the LO part of the
> > PIDF.
> > Also note that the form of the URI itself could inadvertently provide
> > identity information (prez:brian.rosen@neustar.biz), which might not be
> > desirable.  If the idea to allow sip: and prez: is acceptable in the
> > Location header URI, the text would contain a discussion of this issue.
> >
> > Brian
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> --------------------------------------------------------------------------
> ----------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
> --------------------------------------------------------------------------
> ----------------------
> [mf2]


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Wed Jul 19 20:15:54 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: <sip@ietf.org>,
	"'Winterbottom, James'" <James.Winterbottom@andrew.com>
Subject: Re: [Sip] RE: [Geopriv] More on the contents of the Location header
	URI- http
Date: Wed, 19 Jul 2006 20:12:52 -0400
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Actually, it means that a base GET/return PIDF is ALL you could implement;
you can't get anything back from the PIDF that would tell you anything else
is acceptable, and you can't tell from the URI.  

That would mean a SOAP based mechanism could never be specified, for
example.  You couldn't have an optional parameter unless you were prepared
to try it, have it fail, and then try something else.

Brian


> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Wednesday, July 19, 2006 8:00 PM
> To: Brian Rosen; Rosen, Brian; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location
> header URI- http
> 
> Providing all mechanisms support a base GET AND they return a PIDF-LO
> there is no problem. That is all that needs to be said.
> 
> 
> 
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Thursday, 20 July 2006 9:52 AM
> > To: Winterbottom, James; Rosen, Brian; sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location
> > header URI- http
> >
> > The problem is that HTTP is used as a transport protocol by several
> > mechanisms, and you need to know from the contents of the location
> header
> > which mechanism is used.  Is it SOAP, raw GET, LCP, RELO, HELD or
> > something
> > else, all of which would say "https:blahblah"?
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > Sent: Wednesday, July 19, 2006 7:26 PM
> > > To: Rosen, Brian; sip@ietf.org
> > > Cc: geopriv@ietf.org
> > > Subject: [Sip] RE: [Geopriv] More on the contents of the Location
> header
> > > URI- http
> > >
> > > Brian,
> > >
> > > I am not sure that I agree with you on this one.
> > > The problem is not in the request protocol at all, the GET works
> fine
> > > providing the return type is always a PIDF-LO.
> > >
> > > Cheers
> > > James
> > >
> > > > -----Original Message-----
> > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > Sent: Thursday, 20 July 2006 7:33 AM
> > > > To: sip@ietf.org
> > > > Cc: geopriv@ietf.org
> > > > Subject: [Geopriv] More on the contents of the Location header URI
> -
> > > http
> > > >
> > > > I started a discussion of what can be in the location header
> besides a
> > > > cid:.
> > > > We remain convinced that AbsoluteURI is a poor choice; we must be
> more
> > > > specific, including privacy concerns.
> > > >
> > > > It is now clear that simply saying "https:" is not enough.  There
> > > could
> > > > be multiple location resolution protocols that use http transport.
> We
> > > > need something else that would differentiate which one.  The
> simple
> > > GET
> > > > mechanism we proposed would need at least a parameter of some
> sort.
> > > >
> > > > After thinking about it for a while, I think the best resolution
> is
> > > > Andy's suggestion of a registry, used in conjunction with a
> parameter.
> > > > I'm ambivalent on whether the parameter is a URI parameter or a
> > > > parameter on the Location header.  We would define the registry
> (and
> > > the
> > > > parameter) in location-conveyance and create one entry (the simple
> GET
> > > > with PIDF-LO return).  It may be, for example, that the proposed
> L7
> > > > location acquisition protocol defines another http based
> resolution
> > > > protocol.  That document could define another entry in the
> registry.
> > > >
> > > > Brian
> > > >
> > > > _______________________________________________
> > > > Geopriv mailing list
> > > > Geopriv@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > >
> > >
> ------------------------------------------------------------------------
> > --
> > > ----------------------
> > > This message is for the designated recipient only and may
> > > contain privileged, proprietary, or otherwise private information.
> > > If you have received it in error, please notify the sender
> > > immediately and delete the original.  Any unauthorized use of
> > > this email is prohibited.
> > >
> ------------------------------------------------------------------------
> > --
> > > ----------------------
> > > [mf2]
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> --------------------------------------------------------------------------
> ----------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
> --------------------------------------------------------------------------
> ----------------------
> [mf2]


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Wed Jul 19 20:16:27 2006
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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Rosen, Brian" <Brian.Rosen@neustar.biz>,
	"Brian Rosen" <br@brianrosen.net>, <sip@ietf.org>
Date: Wed, 19 Jul 2006 19:14:38 -0500
Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location header
	URI- http
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What it means is that you MUST provide a Get mechanism with no
parameters, you MAY provide anything you like on top of that. If you are
prepared to take the hit with trying something specific first then that
is your prerogative, or you may decide to do the GET first, and
something special as a second-try, which is also okay. I simply don't
agree with your reasoning to exclude a simple GET mechanism now.=20

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Thursday, 20 July 2006 10:09 AM
> To: Winterbottom, James; Brian Rosen; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location
> header URI- http
>=20
> Actually, it means that a base GET/return PIDF is ALL you could
> implement; you can't get anything back from the PIDF that would tell
you
> anything else is acceptable, and you can't tell from the URI.
>=20
> That would mean a SOAP based mechanism could never be specified, for
> example.  You couldn't have an optional parameter unless you were
> prepared to try it, have it fail, and then try something else.
>=20
> Brian
>=20
>=20
> > -----Original Message-----
> > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > Sent: Wednesday, July 19, 2006 8:00 PM
> > To: Brian Rosen; Rosen, Brian; sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: RE: [Sip] RE: [Geopriv] More on the contents of the
Location
> > header URI- http
> >
> > Providing all mechanisms support a base GET AND they return a
PIDF-LO
> > there is no problem. That is all that needs to be said.
> >
> >
> >
> > > -----Original Message-----
> > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > Sent: Thursday, 20 July 2006 9:52 AM
> > > To: Winterbottom, James; Rosen, Brian; sip@ietf.org
> > > Cc: geopriv@ietf.org
> > > Subject: RE: [Sip] RE: [Geopriv] More on the contents of the
> Location
> > > header URI- http
> > >
> > > The problem is that HTTP is used as a transport protocol by
several
> > > mechanisms, and you need to know from the contents of the location
> > header
> > > which mechanism is used.  Is it SOAP, raw GET, LCP, RELO, HELD or
> > > something
> > > else, all of which would say "https:blahblah"?
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > > Sent: Wednesday, July 19, 2006 7:26 PM
> > > > To: Rosen, Brian; sip@ietf.org
> > > > Cc: geopriv@ietf.org
> > > > Subject: [Sip] RE: [Geopriv] More on the contents of the
Location
> > header
> > > > URI- http
> > > >
> > > > Brian,
> > > >
> > > > I am not sure that I agree with you on this one.
> > > > The problem is not in the request protocol at all, the GET works
> > fine
> > > > providing the return type is always a PIDF-LO.
> > > >
> > > > Cheers
> > > > James
> > > >
> > > > > -----Original Message-----
> > > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > > Sent: Thursday, 20 July 2006 7:33 AM
> > > > > To: sip@ietf.org
> > > > > Cc: geopriv@ietf.org
> > > > > Subject: [Geopriv] More on the contents of the Location header
> URI
> > -
> > > > http
> > > > >
> > > > > I started a discussion of what can be in the location header
> > besides a
> > > > > cid:.
> > > > > We remain convinced that AbsoluteURI is a poor choice; we must
> be
> > more
> > > > > specific, including privacy concerns.
> > > > >
> > > > > It is now clear that simply saying "https:" is not enough.
> There
> > > > could
> > > > > be multiple location resolution protocols that use http
> transport.
> > We
> > > > > need something else that would differentiate which one.  The
> > simple
> > > > GET
> > > > > mechanism we proposed would need at least a parameter of some
> > sort.
> > > > >
> > > > > After thinking about it for a while, I think the best
resolution
> > is
> > > > > Andy's suggestion of a registry, used in conjunction with a
> > parameter.
> > > > > I'm ambivalent on whether the parameter is a URI parameter or
a
> > > > > parameter on the Location header.  We would define the
registry
> > (and
> > > > the
> > > > > parameter) in location-conveyance and create one entry (the
> simple
> > GET
> > > > > with PIDF-LO return).  It may be, for example, that the
proposed
> > L7
> > > > > location acquisition protocol defines another http based
> > resolution
> > > > > protocol.  That document could define another entry in the
> > registry.
> > > > >
> > > > > Brian
> > > > >
> > > > > _______________________________________________
> > > > > Geopriv mailing list
> > > > > Geopriv@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > >
> > > >
> >
>
------------------------------------------------------------------------
> > > --
> > > > ----------------------
> > > > This message is for the designated recipient only and may
> > > > contain privileged, proprietary, or otherwise private
information.
> > > > If you have received it in error, please notify the sender
> > > > immediately and delete the original.  Any unauthorized use of
> > > > this email is prohibited.
> > > >
> >
>
------------------------------------------------------------------------
> > > --
> > > > ----------------------
> > > > [mf2]
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current
sip
> > > > Use sipping@ietf.org for new developments on the application of
> sip
> > >
> >
> >
>
------------------------------------------------------------------------
> --
> > ----------------------
> > This message is for the designated recipient only and may
> > contain privileged, proprietary, or otherwise private information.
> > If you have received it in error, please notify the sender
> > immediately and delete the original.  Any unauthorized use of
> > this email is prohibited.
> >
>
------------------------------------------------------------------------
> --
> > ----------------------
> > [mf2]

---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
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If you have received it in error, please notify the sender
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---------------------------------------------------------------------------=
---------------------
[mf2]

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Wed Jul 19 20:35:54 2006
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From: "Thomson, Martin" <Martin.Thomson@andrew.com>
To: "Brian Rosen" <br@brianrosen.net>,
	<sip@ietf.org>
Date: Wed, 19 Jul 2006 19:33:17 -0500
Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1193493571==--



From sip-bounces@ietf.org Wed Jul 19 20:37:39 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Winterbottom, James'" <James.Winterbottom@andrew.com>,
	"Rosen, Brian" <Brian.Rosen@neustar.biz>, <sip@ietf.org>
Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location header
	URI- http
Date: Wed, 19 Jul 2006 20:33:17 -0400
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Suppose the L7 protocol defines an optional parameter foo.

What is the dereferencer to do?  If it tries foo, but the URI is to a raw
GET processor, it will get a generic error.  There is nothing else it could
do though, since it can't tell from anything it has that foo is optional,
but legal.

A registry and a parameter would tell it that foo is legal (if it was an L7
protocol URI).

I'm not excluding a simple GET; I'm just advocating a parameter and a
registry that says it IS a simple GET.

Brian

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Wednesday, July 19, 2006 8:15 PM
> To: Rosen, Brian; Brian Rosen; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location
> header URI- http
> 
> 
> What it means is that you MUST provide a Get mechanism with no
> parameters, you MAY provide anything you like on top of that. If you are
> prepared to take the hit with trying something specific first then that
> is your prerogative, or you may decide to do the GET first, and
> something special as a second-try, which is also okay. I simply don't
> agree with your reasoning to exclude a simple GET mechanism now.
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Thursday, 20 July 2006 10:09 AM
> > To: Winterbottom, James; Brian Rosen; sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: RE: [Sip] RE: [Geopriv] More on the contents of the Location
> > header URI- http
> >
> > Actually, it means that a base GET/return PIDF is ALL you could
> > implement; you can't get anything back from the PIDF that would tell
> you
> > anything else is acceptable, and you can't tell from the URI.
> >
> > That would mean a SOAP based mechanism could never be specified, for
> > example.  You couldn't have an optional parameter unless you were
> > prepared to try it, have it fail, and then try something else.
> >
> > Brian
> >
> >
> > > -----Original Message-----
> > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > Sent: Wednesday, July 19, 2006 8:00 PM
> > > To: Brian Rosen; Rosen, Brian; sip@ietf.org
> > > Cc: geopriv@ietf.org
> > > Subject: RE: [Sip] RE: [Geopriv] More on the contents of the
> Location
> > > header URI- http
> > >
> > > Providing all mechanisms support a base GET AND they return a
> PIDF-LO
> > > there is no problem. That is all that needs to be said.
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > Sent: Thursday, 20 July 2006 9:52 AM
> > > > To: Winterbottom, James; Rosen, Brian; sip@ietf.org
> > > > Cc: geopriv@ietf.org
> > > > Subject: RE: [Sip] RE: [Geopriv] More on the contents of the
> > Location
> > > > header URI- http
> > > >
> > > > The problem is that HTTP is used as a transport protocol by
> several
> > > > mechanisms, and you need to know from the contents of the location
> > > header
> > > > which mechanism is used.  Is it SOAP, raw GET, LCP, RELO, HELD or
> > > > something
> > > > else, all of which would say "https:blahblah"?
> > > >
> > > > Brian
> > > >
> > > > > -----Original Message-----
> > > > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > > > Sent: Wednesday, July 19, 2006 7:26 PM
> > > > > To: Rosen, Brian; sip@ietf.org
> > > > > Cc: geopriv@ietf.org
> > > > > Subject: [Sip] RE: [Geopriv] More on the contents of the
> Location
> > > header
> > > > > URI- http
> > > > >
> > > > > Brian,
> > > > >
> > > > > I am not sure that I agree with you on this one.
> > > > > The problem is not in the request protocol at all, the GET works
> > > fine
> > > > > providing the return type is always a PIDF-LO.
> > > > >
> > > > > Cheers
> > > > > James
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > > > Sent: Thursday, 20 July 2006 7:33 AM
> > > > > > To: sip@ietf.org
> > > > > > Cc: geopriv@ietf.org
> > > > > > Subject: [Geopriv] More on the contents of the Location header
> > URI
> > > -
> > > > > http
> > > > > >
> > > > > > I started a discussion of what can be in the location header
> > > besides a
> > > > > > cid:.
> > > > > > We remain convinced that AbsoluteURI is a poor choice; we must
> > be
> > > more
> > > > > > specific, including privacy concerns.
> > > > > >
> > > > > > It is now clear that simply saying "https:" is not enough.
> > There
> > > > > could
> > > > > > be multiple location resolution protocols that use http
> > transport.
> > > We
> > > > > > need something else that would differentiate which one.  The
> > > simple
> > > > > GET
> > > > > > mechanism we proposed would need at least a parameter of some
> > > sort.
> > > > > >
> > > > > > After thinking about it for a while, I think the best
> resolution
> > > is
> > > > > > Andy's suggestion of a registry, used in conjunction with a
> > > parameter.
> > > > > > I'm ambivalent on whether the parameter is a URI parameter or
> a
> > > > > > parameter on the Location header.  We would define the
> registry
> > > (and
> > > > > the
> > > > > > parameter) in location-conveyance and create one entry (the
> > simple
> > > GET
> > > > > > with PIDF-LO return).  It may be, for example, that the
> proposed
> > > L7
> > > > > > location acquisition protocol defines another http based
> > > resolution
> > > > > > protocol.  That document could define another entry in the
> > > registry.
> > > > > >
> > > > > > Brian
> > > > > >
> > > > > > _______________________________________________
> > > > > > Geopriv mailing list
> > > > > > Geopriv@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > > >
> > > > >
> > >
> >
> ------------------------------------------------------------------------
> > > > --
> > > > > ----------------------
> > > > > This message is for the designated recipient only and may
> > > > > contain privileged, proprietary, or otherwise private
> information.
> > > > > If you have received it in error, please notify the sender
> > > > > immediately and delete the original.  Any unauthorized use of
> > > > > this email is prohibited.
> > > > >
> > >
> >
> ------------------------------------------------------------------------
> > > > --
> > > > > ----------------------
> > > > > [mf2]
> > > > >
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol
> > > > > Use sip-implementors@cs.columbia.edu for questions on current
> sip
> > > > > Use sipping@ietf.org for new developments on the application of
> > sip
> > > >
> > >
> > >
> >
> ------------------------------------------------------------------------
> > --
> > > ----------------------
> > > This message is for the designated recipient only and may
> > > contain privileged, proprietary, or otherwise private information.
> > > If you have received it in error, please notify the sender
> > > immediately and delete the original.  Any unauthorized use of
> > > this email is prohibited.
> > >
> >
> ------------------------------------------------------------------------
> > --
> > > ----------------------
> > > [mf2]
> 
> --------------------------------------------------------------------------
> ----------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
> --------------------------------------------------------------------------
> ----------------------
> [mf2]


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Thu Jul 20 02:06:34 2006
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Subject: RE: [Sip] SIP Security mechanism?
Date: Thu, 20 Jul 2006 08:04:25 +0200
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Thread-Topic: [Sip] SIP Security mechanism?
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From: "Fries, Steffen" <steffen.fries@siemens.com>
To: "Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>, <fwmiller@cornfed.com>
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Hi,

there have been discussions about reusing a symmetric key for S/MIME in
the SIPPING area a while ago. The draft is meanwhile expired but maybe
interesting for this. Ist draft-ono-sipping-smime-keyreuse-01.txt =20

Ciao
	Steffen

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: Wednesday, July 19, 2006 11:03 PM
> To: fwmiller@cornfed.com
> Cc: sip@ietf.org
> Subject: Re: [Sip] SIP Security mechanism?
>=20
> Hi,
>=20
> it is true that the recent security work in SIP focused on=20
> media security.
>=20
> Some of the proposed mechanisms could also provide security=20
> for SIP signaling if signaling messages are exchanged between=20
> the end points (after the initial exchange via proxies that=20
> carries, for example, a finger print attribute).
>=20
> Ciao
> Hannes
>=20
>=20
> fwmiller@cornfed.com wrote:
> > The idea is the general concept of doing some kind of key exchange=20
> > between the UAs prior to doing signaling.  This would address the=20
> > problem of maintaining the public key infrastructure.
> >=20
> > If that particular algorithm and key set does not work,=20
> perhaps some=20
> > algoritm that had the "UAS" generate an asymmetric public key=20
> > dynamically would work.  The key exchange could be for each=20
> endpoint=20
> > to have the other endpoint generate an asymmetric key?  I'm not a=20
> > crypto person but this seems like it might be doable, does=20
> any crypto=20
> > person here know of how that could be done?
> >=20
> >=20
> > FM
> >=20
> >=20
> >=20
> > ------------------------------------------------ On Wed, 19 Jul 2006
> > 21:37:47 +0200, Hannes Tschofenig <Hannes.Tschofenig@gmx.net> wrote:
> >=20
> >=20
> >> If you run a DH then the outcome is a symmetric key.=20
> S/MIME does not=20
> >> work with symmetric keys.
> >>=20
> >> Ciao Hannes
> >>=20
> >> fwmiller@cornfed.com wrote:
> >>=20
> >>> I've been reading about the ZRTP lately.  I read=20
> Henning's comment=20
> >>> at the IETF meeting about how S/MIME relies on the=20
> receiver's public=20
> >>> key to be available to allow for encryption/decryption to work.
> >>>=20
> >>> Why couldnt we put in place a mechanism that did a key=20
> exchange for=20
> >>> SIP signaling similar to whats done in ZRTP (i.e. modified
> >>> Diffie-Hellman) prior to any signaling traffic, then use the=20
> >>> exchanged keys as the basis for S/MIME encryption of the bodies?
> >>> Or we could even get really carried away and use it to do=20
> encryption=20
> >>> of the entire SIP message if the signaling did not=20
> traverse proxies?=20
> >>> This could be useful for encrypting UDP signaling, again=20
> if no proxy=20
> >>> were involved.
> >>>=20
> >>> Thoughts? FM
> >>>=20
> >>>=20
> >>> _______________________________________________ Sip mailing list=20
> >>> https://www1.ietf.org/mailman/listinfo/sip This list is for NEW=20
> >>> development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current=20
> sip Use =20
> >>> sipping@ietf.org for new developments on the application of sip
> >>>=20
> >>>=20
> >>=20
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From sip-bounces@ietf.org Thu Jul 20 04:21:43 2006
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Subject: [Sip] Re: [Geopriv] Routing of calls based on location
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Hi Brian,

thanks for raising this issue. Please find my comments below:

Brian Rosen wrote:
> NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU REPLY, SEND IT ONLY TO
> GEOPRIV
> 
> The SIP Location Conveyance document anticipates the ability to route a call
> based on the location of the caller.  The original impetus for this is
> emergency calling.  In ecrit, a mechanism has been defined (LoST) that takes
> a service request (in the form of a newly defined service urn) and a
> location and returns a URI.  This mechanism is not limited to emergency
> calling, and could be used for arbitrary location based routing by defining
> new service urns.
> 
> However, this opens up the issue of geopriv privacy concerns because the
> sender does not know the identity of the element (proxy) that will route the
> call using LoST.  It may be that the UAC itself does this, but it may be
> some other intermediary.

Actually, I think we need to be a bit more precise here. There are two 
cases:

(a) End Host runs LoST

(b) Proxy runs LoST

In case of (a) there is no problem. The problem starts with (b).

This was actually one of the reasons why I started the mailing list 
threat about ' Explicit Indication of Location Object  Consumer' (see
http://www1.ietf.org/mail-archive/web/sip/current/msg14356.html).

> 
> If the identity of the intermediary is not known, then protecting the
> location information becomes more difficult, because we cannot encrypt using
> keying material from the intended recipient.

In many cases the identity of the intermediary is known with (b) since 
it is the outbound proxy. The identity of the final recipient is for 
sure not known.

In fact there are at least two intendent recipients here:

* final recipient (i.e., the entity behind the URI returned by LoST)
* proxy that needs to inspect location information in order to perform 
location based routing.

So, in previous mailing list conversations we discussed whether these 
two entities should be treated separately, i.e., by putting two PIDF-LOs 
in the message; one for the intermediary and one for the end point. From 
a security point of view it seems to be a waste because you might not 
know both entities in advance anyway.


   In fact, we would need to use
> hop-by-hop security, with transitive trust.

You might be able to know the location based routing proxy.

What happens if you send a SIP signaling message with a service URN 
parameter to your outbound proxy but it does not have the location-based 
routing capability?

In some sense you need to know already ahead of time whether this 
functionality is support.


   This is what the emergency case
> does, but it also allows no channel security if necessary (usually meaning,
> try it with TLS and if that fails, try again without TLS).

The emergency case is somewhat special because you additionally have to 
consider what happens if none of the security works/can be used.

> 
> We propose to explicitly allow hop-by-hop security when sending location
> using SIP for the purpose of location based routing.



   We will specify you
> MUST use TLS if sending location by reference (and there will be some words
> about the choice of URI construction which is raised in another email I am
> about to send).

I think that this is largely orthogonal issue.

> 
> An important issue that we need to discuss is what "Do Not Distribute" means
> in this case.  We would like it to allow the data to be passed onward.
> Indeed, SIP proxies shouldn't be dropping headers.  We propose that it means
> that it cannot be distributed except that the location header (and body of
> course) is passed as it normally would.

Let us look at the scenario where the end host attaches location 
information.

It would set the policies and then it knows that he is going to request 
an intermediary to perform location-based routing.

In this case preventing further distribution of the PIDF-LO would 
essentially mean that the proxy removes it and sends the message to the 
  URI provided by LoST. Now, it depends on the application whether this 
is a desired functionality but I could imagine it is. I might want to 
find the nearest "something" but I don't want to tell them where I am 
because it is not relevant for them. This does not apply to emergency 
calls but that's a separate story.

In fact there more than just remove the PIDF-LO or keep it. In some 
sense this aspect is similar to the discussed usage of 2 PIDF-LOs in a 
message where one is used for routing and the other one is used for the 
final recipient (both might have a different precision). The proxy might 
also be told to reduce the granularity of location information by the 
user before forwarding the message.

> Note that although this is an issue that concerns the SIP working group,
> because the location conveyance document is a sip document, it really is a
> geopriv issue.  It also affects ecrit, since we are using its protocol
> beyond its initial purpose.  We ask that this discussion be held on the
> geopriv list, and not copied to either sip or ecrit.
> 
> Brian

Ciao
Hannes

> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From sip-bounces@ietf.org Thu Jul 20 05:34:51 2006
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Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
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Hi Brian,

in a previous mail discussion you told me that we should not create new 
requirements on the fly.

We are very early in the discussion about the Subscription URI concept 
in the Geopriv-L7 design team. It will take much longer to discuss it. 
It would make more sense to compile a separate draft, if someone wants it.

In the location conveyance document I would like to see a simple HTTPS 
based mechanism (if you think that the proxy behavior belongs into this 
document). This would prevent further delaying the document.

Ciao
Hannes

Brian Rosen wrote:
> Suppose we allowed sip: or prez: as the location header URI.  The resolution
> of this scheme would be "subscribe to presence".
> 
> Note that if you asked for immediate expiration (0), you will get the
> immediate NOTIFY.  This constitutes one reasonable way to provide
> location-by-reference.  It is a form of "pull" (you send a SUBSCRIBE, you
> get one NOTIFY in return with the PIDF-LO).  If your subscription has a
> longer expiration, it is a form of "push", and you might want to have a
> filter to limit how many pushes you get.
> 
> Note that normally, presence provides identity as well as location.  It may
> be that in some uses for the Location header, such as a location based
> routing, the identity information is not appropriate.  The policy of the
> presence server for that use may be to only return the LO part of the PIDF.
> Also note that the form of the URI itself could inadvertently provide
> identity information (prez:brian.rosen@neustar.biz), which might not be
> desirable.  If the idea to allow sip: and prez: is acceptable in the
> Location header URI, the text would contain a discussion of this issue.
> 
> Brian
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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From sip-bounces@ietf.org Thu Jul 20 06:02:37 2006
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Subject: Re: [Sip] RE: [Geopriv] More on the contents of the Location header
	URI- http
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That's fine and there are ways to address this issue. A registry would 
be fine for me with a single mandatory to implement mechanism.

Ciao
Hannes

Brian Rosen wrote:
> The problem is that HTTP is used as a transport protocol by several
> mechanisms, and you need to know from the contents of the location header
> which mechanism is used.  Is it SOAP, raw GET, LCP, RELO, HELD or something
> else, all of which would say "https:blahblah"?
> 
> Brian
> 
> 
>>-----Original Message-----
>>From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
>>Sent: Wednesday, July 19, 2006 7:26 PM
>>To: Rosen, Brian; sip@ietf.org
>>Cc: geopriv@ietf.org
>>Subject: [Sip] RE: [Geopriv] More on the contents of the Location header
>>URI- http
>>
>>Brian,
>>
>>I am not sure that I agree with you on this one.
>>The problem is not in the request protocol at all, the GET works fine
>>providing the return type is always a PIDF-LO.
>>
>>Cheers
>>James
>>
>>
>>>-----Original Message-----
>>>From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
>>>Sent: Thursday, 20 July 2006 7:33 AM
>>>To: sip@ietf.org
>>>Cc: geopriv@ietf.org
>>>Subject: [Geopriv] More on the contents of the Location header URI -
>>
>>http
>>
>>>I started a discussion of what can be in the location header besides a
>>>cid:.
>>>We remain convinced that AbsoluteURI is a poor choice; we must be more
>>>specific, including privacy concerns.
>>>
>>>It is now clear that simply saying "https:" is not enough.  There
>>
>>could
>>
>>>be multiple location resolution protocols that use http transport.  We
>>>need something else that would differentiate which one.  The simple
>>
>>GET
>>
>>>mechanism we proposed would need at least a parameter of some sort.
>>>
>>>After thinking about it for a while, I think the best resolution is
>>>Andy's suggestion of a registry, used in conjunction with a parameter.
>>>I'm ambivalent on whether the parameter is a URI parameter or a
>>>parameter on the Location header.  We would define the registry (and
>>
>>the
>>
>>>parameter) in location-conveyance and create one entry (the simple GET
>>>with PIDF-LO return).  It may be, for example, that the proposed L7
>>>location acquisition protocol defines another http based resolution
>>>protocol.  That document could define another entry in the registry.
>>>
>>>Brian
>>>
>>>_______________________________________________
>>>Geopriv mailing list
>>>Geopriv@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>
>>--------------------------------------------------------------------------
>>----------------------
>>This message is for the designated recipient only and may
>>contain privileged, proprietary, or otherwise private information.
>>If you have received it in error, please notify the sender
>>immediately and delete the original.  Any unauthorized use of
>>this email is prohibited.
>>--------------------------------------------------------------------------
>>----------------------
>>[mf2]
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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From sip-bounces@ietf.org Thu Jul 20 08:12:40 2006
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Hi,

there is some weird thing aroud To header (analogously all headers of
the similar form). Consider now To header. Its grammar is the following:

To =3D ( "To" / "t" ) HCOLON ( name-addr / addr-spec ) *( SEMI to-param =
)

where
to-param  =3D tag-param / generic-param
tag-param =3D "tag" EQUAL token
name-addr =3D [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec =3D SIP-URI / SIPS-URI / absoluteURI

SIP-URI =3D "sip:" [ userinfo ] hostport uri-parameters [ headers ]
uri-parameters =3D *( ";" uri-parameter)


The problem lies in the fact that both To header and SIP-URI may have
own parameters. Using above mentioned grammar there is possibility to
construct To header where there is no chance to distinguish between
SIP-URI parameters and To header parameters. Se example bellow:

SIP-URI =3D sip:wolf@fairytale.com;uriparam=3Dvalue
Additional To header param =3D tparam=3Dvalue

and the complete and valid header is
To: sip:wolf@fairytale.com;uriparam=3Dvalue;tparam=3Dvalue

and again, there is now no chance to distinguish whether
- all parameters are SIP-URI specific
- one parameter is SIP-URI specific and one To header specific
- all parameters are To header specific


br, Martin



+-------------------------------------+
  Martin Hynar
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  TietoEnator
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From sip-bounces@ietf.org Thu Jul 20 08:13:03 2006
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(As WG chair)

Previous mails on the SIP list have expressed the view that the STUN keepalive mechanism should be documented separately as it is a useful mechanism in its own right.

See http://www3.ietf.org/proceedings/06jul/slides/sip-5.pdf slide 5 for the issue as raised at the IETF #66 meeting.	

After discussion with the editor, the WG chairs propose the following:

-	that the documentation on STUN keepalive will REMAIN in the outbound document, and WILL NOT be split to a separate specification;

-	that the STUN keepalive mechanism WILL BE separated within the outbound specification, such that it is clear that it can be implemented independently of the rest of the outbound specification, and so that the keepalive material is clearly referencable in its own right.

We believe that this meets the requirements of at least most of the working group, and for pragmatic reasons we want to work against more fragmentation of the SIP documentation.

If you have problems with this proposal and desire a different solution, then please raise your concerns to the list, or direct to the WG chairs, in response to this mail.

regards

Keith

Keith Drage
drage@lucent.com
tel: +44 1793 776249

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From sip-bounces@ietf.org Thu Jul 20 08:31:00 2006
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Subject: Re: [Sip] How to distinguish parameters in To header
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Martin.Hynar@tietoenator.com ha scritto:
> Hi,
>
> there is some weird thing aroud To header (analogously all headers of
> the similar form). Consider now To header. Its grammar is the following:
>
> To = ( "To" / "t" ) HCOLON ( name-addr / addr-spec ) *( SEMI to-param )
>
> where
> to-param  = tag-param / generic-param
> tag-param = "tag" EQUAL token
> name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
> addr-spec = SIP-URI / SIPS-URI / absoluteURI
>
> SIP-URI = "sip:" [ userinfo ] hostport uri-parameters [ headers ]
> uri-parameters = *( ";" uri-parameter)
>
>
> The problem lies in the fact that both To header and SIP-URI may have
> own parameters. Using above mentioned grammar there is possibility to
> construct To header where there is no chance to distinguish between
> SIP-URI parameters and To header parameters. Se example bellow:
>
> SIP-URI = sip:wolf@fairytale.com;uriparam=value
> Additional To header param = tparam=value
>
> and the complete and valid header is
> To: sip:wolf@fairytale.com;uriparam=value;tparam=value
>
> and again, there is now no chance to distinguish whether
> - all parameters are SIP-URI specific
> - one parameter is SIP-URI specific and one To header specific
> - all parameters are To header specific
>
>
> br, Martin
>
>   
Fortunately the way to distinguish URI parameters from header parameters 
exists!
This thread has been already discussed in sip-implementors mailing list, 
so you can check the mailing list archives.

RFC 3261 - 20.10
"When the header field value contains a display name, the URI
including all URI parameters is enclosed in "<" and ">". If no "<"
and ">" are present, all parameters after the URI are header
parameters, not URI parameters."
[cut]
"Even if the "display-name" is empty, the "name-addr" form MUST be
used if the "addr-spec" contains a comma, semicolon, or question
mark."
[cut]
"These rules for parsing a display name, URI and URI parameters, and
header parameters also apply for the header fields To and From."

So if URI is in the addr-spec form (is not encolsed in "<" and ">") all 
the parameters are header parameters.

To: sip:wolf@fairytale.com;uriparam=value -> "uriparam" is a header parameter
To: <sip:wolf@fairytale.com;uriparam=value>;headerparam=value2 -> 
"uriparam" is an URI parameter and "headerparam" is a header parameter.

-- 
Marco Ambu
Abbeynet S.p.a. <http://www.abbeynet.it>
E-mail: marco.ambu@abbeynet.it
VoIP address: <sip:210000016@sip.abbeyphone.com>

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From sip-bounces@ietf.org Thu Jul 20 08:31:50 2006
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This looks like a good solution to me.
Thank you.
Joel

At 08:11 AM 7/20/2006, Drage, Keith (Keith) wrote:
>(As WG chair)
>
>Previous mails on the SIP list have expressed the view that the STUN 
>keepalive mechanism should be documented separately as it is a 
>useful mechanism in its own right.
>
>See http://www3.ietf.org/proceedings/06jul/slides/sip-5.pdf slide 5 
>for the issue as raised at the IETF #66 meeting.
>
>After discussion with the editor, the WG chairs propose the following:
>
>-       that the documentation on STUN keepalive will REMAIN in the 
>outbound document, and WILL NOT be split to a separate specification;
>
>-       that the STUN keepalive mechanism WILL BE separated within 
>the outbound specification, such that it is clear that it can be 
>implemented independently of the rest of the outbound specification, 
>and so that the keepalive material is clearly referencable in its own right.
>
>We believe that this meets the requirements of at least most of the 
>working group, and for pragmatic reasons we want to work against 
>more fragmentation of the SIP documentation.
>
>If you have problems with this proposal and desire a different 
>solution, then please raise your concerns to the list, or direct to 
>the WG chairs, in response to this mail.
>
>regards
>
>Keith
>
>Keith Drage
>drage@lucent.com
>tel: +44 1793 776249


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From sip-bounces@ietf.org Thu Jul 20 08:31:50 2006
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Subject: [Sip] draft-ietf-sip-outbound and
	draft-johns-sip-outbound-middialog-dr aft
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(As WG chair)

We discussed this document at IETF#66.

See http://www3.ietf.org/proceedings/06jul/slides/sip-4.pdf for the presentation and http://www.ietf.org/internet-drafts/draft-johns-sip-outbound-middialog-draft-00.txt for the draft itself.

The chairs have discussed this draft with the outbound editor, and we came to the conclusion that this problem is separable from the general outbound discussion, and that outbound is still useful without addressing this problem. 

We therefore propose that the existing charter item on outbound WILL NOT address this issue.

If anyone has issues with this proposal, can they please make this known direct to the SIP WG list, or direct to the SIP WG chairs, in response to this mail.

This in no way prevents the author of http://www.ietf.org/internet-drafts/draft-johns-sip-outbound-middialog-draft-00.txt continuing to progress the work with a view to a future SIP WG charter item, and we encourage SIP WG members to review the draft and make comments on the document.

regards

Keith

Keith Drage
drage@lucent.com
tel: +44 1793 776249

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At the moment, I do not see the location by reference as unsolveable. 

There are specific security considerations that need to be taken into account, and these are by no means complete at the moment. 

I do not at the moment see putting a location within a SIP header as an alternative to location by reference; rather I see it as another potential solution which SIP have discussed and rejected in the past.

regards

Keith

> -----Original Message-----
> From: Marc Linsner [mailto:mlinsner@cisco.com]
> Sent: 19 July 2006 14:32
> To: 'Drage, Keith (Keith)'
> Cc: sip@ietf.org; geopriv@ietf.org
> Subject: RE: [Geopriv] Location-by-value in a SIP Location Header
> 
> 
> Keith,
> 
> A couple of issues with your statements below:
> 
> - As you point out in #2, the SIP group has identified two mechanisms,
> location by value in the body and location by reference in 
> the header.  The
> issue stems from the fact that location by reference is yet to be a
> mechanism.  This has been mired down with technical 
> discussions in GeoPriv
> for a long time.
> 
> - Your list of reasons needs to be expanded to include 'the identified
> mechanism has no technical solution'.
> 
> As you express, there are many reasons to complete this work 
> quickly, but to
> have a total solution it is important to examine these alternatives
> mechanisms.  Like you, I certainly hope this comes to a 
> resolution very
> soon.
> 
> Thanks,
> 
> -Marc-
> 
> 
> 
> > 
> > (As SIP WG chair)
> > 
> > Just to set some limits to this discussion, let me state 
> > where we are in the progression of this work item. Remember 
> > it is now being classed as urgent and we need to get an RFC 
> > out of the door, and the SIP chairs are working with the 
> > GEOPRIV chairs as to the quickest way to do this:
> > 
> > 1)	The requirements in the document have been approved by 
> > the SIPPING group (a long time ago) and therefore we should 
> > not be inventing new requirements without a clear use case 
> > that has clear agreement.
> > 
> > 2)	The SIP group have already identified that two 
> > mechanisms to support these requirements should be documented 
> > (e.g. the location by value in a body, and the location by 
> > reference in a header. These two mechanisms appear to support 
> > all the current requirements. In other words, SIP have agreed 
> > the technical solution, and we are in the stage of trying to 
> > document it.
> > 
> > 3)	Using a data URL appears to add a third mechanism, if 
> > only because it will at least need to have a different set of 
> > security considerations documented for it. 
> > 
> > We should only be adding a third mechanism if:
> > 
> > a)	the existing mechanism can be shown not to meet all the 
> > requirements; or
> > b)	the use case where it offers advantages is clearly 
> > identified; and
> > c)	the compatibility issues are addressed (i.e. does the 
> > UAS need to support all of what is now three mechanisms or is 
> > there some other support requirement); and
> > d)	this can be shown as impossible to be discussed as a 
> > future extension.
> > 
> > Could I also suggest (in addition to the above) that if 
> > people want to proceed with this discussion, it would be 
> > appropriate to submit an i-d targeted at the SIP WG 
> > identifying the technical solution proposed (with as much 
> > completeness as is possible). And remember if you think this 
> > discussion is needed, it needs to occur quickly. 
> > 
> > regards
> > 
> > Keith
> > 
> > > -----Original Message-----
> > > From: Thomson, Martin [mailto:Martin.Thomson@andrew.com]
> > > Sent: 19 July 2006 00:15
> > > To: Rosen, Brian; Henning Schulzrinne
> > > Cc: sip@ietf.org; geopriv@ietf.org
> > > Subject: RE: [Geopriv] Location-by-value in a SIP Location Header
> > > 
> > > 
> > > If you want technically feasible, I can post a URI to the 
> list that 
> > > includes a complete PIDF-LO with DSig.  It's a little long though.
> > > 
> > > I'm not suggesting that this a good idea, but I see no reason to 
> > > prevent it.
> > > 
> > > > -----Original Message-----
> > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > Sent: Tuesday, 18 July 2006 11:07 PM
> > > > To: Henning Schulzrinne
> > > > Cc: sip@ietf.org; geopriv@ietf.org
> > > > Subject: RE: [Geopriv] Location-by-value in a SIP 
> Location Header
> > > > 
> > > > Perhaps some enlightenment is in order.  It's not clear to
> > > me how you
> > > > dsig a Data uri in a Location header (or maybe you mean
> > > dsig the pidf-lo
> > > > and put both the pidf-lo and the signature in the Data 
> > URI) somehow?
> > > > Are there some examples of this kind of thing elsewhere?
> > > > 
> > > > If we are ignoring technical arguments (as opposed to use
> > > cases), it's
> > > > out of forgetfulness, and not willfulness.  What were you
> > > referring to?
> > > > 
> > > > Brian
> > > > 
> > > > 
> > > > 
> > > > -----Original Message-----
> > > > From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> > > > Sent: Tuesday, July 18, 2006 8:59 AM
> > > > To: Rosen, Brian
> > > > Cc: sip@ietf.org; geopriv@ietf.org
> > > > Subject: Re: [Geopriv] Location-by-value in a SIP 
> Location Header
> > > > 
> > > > Brian, James:
> > > > 
> > > > this is simply false. If desired, XML-DSIG is a well-known and 
> > > > accepted mechanism to achieve integrity. Such 
> specification is no 
> > > > longer than the MUST NOT that you propose. You are also 
> willfully 
> > > > ignoring the other technical arguments made as part of this
> > > discussion.
> > > > 
> > > > Henning
> > > > 
> > > > On Jul 18, 2006, at 8:48 AM, Rosen, Brian wrote:
> > > > 
> > > > > There was a desire expressed by Hannes to provide a way
> > > for a proxy to
> > > > > insert location-by-value.  He had proposed a SIP 
> > Location header 
> > > > > specific way.  In IETF66, Henning suggested just using 
> > a data URI.
> > > > >
> > > > > Doing this may run afoul of the geopriv location 
> > privacy concerns, 
> > > > > because there is no way for the user to sign the header 
> > to provide 
> > > > > assurance that the PIDF, and specifically the retention 
> > and other 
> > > > > policy bits are preserved.  Note that S/MIME provides 
> sufficient
> > > integrity
> > > > > protection for a body.  It's probably okay to use TLS 
> > per hop to 
> > > > > provide privacy, but not integrity protection.  The authors
> > > believe that there
> > > > > is no really compelling use case for this feature, and
> > > the effort to
> > > > > define acceptable security mechanisms for location data
> > > in a header
> > > > > would be a significant effort, further delaying the draft.
> > > > >
> > > > > We propose, therefore, to state in 
> > sip-location-conveyance, that a 
> > > > > Data URI MUST NOT be used in the Location header until a
> > > standards track
> > > > > RFC
> > > > > defines a suitable security mechanism to protect the 
> > PIDF in the 
> > > > > header.
> > > > >
> > > > > Brian and James
> > > > >
> > > > > _______________________________________________
> > > > > Geopriv mailing list
> > > > > Geopriv@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > > 
> > > > 
> > > > _______________________________________________
> > > > Geopriv mailing list
> > > > Geopriv@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > 
> > > --------------------------------------------------------------
> > > ----------------------------------
> > > This message is for the designated recipient only and may contain 
> > > privileged, proprietary, or otherwise private information.
> > > If you have received it in error, please notify the sender 
> > immediately 
> > > and delete the original.  Any unauthorized use of this email is 
> > > prohibited.
> > > --------------------------------------------------------------
> > > ----------------------------------
> > > [mf2]
> > > 
> > 
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> 

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From sip-bounces@ietf.org Thu Jul 20 11:00:56 2006
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Henning Schulzrinne wrote:
> a four-step process:
> 
> - take any PIDF-LO
> - wrap in XML-DSIG as specified by the relevant W3C recommendation
> - convert to base-64
> - make into a data URL, as in the example I gave in an earlier message
> 
>   data:application/pidf+xml;base64,<base64-data>

Here's a rather naive question... what's the point of XML-DSIG?

-andy


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Drage, Keith (Keith) wrote:
> We will need to consider than outside this mail. Personally I think by reference works better for the 3G architecture in this use case. There is certainly no explicit 3GPP requirement at the moment.


Actually, there are many environments where reference works better.

-andy


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Hi All,

In RFC 3261 the construction of ACK requestURI MUST conatain original
request as it said in below paragraph
*17.1.1.3 Construction of the ACK Request*

The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request").

But, in RFC 3665 the contruction of RequestURI for ACK is taken from
ConatctHeader of 200 OK as shown below. Could I know which is the
approipraite one. Should I go for 3261 or Informational RFC 3665

*RFC 3665 3.1. Successful Session Establishment*
INVITE sip:bob@biloxi.example.com SIP/2.0

200 OK contact Header: <sip:bob@client.biloxi.example.com>
ACK sip:bob@client.biloxi.example.com SIP/2.0

thanks in advance

regards

Sudhir

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<div>Hi All,</div>
<div>&nbsp;</div>
<div>In RFC 3261 the construction of ACK requestURI&nbsp;MUST conatain original request as it said in below paragraph&nbsp;</div>
<div><strong><a href="http://17.1.1.3">17.1.1.3</a> Construction of the ACK Request</strong></div>
<div>
<p>The ACK request constructed by the client transaction MUST contain<br>values for the Call-ID, From, and Request-URI that are equal to the<br>values of those header fields in the request passed to the transport<br>by the client transaction (call this the &quot;original request&quot;).
</p>
<p>But, in RFC 3665 the contruction of RequestURI for ACK is taken from ConatctHeader of 200 OK as shown below. Could I know which is the approipraite one. Should I go for 3261 or Informational RFC 3665</p>
<p><strong>RFC 3665 3.1. Successful Session Establishment</strong><br>INVITE <a href="mailto:sip:bob@biloxi.example.com">sip:bob@biloxi.example.com</a> SIP/2.0</p>
<p>200 OK contact Header: &lt;<a href="mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.example.com</a>&gt;<br>ACK <a href="mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.example.com</a> SIP/2.0
</p>
<p>thanks in advance</p>
<p>regards</p>
<p>Sudhir</p>
<p>&nbsp;</p>
<p>&nbsp;</p></div>

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Much as I'd love to dispute this, I think it's a settled issue; we're
supporting location-by-reference in location-conveyance.  I don't think we
need to continue to debate this.  It's in there.

What is still at issue is whether location-by-value in a header is in there.
We need a real use case.  I'm having trouble understanding how you can have
a SIP proxy that can get a call to the outside of some domain, but can't
provide a server to resolve a reference.

If we have bought off on location-by-value in a body, and
location-by-reference in a header, what case is not covered by those two
options?
Much as I would prefer proxy insertion by value over proxy insertion by
reference (even if I think proxy insertion is a bad idea, that's another
settled issue), I think a proxy can always use by-reference.  Fewer options
are better.

Brian



> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Thursday, July 20, 2006 10:25 AM
> To: Drage, Keith (Keith)
> Cc: sip@ietf.org; geopriv@ietf.org
> Subject: Re: [Geopriv] Location-by-value in a SIP Location Header
> 
> Drage, Keith (Keith) wrote:
> > We will need to consider than outside this mail. Personally I think by
> reference works better for the 3G architecture in this use case. There is
> certainly no explicit 3GPP requirement at the moment.
> 
> 
> Actually, there are many environments where reference works better.
> 
> -andy
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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Brian Rosen wrote:
> Much as I'd love to dispute this, I think it's a settled issue; we're
> supporting location-by-reference in location-conveyance.  I don't think we
> need to continue to debate this.  It's in there.
> 
> What is still at issue is whether location-by-value in a header is in there.
> We need a real use case.  I'm having trouble understanding how you can have
> a SIP proxy that can get a call to the outside of some domain, but can't
> provide a server to resolve a reference.
> 
> If we have bought off on location-by-value in a body, and
> location-by-reference in a header, what case is not covered by those two
> options?
> Much as I would prefer proxy insertion by value over proxy insertion by
> reference (even if I think proxy insertion is a bad idea, that's another
> settled issue), I think a proxy can always use by-reference.  Fewer options
> are better.

I agree.  Fewer options are better.

-andy


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From sip-bounces@ietf.org Thu Jul 20 13:16:23 2006
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Date: Thu, 20 Jul 2006 19:02:45 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
To: Brian Rosen <br@brianrosen.net>,
	"'Thomson, Martin'" <Martin.Thomson@andrew.com>, sip@ietf.org
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Brian,

To me using ABNF rules to restrict allowable schemes is not a good idea. AB=
NF defines syntax, not semantics

Compare this with the Route header in SIP. The ABNF allows any scheme, text=
 says it MUST be sip: or sips:. That suffices (you could define which error=
 response must be sent)

Moreover, the protocol to use is a policy decision. You can make recommenda=
tions, even define new mechanisms, but I think the choice should be left to=
 implementors

Regards,

Jeroen
-----Oorspronkelijk bericht -----
Van: "Brian Rosen" <br@brianrosen.net>
Aan: "'Thomson, Martin'" <Martin.Thomson@andrew.com>; sip@ietf.org
Verzonden: 20-7-06 18:19
Onderwerp: RE: [Sip] Allowing sip: (or prez:) in the location header

There is a point to limiting what can be sent in the Location header to
specific protocols which has received adequate review in both sip and
geopriv.

So, here is my concrete proposal:
Location       =3D "Location" HCOLON location-param *(COMMA location-param)
location-param =3D LAQUOT allowed-scheme HCOLON ":" ( hier-part / opaque-pa=
rt
) RAQUOT *( SEMI loc-param ) ; hier-part and opaque-part from 3261
allowed-scheme =3D 'cid' / 'http' / 'https' / 'sip' / 'sips' / 'prez'
loc-param      =3D 'location-dereference-protocol=3D' deref-protocol /
generic-param
deref-protocol =3D token

allowed-scheme may be extended by a standards-track RFC.  Such an extension
is NOT considered a SIP extension and is NOT subject to the process defined
by RFC3427.  Corresponding resolution processes for such schemes MUST be
included in the RFC, and MAY be subject to review against RFC3693.

This document (in Section ?) defines a new IANA registry for deref-protocol=
,
and populates it with one initial value, 'rawGET'.  rawGET indicates that
the location reference may be dereferenced by sending a GET with no
parameters on the URI, using HTTPS (except in the emergency case, where HTT=
P
is permitted), and expects to receive one PIDF-LO document (with a MIME-typ=
e
of "application/pidf+xml")in response.  deref-protocol MUST be included if
the scheme is HTTPS (or HTTP in the emergency case).  Other schemes may or
may not require deref-protocol.  A standards-track RFC is required to add a


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From sip-bounces@ietf.org Thu Jul 20 13:38:55 2006
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Jeroen Van Bemmel wrote:
> Brian,
> 
> To me using ABNF rules to restrict allowable schemes is not a good idea. ABNF defines syntax, not semantics
> 
> Compare this with the Route header in SIP. The ABNF allows any scheme, text says it MUST be sip: or sips:. That suffices (you could define which error response must be sent)

I agree.  The same is also true for a lot of specs that use XML Schema 
but then use 2119 language on top of that.

> Moreover, the protocol to use is a policy decision. You can make recommendations, even define new mechanisms, but I think the choice should be left to implementors

I disagree with this.  What is my server gonna do with a mailto: or dns: 
URI?  There should be a small list of acceptable choices if we want 
interoperability.

-andy


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Date: Thu, 20 Jul 2006 19:37:08 +0200
From: =?iso-8859-1?Q?Gustavo_Garc=EDa_Bernardo?= <ggb@tid.es>
Subject: RE: [Sip] Query reagrding RFC 3665
In-reply-to: <52db68bc0607200619o50341173uabe9f9e29881e0a4@mail.gmail.com>
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	boundary="Boundary_(ID_eHnJxWg+rsXEe7OaPzgi0w)"

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The rules in RFC3261 17.1.1.3 are for ACKs created for non 2XX INVITE
responses (ACKs created in transactions layer).  For 2XX responses, the =
ACKs
are created in UA layer, using the same rules as any other request =
inside a
established dialog.

=20

=93The UAC uses the remote target and route set to build the Request-URI

   and Route header field of the request.=94 (RFC3261 12.2.1.1 =
Generating the
Request)

=20

Where the remote target is the Contact received in 2XX response.

=20

So, the second one (according to RFC3665 too), is the correct.

=20

G.

=20

  _____ =20

De: Sudhir [mailto:t.sudhirkumar@gmail.com]=20
Enviado el: jueves, 20 de julio de 2006 15:19
Para: sip@ietf.org
Asunto: [Sip] Query reagrding RFC 3665

=20

Hi All,

=20

In RFC 3261 the construction of ACK requestURI MUST conatain original
request as it said in below paragraph=20

17.1.1.3 Construction of the ACK Request

The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request").=20

But, in RFC 3665 the contruction of RequestURI for ACK is taken from
ConatctHeader of 200 OK as shown below. Could I know which is the
approipraite one. Should I go for 3261 or Informational RFC 3665

RFC 3665 3.1. Successful Session Establishment
INVITE sip:bob@biloxi.example.com SIP/2.0

200 OK contact Header: <sip:bob@client.biloxi.example.com>
ACK sip:bob@client.biloxi.example.com SIP/2.0=20

thanks in advance

regards

Sudhir

=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>The rules in =
RFC3261
17.1.1.3 are for ACKs created for non 2XX INVITE responses (ACKs created =
in
transactions layer). =A0For 2XX responses, the ACKs are created in UA =
layer,
using the same rules as any other request inside a established =
dialog.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>&#8220;The UAC =
uses the
remote target and route set to build the =
Request-URI<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>=A0=A0 and Route =
header field
of the request.&#8221; (RFC3261 12.2.1.1 Generating the =
Request)<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Where the remote =
target
is the Contact received in 2XX response.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>So, the second =
one (according
to RFC3665 too), is the correct.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>G.<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

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

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>De:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
Sudhir
[mailto:t.sudhirkumar@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Enviado el:</span></b> jueves, 20 de =
julio de
2006 15:19<br>
<b><span style=3D'font-weight:bold'>Para:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Asunto:</span></b> [Sip] Query =
reagrding RFC
3665</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Hi All,<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>In RFC 3261 the construction of ACK requestURI&nbsp;MUST =
conatain
original request as it said in below =
paragraph&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><strong><b><font size=3D3 face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt'><a href=3D"http://17.1.1.3">17.1.1.3</a> =
Construction of
the ACK Request</span></font></b></strong><o:p></o:p></p>

</div>

<div>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>The ACK
request constructed by the client transaction MUST contain<br>
values for the Call-ID, From, and Request-URI that are equal to the<br>
values of those header fields in the request passed to the transport<br>
by the client transaction (call this the &quot;original request&quot;). =
<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>But, in
RFC 3665 the contruction of RequestURI for ACK is taken from =
ConatctHeader of
200 OK as shown below. Could I know which is the approipraite one. =
Should I go
for 3261 or Informational RFC 3665<o:p></o:p></span></font></p>

<p><strong><b><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>RFC
3665 3.1. Successful Session =
Establishment</span></font></b></strong><br>
INVITE <a =
href=3D"mailto:sip:bob@biloxi.example.com">sip:bob@biloxi.example.com</a>=

SIP/2.0<o:p></o:p></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>200 OK
contact Header: &lt;<a =
href=3D"mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.e=
xample.com</a>&gt;<br>
ACK <a =
href=3D"mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.e=
xample.com</a>
SIP/2.0 <o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>thanks in
advance<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>regards<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>Sudhir<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

</div>

</div>

</body>

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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Date: Thu, 20 Jul 2006 13:28:52 -0400
From: Andrew Newton <andy@hxr.us>
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To: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
References: <0J2P002Y5R8KS990@jes-fe1.zx.nl>
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Jeroen Van Bemmel wrote:
> Brian,
> 
> To me using ABNF rules to restrict allowable schemes is not a good idea. ABNF defines syntax, not semantics
> 
> Compare this with the Route header in SIP. The ABNF allows any scheme, text says it MUST be sip: or sips:. That suffices (you could define which error response must be sent)

I agree.  The same is also true for a lot of specs that use XML Schema 
but then use 2119 language on top of that.

> Moreover, the protocol to use is a policy decision. You can make recommendations, even define new mechanisms, but I think the choice should be left to implementors

I disagree with this.  What is my server gonna do with a mailto: or dns: 
URI?  There should be a small list of acceptable choices if we want 
interoperability.

-andy


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Date: Thu, 20 Jul 2006 19:37:08 +0200
From: =?iso-8859-1?Q?Gustavo_Garc=EDa_Bernardo?= <ggb@tid.es>
Subject: RE: [Sip] Query reagrding RFC 3665
In-reply-to: <52db68bc0607200619o50341173uabe9f9e29881e0a4@mail.gmail.com>
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The rules in RFC3261 17.1.1.3 are for ACKs created for non 2XX INVITE
responses (ACKs created in transactions layer).  For 2XX responses, the =
ACKs
are created in UA layer, using the same rules as any other request =
inside a
established dialog.

=20

=93The UAC uses the remote target and route set to build the Request-URI

   and Route header field of the request.=94 (RFC3261 12.2.1.1 =
Generating the
Request)

=20

Where the remote target is the Contact received in 2XX response.

=20

So, the second one (according to RFC3665 too), is the correct.

=20

G.

=20

  _____ =20

De: Sudhir [mailto:t.sudhirkumar@gmail.com]=20
Enviado el: jueves, 20 de julio de 2006 15:19
Para: sip@ietf.org
Asunto: [Sip] Query reagrding RFC 3665

=20

Hi All,

=20

In RFC 3261 the construction of ACK requestURI MUST conatain original
request as it said in below paragraph=20

17.1.1.3 Construction of the ACK Request

The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request").=20

But, in RFC 3665 the contruction of RequestURI for ACK is taken from
ConatctHeader of 200 OK as shown below. Could I know which is the
approipraite one. Should I go for 3261 or Informational RFC 3665

RFC 3665 3.1. Successful Session Establishment
INVITE sip:bob@biloxi.example.com SIP/2.0

200 OK contact Header: <sip:bob@client.biloxi.example.com>
ACK sip:bob@client.biloxi.example.com SIP/2.0=20

thanks in advance

regards

Sudhir

=20

=20


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<body lang=3DES link=3Dblue vlink=3Dblue>

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>The rules in =
RFC3261
17.1.1.3 are for ACKs created for non 2XX INVITE responses (ACKs created =
in
transactions layer). =A0For 2XX responses, the ACKs are created in UA =
layer,
using the same rules as any other request inside a established =
dialog.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>&#8220;The UAC =
uses the
remote target and route set to build the =
Request-URI<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>=A0=A0 and Route =
header field
of the request.&#8221; (RFC3261 12.2.1.1 Generating the =
Request)<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Where the remote =
target
is the Contact received in 2XX response.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>So, the second =
one (according
to RFC3665 too), is the correct.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>G.<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

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

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>De:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
Sudhir
[mailto:t.sudhirkumar@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Enviado el:</span></b> jueves, 20 de =
julio de
2006 15:19<br>
<b><span style=3D'font-weight:bold'>Para:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Asunto:</span></b> [Sip] Query =
reagrding RFC
3665</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Hi All,<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>In RFC 3261 the construction of ACK requestURI&nbsp;MUST =
conatain
original request as it said in below =
paragraph&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><strong><b><font size=3D3 face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt'><a href=3D"http://17.1.1.3">17.1.1.3</a> =
Construction of
the ACK Request</span></font></b></strong><o:p></o:p></p>

</div>

<div>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>The ACK
request constructed by the client transaction MUST contain<br>
values for the Call-ID, From, and Request-URI that are equal to the<br>
values of those header fields in the request passed to the transport<br>
by the client transaction (call this the &quot;original request&quot;). =
<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>But, in
RFC 3665 the contruction of RequestURI for ACK is taken from =
ConatctHeader of
200 OK as shown below. Could I know which is the approipraite one. =
Should I go
for 3261 or Informational RFC 3665<o:p></o:p></span></font></p>

<p><strong><b><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>RFC
3665 3.1. Successful Session =
Establishment</span></font></b></strong><br>
INVITE <a =
href=3D"mailto:sip:bob@biloxi.example.com">sip:bob@biloxi.example.com</a>=

SIP/2.0<o:p></o:p></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>200 OK
contact Header: &lt;<a =
href=3D"mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.e=
xample.com</a>&gt;<br>
ACK <a =
href=3D"mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.e=
xample.com</a>
SIP/2.0 <o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>thanks in
advance<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>regards<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>Sudhir<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

</div>

</div>

</body>

</html>

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From sip-bounces@ietf.org Thu Jul 20 14:58:55 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Jeroen Van Bemmel'" <jbemmel@zonnet.nl>,
	"'Thomson, Martin'" <Martin.Thomson@andrew.com>, <sip@ietf.org>
Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
Date: Thu, 20 Jul 2006 14:57:00 -0400
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Jeroen

I don't agree that this is a semantic issue; we are restricting precisely
what may appear in allowed-scheme.  I usually think it's better to have the
ABNF precise, rather than relying on text to state what is actually
permitted on the wire.  We are talking about what is syntactically allowed,
not how you interpret it.

The ABNF =/ syntax (from RFC4234) is a convenient way to add a scheme to the
list, even when used by independently created RFCs.

I, like Andy, don't believe the decision on which protocols can appear in
allowed-scheme can be left to implementation.  Every dereferencer has to be
prepared to use every possible protocol, especially on the Internet, because
there is no way to ask for a specific URI or limit the schemes the location
acquisition mechanism may obtain.

Brian

> -----Original Message-----
> From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Thursday, July 20, 2006 1:03 PM
> To: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
> Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
> 
> Brian,
> 
> To me using ABNF rules to restrict allowable schemes is not a good idea.
> ABNF defines syntax, not semantics
> 
> Compare this with the Route header in SIP. The ABNF allows any scheme,
> text says it MUST be sip: or sips:. That suffices (you could define which
> error response must be sent)
> 
> Moreover, the protocol to use is a policy decision. You can make
> recommendations, even define new mechanisms, but I think the choice should
> be left to implementors
> 
> Regards,
> 
> Jeroen
> -----Oorspronkelijk bericht -----
> Van: "Brian Rosen" <br@brianrosen.net>
> Aan: "'Thomson, Martin'" <Martin.Thomson@andrew.com>; sip@ietf.org
> Verzonden: 20-7-06 18:19
> Onderwerp: RE: [Sip] Allowing sip: (or prez:) in the location header
> 
> There is a point to limiting what can be sent in the Location header to
> specific protocols which has received adequate review in both sip and
> geopriv.
> 
> So, here is my concrete proposal:
> Location       = "Location" HCOLON location-param *(COMMA location-param)
> location-param = LAQUOT allowed-scheme HCOLON ":" ( hier-part / opaque-
> part
> ) RAQUOT *( SEMI loc-param ) ; hier-part and opaque-part from 3261
> allowed-scheme = 'cid' / 'http' / 'https' / 'sip' / 'sips' / 'prez'
> loc-param      = 'location-dereference-protocol=' deref-protocol /
> generic-param
> deref-protocol = token
> 
> allowed-scheme may be extended by a standards-track RFC.  Such an
> extension
> is NOT considered a SIP extension and is NOT subject to the process
> defined
> by RFC3427.  Corresponding resolution processes for such schemes MUST be
> included in the RFC, and MAY be subject to review against RFC3693.
> 
> This document (in Section ?) defines a new IANA registry for deref-
> protocol,
> and populates it with one initial value, 'rawGET'.  rawGET indicates that
> the location reference may be dereferenced by sending a GET with no
> parameters on the URI, using HTTPS (except in the emergency case, where
> HTTP
> is permitted), and expects to receive one PIDF-LO document (with a MIME-
> type
> of "application/pidf+xml")in response.  deref-protocol MUST be included if
> the scheme is HTTPS (or HTTP in the emergency case).  Other schemes may or
> may not require deref-protocol.  A standards-track RFC is required to add
> a


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From sip-bounces@ietf.org Thu Jul 20 16:42:09 2006
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This is a multi-part message in MIME format.

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For scenario 1 (seeking nearest Pizza Hut) I think one would want to =
provide the location to the proxy, such that it can locate the nearest =
X. However, I may not want that X to know my location. So I would like =
to request the proxy to use my location for the lookup, and then remove =
it from the request

Hence a new requirement: it must be possible for the UAC to indicate to =
a proxy that it should remove the location information

Regards,

Jeroen
------=_NextPart_000_0037_01C6AC4C.D45FD4E0
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5450.4" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>For scenario 1 (seeking nearest Pizza =
Hut) I think=20
one would want to provide the location to the proxy, such that it can =
locate the=20
nearest X. However, I may not want that X to know my location. So I =
would like=20
to request the proxy to use my location for the lookup, and then remove =
it from=20
the request</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Hence a new requirement: it must be =
possible for=20
the UAC to indicate to a proxy that it should remove the location=20
information</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Regards,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Jeroen</FONT></DIV></BODY></HTML>

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From sip-bounces@ietf.org Thu Jul 20 16:42:21 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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References: <0J2P002Y5R8KS990@jes-fe1.zx.nl> <44BFBD54.5020108@hxr.us>
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Andrew,

>> Moreover, the protocol to use is a policy decision. You can make 
>> recommendations, even define new mechanisms, but I think the choice 
>> should be left to implementors
>
> I disagree with this.  What is my server gonna do with a mailto: or dns: 
> URI?  There should be a small list of acceptable choices if we want 
> interoperability.

In such a case your server should probably respond with "4xx Location URI 
Scheme not supported", where 4xx is TBD. However, if retrieving the location 
is not critical to completing the call, then it should probably ignore the 
header. So there seems to be a need for a parameter indicating "optional" 
versus "required"

IMO the text in the draft should say:
- proxies MUST support HTTP, SHOULD support HTTPS, MAY support cid (for 
dereferencing)
- UAS MUST support HTTP and cid, SHOULD support HTTPS (for dereferencing)
- TBD: define whether and when proxy/UAS should generate error for 
unsupported URI scheme (and then indicate which schemes are supported)
- UAC/Proxy SHOULD use HTTPS when adding Location by reference
- UAC MAY use cid for location by value, but SHOULD then ensure 
confidentiality

But above all: the draft should specify what is minimally needed to meet the 
stated requirements, while not unnecessarily restricting solutions that 
might have additional requirements which are not currently foreseen

Regards,

jeroen 


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From sip-bounces@ietf.org Thu Jul 20 17:29:25 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>, "'Andrew Newton'" <andy@hxr.us>
Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
Date: Thu, 20 Jul 2006 17:27:21 -0400
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I don't think you have the big picture.

Location comes from the access network, via some acquisition protocol, or in
some cases, can be self generated in the endpoint.   The access network
decides what the representation of the location is if it's supplying
location, although it may give an endpoint some options (value/reference,
civic/geo).

The UAC gets this location and puts it in the SIP message.

Some entity, could be a proxy for location based routing, or the UAS, gets
the location and does something with it.

For the sake of argument, assume its location-by-reference.
Call the proxy or UAS the "de-referencer".

It's the access network (could be a broadband carrier, could be an
enterprise) that is putting up the server that provides the de-reference
service.  Call the server a "Location Information Server" = LIS

Okay, so the access network chooses some protocol, makes up a suitable URI
and passes it via the acquisition protocol to the UAC.  The UAC passes it to
the de-referencer, who presents it to the LIS.  The LIS responds with the
location value.

The problem is that the de-referencer can't control which LIS, and can't
control what protocol decision the LIS made.  It has to implement EVERY ONE
of the protocols that could appear in the Location header.  If the LIS
implements a protocol the de-referencer doesn't understand, the whole
mechanism fails, with no recourse.  On the Internet, which is where we want
the protocols to work, you can't control which access network the UAC is
coming from.

So, instead, we restrict the list of allowed de-referencing protocols to a
small number, and make that list very explicit.  Now a de-referencer knows
that if he implements ALL of the listed protocols, he will be able to
dereference any location reference he gets.



There is another big picture aspect.  The whole location thing comes under
geopriv.  We have to do a sip Location header in sip, but geopriv has to
review our work.  The "priv" part of geopriv is privacy.  They care, a whole
lot, about the privacy aspects of location.  One expression of that concern
is to have the security posture of "what is not explicitly permitted is
prohibited".  They really truly want to look at every IETF protocol that is
used to de-reference location, and decide if it provides sufficient privacy
protection for location.  They really do want an explicit list, and they
want to scrutinize every protocol on that list to see how it addresses their
concerns.  SIP is what geopriv calls a "using protocol" and they review them
seriously.  This means, for example, that it's probably not okay to use HTTP
to de-reference.  There is a specific, limited exemption for emergency calls
on that.  So HTTPS is a MUST, and HTTP is MUST NOT except for the emergency
case.  I believe that "AbsoluteURI" would not be acceptable to geopriv, but
of course I can't speak for the group.

Brian

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Thursday, July 20, 2006 4:31 PM
> To: Andrew Newton
> Cc: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
> Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
> 
> Andrew,
> 
> >> Moreover, the protocol to use is a policy decision. You can make
> >> recommendations, even define new mechanisms, but I think the choice
> >> should be left to implementors
> >
> > I disagree with this.  What is my server gonna do with a mailto: or dns:
> > URI?  There should be a small list of acceptable choices if we want
> > interoperability.
> 
> In such a case your server should probably respond with "4xx Location URI
> Scheme not supported", where 4xx is TBD. However, if retrieving the
> location
> is not critical to completing the call, then it should probably ignore the
> header. So there seems to be a need for a parameter indicating "optional"
> versus "required"
> 
> IMO the text in the draft should say:
> - proxies MUST support HTTP, SHOULD support HTTPS, MAY support cid (for
> dereferencing)
> - UAS MUST support HTTP and cid, SHOULD support HTTPS (for dereferencing)
> - TBD: define whether and when proxy/UAS should generate error for
> unsupported URI scheme (and then indicate which schemes are supported)
> - UAC/Proxy SHOULD use HTTPS when adding Location by reference
> - UAC MAY use cid for location by value, but SHOULD then ensure
> confidentiality
> 
> But above all: the draft should specify what is minimally needed to meet
> the
> stated requirements, while not unnecessarily restricting solutions that
> might have additional requirements which are not currently foreseen
> 
> Regards,
> 
> jeroen


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At 10:35 PM 7/20/2006 +0200, Jeroen van Bemmel wrote:
>For scenario 1 (seeking nearest Pizza Hut) I think one would want to 
>provide the location to the proxy, such that it can locate the nearest X. 
>However, I may not want that X to know my location. So I would like to 
>request the proxy to use my location for the lookup, and then remove it 
>from the request

This can be solved by Pizza Hut responding with a 302 with the URI of 
nearest X (store, I assume).  This way, I send the new SIP Request with or 
without my Location (which will probably be needed if I want the pizza 
delivered, but not needed if I'm traveling to the store's location).

>
>Hence a new requirement: it must be possible for the UAC to indicate to a 
>proxy that it should remove the location information

given the above, I'm not sure this requirement is needed.

But it might be interesting if I can ask Pizza Hut to 3XX me to the nearest 
store.  I don't know how we would indicate that in a request to Pizza-Hut.com

>
>Regards,
>
>Jeroen
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

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At 12:19 PM 7/20/2006 -0400, Brian Rosen wrote:
>There is a point to limiting what can be sent in the Location header to
>specific protocols which has received adequate review in both sip and
>geopriv.
>
>So, here is my concrete proposal:
>Location       = "Location" HCOLON location-param *(COMMA location-param)
>location-param = LAQUOT allowed-scheme HCOLON ":" ( hier-part / opaque-part
>) RAQUOT *( SEMI loc-param ) ; hier-part and opaque-part from 3261
>allowed-scheme = 'cid' / 'http' / 'https' / 'sip' / 'sips' / 'prez'
>loc-param      = 'location-dereference-protocol=' deref-protocol /
>generic-param
>deref-protocol = token

I'm still not a fan of using 'sip' here. 'sip' or 'sips' means using 
SUBSCRIBE and that's a PULL mechanism in this scenario; i.e. the UAC that's 
subscribing to another SIP element is not using its location to do anything 
(that will come back in the NOTIFY).  The UAC (in the SUB transaction) in 
this scenario is PULLING another UA's location from somewhere, and that's 
not "Location Conveyance" - which is what I got hammered on in the Paris 
meeting.

Clear consensus from that meeting was that this document describes PUSHING 
my UAC's location to somewhere else.

Using HTTP(S) to fetch (presumably with a GET request) my 
location-by-reference once from a server is not a pull, it is a 
Content-Indirection operation, which is a fake of a PUSH, IMO.

Using a SUBSCRIBE, you are always receiving something in the NOTIFY.  If 
"location" is what's returned in that NOTIFY, this is a "Location 
Retreaval" PULL, and against what the Paris meeting said to do, and is for 
another document, a PULL mechanism document that hasn't been written yet.

Therefore, I recommend we remove sip: and sips: from the BNF above.  I'm 
still thinking about what to do about the 'pres' proposal.

Adam's scenario (on the mike in Montreal) about using SUBSCRIBE to a 
weather service did not have Adam's location coming back in the NOTIFY, so 
that use-case doesn't apply to this discussion (but is a great use-case!)

comments

James

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At 06:25 PM 7/20/2006 -0400, Brian Rosen wrote:
>Dueling authors.  Pistols at 40 paces?  Have you a second?
>
>Remind me what the religious argument against pull is?

Jon Peterson had the strongest opinion on the list, but there were several 
opinions at the Paris meeting, including Rohan (especially brutal about 
this point) and Dean.

Most seemed to hang on the idea that our doc is all about "conveying" - 
which means giving or showing something else what I have.

SUB and NOT have to do with me receiving what your location is, which is 
not conveying, by definition.

Before Paris, our doc (dear co-author  :-) discussed both conveying and 
receiving location.  I can't remember who from the SIMPLE WG got on me for 
trying to have SUB/NOT or PUBLISH in our ID.


>I'm conveying location by reference using an IETF procedure designed
>expressly to convey location: presence.

I think if you looked at this, you'd likely see that I push my presence 
location up to a compositor server, and others subscribe to that server to 
learn my presence.  Those others are pulling my location from the compositor.

>Whether I get one NOTIFY or
>multiple ones can't possibly matter to the sip protocol, the Location
>header, the security considerations or anything else I can think of.

...except the pov of the document (being about conveying).


>Look at it this way: suppose I were to say if you use sip/sips, you MUST set
>Expiry to zero.

Then, I think, the Geopriv folks will say that LoST cannot then take that 
location (in or out of a PIDF-LO) and send it to a LoST server - because it 
was distributed after the <retransmission-allowed> flag was set to "no"....

>That's plain and simple location by reference.

I completely understand the idea, but I think we may have unearthed another 
problem in the last 24 hours.

If the INVITE has a PIDF-LO with this element

         <retransmission-allowed>no</retransmission-allowed>

Can a ESRP (emergency services routing proxy that does the LoST query for 
the PSAP URI) send location anywhere?  Geopriv might say no, because it 
violates one of its "priv" rules. Now what? Do we make an exception?  We 
have to get their buy-in for this. We don't have it yet.

>You get a
>URI, you send it using a defined protocol to a server, it sends you a PIDF.

..-LO

>If you buy that, what in the document prevents me from allowing any value I
>want in Expiry?

The <retransmission-allowed> is a binary value: yes or no, not a hop-count 
field... this may have us in a pickle in this use case, even if most don't 
believe it's a problem.

At the end of the day, I think we should have

         allowed-scheme = 'cid' / 'http' / 'https' / 'pres'

                 (and I'm not so sure about 'pres')

until the location-retreavel ID is written that extends this document 
(which will define how to SUBSCRIBE/NOTIFY and PUBLISH location).

I'm fine with defining the additional sematics surrounding how to use HTTP 
and HTTPS (i.e. with request type and so on), as I consider that part of 
the dereferencing part of conveyance.

That said, once you have SUBSCRIBE, you have tracking with multiple 
NOTIFYs, and that's an event package that isn't proposed yet, because it 
needs to have a timing interval and a trigger on what is considered a 
move.  This would be a more general use-case that Rohan's movement trigger 
event package ID.


>Brian
>
>
>
> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: Thursday, July 20, 2006 5:59 PM
> > To: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
> > Subject: RE: [Sip] Allowing sip: in location header not good
> >
> > At 12:19 PM 7/20/2006 -0400, Brian Rosen wrote:
> > >There is a point to limiting what can be sent in the Location header to
> > >specific protocols which has received adequate review in both sip and
> > >geopriv.
> > >
> > >So, here is my concrete proposal:
> > >Location       = "Location" HCOLON location-param *(COMMA location-param)
> > >location-param = LAQUOT allowed-scheme HCOLON ":" ( hier-part / opaque-
> > part
> > >) RAQUOT *( SEMI loc-param ) ; hier-part and opaque-part from 3261
> > >allowed-scheme = 'cid' / 'http' / 'https' / 'sip' / 'sips' / 'prez'
> > >loc-param      = 'location-dereference-protocol=' deref-protocol /
> > >generic-param
> > >deref-protocol = token
> >
> > I'm still not a fan of using 'sip' here. 'sip' or 'sips' means using
> > SUBSCRIBE and that's a PULL mechanism in this scenario; i.e. the UAC
> > that's
> > subscribing to another SIP element is not using its location to do
> > anything
> > (that will come back in the NOTIFY).  The UAC (in the SUB transaction) in
> > this scenario is PULLING another UA's location from somewhere, and that's
> > not "Location Conveyance" - which is what I got hammered on in the Paris
> > meeting.
> >
> > Clear consensus from that meeting was that this document describes PUSHING
> > my UAC's location to somewhere else.
> >
> > Using HTTP(S) to fetch (presumably with a GET request) my
> > location-by-reference once from a server is not a pull, it is a
> > Content-Indirection operation, which is a fake of a PUSH, IMO.
> >
> > Using a SUBSCRIBE, you are always receiving something in the NOTIFY.  If
> > "location" is what's returned in that NOTIFY, this is a "Location
> > Retreaval" PULL, and against what the Paris meeting said to do, and is for
> > another document, a PULL mechanism document that hasn't been written yet.
> >
> > Therefore, I recommend we remove sip: and sips: from the BNF above.  I'm
> > still thinking about what to do about the 'pres' proposal.
> >
> > Adam's scenario (on the mike in Montreal) about using SUBSCRIBE to a
> > weather service did not have Adam's location coming back in the NOTIFY, so
> > that use-case doesn't apply to this discussion (but is a great use-case!)
> >
> > comments
> >
> > James

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From sip-bounces@ietf.org Thu Jul 20 19:30:05 2006
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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Brian Rosen <br@brianrosen.net>
Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
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Brian,

In general I could go with either a syntactic or semantic constraint. In 
general I think a syntactic constraint is preferred if it is possible, 
and the fallback to semantic constraint comes when the concept can't be 
represented precisely in the syntax, or is too cumbersome to do in practice.

But I am concerned with how you have tried to do this in syntax. What 
you have ended up with is a looser definition for a SIP URI than is 
present in 3261. That at least "smells a little funny".

If you want to do it in syntax, then I think you should enumerate the 
syntactic type of each kind of URI you want to permit. That still allows 
new ones to be added, but then at least each one will be properly 
constrained.

	Paul

Brian Rosen wrote:
> Jeroen
> 
> I don't agree that this is a semantic issue; we are restricting precisely
> what may appear in allowed-scheme.  I usually think it's better to have the
> ABNF precise, rather than relying on text to state what is actually
> permitted on the wire.  We are talking about what is syntactically allowed,
> not how you interpret it.
> 
> The ABNF =/ syntax (from RFC4234) is a convenient way to add a scheme to the
> list, even when used by independently created RFCs.
> 
> I, like Andy, don't believe the decision on which protocols can appear in
> allowed-scheme can be left to implementation.  Every dereferencer has to be
> prepared to use every possible protocol, especially on the Internet, because
> there is no way to ask for a specific URI or limit the schemes the location
> acquisition mechanism may obtain.
> 
> Brian
> 
>> -----Original Message-----
>> From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: Thursday, July 20, 2006 1:03 PM
>> To: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
>> Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
>>
>> Brian,
>>
>> To me using ABNF rules to restrict allowable schemes is not a good idea.
>> ABNF defines syntax, not semantics
>>
>> Compare this with the Route header in SIP. The ABNF allows any scheme,
>> text says it MUST be sip: or sips:. That suffices (you could define which
>> error response must be sent)
>>
>> Moreover, the protocol to use is a policy decision. You can make
>> recommendations, even define new mechanisms, but I think the choice should
>> be left to implementors
>>
>> Regards,
>>
>> Jeroen
>> -----Oorspronkelijk bericht -----
>> Van: "Brian Rosen" <br@brianrosen.net>
>> Aan: "'Thomson, Martin'" <Martin.Thomson@andrew.com>; sip@ietf.org
>> Verzonden: 20-7-06 18:19
>> Onderwerp: RE: [Sip] Allowing sip: (or prez:) in the location header
>>
>> There is a point to limiting what can be sent in the Location header to
>> specific protocols which has received adequate review in both sip and
>> geopriv.
>>
>> So, here is my concrete proposal:
>> Location       = "Location" HCOLON location-param *(COMMA location-param)
>> location-param = LAQUOT allowed-scheme HCOLON ":" ( hier-part / opaque-
>> part
>> ) RAQUOT *( SEMI loc-param ) ; hier-part and opaque-part from 3261
>> allowed-scheme = 'cid' / 'http' / 'https' / 'sip' / 'sips' / 'prez'
>> loc-param      = 'location-dereference-protocol=' deref-protocol /
>> generic-param
>> deref-protocol = token
>>
>> allowed-scheme may be extended by a standards-track RFC.  Such an
>> extension
>> is NOT considered a SIP extension and is NOT subject to the process
>> defined
>> by RFC3427.  Corresponding resolution processes for such schemes MUST be
>> included in the RFC, and MAY be subject to review against RFC3693.
>>
>> This document (in Section ?) defines a new IANA registry for deref-
>> protocol,
>> and populates it with one initial value, 'rawGET'.  rawGET indicates that
>> the location reference may be dereferenced by sending a GET with no
>> parameters on the URI, using HTTPS (except in the emergency case, where
>> HTTP
>> is permitted), and expects to receive one PIDF-LO document (with a MIME-
>> type
>> of "application/pidf+xml")in response.  deref-protocol MUST be included if
>> the scheme is HTTPS (or HTTP in the emergency case).  Other schemes may or
>> may not require deref-protocol.  A standards-track RFC is required to add
>> a
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From sip-bounces@ietf.org Thu Jul 20 19:55:29 2006
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_______________________________________________
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From sip-bounces@ietf.org Thu Jul 20 20:22:14 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
Date: Thu, 20 Jul 2006 20:20:06 -0400
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Just to make sure I'm following you, the part you are unhappy with would be
the "heir-part / opaque-part" which is the definition of AbsoluteURI in
3261?  You would want me to define something that was closer to what each
URI scheme uses for what comes after the colon? 

So maybe:
location-param = LAQUOT allowed-URI RAQUOT *( SEMI loc-param )
;cid-url from RFC2111
;http_URL from RFC2616;
;SIP/SIPS-URI from RFC3261
;PRES-URI from RFC3859 
allowed-URI = cid-uri / http_URL / https_URL / SIP-URI / SIPS-URI
              / PRES-URI


Amusing side note on this.  I don't believe there is any actual definition
of an https_URL.

Brian

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Thursday, July 20, 2006 7:28 PM
> To: Brian Rosen
> Cc: 'Jeroen Van Bemmel'; 'Thomson, Martin'; sip@ietf.org
> Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
> 
> Brian,
> 
> In general I could go with either a syntactic or semantic constraint. In
> general I think a syntactic constraint is preferred if it is possible,
> and the fallback to semantic constraint comes when the concept can't be
> represented precisely in the syntax, or is too cumbersome to do in
> practice.
> 
> But I am concerned with how you have tried to do this in syntax. What
> you have ended up with is a looser definition for a SIP URI than is
> present in 3261. That at least "smells a little funny".
> 
> If you want to do it in syntax, then I think you should enumerate the
> syntactic type of each kind of URI you want to permit. That still allows
> new ones to be added, but then at least each one will be properly
> constrained.
> 
> 	Paul
> 
> Brian Rosen wrote:
> > Jeroen
> >
> > I don't agree that this is a semantic issue; we are restricting
> precisely
> > what may appear in allowed-scheme.  I usually think it's better to have
> the
> > ABNF precise, rather than relying on text to state what is actually
> > permitted on the wire.  We are talking about what is syntactically
> allowed,
> > not how you interpret it.
> >
> > The ABNF =/ syntax (from RFC4234) is a convenient way to add a scheme to
> the
> > list, even when used by independently created RFCs.
> >
> > I, like Andy, don't believe the decision on which protocols can appear
> in
> > allowed-scheme can be left to implementation.  Every dereferencer has to
> be
> > prepared to use every possible protocol, especially on the Internet,
> because
> > there is no way to ask for a specific URI or limit the schemes the
> location
> > acquisition mechanism may obtain.
> >
> > Brian
> >
> >> -----Original Message-----
> >> From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
> >> Sent: Thursday, July 20, 2006 1:03 PM
> >> To: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
> >> Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
> >>
> >> Brian,
> >>
> >> To me using ABNF rules to restrict allowable schemes is not a good
> idea.
> >> ABNF defines syntax, not semantics
> >>
> >> Compare this with the Route header in SIP. The ABNF allows any scheme,
> >> text says it MUST be sip: or sips:. That suffices (you could define
> which
> >> error response must be sent)
> >>
> >> Moreover, the protocol to use is a policy decision. You can make
> >> recommendations, even define new mechanisms, but I think the choice
> should
> >> be left to implementors
> >>
> >> Regards,
> >>
> >> Jeroen
> >> -----Oorspronkelijk bericht -----
> >> Van: "Brian Rosen" <br@brianrosen.net>
> >> Aan: "'Thomson, Martin'" <Martin.Thomson@andrew.com>; sip@ietf.org
> >> Verzonden: 20-7-06 18:19
> >> Onderwerp: RE: [Sip] Allowing sip: (or prez:) in the location header
> >>
> >> There is a point to limiting what can be sent in the Location header to
> >> specific protocols which has received adequate review in both sip and
> >> geopriv.
> >>
> >> So, here is my concrete proposal:
> >> Location       = "Location" HCOLON location-param *(COMMA location-
> param)
> >> location-param = LAQUOT allowed-scheme HCOLON ":" ( hier-part / opaque-
> >> part
> >> ) RAQUOT *( SEMI loc-param ) ; hier-part and opaque-part from 3261
> >> allowed-scheme = 'cid' / 'http' / 'https' / 'sip' / 'sips' / 'prez'
> >> loc-param      = 'location-dereference-protocol=' deref-protocol /
> >> generic-param
> >> deref-protocol = token
> >>
> >> allowed-scheme may be extended by a standards-track RFC.  Such an
> >> extension
> >> is NOT considered a SIP extension and is NOT subject to the process
> >> defined
> >> by RFC3427.  Corresponding resolution processes for such schemes MUST
> be
> >> included in the RFC, and MAY be subject to review against RFC3693.
> >>
> >> This document (in Section ?) defines a new IANA registry for deref-
> >> protocol,
> >> and populates it with one initial value, 'rawGET'.  rawGET indicates
> that
> >> the location reference may be dereferenced by sending a GET with no
> >> parameters on the URI, using HTTPS (except in the emergency case, where
> >> HTTP
> >> is permitted), and expects to receive one PIDF-LO document (with a
> MIME-
> >> type
> >> of "application/pidf+xml")in response.  deref-protocol MUST be included
> if
> >> the scheme is HTTPS (or HTTP in the emergency case).  Other schemes may
> or
> >> may not require deref-protocol.  A standards-track RFC is required to
> add
> >> a
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


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From: Andrew Newton <andy@hxr.us>
Subject: Re: [Geopriv] RE: [Sip] Allowing sip: in location header not good
Date: Thu, 20 Jul 2006 20:53:45 -0400
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On Jul 20, 2006, at 7:55 PM, Thomson, Martin wrote:
> That is to say - the location conveyance document doesn't cover its  
> use, and nothing else does...yet.  When someone does a using- 
> protocol specification for SUB/NOT, then we have a place to  
> describe how to use sip[s]: in the Location header.

I guess we've all forgotten about RFC 4079.  Here's the link:
http://www.ietf.org/rfc/rfc4079.txt

-andy

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From sip-bounces@ietf.org Thu Jul 20 20:56:29 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Thomson, Martin'" <Martin.Thomson@andrew.com>,
	"'James M. Polk'" <jmpolk@cisco.com>, <sip@ietf.org>, <geopriv@ietf.org>
Subject: RE: [Sip] Allowing sip: in location header not good
Date: Thu, 20 Jul 2006 20:54:35 -0400
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I think that ship has sailed.  PIDF-LO is precisely defined in presence, and
we don't need any documents that further describe how location is conveyed
in a presence subscription.  All this document would do is describe how you
might go about specifying a presence uri that would be a little less general
(provide less information) than the one you would get by using the "From:"
URI.  The ruleset for this presence uri would be different from the more
general uri.  

Now, it does occur to me that if we allow sip/sips/pres, then we have a
minimal resolution protocol for location by reference.  It might be then
acceptable to drop the whole raw GET idea completely, and leave that to the
L7 protocol.  I kind of like that idea, because the raw GET is really
defining a whole new protocol and I don't think we want to, or need to, do
that in this document.  We brought it up because without at least a minimal
dereference solution, the mechanism does not meet the requirements.  A
presence subscription would do that, with no need for more.

And if we do that, we could drop the resolution-protocol parameter and
registry, because sip/sips/pres doesn't need it.

Brian

> -----Original Message-----
> From: Thomson, Martin [mailto:Martin.Thomson@andrew.com]
> Sent: Thursday, July 20, 2006 7:56 PM
> To: James M. Polk; Brian Rosen; sip@ietf.org; geopriv@ietf.org
> Subject: RE: [Sip] Allowing sip: in location header not good
> 
> I don't think that you need to preclude the use of SIP here, but there are
> a lot of things that need to be considered before SUB/NOT are allowed.
> 
> That is to say - the location conveyance document doesn't cover its use,
> and nothing else does...yet.  When someone does a using-protocol
> specification for SUB/NOT, then we have a place to describe how to use
> sip[s]: in the Location header.
> 
> Fact is - I don't think that we have ANY using-protocol defined, hence the
> problems we are having in deciding what goes in the header.
> 
> 
> >                  (and I'm not so sure about 'pres')
> 
> A pres: URI is in effect a sip[s]: URI for these purposes.
> 
> 
> Cheers,
> Martin
> --------------------------------------------------------------------------
> ----------------------
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> [mf2]


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Date: Fri, 21 Jul 2006 09:27:17 +0800
From: Stephen Zhou <zhouhong@huawei.com>
Subject: RE: [Sip] Query reagrding RFC 3665
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Hi Sudhir,

 

The contact is the last-hop route.

So if no Route header in 200OK message, you should put the last-hop route to
Request-URI.

 

Stephen

 

  _____  

From: Sudhir [mailto:t.sudhirkumar@gmail.com] 
Sent: Thursday, July 20, 2006 9:19 PM
To: sip@ietf.org
Subject: [Sip] Query reagrding RFC 3665

 

Hi All,

 

In RFC 3261 the construction of ACK requestURI MUST conatain original
request as it said in below paragraph 

17.1.1.3 Construction of the ACK Request

The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request"). 

But, in RFC 3665 the contruction of RequestURI for ACK is taken from
ConatctHeader of 200 OK as shown below. Could I know which is the
approipraite one. Should I go for 3261 or Informational RFC 3665

RFC 3665 3.1. Successful Session Establishment
INVITE sip:bob@biloxi.example.com SIP/2.0

200 OK contact Header: <sip:bob@client.biloxi.example.com>
ACK sip:bob@client.biloxi.example.com SIP/2.0 

thanks in advance

regards

Sudhir

 

 


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<div>

<p class=MsoNormal><font size=2 color=navy face="Times New Roman"><span
lang=EN-US style='font-size:10.0pt;color:navy'>Hi Sudhir,<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face="Times New Roman"><span
lang=EN-US style='font-size:10.0pt;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>The contact is the
last-hop route.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>So if no Route header in
200OK message, you should put the last-hop route to Request-URI.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>Stephen<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face="Times New Roman"><span
lang=EN-US style='font-size:10.0pt;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>

<hr size=2 width="100%" align=center tabindex=-1>

</span></font></div>

<p class=MsoNormal><b><font size=2 face=Tahoma><span lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
Sudhir [mailto:t.sudhirkumar@gmail.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, July 20, 2006 9:19
PM<br>
<b><span style='font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Sip] Query reagrding RFC
3665</span></font><span lang=EN-US><o:p></o:p></span></p>

</div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>Hi All,<o:p></o:p></span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>In RFC 3261 the construction of ACK
requestURI&nbsp;MUST conatain original request as it said in below
paragraph&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=MsoNormal><strong><b><font size=3 face="Times New Roman"><span
lang=EN-US style='font-size:12.0pt'><a href="http://17.1.1.3">17.1.1.3</a>
Construction of the ACK Request</span></font></b></strong><span lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>The
ACK request constructed by the client transaction MUST contain<br>
values for the Call-ID, From, and Request-URI that are equal to the<br>
values of those header fields in the request passed to the transport<br>
by the client transaction (call this the &quot;original request&quot;). <o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>But,
in RFC 3665 the contruction of RequestURI for ACK is taken from ConatctHeader
of 200 OK as shown below. Could I know which is the approipraite one. Should I
go for 3261 or Informational RFC 3665<o:p></o:p></span></font></p>

<p><strong><b><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>RFC 3665 3.1. Successful Session Establishment</span></font></b></strong><span
lang=EN-US><br>
INVITE <a href="mailto:sip:bob@biloxi.example.com">sip:bob@biloxi.example.com</a>
SIP/2.0<o:p></o:p></span></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>200
OK contact Header: &lt;<a href="mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.example.com</a>&gt;<br>
ACK <a href="mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.example.com</a>
SIP/2.0 <o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>thanks
in advance<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>regards<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>Sudhir<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

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Date: Fri, 21 Jul 2006 09:42:06 +0800
From: Stephen Zhou <zhouhong@huawei.com>
Subject: RE: [Sip] Query reagrding RFC 3665
In-reply-to: <52db68bc0607200619o50341173uabe9f9e29881e0a4@mail.gmail.com>
To: 'Sudhir' <t.sudhirkumar@gmail.com>, sip@ietf.org
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Hi,

 

Omit my previous letter.

 

The contact is the last-hop route.

So if no Record-Route header in 200OK message, you should put the last-hop
route to Request-URI.

 

Stephen

 

  _____  

From: Sudhir [mailto:t.sudhirkumar@gmail.com] 
Sent: Thursday, July 20, 2006 9:19 PM
To: sip@ietf.org
Subject: [Sip] Query reagrding RFC 3665

 

Hi All,

 

In RFC 3261 the construction of ACK requestURI MUST conatain original
request as it said in below paragraph 

17.1.1.3 Construction of the ACK Request

The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request"). 

But, in RFC 3665 the contruction of RequestURI for ACK is taken from
ConatctHeader of 200 OK as shown below. Could I know which is the
approipraite one. Should I go for 3261 or Informational RFC 3665

RFC 3665 3.1. Successful Session Establishment
INVITE sip:bob@biloxi.example.com SIP/2.0

200 OK contact Header: <sip:bob@client.biloxi.example.com>
ACK sip:bob@client.biloxi.example.com SIP/2.0 

thanks in advance

regards

Sudhir

 

 


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<div class=Section1>

<div>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>Hi,<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>Omit my previous letter.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face="Times New Roman"><span
lang=EN-US style='font-size:10.0pt;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>The contact is the
last-hop route.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>So if no </span></font><font
size=1 color="#ff9900" face=Arial><span lang=EN-US style='font-size:9.0pt;
font-family:Arial;color:#FF9900'>Record-Route</span></font><font size=1
color=navy face=Arial><span lang=EN-US style='font-size:9.0pt;font-family:Arial;
color:navy'> header in 200OK message, you should put the last-hop route to
Request-URI.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>Stephen<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>

<hr size=2 width="100%" align=center tabindex=-1>

</span></font></div>

<p class=MsoNormal><b><font size=2 face=Tahoma><span lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
Sudhir [mailto:t.sudhirkumar@gmail.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, July 20, 2006 9:19
PM<br>
<b><span style='font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Sip] Query reagrding RFC
3665</span></font><span lang=EN-US><o:p></o:p></span></p>

</div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>Hi All,<o:p></o:p></span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>In RFC 3261 the construction of ACK
requestURI&nbsp;MUST conatain original request as it said in below
paragraph&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=MsoNormal><strong><b><font size=3 face="Times New Roman"><span
lang=EN-US style='font-size:12.0pt'><a href="http://17.1.1.3">17.1.1.3</a>
Construction of the ACK Request</span></font></b></strong><span lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>The
ACK request constructed by the client transaction MUST contain<br>
values for the Call-ID, From, and Request-URI that are equal to the<br>
values of those header fields in the request passed to the transport<br>
by the client transaction (call this the &quot;original request&quot;). <o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>But,
in RFC 3665 the contruction of RequestURI for ACK is taken from ConatctHeader
of 200 OK as shown below. Could I know which is the approipraite one. Should I
go for 3261 or Informational RFC 3665<o:p></o:p></span></font></p>

<p><strong><b><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>RFC 3665 3.1. Successful Session Establishment</span></font></b></strong><span
lang=EN-US><br>
INVITE <a href="mailto:sip:bob@biloxi.example.com">sip:bob@biloxi.example.com</a>
SIP/2.0<o:p></o:p></span></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>200
OK contact Header: &lt;<a href="mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.example.com</a>&gt;<br>
ACK <a href="mailto:sip:bob@client.biloxi.example.com">sip:bob@client.biloxi.example.com</a>
SIP/2.0 <o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>thanks
in advance<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>regards<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>Sudhir<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

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Brian Rosen wrote:
> I think that ship has sailed.  PIDF-LO is precisely defined in presence, and
> we don't need any documents that further describe how location is conveyed
> in a presence subscription.  All this document would do is describe how you
> might go about specifying a presence uri that would be a little less general
> (provide less information) than the one you would get by using the "From:"
> URI.  The ruleset for this presence uri would be different from the more
> general uri.  
>
> Now, it does occur to me that if we allow sip/sips/pres, then we have a
> minimal resolution protocol for location by reference.  It might be then
> acceptable to drop the whole raw GET idea completely, and leave that to the
> L7 protocol.  I kind of like that idea, because the raw GET is really
> defining a whole new protocol and I don't think we want to, or need to, do
> that in this document.  We brought it up because without at least a minimal
> dereference solution, the mechanism does not meet the requirements.  A
> presence subscription would do that, with no need for more.
>
> And if we do that, we could drop the resolution-protocol parameter and
> registry, because sip/sips/pres doesn't need it.
>   

Brian's words here are the first to make me say "Aha! I think we have 
something new" that I've read so far this evening. There may be some 
merit in this approach -- please consider it carefully.

--
Dean

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From: "Thomson, Martin" <Martin.Thomson@andrew.com>
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Subject: RE: [Geopriv] RE: [Sip] Allowing sip: in location header not good
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_______________________________________________
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From sip-bounces@ietf.org Fri Jul 21 09:27:03 2006
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Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
Date: Fri, 21 Jul 2006 15:12:28 +0200
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Thread-Topic: [Sip] Allowing sip: (or prez:) in the location header
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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
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(As individual)

In this discussion it is probably worth remembering the endpoint we are
trying to get to.

There should be a limitation on the number of protocols specified
because:

-	the consumer of the location header needs to support all the
protocols indicated in order to fetch the location. In the case of
emergency calls that must happen first time, i.e. without sending the
424 response code back to the UAC, and getting a new request containing
a new location header with a new URI which we then might support.

-	GEOPRIV wants to maintain supervision of all GEOPRIV using
protocols in order to ensure the privacy requirements are met, and
remember, such privacy requirements are only overridable in certain
jurisdictions for emergency call, not necessarily in all jurisdictions.
Therefore any potential extension is going to need GEOPRIV WG decision.

Therefore any extension to such a list is going to be "RFC required" or
possibly even "standards track RFC required". Therefore both precise
ABNF and RFC 2119 text can be updated by such an RFC, and we just need
to decide which one presents the clearer requirement to the reader, both
in its original form, and in any later potential updates RFC that
modifies the location-conveyance RFC.

Regards

Keith
=20

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]=20
> Sent: 21 July 2006 01:43
> To: 'Thomson, Martin'; 'Jeroen van Bemmel'; 'Andrew Newton'
> Cc: sip@ietf.org
> Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
>=20
> We're dealing with a couple of issues here, so I'll take them=20
> one at a time.
>=20
> There is precedent for ABNF precise, and precedent for text=20
> explanations.  I can tell you from long experience that=20
> precise ABNF produces more interoperable implementations. =20
> We're forever correcting people that the ABNF is not=20
> definitive, and the text restricts what you can do.  If the=20
> ABNF gets out of hand, then we often take short cuts using=20
> normative text.  I'd prefer to avoid that.
>=20
> I think it's inescapable that we are creating a precise list=20
> of protocols, and prohibiting that which is not explicitly=20
> allowed.  I don't see you arguing against that, just how it's=20
> represented in the document.
>=20
> Generally, I think the parameter and registry is a better=20
> solution than squandering the GET method for a no parameter=20
> resolution.  It works, and it's not terrible, but I think=20
> explicitly stating the resolution protocol is much clearer. =20
> I have another thought on that whole subject, but that will=20
> wait for another email.
>=20
> I think the text says "HTTPS must be used, except for the=20
> emergency case".
> I don't think the draft needs to talk about HTTP headers for=20
> the simple GET case, which is all we allow.  I think=20
> authentication and cipher suites must be covered for all TLS=20
> uses alike, so no sweat there.
>=20
> Brian
>=20
> > -----Original Message-----
> > From: Thomson, Martin [mailto:Martin.Thomson@andrew.com]
> > Sent: Thursday, July 20, 2006 7:45 PM
> > To: Brian Rosen; Jeroen van Bemmel; Andrew Newton
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
> >=20
> > No, I think that Jeroen has exactly the right idea. =20
> Restriction can=20
> > exist at two levels, and in this case, text is the correct=20
> place for=20
> > those choices.  I don't like to argue from precedent, but=20
> there is a=20
> > clear precedent for this.
> >=20
> > The issue of attaching additional parameters to a URI ("URI=20
> metadata")=20
> > is a good one, and permits the use of value-add protocols. =20
> But that=20
> > should not affect the basic specification, it can remain optional.
> >=20
> > This can be very simply specified: If the https: URI exists=20
> a raw GET=20
> > MUST be possible.  An https: URI is widely understood, and I don't=20
> > think that this behaviour is contradictory to any=20
> expectations.  The=20
> > value-add protocols can be exercised by a POST, if the parameters=20
> > indicate a specific protocol is supported and the requester=20
> > understands the parameter and the protocol.  A GET should=20
> still be possible regardless.
> >=20
> > You don't need to create a registry for protocols right=20
> now; let those=20
> > protocol specifications (please, let there be only one)=20
> deal with that.
> >=20
> > One problem with limiting as you have is that you have to be very=20
> > clear in how each of those protocols are used.  Now, given Paul's=20
> > comments, I think that sip[s] and pres need a little more info - do=20
> > you want to include that as well?
> >=20
> > Ultimately, I am arguing for a smaller location conveyance=20
> draft.  The=20
> > problem with these discussions is that you can, effectively, refine=20
> > indefinitely.  Do we start to discuss what authentication=20
> methods are=20
> > appropriate in the HTTP GET?  What about HTTP headers?  How we deal=20
> > with caching?  TLS cipher-suites?  I think that it's=20
> important to know=20
> > when to draw the line.
> >=20
> > Regards,
> > Martin
> >=20
> > p.s... s/prez/pres/
> >=20
> > > -----Original Message-----
> > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > Sent: Friday, 21 July 2006 7:27 AM
> > > To: 'Jeroen van Bemmel'; 'Andrew Newton'
> > > Cc: Thomson, Martin; sip@ietf.org
> > > Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
> > >
> > > I don't think you have the big picture.
> > >
> > > Location comes from the access network, via some acquisition=20
> > > protocol,
> > or
> > > in
> > > some cases, can be self generated in the endpoint.   The=20
> access network
> > > decides what the representation of the location is if=20
> it's supplying=20
> > > location, although it may give an endpoint some options
> > (value/reference,
> > > civic/geo).
> > >
> > > The UAC gets this location and puts it in the SIP message.
> > >
> > > Some entity, could be a proxy for location based routing, or the=20
> > > UAS,
> > gets
> > > the location and does something with it.
> > >
> > > For the sake of argument, assume its location-by-reference.
> > > Call the proxy or UAS the "de-referencer".
> > >
> > > It's the access network (could be a broadband carrier, could be an
> > > enterprise) that is putting up the server that provides the=20
> > > de-reference service.  Call the server a "Location Information=20
> > > Server" =3D LIS
> > >
> > > Okay, so the access network chooses some protocol, makes up a=20
> > > suitable
> > URI
> > > and passes it via the acquisition protocol to the UAC.  The UAC=20
> > > passes
> > it
> > > to
> > > the de-referencer, who presents it to the LIS.  The LIS responds=20
> > > with
> > the
> > > location value.
> > >
> > > The problem is that the de-referencer can't control which=20
> LIS, and=20
> > > can't control what protocol decision the LIS made.  It has to=20
> > > implement EVERY ONE of the protocols that could appear in the=20
> > > Location header.  If the LIS implements a protocol the=20
> de-referencer=20
> > > doesn't understand, the whole mechanism fails, with no=20
> recourse.  On=20
> > > the Internet, which is where we want the protocols to work, you=20
> > > can't control which access network the UAC is coming from.
> > >
> > > So, instead, we restrict the list of allowed de-referencing=20
> > > protocols to
> > a
> > > small number, and make that list very explicit.  Now a=20
> de-referencer
> > knows
> > > that if he implements ALL of the listed protocols, he=20
> will be able=20
> > > to dereference any location reference he gets.
> > >
> > >
> > >
> > > There is another big picture aspect.  The whole location=20
> thing comes
> > under
> > > geopriv.  We have to do a sip Location header in sip, but geopriv=20
> > > has to review our work.  The "priv" part of geopriv is privacy. =20
> > > They care, a whole lot, about the privacy aspects of=20
> location.  One=20
> > > expression of that concern is to have the security=20
> posture of "what=20
> > > is not explicitly permitted is prohibited".  They really=20
> truly want=20
> > > to look at every IETF protocol that is used to de-reference=20
> > > location, and decide if it provides sufficient privacy protection=20
> > > for location.  They really do want an explicit list, and=20
> they want=20
> > > to scrutinize every protocol on that list to see how it addresses=20
> > > their concerns.  SIP is what geopriv calls a "using protocol" and=20
> > > they review them seriously.  This means, for example, that it's=20
> > > probably not okay to use HTTP to de-reference.  There is=20
> a specific,=20
> > > limited exemption for emergency calls on that.  So HTTPS=20
> is a MUST,=20
> > > and HTTP is MUST NOT except for the emergency case.  I=20
> believe that=20
> > > "AbsoluteURI" would not be acceptable to geopriv, but of course I=20
> > > can't speak for the group.
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> > > > Sent: Thursday, July 20, 2006 4:31 PM
> > > > To: Andrew Newton
> > > > Cc: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
> > > > Subject: Re: [Sip] Allowing sip: (or prez:) in the=20
> location header
> > > >
> > > > Andrew,
> > > >
> > > > >> Moreover, the protocol to use is a policy decision. You can=20
> > > > >> make recommendations, even define new mechanisms,=20
> but I think=20
> > > > >> the choice should be left to implementors
> > > > >
> > > > > I disagree with this.  What is my server gonna do=20
> with a mailto:=20
> > > > > or
> > > dns:
> > > > > URI?  There should be a small list of acceptable=20
> choices if we=20
> > > > > want interoperability.
> > > >
> > > > In such a case your server should probably respond with "4xx=20
> > > > Location
> > > URI
> > > > Scheme not supported", where 4xx is TBD. However, if retrieving=20
> > > > the location is not critical to completing the call, then it=20
> > > > should probably ignore
> > > the
> > > > header. So there seems to be a need for a parameter indicating
> > > "optional"
> > > > versus "required"
> > > >
> > > > IMO the text in the draft should say:
> > > > - proxies MUST support HTTP, SHOULD support HTTPS, MAY=20
> support cid
> > (for
> > > > dereferencing)
> > > > - UAS MUST support HTTP and cid, SHOULD support HTTPS (for
> > > dereferencing)
> > > > - TBD: define whether and when proxy/UAS should=20
> generate error for=20
> > > > unsupported URI scheme (and then indicate which schemes are=20
> > > > supported)
> > > > - UAC/Proxy SHOULD use HTTPS when adding Location by reference
> > > > - UAC MAY use cid for location by value, but SHOULD then ensure=20
> > > > confidentiality
> > > >
> > > > But above all: the draft should specify what is=20
> minimally needed=20
> > > > to
> > meet
> > > > the
> > > > stated requirements, while not unnecessarily=20
> restricting solutions
> > that
> > > > might have additional requirements which are not currently=20
> > > > foreseen
> > > >
> > > > Regards,
> > > >
> > > > jeroen
> > >
> >=20
> >=20
> ----------------------------------------------------------------------
> > ----
> > ----------------------
> > This message is for the designated recipient only and may contain=20
> > privileged, proprietary, or otherwise private information.
> > If you have received it in error, please notify the sender=20
> immediately=20
> > and delete the original.  Any unauthorized use of this email is=20
> > prohibited.
> >=20
> ----------------------------------------------------------------------
> > ----
> > ----------------------
> > [mf2]
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Brian Rosen <br@brianrosen.net>
Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
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Brian Rosen wrote:
> I don't think you have the big picture.

Nice explanation. More below.

> Location comes from the access network, via some acquisition protocol, or in
> some cases, can be self generated in the endpoint.   The access network
> decides what the representation of the location is if it's supplying
> location, although it may give an endpoint some options (value/reference,
> civic/geo).
> 
> The UAC gets this location and puts it in the SIP message.
> 
> Some entity, could be a proxy for location based routing, or the UAS, gets
> the location and does something with it.
> 
> For the sake of argument, assume its location-by-reference.
> Call the proxy or UAS the "de-referencer".
> 
> It's the access network (could be a broadband carrier, could be an
> enterprise) that is putting up the server that provides the de-reference
> service.  Call the server a "Location Information Server" = LIS
> 
> Okay, so the access network chooses some protocol, makes up a suitable URI
> and passes it via the acquisition protocol to the UAC.  The UAC passes it to
> the de-referencer, who presents it to the LIS.  The LIS responds with the
> location value.
> 
> The problem is that the de-referencer can't control which LIS, and can't
> control what protocol decision the LIS made.  It has to implement EVERY ONE
> of the protocols that could appear in the Location header.  If the LIS
> implements a protocol the de-referencer doesn't understand, the whole
> mechanism fails, with no recourse.  On the Internet, which is where we want
> the protocols to work, you can't control which access network the UAC is
> coming from.
> 
> So, instead, we restrict the list of allowed de-referencing protocols to a
> small number, and make that list very explicit.  Now a de-referencer knows
> that if he implements ALL of the listed protocols, he will be able to
> dereference any location reference he gets.
> 
> There is another big picture aspect.  The whole location thing comes under
> geopriv.  We have to do a sip Location header in sip, but geopriv has to
> review our work.  The "priv" part of geopriv is privacy.  They care, a whole
> lot, about the privacy aspects of location.  One expression of that concern
> is to have the security posture of "what is not explicitly permitted is
> prohibited".  They really truly want to look at every IETF protocol that is
> used to de-reference location, and decide if it provides sufficient privacy
> protection for location.  They really do want an explicit list, and they
> want to scrutinize every protocol on that list to see how it addresses their
> concerns.  SIP is what geopriv calls a "using protocol" and they review them
> seriously.  This means, for example, that it's probably not okay to use HTTP
> to de-reference.  There is a specific, limited exemption for emergency calls
> on that.  So HTTPS is a MUST, and HTTP is MUST NOT except for the emergency
> case.  I believe that "AbsoluteURI" would not be acceptable to geopriv, but
> of course I can't speak for the group.

I fail to see how the exception could work in practice. How can the LIS 
know when it should accept an HTTP query and when it should not? What is 
to prevent somebody from changing an HTTPS URI to an HTTP URI and then 
dereferencing it?

	Paul

> Brian
> 
>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: Thursday, July 20, 2006 4:31 PM
>> To: Andrew Newton
>> Cc: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
>> Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
>>
>> Andrew,
>>
>>>> Moreover, the protocol to use is a policy decision. You can make
>>>> recommendations, even define new mechanisms, but I think the choice
>>>> should be left to implementors
>>> I disagree with this.  What is my server gonna do with a mailto: or dns:
>>> URI?  There should be a small list of acceptable choices if we want
>>> interoperability.
>> In such a case your server should probably respond with "4xx Location URI
>> Scheme not supported", where 4xx is TBD. However, if retrieving the
>> location
>> is not critical to completing the call, then it should probably ignore the
>> header. So there seems to be a need for a parameter indicating "optional"
>> versus "required"
>>
>> IMO the text in the draft should say:
>> - proxies MUST support HTTP, SHOULD support HTTPS, MAY support cid (for
>> dereferencing)
>> - UAS MUST support HTTP and cid, SHOULD support HTTPS (for dereferencing)
>> - TBD: define whether and when proxy/UAS should generate error for
>> unsupported URI scheme (and then indicate which schemes are supported)
>> - UAC/Proxy SHOULD use HTTPS when adding Location by reference
>> - UAC MAY use cid for location by value, but SHOULD then ensure
>> confidentiality
>>
>> But above all: the draft should specify what is minimally needed to meet
>> the
>> stated requirements, while not unnecessarily restricting solutions that
>> might have additional requirements which are not currently foreseen
>>
>> Regards,
>>
>> jeroen
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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From sip-bounces@ietf.org Fri Jul 21 09:46:10 2006
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Subject: RE: [Sip] draft-ietf-sip-outbound and STUN keepalives
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Hi Keith,

I am ok with the proposal.

In the "STUN keep-alive: Where to document it?" thread =
(http://www1.ietf.org/mail-archive/web/sip/current/msg15227.html) it is =
discussed where different things related to STUN keep-alive should be =
documented. Some things MAY have to be documented in 3489bis.

Regards,

Christer
=20

-----Original Message-----
From: Drage, Keith (Keith) [mailto:drage@lucent.com]=20
Sent: 20. hein=E4kuuta 2006 15:11
To: 'sip@ietf.org'
Subject: [Sip] draft-ietf-sip-outbound and STUN keepalives

(As WG chair)

Previous mails on the SIP list have expressed the view that the STUN =
keepalive mechanism should be documented separately as it is a useful =
mechanism in its own right.

See http://www3.ietf.org/proceedings/06jul/slides/sip-5.pdf slide 5 for =
the issue as raised at the IETF #66 meeting.=09

After discussion with the editor, the WG chairs propose the following:

-	that the documentation on STUN keepalive will REMAIN in the outbound =
document, and WILL NOT be split to a separate specification;

-	that the STUN keepalive mechanism WILL BE separated within the =
outbound specification, such that it is clear that it can be implemented =
independently of the rest of the outbound specification, and so that the =
keepalive material is clearly referencable in its own right.

We believe that this meets the requirements of at least most of the =
working group, and for pragmatic reasons we want to work against more =
fragmentation of the SIP documentation.

If you have problems with this proposal and desire a different solution, =
then please raise your concerns to the list, or direct to the WG chairs, =
in response to this mail.

regards

Keith

Keith Drage
drage@lucent.com
tel: +44 1793 776249

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use =
sip-implementors@cs.columbia.edu for questions on current sip Use =
sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Fri Jul 21 10:44:59 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Subject: RE: [Sip] Allowing sip: (or prez:) in the location header
Date: Fri, 21 Jul 2006 10:42:51 -0400
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There are no protocol police.

There is nothing to stop it from doing so.
There is nothing to stop a server from not even implementing HTTPS.  They
seem to do it with sip TLS all the time.

All we can do is write the spec the right way, and ask the implementers to
please follow the spec.

This observation is one reason why we want the reference itself to be
protected in transit. 

Of course, mere possession of a location is not really useful.  You need to
be able to associate the location with some person (or thing).  The
reference, therefore, might want to be opaque, and you again want to protect
the signaling that would show an observer the relationship between an
identity and a location.

Brian

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Friday, July 21, 2006 9:28 AM
> To: Brian Rosen
> Cc: 'Jeroen van Bemmel'; 'Andrew Newton'; sip@ietf.org; 'Thomson, Martin'
> Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
> 
> 
> 
> Brian Rosen wrote:
> > I don't think you have the big picture.
> 
> Nice explanation. More below.
> 
> > Location comes from the access network, via some acquisition protocol,
> or in
> > some cases, can be self generated in the endpoint.   The access network
> > decides what the representation of the location is if it's supplying
> > location, although it may give an endpoint some options
> (value/reference,
> > civic/geo).
> >
> > The UAC gets this location and puts it in the SIP message.
> >
> > Some entity, could be a proxy for location based routing, or the UAS,
> gets
> > the location and does something with it.
> >
> > For the sake of argument, assume its location-by-reference.
> > Call the proxy or UAS the "de-referencer".
> >
> > It's the access network (could be a broadband carrier, could be an
> > enterprise) that is putting up the server that provides the de-reference
> > service.  Call the server a "Location Information Server" = LIS
> >
> > Okay, so the access network chooses some protocol, makes up a suitable
> URI
> > and passes it via the acquisition protocol to the UAC.  The UAC passes
> it to
> > the de-referencer, who presents it to the LIS.  The LIS responds with
> the
> > location value.
> >
> > The problem is that the de-referencer can't control which LIS, and can't
> > control what protocol decision the LIS made.  It has to implement EVERY
> ONE
> > of the protocols that could appear in the Location header.  If the LIS
> > implements a protocol the de-referencer doesn't understand, the whole
> > mechanism fails, with no recourse.  On the Internet, which is where we
> want
> > the protocols to work, you can't control which access network the UAC is
> > coming from.
> >
> > So, instead, we restrict the list of allowed de-referencing protocols to
> a
> > small number, and make that list very explicit.  Now a de-referencer
> knows
> > that if he implements ALL of the listed protocols, he will be able to
> > dereference any location reference he gets.
> >
> > There is another big picture aspect.  The whole location thing comes
> under
> > geopriv.  We have to do a sip Location header in sip, but geopriv has to
> > review our work.  The "priv" part of geopriv is privacy.  They care, a
> whole
> > lot, about the privacy aspects of location.  One expression of that
> concern
> > is to have the security posture of "what is not explicitly permitted is
> > prohibited".  They really truly want to look at every IETF protocol that
> is
> > used to de-reference location, and decide if it provides sufficient
> privacy
> > protection for location.  They really do want an explicit list, and they
> > want to scrutinize every protocol on that list to see how it addresses
> their
> > concerns.  SIP is what geopriv calls a "using protocol" and they review
> them
> > seriously.  This means, for example, that it's probably not okay to use
> HTTP
> > to de-reference.  There is a specific, limited exemption for emergency
> calls
> > on that.  So HTTPS is a MUST, and HTTP is MUST NOT except for the
> emergency
> > case.  I believe that "AbsoluteURI" would not be acceptable to geopriv,
> but
> > of course I can't speak for the group.
> 
> I fail to see how the exception could work in practice. How can the LIS
> know when it should accept an HTTP query and when it should not? What is
> to prevent somebody from changing an HTTPS URI to an HTTP URI and then
> dereferencing it?
> 
> 	Paul
> 
> > Brian
> >
> >> -----Original Message-----
> >> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> >> Sent: Thursday, July 20, 2006 4:31 PM
> >> To: Andrew Newton
> >> Cc: Brian Rosen; 'Thomson, Martin'; sip@ietf.org
> >> Subject: Re: [Sip] Allowing sip: (or prez:) in the location header
> >>
> >> Andrew,
> >>
> >>>> Moreover, the protocol to use is a policy decision. You can make
> >>>> recommendations, even define new mechanisms, but I think the choice
> >>>> should be left to implementors
> >>> I disagree with this.  What is my server gonna do with a mailto: or
> dns:
> >>> URI?  There should be a small list of acceptable choices if we want
> >>> interoperability.
> >> In such a case your server should probably respond with "4xx Location
> URI
> >> Scheme not supported", where 4xx is TBD. However, if retrieving the
> >> location
> >> is not critical to completing the call, then it should probably ignore
> the
> >> header. So there seems to be a need for a parameter indicating
> "optional"
> >> versus "required"
> >>
> >> IMO the text in the draft should say:
> >> - proxies MUST support HTTP, SHOULD support HTTPS, MAY support cid (for
> >> dereferencing)
> >> - UAS MUST support HTTP and cid, SHOULD support HTTPS (for
> dereferencing)
> >> - TBD: define whether and when proxy/UAS should generate error for
> >> unsupported URI scheme (and then indicate which schemes are supported)
> >> - UAC/Proxy SHOULD use HTTPS when adding Location by reference
> >> - UAC MAY use cid for location by value, but SHOULD then ensure
> >> confidentiality
> >>
> >> But above all: the draft should specify what is minimally needed to
> meet
> >> the
> >> stated requirements, while not unnecessarily restricting solutions that
> >> might have additional requirements which are not currently foreseen
> >>
> >> Regards,
> >>
> >> jeroen
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


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From sip-bounces@ietf.org Fri Jul 21 13:12:18 2006
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Subject: RE: [Sip] draft-ietf-sip-outbound and STUN keepalives
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This is perfect.

In fact, I tought that this is what has been proposed all along: that=20
is certainly what I personally wanted.=20

FA> -----Original Message-----
FA> From: Drage, Keith (Keith) [mailto:drage@lucent.com]=20
FA> Sent: Thursday, July 20, 2006 5:11 AM
FA> To: 'sip@ietf.org'
FA> Subject: [Sip] draft-ietf-sip-outbound and STUN keepalives
FA>=20
FA> (As WG chair)
FA>=20
FA> Previous mails on the SIP list have expressed the view that=20
FA> the STUN keepalive mechanism should be documented=20
FA> separately as it is a useful mechanism in its own right.
FA>=20
FA> See http://www3.ietf.org/proceedings/06jul/slides/sip-5.pdf=20
FA> slide 5 for the issue as raised at the IETF #66 meeting.=09
FA>=20
FA> After discussion with the editor, the WG chairs propose the=20
FA> following:
FA>=20
FA> -	that the documentation on STUN keepalive will REMAIN in=20
FA> the outbound document, and WILL NOT be split to a separate=20
FA> specification;
FA>=20
FA> -	that the STUN keepalive mechanism WILL BE separated=20
FA> within the outbound specification, such that it is clear=20
FA> that it can be implemented independently of the rest of the=20
FA> outbound specification, and so that the keepalive material=20
FA> is clearly referencable in its own right.
FA>=20
FA> We believe that this meets the requirements of at least=20
FA> most of the working group, and for pragmatic reasons we=20
FA> want to work against more fragmentation of the SIP documentation.
FA>=20
FA> If you have problems with this proposal and desire a=20
FA> different solution, then please raise your concerns to the=20
FA> list, or direct to the WG chairs, in response to this mail.
FA>=20
FA> regards
FA>=20
FA> Keith
FA>=20
FA> Keith Drage
FA> drage@lucent.com
FA> tel: +44 1793 776249
FA>=20
FA> _______________________________________________
FA> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
FA> This list is for NEW development of the core SIP Protocol=20
FA> Use sip-implementors@cs.columbia.edu for questions on=20
FA> current sip Use sipping@ietf.org for new developments on=20
FA> the application of sip
FA>=20

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To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
Subject: Re: [Sip] draft-ietf-sip-outbound and STUN keepalives
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Right. I mentioned this in the behave meeting. The idea is that 
rfc3489bis formally defines the usage, and calls out additional things 
that an actual protocol that uses it, would have to say. Then 
sip-outbound says those things.

-Jonathan R.

Christer Holmberg (JO/LMF) wrote:

> Hi Keith,
> 
> I am ok with the proposal.
> 
> In the "STUN keep-alive: Where to document it?" thread (http://www1.ietf.org/mail-archive/web/sip/current/msg15227.html) it is discussed where different things related to STUN keep-alive should be documented. Some things MAY have to be documented in 3489bis.
> 
> Regards,
> 
> Christer
>  
> 
> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com] 
> Sent: 20. heinäkuuta 2006 15:11
> To: 'sip@ietf.org'
> Subject: [Sip] draft-ietf-sip-outbound and STUN keepalives
> 
> (As WG chair)
> 
> Previous mails on the SIP list have expressed the view that the STUN keepalive mechanism should be documented separately as it is a useful mechanism in its own right.
> 
> See http://www3.ietf.org/proceedings/06jul/slides/sip-5.pdf slide 5 for the issue as raised at the IETF #66 meeting.	
> 
> After discussion with the editor, the WG chairs propose the following:
> 
> -	that the documentation on STUN keepalive will REMAIN in the outbound document, and WILL NOT be split to a separate specification;
> 
> -	that the STUN keepalive mechanism WILL BE separated within the outbound specification, such that it is clear that it can be implemented independently of the rest of the outbound specification, and so that the keepalive material is clearly referencable in its own right.
> 
> We believe that this meets the requirements of at least most of the working group, and for pragmatic reasons we want to work against more fragmentation of the SIP documentation.
> 
> If you have problems with this proposal and desire a different solution, then please raise your concerns to the list, or direct to the WG chairs, in response to this mail.
> 
> regards
> 
> Keith
> 
> Keith Drage
> drage@lucent.com
> tel: +44 1793 776249
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] SIP Security mechanism?
Date: Fri, 21 Jul 2006 00:17:03 -0700
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you might want to have a read of draft-ietf-sipping-certs-04 if your  
goal is to make sure that users can find the keys of other users for  
use with S/MIME

On Jul 19, 2006, at 12:17 PM, fwmiller@cornfed.com wrote:

>
>
> I've been reading about the ZRTP lately.  I read Henning's comment  
> at the IETF meeting about how S/MIME relies on the receiver's  
> public key to be available to allow for encryption/decryption to work.
>
> Why couldnt we put in place a mechanism that did a key exchange for  
> SIP signaling similar to whats done in ZRTP (i.e. modified Diffie- 
> Hellman) prior to any signaling traffic, then use the exchanged  
> keys as the basis for S/MIME encryption of the bodies?  Or we could  
> even get really carried away and use it to do encryption of the  
> entire SIP message if the signaling did not traverse proxies?  This  
> could be useful for encrypting UDP signaling, again if no proxy  
> were involved.
>
> Thoughts?
> FM
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: Re: [Sip] SIP Security mechanism?
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Thanks for the pointer.  I have scanned this draft over.

The goal of your draft is similar to what I'm targeting but slightly
different.  Please correct me if I am misunderstanding something.  The
draft (which is now draft-ietf-sip-certs-01 I believe) has the goal is
to provide a way to make the distribution of certificates more automated
by providing specific infrastructure that is based on simple user
actions.  The draft specifies publication and an event package that
allows for notification to interested User Agents of certificates.

The great benefit to this approach is its preservation of user
authentication.

My thought was much simpler.  Let me drop in a quote from the ZRTP
draft, draft-zimmermann-avt-zrtp-01.txt:

ZRTP is key agreement protocol which performs Diffie-Hellman key
   exchange during call setup in-band in the Real-time Transport
   Protocol (RTP) [1] media stream which has been established using some
   other signaling protocol such as Session Initiation Protocol (SIP)
   [11].  This generates a shared secret which is then used to generate
   keys and salt for a Secure RTP (SRTP) [2] session.

Although it uses
   a public key algorithm, it does not rely on a public key
   infrastructure (PKI).

Were we to put in place some sort of simple transaction that did this
key exchange before any other signaling, we could use the keys to do
encryption of the message bodies (similar to what S/MIME is doing)
without having to deal with PKI at all.  Since the main problem with the
use of S/MIME to date has been the certificate management, this approach
is also targeting the problem that your draft is.

However, there are a couple of differences.  First, your draft approach
preserves full S/MIME certificate based authentication, this approach
would likely not do that, it would be to allow for encryption only.  The
ZRTP draft proposes prompting the user with a short authentication
string for the media but I don't know if that would be useful for the
signaling.  Second, this approach would be much simpler, likely
requiring only a couple of messages between the UA endpoints.  It would
not require the centralized server elements that would be necessary for
the PUBLISH/SUBSCRIBE/NOTIFY components of your draft.

Consider the messages used to do key exchange in ZRTP (again from the
ZRTP draft):

 Alice                                      Bob
     |                                         |
     | Alice and Bob establish a media session.|
     |                                         |
     |                   RTP                   |
     |<=======================================>|
     |                                         |
     | Hello (ver,cid,hash,cipher,pkt,sas,Alice's ZID) F1
     |---------------------------------------->|
     |                             HelloACK F2 |
     |<----------------------------------------|
     | Hello (ver,cid,hash,cipher,pkt,sas,Bob's ZID) F3
     |<----------------------------------------|
     | HelloACK F4                             |
     |---------------------------------------->|
     |                                         |
     |        Bob acts as the initiator        |
     |                                         |
     |   Commit (Bob's ZID,hash,cipher,pkt,hvi) F5
     |<----------------------------------------|
     | DHPart1 (pvr,rs1IDr,rs2IDr,sigsIDr,srtpsIDr,other_secretIDr) F6
     |---------------------------------------->|
     | DHPart2 (pvi,rs1IDi,rs2IDi,sigsIDi,ssrtpIDi,other_secretIDi) F7
     |<----------------------------------------|
     |                                         |
     | Alice and Bob generate SRTP session key.|
     |                                         |
     |               SRTP begins               |
     |<=======================================>|
     |                                         |
     | Confirm1 (plaintext,sasflag,hmac) F8    |
     |---------------------------------------->|
     |    Confirm2 (plaintext,sasflag,hmac) F9 |
     |<----------------------------------------|
     | Confirm2AK F10                          |
     |---------------------------------------->|


They are generating keys for each of media streams here, i.e. an inband
message exchange is happening in each direction.  Something like this
could probably be simplified to just two messages for the signaling,
something like:

    Bob                             Alice
     |           HELLO                |
     |------------------------------->|
     |           200 OK               |
     |<-------------------------------|
     |                                |

This example uses a new method I made up called HELLO but I suppose this
could be folded into the INVITE transaction too, using a mechanism
similar to the 407.  The key that Alice is supposed to use to encrypt
the bodies sent to Bob would be in the HELLO and the key that Bob is
supposed to use encrypt the bodies sent to Alice would be in the 200 OK.

Is this clearer?  Does it make sense?  Its simpler but the
authentication may not be there.  That works for the media because you
can possibly recognize someone's voice.  In the signaling the
authentication thing may be a bigger issue.

FM


On Fri, 2006-07-21 at 00:17 -0700, Cullen Jennings wrote:
> you might want to have a read of draft-ietf-sipping-certs-04 if your  
> goal is to make sure that users can find the keys of other users for  
> use with S/MIME
> 
> On Jul 19, 2006, at 12:17 PM, fwmiller@cornfed.com wrote:
> 
> >
> >
> > I've been reading about the ZRTP lately.  I read Henning's comment  
> > at the IETF meeting about how S/MIME relies on the receiver's  
> > public key to be available to allow for encryption/decryption to work.
> >
> > Why couldnt we put in place a mechanism that did a key exchange for  
> > SIP signaling similar to whats done in ZRTP (i.e. modified Diffie- 
> > Hellman) prior to any signaling traffic, then use the exchanged  
> > keys as the basis for S/MIME encryption of the bodies?  Or we could  
> > even get really carried away and use it to do encryption of the  
> > entire SIP message if the signaling did not traverse proxies?  This  
> > could be useful for encrypting UDP signaling, again if no proxy  
> > were involved.
> >
> > Thoughts?
> > FM
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip


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Subject: Re: [Geopriv] RE: [Sip] Allowing sip: in location header not good
Date: Thu, 20 Jul 2006 20:53:45 -0400
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On Jul 20, 2006, at 7:55 PM, Thomson, Martin wrote:
> That is to say - the location conveyance document doesn't cover its  
> use, and nothing else does...yet.  When someone does a using- 
> protocol specification for SUB/NOT, then we have a place to  
> describe how to use sip[s]: in the Location header.

I guess we've all forgotten about RFC 4079.  Here's the link:
http://www.ietf.org/rfc/rfc4079.txt

-andy

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From sip-bounces@ietf.org Mon Jul 24 10:21:20 2006
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Subject: [Sip] modification of a P-Asserted-Identity by a proxy
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Hello,

I am designing a CNIP service in a proxy and I plan to remove the 
display name of the P-Asserted-Identity if the called party has not 
subscribed to the CNIP. But I am not sure if my proxy handling the 
services of the called party is allowed to perform this modification.

I was reading the RFC3325 and I was wondering whether if in the case a 
proxy receives a P-Asserted-Identity from a trusted node and it forwards 
it to a trusted node if he is allowed to modify it. In the chapter 5 of 
the FRC, I do not see a mention that the proxy must copy the header 
transparently from one leg to the other in this case, so can I consider 
that this modification is allowed as it is not forbidden.

Thanks is advance,
Olivier



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From sip-bounces@ietf.org Mon Jul 24 10:39:18 2006
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Subject: [Sip] Regarding non-INVITE server transaction
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Hi,
        If the non-IST doesn't receive any RESPONSE message then what will 
happen?
RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention 
properly 
what will happen if the  non-IST doesn't receive any provisional 
response/final response.
 
Regards,
S.Kingston Smiler.
 


***********************  FSS-Unclassified   ***********************
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<br><font size=2 face="sans-serif"><br>
Hi,<br>
 &nbsp; &nbsp; &nbsp; &nbsp;If the non-IST doesn't receive any
RESPONSE message then what will happen?</font>
<br><font size=2 face="sans-serif">RFC 3261 (in </font><font size=2 face="Courier New">Figure
8: non-INVITE server transaction</font><font size=2 face="sans-serif">
) doesn't mention properly </font>
<br><font size=2 face="sans-serif">what will happen if the &nbsp;non-IST
doesn't receive any provisional response/final response.</font>
<br><font size=2 face="sans-serif">&nbsp;<br>
Regards,<br>
S.Kingston Smiler.<br>
 <br>
<br>
<br>
*********************** &nbsp;FSS-Unclassified &nbsp; ***********************</font>
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From sip-bounces@ietf.org Mon Jul 24 11:45:13 2006
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From: John Schnizlein <jschnizl@cisco.com>
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Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
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 From where do you draw the idea that the WG agreed to location by 
reference?  There were several individual submissions on this subject, 
none of which reached consensus to even include as a WG item, much less 
as a WG consensus in favor.

John

On Jul 20, 2006, at 9:24 AM, Drage, Keith (Keith) wrote:

> In terms of previous WG agreements we have only two mechanisms:
> -	location by value in a body
> -	location by reference in a header

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Subject: [Geopriv] RE: [Sip] Allowing sip: in location header not good
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At 6:19 PM -0500 7/20/06, James M. Polk wrote:
>
>If the INVITE has a PIDF-LO with this element
>
>        <retransmission-allowed>no</retransmission-allowed>
>
>Can a ESRP (emergency services routing proxy that does the LoST query for the PSAP URI) send location anywhere?  Geopriv might say no, because it violates one of its "priv" rules. Now what? Do we make an exception?  We have to get their buy-in for this. We don't have it yet.


Do you mean "send onward in the SIP signaling path" or "send to the LoST server"?

The problem here is that if the originating client has to allow all retransmission
in order to get the call through, their privacy protection has gone to nil; in
places where you have to be able to protect privacy during the emergency
call, that's a problem.  So just saying "it must be set to 'yes'" doesn't solve
the problem.  I think we really need to be able to distinguish between
using the location to route the call and forwarding the target of the call
forwarding the location; I believe that means we need to be able to break
the first out of "retransmission-allowed" into a separate permission.

My two cents,
				Ted

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Subject: [Geopriv] RE: [Sip] Allowing sip: in location header not good
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At 6:19 PM -0500 7/20/06, James M. Polk wrote:
>
>If the INVITE has a PIDF-LO with this element
>
>        <retransmission-allowed>no</retransmission-allowed>
>
>Can a ESRP (emergency services routing proxy that does the LoST query for the PSAP URI) send location anywhere?  Geopriv might say no, because it violates one of its "priv" rules. Now what? Do we make an exception?  We have to get their buy-in for this. We don't have it yet.


Do you mean "send onward in the SIP signaling path" or "send to the LoST server"?

The problem here is that if the originating client has to allow all retransmission
in order to get the call through, their privacy protection has gone to nil; in
places where you have to be able to protect privacy during the emergency
call, that's a problem.  So just saying "it must be set to 'yes'" doesn't solve
the problem.  I think we really need to be able to distinguish between
using the location to route the call and forwarding the target of the call
forwarding the location; I believe that means we need to be able to break
the first out of "retransmission-allowed" into a separate permission.

My two cents,
				Ted

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From sip-bounces@ietf.org Mon Jul 24 14:03:46 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'John Schnizlein'" <jschnizl@cisco.com>,
	"'Drage, Keith \(Keith\)'" <drage@lucent.com>
Subject: RE: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
Date: Mon, 24 Jul 2006 12:04:36 -0400
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Well, to be precise, Location by reference has been in the SIP Location
header for a while, and no one has objected to that part of the text.

It's not in any location acquisition protocol yet, although EVERY ONE of the
L7 protocols has included it.

I, personally, would prefer if the only location-by-reference was a presence
subscription.  I think we have to allow that.  It would be nice if that was
it; one mechanism is always better than many if you can get away with it.

Brian

> -----Original Message-----
> From: John Schnizlein [mailto:jschnizl@cisco.com]
> Sent: Thursday, July 20, 2006 10:19 AM
> To: Drage, Keith (Keith)
> Cc: geopriv@ietf.org; sip@ietf.org
> Subject: [Sip] Re: [Geopriv] Location-by-value in a SIP Location Header
> 
>  From where do you draw the idea that the WG agreed to location by
> reference?  There were several individual submissions on this subject,
> none of which reached consensus to even include as a WG item, much less
> as a WG consensus in favor.
> 
> John
> 
> On Jul 20, 2006, at 9:24 AM, Drage, Keith (Keith) wrote:
> 
> > In terms of previous WG agreements we have only two mechanisms:
> > -	location by value in a body
> > -	location by reference in a header
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From sip-bounces@ietf.org Mon Jul 24 16:36:07 2006
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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
To: "John Schnizlein" <jschnizl@cisco.com>,
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Well all I can say is that it has been there from the
draft-ietf-sip-location-conveyance-01 version (dated July 17th 2005)
which was explicitly presented in the Paris IETF#63 SIP WG meeting in
August 2005.=20

Unfortunately the minutes are not particularly helpful in terms of WG
discussion at that time. The concept also had discussion on the SIP list
at that time, with nobody proposing to take it out after it was added,
although there was some call for more discussion of the various location
provision capabilities.

Regards

Keith
=20

> -----Original Message-----
> From: John Schnizlein [mailto:jschnizl@cisco.com]=20
> Sent: 20 July 2006 15:19
> To: Drage, Keith (Keith)
> Cc: geopriv@ietf.org; 'Hannes Tschofenig'; sip@ietf.org
> Subject: Re: [Geopriv] Location-by-value in a SIP Location Header
>=20
>  From where do you draw the idea that the WG agreed to=20
> location by reference?  There were several individual=20
> submissions on this subject, none of which reached consensus=20
> to even include as a WG item, much less as a WG consensus in favor.
>=20
> John
>=20
> On Jul 20, 2006, at 9:24 AM, Drage, Keith (Keith) wrote:
>=20
> > In terms of previous WG agreements we have only two mechanisms:
> > -	location by value in a body
> > -	location by reference in a header
>=20

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On Jul 13, 2006, at 11:12 AM, Scott Lawrence wrote:

>> gzip on the body might help too
>
> If I'm not mistaken, it's not trivial to figure out how much any given
> text will compress with gzip unless you actually try each possibility.
>

I suspect you could use LZ77 (the gzip algo) in a streaming mode or  
select some other streaming compression technique to solve this so  
you add each header in order until it gets too big then use the  
previous result.

However, I'm not arguing for this one way or the other - since this  
is meant to help humans debug, it seems like having it be easily  
human readable would be best.

Cullen (with my individual hat on)

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From sip-bounces@ietf.org Tue Jul 25 05:00:42 2006
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 Mon, Jul 24, 2006 at 20:08:13, Kingston.Smiler wrote about "[Sip-implementors] Regarding non-INVITE server transaction": 

>         If the non-IST doesn't receive any RESPONSE message then what will 
> happen?
> RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention 
> properly 
> what will happen if the  non-IST doesn't receive any provisional 
> response/final response.

==={{{
8.1.3.1 Transaction Layer Errors

   In some cases, the response returned by the transaction layer will
   not be a SIP message, but rather a transaction layer error.  When a
   timeout error is received from the transaction layer, it MUST be
   treated as if a 408 (Request Timeout) status code has been received.
   If a fatal transport error is reported by the transport layer
   (generally, due to fatal ICMP errors in UDP or connection failures in
   TCP), the condition MUST be treated as a 503 (Service Unavailable)
   status code.
===}}}

-- 
Valentin Nechayev
PortaOne Inc., Software Engineer
mailto:netch@portaone.com

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From: Manjunath Warad <manjunathwarad@huawei.com>
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Hi,
	Below mentioned behaviour is for client transaction, not for
server transaction.

Rgds,
Manju

-----Original Message-----
From: sip-implementors-bounces@cs.columbia.edu
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of Valentin
Nechayev
Sent: Tuesday, July 25, 2006 4:55 PM
To: Kingston.Smiler@flextronicssoftware.com
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: Re: [Sip-implementors] Regarding non-INVITE server transaction

 Mon, Jul 24, 2006 at 20:08:13, Kingston.Smiler wrote about
"[Sip-implementors] Regarding non-INVITE server transaction": 

>         If the non-IST doesn't receive any RESPONSE message then what
will 
> happen?
> RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention

> properly 
> what will happen if the  non-IST doesn't receive any provisional 
> response/final response.

==={{{
8.1.3.1 Transaction Layer Errors

   In some cases, the response returned by the transaction layer will
   not be a SIP message, but rather a transaction layer error.  When a
   timeout error is received from the transaction layer, it MUST be
   treated as if a 408 (Request Timeout) status code has been received.
   If a fatal transport error is reported by the transport layer
   (generally, due to fatal ICMP errors in UDP or connection failures in
   TCP), the condition MUST be treated as a 503 (Service Unavailable)
   status code.
===}}}

-- 
Valentin Nechayev
PortaOne Inc., Software Engineer
mailto:netch@portaone.com
_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Tue Jul 25 08:55:26 2006
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Subject: RE: [Sip] RE: [Sip-implementors] Regarding non-INVITE
	servertransaction
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Hi,
        I agree with Stephen because it is not a problem in non - IST; 
This is generic to non-IST as well as IST. 

But consider a scenario like this 

UAC of        UAS of                     UAS of UE2
      UE1                proxy 
 |              |      |                        |
 |         non-INV      |      |     non-INV            |
 |--------------------->|      |---------------------------->|
 |              |      |                2xx     |
 |              |      |                *<---------------|
 |              |      |                        |
                    UAC of 
                                     Proxy
 

         Here UAS of UE2 is re-transmitting the 2xx message toward proxy.
But the message that has been sent by UAS of UE2 hasn't reached UAC of 
proxy
(due to network breakage or some other means). So the proxy core (TU) 
doesn't 
send the response to UAS of proxy.  In this case how will UAS txn of proxy 
can be released.

Regards,
S.Kingston Smiler.
 




Stephen Zhou <zhouhong@huawei.com> 
07/25/2006 03:21 PM


To
"'Manjunath Warad'" <manjunathwarad@huawei.com>, netch@portaone.com, 
Kingston Smiler/BLR/HSS@HSS
cc
sip@ietf.org, sip-implementors@cs.columbia.edu
Subject
RE: [Sip] RE: [Sip-implementors] Regarding non-INVITE   servertransaction






Hi,

Manjunath is right.

We MUST assume the communication between TU layer and transation lay in a
sip stack  is reliable. If there have problems, I thinks this is program
problem rather than protocol(transaction layer) problem. 
Additionally, the same issue exist in IST, if the IST cann't receive the
responses from TU.

BR,
Stephen
-----Original Message-----
From: Manjunath Warad [mailto:manjunathwarad@huawei.com] 
Sent: Tuesday, July 25, 2006 5:08 PM
To: netch@portaone.com; Kingston.Smiler@flextronicssoftware.com
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: [Sip] RE: [Sip-implementors] Regarding non-INVITE server
transaction

Hi,
                 Below mentioned behaviour is for client transaction, not 
for
server transaction.

Rgds,
Manju

-----Original Message-----
From: sip-implementors-bounces@cs.columbia.edu
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of Valentin
Nechayev
Sent: Tuesday, July 25, 2006 4:55 PM
To: Kingston.Smiler@flextronicssoftware.com
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: Re: [Sip-implementors] Regarding non-INVITE server transaction

 Mon, Jul 24, 2006 at 20:08:13, Kingston.Smiler wrote about
"[Sip-implementors] Regarding non-INVITE server transaction": 

>         If the non-IST doesn't receive any RESPONSE message then what
will 
> happen?
> RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention

> properly 
> what will happen if the  non-IST doesn't receive any provisional 
> response/final response.

==={{{
8.1.3.1 Transaction Layer Errors

   In some cases, the response returned by the transaction layer will
   not be a SIP message, but rather a transaction layer error.  When a
   timeout error is received from the transaction layer, it MUST be
   treated as if a 408 (Request Timeout) status code has been received.
   If a fatal transport error is reported by the transport layer
   (generally, due to fatal ICMP errors in UDP or connection failures in
   TCP), the condition MUST be treated as a 503 (Service Unavailable)
   status code.
===}}}

-- 
Valentin Nechayev
PortaOne Inc., Software Engineer
mailto:netch@portaone.com
_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





***********************  FSS-Unclassified   ***********************
--=_alternative 004671B6652571B6_=
Content-Type: text/html; charset="US-ASCII"


<br><font size=2 face="sans-serif">Hi,<br>
 &nbsp; &nbsp; &nbsp; &nbsp;I agree with </font><font size=1 face="sans-serif"><b>Stephen</b></font><font size=2 face="sans-serif">
because it is not a problem in non - IST; </font>
<br><font size=2 face="sans-serif">This is generic to non-IST as well as
IST. </font>
<br>
<br><font size=2 face="sans-serif">But consider a scenario like this </font>
<br>
<br><font size=3 face="Times New Roman">UAC of &nbsp; &nbsp; &nbsp; &nbsp;UAS
of &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
UAS of UE2</font>
<br><font size=3 face="Times New Roman">&nbsp; &nbsp; &nbsp; UE1 &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; proxy &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br><font size=2 face="sans-serif">&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;|</font>
<br><font size=2 face="sans-serif">&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;
non-INV &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp;
&nbsp; non-INV &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;|</font>
<br><font size=2 face="sans-serif">&nbsp;|---------------------&gt;| &nbsp;
&nbsp; &nbsp;|----------------------------&gt;|</font>
<br><font size=2 face="sans-serif">&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;2xx &nbsp; &nbsp;
&nbsp; &nbsp;|</font>
<br><font size=2 face="sans-serif">&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;*&lt;---------------|</font>
<br><font size=2 face="sans-serif">&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;|</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; UAC of </font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;Proxy</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Here
UAS of UE2 is re-transmitting the 2xx message toward proxy.</font>
<br><font size=2 face="sans-serif">But the message that has been sent by
UAS of UE2 hasn't reached UAC of proxy</font>
<br><font size=2 face="sans-serif">(due to network breakage or some other
means). So the proxy core (TU) doesn't </font>
<br><font size=2 face="sans-serif">send the response to UAS of proxy. &nbsp;In
this case how will UAS txn of proxy can be released.</font>
<br>
<br><font size=2 face="sans-serif">Regards,<br>
S.Kingston Smiler.<br>
 <br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td width=40%><font size=1 face="sans-serif"><b>Stephen Zhou &lt;zhouhong@huawei.com&gt;</b>
</font>
<p><font size=1 face="sans-serif">07/25/2006 03:21 PM</font>
<br>
<td width=59%>
<table width=100%>
<tr>
<td>
<div align=right><font size=1 face="sans-serif">To</font></div>
<td valign=top><font size=1 face="sans-serif">&quot;'Manjunath Warad'&quot;
&lt;manjunathwarad@huawei.com&gt;, netch@portaone.com, Kingston Smiler/BLR/HSS@HSS</font>
<tr>
<td>
<div align=right><font size=1 face="sans-serif">cc</font></div>
<td valign=top><font size=1 face="sans-serif">sip@ietf.org, sip-implementors@cs.columbia.edu</font>
<tr>
<td>
<div align=right><font size=1 face="sans-serif">Subject</font></div>
<td valign=top><font size=1 face="sans-serif">RE: [Sip] RE: [Sip-implementors]
Regarding non-INVITE &nbsp; &nbsp; &nbsp; &nbsp; servertransaction</font></table>
<br>
<table>
<tr valign=top>
<td>
<td></table>
<br></table>
<br>
<br>
<br><font size=2><tt>Hi,<br>
<br>
Manjunath is right.<br>
<br>
We MUST assume the communication between TU layer and transation lay in
a<br>
sip stack &nbsp;is reliable. If there have problems, I thinks this is program<br>
problem rather than protocol(transaction layer) problem. <br>
Additionally, the same issue exist in IST, if the IST cann't receive the<br>
responses from TU.<br>
<br>
BR,<br>
Stephen<br>
-----Original Message-----<br>
From: Manjunath Warad [mailto:manjunathwarad@huawei.com] <br>
Sent: Tuesday, July 25, 2006 5:08 PM<br>
To: netch@portaone.com; Kingston.Smiler@flextronicssoftware.com<br>
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu<br>
Subject: [Sip] RE: [Sip-implementors] Regarding non-INVITE server<br>
transaction<br>
<br>
Hi,<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
Below mentioned behaviour is for client transaction, not for<br>
server transaction.<br>
<br>
Rgds,<br>
Manju<br>
<br>
-----Original Message-----<br>
From: sip-implementors-bounces@cs.columbia.edu<br>
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of Valentin<br>
Nechayev<br>
Sent: Tuesday, July 25, 2006 4:55 PM<br>
To: Kingston.Smiler@flextronicssoftware.com<br>
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu<br>
Subject: Re: [Sip-implementors] Regarding non-INVITE server transaction<br>
<br>
 Mon, Jul 24, 2006 at 20:08:13, Kingston.Smiler wrote about<br>
&quot;[Sip-implementors] Regarding non-INVITE server transaction&quot;:
<br>
<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; If the non-IST doesn't receive any RESPONSE
message then what<br>
will <br>
&gt; happen?<br>
&gt; RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention<br>
<br>
&gt; properly <br>
&gt; what will happen if the &nbsp;non-IST doesn't receive any provisional
<br>
&gt; response/final response.<br>
<br>
==={{{<br>
8.1.3.1 Transaction Layer Errors<br>
<br>
 &nbsp; In some cases, the response returned by the transaction layer will<br>
 &nbsp; not be a SIP message, but rather a transaction layer error. &nbsp;When
a<br>
 &nbsp; timeout error is received from the transaction layer, it MUST be<br>
 &nbsp; treated as if a 408 (Request Timeout) status code has been received.<br>
 &nbsp; If a fatal transport error is reported by the transport layer<br>
 &nbsp; (generally, due to fatal ICMP errors in UDP or connection failures
in<br>
 &nbsp; TCP), the condition MUST be treated as a 503 (Service Unavailable)<br>
 &nbsp; status code.<br>
===}}}<br>
<br>
-- <br>
Valentin Nechayev<br>
PortaOne Inc., Software Engineer<br>
mailto:netch@portaone.com<br>
_______________________________________________<br>
Sip-implementors mailing list<br>
Sip-implementors@cs.columbia.edu<br>
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
<br>
<br>
</tt></font>
<br><font size=2 face="sans-serif"><br>
<br>
*********************** &nbsp;FSS-Unclassified &nbsp; ***********************</font>
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Use sipping@ietf.org for new developments on the application of sip
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Date: Tue, 25 Jul 2006 17:51:27 +0800
From: Stephen Zhou <zhouhong@huawei.com>
Subject: RE: [Sip] RE: [Sip-implementors] Regarding non-INVITE server
	transaction
In-reply-to: <029e01c6afc9$d0bb4920$0eb74e0a@china.huawei.com>
To: 'Manjunath Warad' <manjunathwarad@huawei.com>, netch@portaone.com,
	Kingston.Smiler@flextronicssoftware.com
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Errors-To: sip-bounces@ietf.org

Hi,

Manjunath is right.

We MUST assume the communication between TU layer and transation lay in a
sip stack  is reliable. If there have problems, I thinks this is program
problem rather than protocol(transaction layer) problem. 
Additionally, the same issue exist in IST, if the IST cann't receive the
responses from TU.

BR,
Stephen
-----Original Message-----
From: Manjunath Warad [mailto:manjunathwarad@huawei.com] 
Sent: Tuesday, July 25, 2006 5:08 PM
To: netch@portaone.com; Kingston.Smiler@flextronicssoftware.com
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: [Sip] RE: [Sip-implementors] Regarding non-INVITE server
transaction

Hi,
	Below mentioned behaviour is for client transaction, not for
server transaction.

Rgds,
Manju

-----Original Message-----
From: sip-implementors-bounces@cs.columbia.edu
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of Valentin
Nechayev
Sent: Tuesday, July 25, 2006 4:55 PM
To: Kingston.Smiler@flextronicssoftware.com
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: Re: [Sip-implementors] Regarding non-INVITE server transaction

 Mon, Jul 24, 2006 at 20:08:13, Kingston.Smiler wrote about
"[Sip-implementors] Regarding non-INVITE server transaction": 

>         If the non-IST doesn't receive any RESPONSE message then what
will 
> happen?
> RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention

> properly 
> what will happen if the  non-IST doesn't receive any provisional 
> response/final response.

==={{{
8.1.3.1 Transaction Layer Errors

   In some cases, the response returned by the transaction layer will
   not be a SIP message, but rather a transaction layer error.  When a
   timeout error is received from the transaction layer, it MUST be
   treated as if a 408 (Request Timeout) status code has been received.
   If a fatal transport error is reported by the transport layer
   (generally, due to fatal ICMP errors in UDP or connection failures in
   TCP), the condition MUST be treated as a 503 (Service Unavailable)
   status code.
===}}}

-- 
Valentin Nechayev
PortaOne Inc., Software Engineer
mailto:netch@portaone.com
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From sip-bounces@ietf.org Tue Jul 25 13:43:43 2006
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From: "Brian Rosen" <br@brianrosen.net>
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Subject: [Sip] Consensus on changes to location-conveyance
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I have put out a number of emails on proposed changes to location
conveyance.  Some of them have sparked a number of comments.  My summary of
consensus on these issues is:

1. We proposed to explicitly disallow use of a Data URI, which would allow
specification of a Location-by-value in a header.  There was a lot of list
traffic on this, currently in the state where Keith asked for a real use
case for it.  So far, we don't see a use case that cannot be provided by
location-by-reference in a header.  If we don't get a good use case, I think
consensus is to disallow location-by-value in a header.

2. Dereferencing location-by-reference.  We had initially proposed to
document a process to use a raw HTTP(S) Get to dereference.  There was a lot
of discussion, and then I made a new proposal, see item 6. 

3. We proposed to allow a Reason header with a 424 Bad Location response.
This seems to be acceptable to the WG.  Geopriv will create and populate the
registry in a separate draft.

4. We proposed to explicitly allow hop-by-hop TLS security for location when
the endpoint wished to allow routing based on location.  We further proposed
to interpret "Do not Distribute" flags as allowing sending of the location
to the routing database (LoST for example).  Hannes pointed out that there
was a proposal to allow two Location headers, one of which could be used for
routing and another for onward forwarding to the recipient.  Hannes' message
seemed to agree that "Do not distribute" would allow sending of the location
to the routing-by-location database, but would obligate the proxy that did
that to remove the location before it forwarded the message.  There were no
other messages on this thread.  We conclude that allowing hop-by-hop
security for location based routing is acceptable, and Do Not Distribute
DOES allow forwarding to a routing database, but requires that the proxy
doing the location based route to remove the location it uses for that
purpose.

5. We proposed to create a parameter and a registry to distinguish multiple
ways to dereference a location where HTTP(S) was the transport.  Resolution
of this issue depends on resolution of issue 6.

6. We proposed to allow sip/sips and pres: in the location header.  I later
speculated that if we allow this, it forms a simple way to do
location-by-reference, and we could eliminate the raw HTTP(S) Get proposal,
with the parameter and registry entirely.  There was some list traffic in
favor of this.  Martin Thompson is skeptical that SIP Presence
Subscribe/Notify is actually allowed as a geopriv "using protocol".  Several
of think that it is.  If we don't get any more comments on this proposal, we
will allow sip/sips and pres, not have any http(s) resolution text, and not
propose a parameter and accompanying registry.

If this does not reflect your position, please speak up now.

Brian


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From sip-bounces@ietf.org Tue Jul 25 13:57:24 2006
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Subject: [Sip] Hitchhikers guide - when is it done?
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During the IETF meeting, I discussed the one current open issue in the 
Hitchhikers guide. The issue is: when does the document finish? When do 
we send to IESG to issue an RFC?

There were two approaches:

1. split the document in two. One part contains only the core specs, and 
issues relatively soon. The other part has everything else, and 
basically ends up being the last RFC produced by the SIP WG before it 
shuts its doors (which has always been 1.5 years from now for the last 5 
years).

2. keep it as one document. Issue an RFC at some point, and then revise 
it a few times as needed. We need to decide when to issue it.


I'd like to propose option 2. I think people get more value from it as a 
consolidated document. Furthermore, I'd propose that we send the 
document to IESG for consideration as informational at the instant all 
of the documents listed as 'core' have RFC numbers issued by RFC-ed.

Comments?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Tue Jul 25 14:00:35 2006
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From: Somesh S Shanbhag <someshss@yahoo.com>
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Subject: [Sip] Re: [Sip-implementors] Regarding non-INVITE server transaction
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Hi Kingston,
  
  RFC 3261 section 8.2.6.1 suggests the following ...
  
  
   One largely non-method-specific guideline for the generation of
   responses is that UASs SHOULD NOT issue a provisional response for a
   non-INVITE request.  Rather, UASs SHOULD generate a final response to
   a non-INVITE request as soon as possible.

   When a 100 (Trying) response is generated, any Timestamp header field
   present in the request MUST be copied into this 100 (Trying)
   response.  If there is a delay in generating the response, the UAS
   SHOULD add a delay value into the Timestamp value in the response.
   This value MUST contain the difference between the time of sending of
   the response and receipt of the request, measured in seconds.

So, Its expected that the UAS respond to non-INVITE requests
immediately with preferably a final response and not a
provisional response...

So, I think if the case that you have mentioned takes place I think
Transaction User (TU) must take care and Transaction Layer
shall remain un-noticed ....

Thanks & Regards,
Somesh S. Shanbhag

  
Kingston.Smiler@flextronicssoftware.com wrote:  Hi,
        If the non-IST doesn't receive any RESPONSE message then what will 
happen?
RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention 
properly 
what will happen if the  non-IST doesn't receive any provisional 
response/final response.
 
Regards,
S.Kingston Smiler.
 


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-----------------------------------------
SIMPLICITY IS THE BEAUTY.
BE NATURAL LIVE NATURAL.
-----------------------------------------
Somesh S. Shanbhag
Focus Area - VoIP Team (FA-VoIP)
Mascon Global Limited
#59/2, 100Ft Ring Road
Banashankari II stage
Bangalore-560070
Karnataka
INDIA
Website: http://www.mgl.com/
-----------------------------------------
 		
---------------------------------
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Hi Kingston,<br>  <br>  RFC 3261 section 8.2.6.1 suggests the following ...<br>  <br>  <pre>   One largely non-method-specific guideline for the generation of<br>   responses is that UASs SHOULD NOT issue a provisional response for a<br>   non-INVITE request.  Rather, UASs SHOULD generate a final response to<br>   a non-INVITE request as soon as possible.<br><br>   When a 100 (Trying) response is generated, any Timestamp header field<br>   present in the request MUST be copied into this 100 (Trying)<br>   response.  If there is a delay in generating the response, the UAS<br>   SHOULD add a delay value into the Timestamp value in the response.<br>   This value MUST contain the difference between the time of sending of<br>   the response and receipt of the request, measured in seconds.<br><br>So, Its expected that the UAS respond to non-INVITE requests<br>immediately with preferably a final response and not a<br>provisional response...<br><br>So, I think if the case that you
 have mentioned takes place I think<br>Transaction User (TU) must take care and Transaction Layer<br>shall remain un-noticed ....<br><br>Thanks &amp; Regards,<br>Somesh S. Shanbhag<br></pre>  <br><b><i>Kingston.Smiler@flextronicssoftware.com</i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">  Hi,<br>        If the non-IST doesn't receive any RESPONSE message then what will <br>happen?<br>RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention <br>properly <br>what will happen if the  non-IST doesn't receive any provisional <br>response/final response.<br> <br>Regards,<br>S.Kingston Smiler.<br> <br><br><br>***********************  FSS-Unclassified   ***********************<br>_______________________________________________<br>Sip-implementors mailing
 list<br>Sip-implementors@cs.columbia.edu<br>https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors<br></blockquote><br><BR><BR>-----------------------------------------<br>SIMPLICITY IS THE BEAUTY.<br>BE NATURAL LIVE NATURAL.<br>-----------------------------------------<br>Somesh S. Shanbhag<br>Focus Area - VoIP Team (FA-VoIP)<br>Mascon Global Limited<br>#59/2, 100Ft Ring Road<br>Banashankari II stage<br>Bangalore-560070<br>Karnataka<br>INDIA<br>Website: http://www.mgl.com/<br>-----------------------------------------<p>&#32;
		<hr size=1>Do you Yahoo!?<br> Next-gen email? Have it all with the <a href="http://us.rd.yahoo.com/evt=42241/*http://advision.webevents.yahoo.com/handraisers"> all-new Yahoo! Mail Beta.</a>
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From sip-bounces@ietf.org Tue Jul 25 14:07:55 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>,
	"'IETF SIP List'" <sip@ietf.org>
Subject: RE: [Sip] Hitchhikers guide - when is it done?
Date: Tue, 25 Jul 2006 14:05:56 -0400
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I like it, but I want to know why you want to wait until you have core RFC
numbers.  Whenever you put it in, you will block on normative references.
SIP will be guaranteed to move on before it issues.  You have to decide if
you try to edit it as that happens (AUTH48) or not. 

I'd submit it when it's done, and not wait. 

Brian

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: Tuesday, July 25, 2006 1:56 PM
> To: IETF SIP List
> Subject: [Sip] Hitchhikers guide - when is it done?
> 
> During the IETF meeting, I discussed the one current open issue in the
> Hitchhikers guide. The issue is: when does the document finish? When do
> we send to IESG to issue an RFC?
> 
> There were two approaches:
> 
> 1. split the document in two. One part contains only the core specs, and
> issues relatively soon. The other part has everything else, and
> basically ends up being the last RFC produced by the SIP WG before it
> shuts its doors (which has always been 1.5 years from now for the last 5
> years).
> 
> 2. keep it as one document. Issue an RFC at some point, and then revise
> it a few times as needed. We need to decide when to issue it.
> 
> 
> I'd like to propose option 2. I think people get more value from it as a
> consolidated document. Furthermore, I'd propose that we send the
> document to IESG for consideration as informational at the instant all
> of the documents listed as 'core' have RFC numbers issued by RFC-ed.
> 
> Comments?
> 
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From sip-bounces@ietf.org Tue Jul 25 14:08:56 2006
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Subject: RE: [Sip] Hitchhikers guide - when is it done?
Date: Tue, 25 Jul 2006 20:07:02 +0200
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Thread-Topic: [Sip] Hitchhikers guide - when is it done?
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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>, "IETF SIP List" <sip@ietf.org>
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(As individual)

I also prefer option 2.

I think the IETF has to solve the problem of more rapid revision of
documents.

Regards

Keith=20

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: 25 July 2006 18:56
> To: IETF SIP List
> Subject: [Sip] Hitchhikers guide - when is it done?
>=20
> During the IETF meeting, I discussed the one current open=20
> issue in the Hitchhikers guide. The issue is: when does the=20
> document finish? When do we send to IESG to issue an RFC?
>=20
> There were two approaches:
>=20
> 1. split the document in two. One part contains only the core=20
> specs, and issues relatively soon. The other part has=20
> everything else, and basically ends up being the last RFC=20
> produced by the SIP WG before it shuts its doors (which has=20
> always been 1.5 years from now for the last 5 years).
>=20
> 2. keep it as one document. Issue an RFC at some point, and=20
> then revise it a few times as needed. We need to decide when=20
> to issue it.
>=20
>=20
> I'd like to propose option 2. I think people get more value=20
> from it as a=20
> consolidated document. Furthermore, I'd propose that we send the=20
> document to IESG for consideration as informational at the=20
> instant all=20
> of the documents listed as 'core' have RFC numbers issued by RFC-ed.
>=20
> Comments?
>=20
> -Jonathan R.
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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From sip-bounces@ietf.org Tue Jul 25 14:15:35 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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	<sip@ietf.org>
References: <013501c6b011$98b8d360$640fa8c0@cis.neustar.com>
Subject: Re: [Sip] Consensus on changes to location-conveyance
Date: Tue, 25 Jul 2006 20:13:41 +0200
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Brian,

Your point 4 below is a concrete use case for point 1 (location by value in 
header). Using location for routing decisions benefits from quick access, a 
lookup by reference introduces delay.

If you want a proxy to remove the location info, you cannot put the location 
info in the body, because proxies are not allowed to modify that. So you are 
left with location by reference. The use of TLS is then only to protect this 
URL?

Regards,

Jeroen

----- Original Message ----- 
From: "Brian Rosen" <br@brianrosen.net>
To: <sip@ietf.org>
Sent: Tuesday, July 25, 2006 7:41 PM
Subject: [Sip] Consensus on changes to location-conveyance


>I have put out a number of emails on proposed changes to location
> conveyance.  Some of them have sparked a number of comments.  My summary 
> of
> consensus on these issues is:
>
> 1. We proposed to explicitly disallow use of a Data URI, which would allow
> specification of a Location-by-value in a header.  There was a lot of list
> traffic on this, currently in the state where Keith asked for a real use
> case for it.  So far, we don't see a use case that cannot be provided by
> location-by-reference in a header.  If we don't get a good use case, I 
> think
> consensus is to disallow location-by-value in a header.
>
> 2. Dereferencing location-by-reference.  We had initially proposed to
> document a process to use a raw HTTP(S) Get to dereference.  There was a 
> lot
> of discussion, and then I made a new proposal, see item 6.
>
> 3. We proposed to allow a Reason header with a 424 Bad Location response.
> This seems to be acceptable to the WG.  Geopriv will create and populate 
> the
> registry in a separate draft.
>
> 4. We proposed to explicitly allow hop-by-hop TLS security for location 
> when
> the endpoint wished to allow routing based on location.  We further 
> proposed
> to interpret "Do not Distribute" flags as allowing sending of the location
> to the routing database (LoST for example).  Hannes pointed out that there
> was a proposal to allow two Location headers, one of which could be used 
> for
> routing and another for onward forwarding to the recipient.  Hannes' 
> message
> seemed to agree that "Do not distribute" would allow sending of the 
> location
> to the routing-by-location database, but would obligate the proxy that did
> that to remove the location before it forwarded the message.  There were 
> no
> other messages on this thread.  We conclude that allowing hop-by-hop
> security for location based routing is acceptable, and Do Not Distribute
> DOES allow forwarding to a routing database, but requires that the proxy
> doing the location based route to remove the location it uses for that
> purpose.
>
> 5. We proposed to create a parameter and a registry to distinguish 
> multiple
> ways to dereference a location where HTTP(S) was the transport. 
> Resolution
> of this issue depends on resolution of issue 6.
>
> 6. We proposed to allow sip/sips and pres: in the location header.  I 
> later
> speculated that if we allow this, it forms a simple way to do
> location-by-reference, and we could eliminate the raw HTTP(S) Get 
> proposal,
> with the parameter and registry entirely.  There was some list traffic in
> favor of this.  Martin Thompson is skeptical that SIP Presence
> Subscribe/Notify is actually allowed as a geopriv "using protocol". 
> Several
> of think that it is.  If we don't get any more comments on this proposal, 
> we
> will allow sip/sips and pres, not have any http(s) resolution text, and 
> not
> propose a parameter and accompanying registry.
>
> If this does not reflect your position, please speak up now.
>
> Brian
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Tue Jul 25 14:32:36 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>,
	<sip@ietf.org>
Subject: RE: [Sip] Consensus on changes to location-conveyance
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Jeroen

I've always disliked location-by-reference because it has delay, and
introduces the possibility (in the emergency case especially) where the
dereferencer has connectivity to the endpoint but not the LIS.

The WG however, has consensus that these issues are not significant, and the
benefits outweigh the problems.

Having said that using a reference is NOT seen as a performance problem then
it's not a performance problem, and you can't argue it's a use case.

Agree that if the endpoint wants location to be removed after routing before
forwarding, then it has to use by-reference.  Don't see that as a problem.

Yes, TLS protects the URL, which is important, because anyone observing the
SIP messaging can connect the identity of the SIP correspondent with the
location (if it is allowed to dereference it), which is bad.

Brian

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Tuesday, July 25, 2006 2:14 PM
> To: Brian Rosen; sip@ietf.org
> Subject: Re: [Sip] Consensus on changes to location-conveyance
> 
> Brian,
> 
> Your point 4 below is a concrete use case for point 1 (location by value
> in
> header). Using location for routing decisions benefits from quick access,
> a
> lookup by reference introduces delay.
> 
> If you want a proxy to remove the location info, you cannot put the
> location
> info in the body, because proxies are not allowed to modify that. So you
> are
> left with location by reference. The use of TLS is then only to protect
> this
> URL?
> 
> Regards,
> 
> Jeroen
> 
> ----- Original Message -----
> From: "Brian Rosen" <br@brianrosen.net>
> To: <sip@ietf.org>
> Sent: Tuesday, July 25, 2006 7:41 PM
> Subject: [Sip] Consensus on changes to location-conveyance
> 
> 
> >I have put out a number of emails on proposed changes to location
> > conveyance.  Some of them have sparked a number of comments.  My summary
> > of
> > consensus on these issues is:
> >
> > 1. We proposed to explicitly disallow use of a Data URI, which would
> allow
> > specification of a Location-by-value in a header.  There was a lot of
> list
> > traffic on this, currently in the state where Keith asked for a real use
> > case for it.  So far, we don't see a use case that cannot be provided by
> > location-by-reference in a header.  If we don't get a good use case, I
> > think
> > consensus is to disallow location-by-value in a header.
> >
> > 2. Dereferencing location-by-reference.  We had initially proposed to
> > document a process to use a raw HTTP(S) Get to dereference.  There was a
> > lot
> > of discussion, and then I made a new proposal, see item 6.
> >
> > 3. We proposed to allow a Reason header with a 424 Bad Location
> response.
> > This seems to be acceptable to the WG.  Geopriv will create and populate
> > the
> > registry in a separate draft.
> >
> > 4. We proposed to explicitly allow hop-by-hop TLS security for location
> > when
> > the endpoint wished to allow routing based on location.  We further
> > proposed
> > to interpret "Do not Distribute" flags as allowing sending of the
> location
> > to the routing database (LoST for example).  Hannes pointed out that
> there
> > was a proposal to allow two Location headers, one of which could be used
> > for
> > routing and another for onward forwarding to the recipient.  Hannes'
> > message
> > seemed to agree that "Do not distribute" would allow sending of the
> > location
> > to the routing-by-location database, but would obligate the proxy that
> did
> > that to remove the location before it forwarded the message.  There were
> > no
> > other messages on this thread.  We conclude that allowing hop-by-hop
> > security for location based routing is acceptable, and Do Not Distribute
> > DOES allow forwarding to a routing database, but requires that the proxy
> > doing the location based route to remove the location it uses for that
> > purpose.
> >
> > 5. We proposed to create a parameter and a registry to distinguish
> > multiple
> > ways to dereference a location where HTTP(S) was the transport.
> > Resolution
> > of this issue depends on resolution of issue 6.
> >
> > 6. We proposed to allow sip/sips and pres: in the location header.  I
> > later
> > speculated that if we allow this, it forms a simple way to do
> > location-by-reference, and we could eliminate the raw HTTP(S) Get
> > proposal,
> > with the parameter and registry entirely.  There was some list traffic
> in
> > favor of this.  Martin Thompson is skeptical that SIP Presence
> > Subscribe/Notify is actually allowed as a geopriv "using protocol".
> > Several
> > of think that it is.  If we don't get any more comments on this
> proposal,
> > we
> > will allow sip/sips and pres, not have any http(s) resolution text, and
> > not
> > propose a parameter and accompanying registry.
> >
> > If this does not reflect your position, please speak up now.
> >
> > Brian
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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*Hi,
**
**	   I am supposed to build a additional features  sip based ip phone for
*the dissertation
**
* Could somebody give me guidance as to how I can go about this?
**	Where can I get a SIP stack ? How can I code (preferably in
*C++) *using this stack? **
*
*SIP stack  which i can run on windows xp *
 *
Anxiously awainting ur responses....*


kind regards,
banu

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<pre><i>Hi,<br></i><i><br></i><i>	   I am supposed to build a additional features  sip based ip phone for<br></i>the dissertation <br><i></i><br><i>&nbsp;Could somebody give me guidance as to how I can go about this?<br></i><i>
	Where can I get a SIP stack ? How can I code (preferably in<br></i>C++) <i>using this stack? </i><i><span style="font-style: italic;"><br><span style="font-style: italic;"></span></span></i><br><i><i>SIP stack  which i can run on windows xp 
</i></i>
 <i><span style="font-style: italic;"><br></span>Anxiously awainting ur responses....</i></pre><br>kind regards,<br>banu<br>

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_______________________________________________
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Hi ,

Thanks a lot for the response.

I am new to SIP's practical applications. Could you please give me
guidance as to how I can use this SIP stack to develop a additional
features like security  and privacy


Thanks in advance..

Regards
Banu

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<pre>Hi ,<br><br>Thanks a lot for the response.<br><br>I am new to SIP's practical applications. Could you please give me<br>guidance as to how I can use this SIP stack to develop a additional features like security  and privacy
<br><br><br>Thanks in advance..<br><br>Regards<br>Banu</pre>

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Subject: RE: [Sip] Hitchhikers guide - when is it done?
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Option 2 also.

Issuing a list of core RFCs as option 1 suggests requires a strict list of 
core vs. non core
sooner then later. Later will be better in this case.

--Avshalom





"Drage, Keith \(Keith\)" <drage@lucent.com> 
25/07/2006 21:07

To
"Jonathan Rosenberg" <jdrosen@cisco.com>, "IETF SIP List" <sip@ietf.org>
cc

Subject
RE: [Sip] Hitchhikers guide - when is it done?






(As individual)

I also prefer option 2.

I think the IETF has to solve the problem of more rapid revision of
documents.

Regards

Keith 

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
> Sent: 25 July 2006 18:56
> To: IETF SIP List
> Subject: [Sip] Hitchhikers guide - when is it done?
> 
> During the IETF meeting, I discussed the one current open 
> issue in the Hitchhikers guide. The issue is: when does the 
> document finish? When do we send to IESG to issue an RFC?
> 
> There were two approaches:
> 
> 1. split the document in two. One part contains only the core 
> specs, and issues relatively soon. The other part has 
> everything else, and basically ends up being the last RFC 
> produced by the SIP WG before it shuts its doors (which has 
> always been 1.5 years from now for the last 5 years).
> 
> 2. keep it as one document. Issue an RFC at some point, and 
> then revise it a few times as needed. We need to decide when 
> to issue it.
> 
> 
> I'd like to propose option 2. I think people get more value 
> from it as a 
> consolidated document. Furthermore, I'd propose that we send the 
> document to IESG for consideration as informational at the 
> instant all 
> of the documents listed as 'core' have RFC numbers issued by RFC-ed.
> 
> Comments?
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From sip-bounces@ietf.org Tue Jul 25 15:42:27 2006
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Subject: Re: [Sip] Hitchhikers guide - when is it done?
From: Daryl Malas <daryl@level3.net>
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I like option 2.  I would like to see it submitted sooner rather than
later.  Any pending or new RFC's can be added to a new revision, which
will obsolete this one down the road. --Daryl

On Tue, 2006-07-25 at 13:55 -0400, Jonathan Rosenberg wrote:
> During the IETF meeting, I discussed the one current open issue in the 
> Hitchhikers guide. The issue is: when does the document finish? When do 
> we send to IESG to issue an RFC?
> 
> There were two approaches:
> 
> 1. split the document in two. One part contains only the core specs, and 
> issues relatively soon. The other part has everything else, and 
> basically ends up being the last RFC produced by the SIP WG before it 
> shuts its doors (which has always been 1.5 years from now for the last 5 
> years).
> 
> 2. keep it as one document. Issue an RFC at some point, and then revise 
> it a few times as needed. We need to decide when to issue it.
> 
> 
> I'd like to propose option 2. I think people get more value from it as a 
> consolidated document. Furthermore, I'd propose that we send the 
> document to IESG for consideration as informational at the instant all 
> of the documents listed as 'core' have RFC numbers issued by RFC-ed.
> 
> Comments?
> 
> -Jonathan R.


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From sip-bounces@ietf.org Tue Jul 25 15:52:27 2006
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Thread-Topic: [Geopriv] Consensus on changes to location-conveyance
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "Marc Linsner" <mlinsner@cisco.com>, <sip@ietf.org>
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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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When the original PIDF-LO RFC came out, location-by-reference was
defined.

You have a URI.  You can exchange the URI for location.  There is a
protocol defined for that purpose (SIP SUBSCRIBE).  That is location by
reference.  It also serves as the location update mechanism, clearly
something we need.

Some day, there may be other location URI references.  We don't need to
cover them in this draft.  The basic location by reference horse has
left the barn.  Were you planning on creating an RFC that specifically
disallows a Subscribe on a sip/sips/pres URI to get location?   THAT WAS
THE POINT OF MAKING IT PART OF A PIDF.

I do think there is some value in defining a way to express a
location-by-reference aka presence URI which is not just the AoR.  That
is one reason why I want to define the Location URI as being able to
contain a sip/sips/pres URI.  The purpose is to allow a UA or a proxy to
insert location where there is no association between the identity of
the SIP user and the location.=20

But that is all I think we need.

We have a requirement for proxy insertion of location.  If we disallow
location-by-reference (which I think is a good idea in general, but
futile in practice), then we would need something like the Data URI to
support location insertion by proxy.  If we DO allow
location-by-reference, then it also serves to meet the location
insertion by proxy requirement, and we don't need the Data URI.  We're
looking for a use case which requires proxy insertion by value, but
haven't heard one yet.

You can make an argument that you can take the From:, assume it's an
AoR, and SUBSCRIBE to the presence of that AoR to get location, thus not
needing a Location header.  I really think letting the Location header
specify an alternate presence URI is a good idea.

Brian

> -----Original Message-----
> From: Marc Linsner [mailto:mlinsner@cisco.com]
> Sent: Tuesday, July 25, 2006 3:29 PM
> To: Rosen, Brian; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>=20
> In-line.....
>=20
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Tuesday, July 25, 2006 1:38 PM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: [Geopriv] Consensus on changes to location-conveyance
> >
> > I have put out a number of emails on proposed changes to
> > location conveyance.  Some of them have sparked a number of
> > comments.  My summary of consensus on these issues is:
> >
> > 1. We proposed to explicitly disallow use of a Data URI,
> > which would allow specification of a Location-by-value in a
> > header.  There was a lot of list traffic on this, currently
> > in the state where Keith asked for a real use case for it.
> > So far, we don't see a use case that cannot be provided by
> > location-by-reference in a header.  If we don't get a good
> > use case, I think consensus is to disallow location-by-value
> > in a header.
> >
>=20
> I see a problem here.  Location-by-reference is nothing more than
'magic
> happens here'!  Please point to a wg item documenting the
> location-by-reference mechanism.  How can you state the
> location-by-reference will cover all the use cases that the Data URI
would
> when location-by-reference hasn't even been defined anywhere?  You're
> loading up the magic with a lot of assumptions.  Location-by-reference
is
> an
> individual submission as a solution to a set of requirements that
haven't
> been fleshed-out in geopriv.
>=20
> I believe that Henning/Hannes also recognized this issue, hence the
> suggestion of a Data URI.  Dismissing this suggestion based on an
already
> accepted non-existent mechanism seems like getting the cart before the
> horse.
>=20
> ..........snip
>=20
> > 6. We proposed to allow sip/sips and pres: in the location
> > header.  I later speculated that if we allow this, it forms a
> > simple way to do location-by-reference, and we could
> > eliminate the raw HTTP(S) Get proposal, with the parameter
> > and registry entirely.  There was some list traffic in favor
> > of this.  Martin Thompson is skeptical that SIP Presence
> > Subscribe/Notify is actually allowed as a geopriv "using
> > protocol".  Several of think that it is.  If we don't get any
> > more comments on this proposal, we will allow sip/sips and
> > pres, not have any
> > http(s) resolution text, and not propose a parameter and
> > accompanying registry.
>=20
> I certainly think it's more prudent to define the dereferencing
mechanism
> prior to making claims about the protocol.
>=20
> Do I believe location-by-reference will happen?  Probably, but making
> assumptions about the mechanism now is quite premature.
>=20
> -Marc-
>=20
> >
> > If this does not reflect your position, please speak up now.
> >

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From sip-bounces@ietf.org Tue Jul 25 16:06:38 2006
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From: Andrew Newton <andy@hxr.us>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Tue, 25 Jul 2006 16:04:48 -0400
To: "Rosen, Brian" <Brian.Rosen@neustar.biz>
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For the record, I agree with Brian.

-andy

On Jul 25, 2006, at 3:50 PM, Rosen, Brian wrote:

> When the original PIDF-LO RFC came out, location-by-reference was
> defined.
>
> You have a URI.  You can exchange the URI for location.  There is a
> protocol defined for that purpose (SIP SUBSCRIBE).  That is  
> location by
> reference.  It also serves as the location update mechanism, clearly
> something we need.
>
> Some day, there may be other location URI references.  We don't  
> need to
> cover them in this draft.  The basic location by reference horse has
> left the barn.  Were you planning on creating an RFC that specifically
> disallows a Subscribe on a sip/sips/pres URI to get location?    
> THAT WAS
> THE POINT OF MAKING IT PART OF A PIDF.
>
> I do think there is some value in defining a way to express a
> location-by-reference aka presence URI which is not just the AoR.   
> That
> is one reason why I want to define the Location URI as being able to
> contain a sip/sips/pres URI.  The purpose is to allow a UA or a  
> proxy to
> insert location where there is no association between the identity of
> the SIP user and the location.
>
> But that is all I think we need.
>
> We have a requirement for proxy insertion of location.  If we disallow
> location-by-reference (which I think is a good idea in general, but
> futile in practice), then we would need something like the Data URI to
> support location insertion by proxy.  If we DO allow
> location-by-reference, then it also serves to meet the location
> insertion by proxy requirement, and we don't need the Data URI.  We're
> looking for a use case which requires proxy insertion by value, but
> haven't heard one yet.
>
> You can make an argument that you can take the From:, assume it's an
> AoR, and SUBSCRIBE to the presence of that AoR to get location,  
> thus not
> needing a Location header.  I really think letting the Location header
> specify an alternate presence URI is a good idea.
>
> Brian
>
>> -----Original Message-----
>> From: Marc Linsner [mailto:mlinsner@cisco.com]
>> Sent: Tuesday, July 25, 2006 3:29 PM
>> To: Rosen, Brian; sip@ietf.org
>> Cc: geopriv@ietf.org
>> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>>
>> In-line.....
>>
>>> -----Original Message-----
>>> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
>>> Sent: Tuesday, July 25, 2006 1:38 PM
>>> To: sip@ietf.org
>>> Cc: geopriv@ietf.org
>>> Subject: [Geopriv] Consensus on changes to location-conveyance
>>>
>>> I have put out a number of emails on proposed changes to
>>> location conveyance.  Some of them have sparked a number of
>>> comments.  My summary of consensus on these issues is:
>>>
>>> 1. We proposed to explicitly disallow use of a Data URI,
>>> which would allow specification of a Location-by-value in a
>>> header.  There was a lot of list traffic on this, currently
>>> in the state where Keith asked for a real use case for it.
>>> So far, we don't see a use case that cannot be provided by
>>> location-by-reference in a header.  If we don't get a good
>>> use case, I think consensus is to disallow location-by-value
>>> in a header.
>>>
>>
>> I see a problem here.  Location-by-reference is nothing more than
> 'magic
>> happens here'!  Please point to a wg item documenting the
>> location-by-reference mechanism.  How can you state the
>> location-by-reference will cover all the use cases that the Data URI
> would
>> when location-by-reference hasn't even been defined anywhere?  You're
>> loading up the magic with a lot of assumptions.  Location-by- 
>> reference
> is
>> an
>> individual submission as a solution to a set of requirements that
> haven't
>> been fleshed-out in geopriv.
>>
>> I believe that Henning/Hannes also recognized this issue, hence the
>> suggestion of a Data URI.  Dismissing this suggestion based on an
> already
>> accepted non-existent mechanism seems like getting the cart before  
>> the
>> horse.
>>
>> ..........snip
>>
>>> 6. We proposed to allow sip/sips and pres: in the location
>>> header.  I later speculated that if we allow this, it forms a
>>> simple way to do location-by-reference, and we could
>>> eliminate the raw HTTP(S) Get proposal, with the parameter
>>> and registry entirely.  There was some list traffic in favor
>>> of this.  Martin Thompson is skeptical that SIP Presence
>>> Subscribe/Notify is actually allowed as a geopriv "using
>>> protocol".  Several of think that it is.  If we don't get any
>>> more comments on this proposal, we will allow sip/sips and
>>> pres, not have any
>>> http(s) resolution text, and not propose a parameter and
>>> accompanying registry.
>>
>> I certainly think it's more prudent to define the dereferencing
> mechanism
>> prior to making claims about the protocol.
>>
>> Do I believe location-by-reference will happen?  Probably, but making
>> assumptions about the mechanism now is quite premature.
>>
>> -Marc-
>>
>>>
>>> If this does not reflect your position, please speak up now.
>>>
>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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On Jul 25, 2006, at 3:29 PM, Marc Linsner wrote:

> Do I believe location-by-reference will happen?  Probably, but making
> assumptions about the mechanism now is quite premature.

Perhaps nothing will save us.  We've had by-value since step one, and  
now by-value take 2 is being proposed.

-andy

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From sip-bounces@ietf.org Tue Jul 25 16:31:57 2006
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Brian,

Three more remarks:

1 - Location by value is much easier to get interoperable than location by 
reference, also because the latter has many semantic issues (in addition to 
performance, chance at failure):
* When dereferencing, for what time instance do I get the location?
* if { moment of call }, how do you guarantee that the dereferenced source 
will have exactly that information?
* if { moment of dereference }, for how long is the URI supposed / assumed 
to be usable?
* how does the user control/limit access to that location information?

All of the above is clear for location-by-value. It is therefore much easier 
to ensure privacy for that, in a way that is easily explained to the user: 
either you give your location at that moment, or you dont. Period

2 - In particular for emergency scenarios, I think you would want both (by 
value and by reference), even within the same request: rough location by 
value (eg cell-id) could be used to route the call to the appropriate 
emergency center, then a Location URI (provided in the same request) can be 
dereferenced from there, which may give a more precise location after some 
time (say 10s of seconds). Moreover, the reference could be used to track 
the location over time

3 Rather than relying on TLS for security of the location-user mapping, I 
always assumed there would be requirements for access control at the 
location server? To me requiring TLS is a definite way to ensure that this 
will not see widespread use...

Regards,

Jeroen


----- Original Message ----- 
From: "Brian Rosen" <br@brianrosen.net>
To: "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>; <sip@ietf.org>
Sent: Tuesday, July 25, 2006 8:30 PM
Subject: RE: [Sip] Consensus on changes to location-conveyance


> Jeroen
>
> I've always disliked location-by-reference because it has delay, and
> introduces the possibility (in the emergency case especially) where the
> dereferencer has connectivity to the endpoint but not the LIS.
>
> The WG however, has consensus that these issues are not significant, and 
> the
> benefits outweigh the problems.
>
> Having said that using a reference is NOT seen as a performance problem 
> then
> it's not a performance problem, and you can't argue it's a use case.
>
> Agree that if the endpoint wants location to be removed after routing 
> before
> forwarding, then it has to use by-reference.  Don't see that as a problem.
>
> Yes, TLS protects the URL, which is important, because anyone observing 
> the
> SIP messaging can connect the identity of the SIP correspondent with the
> location (if it is allowed to dereference it), which is bad.
>
> Brian
>
>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: Tuesday, July 25, 2006 2:14 PM
>> To: Brian Rosen; sip@ietf.org
>> Subject: Re: [Sip] Consensus on changes to location-conveyance
>>
>> Brian,
>>
>> Your point 4 below is a concrete use case for point 1 (location by value
>> in
>> header). Using location for routing decisions benefits from quick access,
>> a
>> lookup by reference introduces delay.
>>
>> If you want a proxy to remove the location info, you cannot put the
>> location
>> info in the body, because proxies are not allowed to modify that. So you
>> are
>> left with location by reference. The use of TLS is then only to protect
>> this
>> URL?
>>
>> Regards,
>>
>> Jeroen
>>
>> ----- Original Message -----
>> From: "Brian Rosen" <br@brianrosen.net>
>> To: <sip@ietf.org>
>> Sent: Tuesday, July 25, 2006 7:41 PM
>> Subject: [Sip] Consensus on changes to location-conveyance
>>
>>
>> >I have put out a number of emails on proposed changes to location
>> > conveyance.  Some of them have sparked a number of comments.  My 
>> > summary
>> > of
>> > consensus on these issues is:
>> >
>> > 1. We proposed to explicitly disallow use of a Data URI, which would
>> allow
>> > specification of a Location-by-value in a header.  There was a lot of
>> list
>> > traffic on this, currently in the state where Keith asked for a real 
>> > use
>> > case for it.  So far, we don't see a use case that cannot be provided 
>> > by
>> > location-by-reference in a header.  If we don't get a good use case, I
>> > think
>> > consensus is to disallow location-by-value in a header.
>> >
>> > 2. Dereferencing location-by-reference.  We had initially proposed to
>> > document a process to use a raw HTTP(S) Get to dereference.  There was 
>> > a
>> > lot
>> > of discussion, and then I made a new proposal, see item 6.
>> >
>> > 3. We proposed to allow a Reason header with a 424 Bad Location
>> response.
>> > This seems to be acceptable to the WG.  Geopriv will create and 
>> > populate
>> > the
>> > registry in a separate draft.
>> >
>> > 4. We proposed to explicitly allow hop-by-hop TLS security for location
>> > when
>> > the endpoint wished to allow routing based on location.  We further
>> > proposed
>> > to interpret "Do not Distribute" flags as allowing sending of the
>> location
>> > to the routing database (LoST for example).  Hannes pointed out that
>> there
>> > was a proposal to allow two Location headers, one of which could be 
>> > used
>> > for
>> > routing and another for onward forwarding to the recipient.  Hannes'
>> > message
>> > seemed to agree that "Do not distribute" would allow sending of the
>> > location
>> > to the routing-by-location database, but would obligate the proxy that
>> did
>> > that to remove the location before it forwarded the message.  There 
>> > were
>> > no
>> > other messages on this thread.  We conclude that allowing hop-by-hop
>> > security for location based routing is acceptable, and Do Not 
>> > Distribute
>> > DOES allow forwarding to a routing database, but requires that the 
>> > proxy
>> > doing the location based route to remove the location it uses for that
>> > purpose.
>> >
>> > 5. We proposed to create a parameter and a registry to distinguish
>> > multiple
>> > ways to dereference a location where HTTP(S) was the transport.
>> > Resolution
>> > of this issue depends on resolution of issue 6.
>> >
>> > 6. We proposed to allow sip/sips and pres: in the location header.  I
>> > later
>> > speculated that if we allow this, it forms a simple way to do
>> > location-by-reference, and we could eliminate the raw HTTP(S) Get
>> > proposal,
>> > with the parameter and registry entirely.  There was some list traffic
>> in
>> > favor of this.  Martin Thompson is skeptical that SIP Presence
>> > Subscribe/Notify is actually allowed as a geopriv "using protocol".
>> > Several
>> > of think that it is.  If we don't get any more comments on this
>> proposal,
>> > we
>> > will allow sip/sips and pres, not have any http(s) resolution text, and
>> > not
>> > propose a parameter and accompanying registry.
>> >
>> > If this does not reflect your position, please speak up now.
>> >
>> > Brian
>> >
>> >
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol
>> > Use sip-implementors@cs.columbia.edu for questions on current sip
>> > Use sipping@ietf.org for new developments on the application of sip
> 


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As Brian points out, the whole point of using PIDF for location =
information was the similarity of the early defined use cases to the =
presence architecture. This certainly included use cases in which one =
wanted to fetch or subscribe to location information. Those operations =
require the existence of a reference, and that reference was commonly =
described as a URI, at the time. Lately, I think the idea of having a =
URI pointing to location information has become associated with a =
specific proposal (i.e. HELD), but clearly RFC4079 sketched out an =
architecture for using presence to acquire location information and it =
specifically mentions URIs as a way to refer to presence.=20

There is no "hidden magic" in any of this. Subscribing to presence, and =
using URIs to indicate presence information, is exhaustively specified =
for SIP. PIDF-LO was designed to reuse all of that work and specifically =
not to require us to go out and reinvent everything in order to be able =
to refer to location information out in the network.

Basically, I support the direction re: bullet 6. I might further suggest =
that the CID URI scheme be permitted in the Location header, in order to =
be able to refer to MIME bodies included in SIP messages. I'm also =
neutral/favorable on using HTTP GET. But I think either of these URI =
schemes and their usage in the Location header could just as well be =
described in subsequent documents.

Jon Peterson
NeuStar, Inc,

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Tuesday, July 25, 2006 1:05 PM
> To: Rosen, Brian
> Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
> location-conveyance
>=20
>=20
> For the record, I agree with Brian.
>=20
> -andy
>=20
> On Jul 25, 2006, at 3:50 PM, Rosen, Brian wrote:
>=20
> > When the original PIDF-LO RFC came out, location-by-reference was
> > defined.
> >
> > You have a URI.  You can exchange the URI for location.  There is a
> > protocol defined for that purpose (SIP SUBSCRIBE).  That is =20
> > location by
> > reference.  It also serves as the location update mechanism, clearly
> > something we need.
> >
> > Some day, there may be other location URI references.  We don't =20
> > need to
> > cover them in this draft.  The basic location by reference horse has
> > left the barn.  Were you planning on creating an RFC that=20
> specifically
> > disallows a Subscribe on a sip/sips/pres URI to get location?   =20
> > THAT WAS
> > THE POINT OF MAKING IT PART OF A PIDF.
> >
> > I do think there is some value in defining a way to express a
> > location-by-reference aka presence URI which is not just the AoR.  =20
> > That
> > is one reason why I want to define the Location URI as being able to
> > contain a sip/sips/pres URI.  The purpose is to allow a UA or a =20
> > proxy to
> > insert location where there is no association between the=20
> identity of
> > the SIP user and the location.
> >
> > But that is all I think we need.
> >
> > We have a requirement for proxy insertion of location.  If=20
> we disallow
> > location-by-reference (which I think is a good idea in general, but
> > futile in practice), then we would need something like the=20
> Data URI to
> > support location insertion by proxy.  If we DO allow
> > location-by-reference, then it also serves to meet the location
> > insertion by proxy requirement, and we don't need the Data=20
> URI.  We're
> > looking for a use case which requires proxy insertion by value, but
> > haven't heard one yet.
> >
> > You can make an argument that you can take the From:, assume it's an
> > AoR, and SUBSCRIBE to the presence of that AoR to get location, =20
> > thus not
> > needing a Location header.  I really think letting the=20
> Location header
> > specify an alternate presence URI is a good idea.
> >
> > Brian
> >
> >> -----Original Message-----
> >> From: Marc Linsner [mailto:mlinsner@cisco.com]
> >> Sent: Tuesday, July 25, 2006 3:29 PM
> >> To: Rosen, Brian; sip@ietf.org
> >> Cc: geopriv@ietf.org
> >> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
> >>
> >> In-line.....
> >>
> >>> -----Original Message-----
> >>> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> >>> Sent: Tuesday, July 25, 2006 1:38 PM
> >>> To: sip@ietf.org
> >>> Cc: geopriv@ietf.org
> >>> Subject: [Geopriv] Consensus on changes to location-conveyance
> >>>
> >>> I have put out a number of emails on proposed changes to
> >>> location conveyance.  Some of them have sparked a number of
> >>> comments.  My summary of consensus on these issues is:
> >>>
> >>> 1. We proposed to explicitly disallow use of a Data URI,
> >>> which would allow specification of a Location-by-value in a
> >>> header.  There was a lot of list traffic on this, currently
> >>> in the state where Keith asked for a real use case for it.
> >>> So far, we don't see a use case that cannot be provided by
> >>> location-by-reference in a header.  If we don't get a good
> >>> use case, I think consensus is to disallow location-by-value
> >>> in a header.
> >>>
> >>
> >> I see a problem here.  Location-by-reference is nothing more than
> > 'magic
> >> happens here'!  Please point to a wg item documenting the
> >> location-by-reference mechanism.  How can you state the
> >> location-by-reference will cover all the use cases that=20
> the Data URI
> > would
> >> when location-by-reference hasn't even been defined=20
> anywhere?  You're
> >> loading up the magic with a lot of assumptions.  Location-by-=20
> >> reference
> > is
> >> an
> >> individual submission as a solution to a set of requirements that
> > haven't
> >> been fleshed-out in geopriv.
> >>
> >> I believe that Henning/Hannes also recognized this issue, hence the
> >> suggestion of a Data URI.  Dismissing this suggestion based on an
> > already
> >> accepted non-existent mechanism seems like getting the=20
> cart before =20
> >> the
> >> horse.
> >>
> >> ..........snip
> >>
> >>> 6. We proposed to allow sip/sips and pres: in the location
> >>> header.  I later speculated that if we allow this, it forms a
> >>> simple way to do location-by-reference, and we could
> >>> eliminate the raw HTTP(S) Get proposal, with the parameter
> >>> and registry entirely.  There was some list traffic in favor
> >>> of this.  Martin Thompson is skeptical that SIP Presence
> >>> Subscribe/Notify is actually allowed as a geopriv "using
> >>> protocol".  Several of think that it is.  If we don't get any
> >>> more comments on this proposal, we will allow sip/sips and
> >>> pres, not have any
> >>> http(s) resolution text, and not propose a parameter and
> >>> accompanying registry.
> >>
> >> I certainly think it's more prudent to define the dereferencing
> > mechanism
> >> prior to making claims about the protocol.
> >>
> >> Do I believe location-by-reference will happen?  Probably,=20
> but making
> >> assumptions about the mechanism now is quite premature.
> >>
> >> -Marc-
> >>
> >>>
> >>> If this does not reflect your position, please speak up now.
> >>>
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

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Thread-Topic: [Geopriv] Consensus on changes to location-conveyance
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "Ted Hardie" <hardie@qualcomm.com>, <sip@ietf.org>
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Thanks.

To define the "retransmission for routing" bit, we need to rev PIDF-LO.
I don't think doing that in location-conveyance is the right place.
Would you agree?

Is there consensus one way or the other on this?  Generally, I do agree
explicit is better than implicit for any kind of permission.  I'm loath
to generate yet one more RFC, but this one looks simple enough.

With respect to an HTTP protocol for location by reference
dereferencing, I have no problem with having some other document specify
this.  We're going to have a simple mechanism defined in
location-conveyance to allow such an extension; it won't require the SIP
change process.  If that is an acceptable compromise, then we can close
this issue, although I'll certainly be holding off a couple of days to
hear from other folks.

Brian

> -----Original Message-----
> From: Ted Hardie [mailto:hardie@qualcomm.com]
> Sent: Tuesday, July 25, 2006 4:34 PM
> To: Rosen, Brian; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>=20
> At 2:45 PM -0400 7/25/06, Rosen, Brian wrote:
> >Summarizing, you think hop-by-hop security (TLS) is acceptable when
the
> >location is sent for the purposes of location based routing, but you
are
> >not in favor of defining "retransmission=3Dno" to permit sending the
> >location to the routing database (e.g. LoST).  You want a more
explicit
> >permission.
>=20
> Yes.
>=20
> >I don't think a cid: with a ruleset is acceptable, as it would not
> >permit location by reference added by a proxy to specify this.
>=20
> Fair enough.
>=20
>=20
>=20
> >If this is the consensus, I think we would have to add a bit to the
PIDF
> >to explicitly control this behavior.
>=20
> I think an extension that says "retransmission for routing=3Dyes" is =
the
> cleanest
> way to go.
>=20
>=20
> >Why do you think we must specify an HTTP dereference procedure in
this
> >document?  What is the purpose when the SIP based method is
available?
> >We had believed we had to have SOME method of dereferencing to make
> >location-by-reference meaningful in the document.  If we have one sip
> >based method, we meet the requirement.  Having more options,
especially
> >completely new HTTP based protocols IN THIS DOCUMENT seems
unnecessary
> >and unwise.  For example, if we define such a protocol, it becomes a
> >geopriv "using protocol" and needs an explicit analysis against the
> >geopriv criteria.
>=20
> I don't think we have to do it in this document, but I read your
previous
> message as a statement of consensus about the HTTP dereference
mechanisms
> that have been discussed.   I don't agree with that statement, since I
> think the
> HTTP discussions somewhat over-thought the potential problems.  In
> particular, a
> lot of the discussion was around the idea that
> http://dereference.example.org/foo-pidf-lo
> might mean lots of different things, depending on what http meant.
There
> are
> already ways to deal with that; none of them are perfect (see, for
> example,
> the discussion in draft-reschke-webdav-mount-05.txt and its appendix A
for
> a similar
> problem in a different context), but they can work, especially if you
set
> a baseline
> approach.
>=20
> 				regards,
> 					Ted Hardie
>=20
>=20
>=20
> >Brian
> >
> >> -----Original Message-----
> >> From: Ted Hardie [mailto:hardie@qualcomm.com]
> >> Sent: Tuesday, July 25, 2006 2:19 PM
> >> To: Rosen, Brian; sip@ietf.org
> >> Cc: geopriv@ietf.org
> >> Subject: Re: [Geopriv] Consensus on changes to location-conveyance
> >>
> >> At 1:38 PM -0400 7/25/06, Rosen, Brian wrote:
> >> >4. We proposed to explicitly allow hop-by-hop TLS security for
> >location
> >> >when the endpoint wished to allow routing based on location.  We
> >further
> >> >proposed to interpret "Do not Distribute" flags as allowing
sending
> >of
> >> >the location to the routing database (LoST for example).  Hannes
> >pointed
> >> >out that there was a proposal to allow two Location headers, one
of
> >> >which could be used for routing and another for onward forwarding
to
> >the
> >> >recipient.  Hannes' message seemed to agree that "Do not
distribute"
> >> >would allow sending of the location to the routing-by-location
> >database,
> >> >but would obligate the proxy that did that to remove the location
> >before
> >> >it forwarded the message.  There were no other messages on this
> >thread.
> >>
> >> Actually, I commented on the same problem, though it may have been
> >> connected to one of the related threads; here's the text of that
> >message:
> >>
> >> >At 5:23 PM -0400 7/19/06, Brian Rosen wrote:
> >> >>NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU REPLY, SEND
IT
> >> ONLY TO
> >> >>GEOPRIV
> >> >>However, this opens up the issue of geopriv privacy concerns
because
> >the
> >> >>sender does not know the identity of the element (proxy) that
will
> >route
> >> the
> >> >>call using LoST.  It may be that the UAC itself does this, but it
> >may be
> >> >>some other intermediary.
> >> >>
> >> >>If the identity of the intermediary is not known, then protecting
> >the
> >> >>location information becomes more difficult, because we cannot
> >encrypt
> >> using
> >> >>keying material from the intended recipient.
> >> >
> >> >Just to mention this, it appears to also rule out the idea of
> >encrypting
> >> >with the user's key, since there is no way of knowing whether the
> >> "intended
> >> >recipient" has that key material either.  Intended recipient is
> >actually
> >> >a bit of a problematic term here, since it's really the on-path
> >network
> >> elements
> >> >which consume the information.  They aren't the intended recipient
of
> >> >the communication, they are the recipients of the signaling (of
which
> >> location
> >> >is now a part).
> >> >
> >> >This is a bit confusing because it seems to be possible to include
a
> >> >"re-transmission allowed"  rule with a value of "no" (intended to
be
> >> >consumed by the "intended recipient" of the underlying
> >communication),
> >> >while still wanting to allow the location to be passed along the
> >path.
> >> >
> >> >>In fact, we would need to use
> >> >>hop-by-hop security, with transitive trust.  This is what the
> >emergency
> >> case
> >> >>does, but it also allows no channel security if necessary
(usually
> >> meaning,
> >> >>try it with TLS and if that fails, try again without TLS).
> >> >
> >> >While I certainly encourage hop-by-hop security, I don't think it
> >really
> >> >meets the geopriv privacy concerns.  I think you  have to have
some
> >way
> >> of
> >> >expressing "re-transmission for routing permitted" (and likely
> >retention
> >> >limited), since that is a separate statement.
> >> >
> >> >Just typing aloud here, but could we come up with a way to use an
> >> >external ruleset to express this?  With an embedded cid, that
would
> >> >not actually require dereferencing.
> >> >					Ted
> >> >
> >>
> >>
> >>
> >> >We conclude that allowing hop-by-hop security for location based
> >routing
> >> >is acceptable, and Do Not Distribute DOES allow forwarding to a
> >routing
> >> >database, but requires that the proxy doing the location based
route
> >to
> >> >remove the location it uses for that purpose.
> >>
> >> To reiterate this, I do not believe that this is sufficient; I
think
> >you
> >> have to
> >> have some way of saying "re-transmission for routing permitted"
before
> >> you meet the geopriv privacy concerns.
> >>
> >> >5. We proposed to create a parameter and a registry to distinguish
> >> >multiple ways to dereference a location where HTTP(S) was the
> >transport.
> >> >Resolution of this issue depends on resolution of issue 6.
> >>
> >> I believe that an HTTP GET of a PIDF-LO object is a reasonable
> >baseline.
> >> Saying that other mechanisms should update this document with their
> >> dereferencing mechanisms if they could be confused with the
baseline
> >> seems to me to be sufficient.  I'm not sure a registry is needed.
In
> >> practice,
> >> I don't think we would hit the interoperability problems often.
These
> >> days,
> >> extensions to HTTP commonly use different  mime types to indicate
the
> >> special
> >> handling, and I expect anyone creating a more feature-rich
> >dereferencing
> >> mechanism built on HTTP would not end up using pidf-lo.
> >>
> >> That means the "uri confusion" aspects of the discussion are
probably
> >not
> >> that severe.  The only issue is if someone supports only the more
> >feature-
> >> rich
> >> alternative and not a baseline GET of the pidf-lo.  I think we
should
> >> discourage that, by establishing a real baseline.
> >>
> >> Note also that HTTP URIs don't indicate the HTTP Method directly;
> >often
> >> they
> >> support multiple methods (e.g. HEAD and GET).
> >>
> >> >6. We proposed to allow sip/sips and pres: in the location header.
I
> >> >later speculated that if we allow this, it forms a simple way to
do
> >> >location-by-reference, and we could eliminate the raw HTTP(S) Get
> >> >proposal, with the parameter and registry entirely.  There was
some
> >list
> >> >traffic in favor of this.  Martin Thompson is skeptical that SIP
> >> >Presence Subscribe/Notify is actually allowed as a geopriv "using
> >> >protocol".  Several of think that it is.  If we don't get any more
> >> >comments on this proposal, we will allow sip/sips and pres, not
have
> >any
> >> >http(s) resolution text, and not propose a parameter and
accompanying
> >> >registry.
> >>
> >> I share some of Martin's skepticism here, but I won't fall on my
sword
> >> if the WG goes in this direction.
> >>
> >> Thanks for the clear set of issues and opportunity to comment,
> >>
> >>					Ted Hardie


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From: "Brian Rosen" <br@brianrosen.net>
To: "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>,
	<sip@ietf.org>
Subject: RE: [Sip] Consensus on changes to location-conveyance
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Jeroen

I'm a big believer in location-by-value over location-by-reference,
especially for the emergency case.  You won't find me arguing for location-
by-reference.  However, there is clear sentiment to support it, and a
requirement to support proxy insertion of location, which I also think is a
bad idea.  Given that we have to support proxy insertion, and the sentiment
to support -by-reference, I'm leaning towards one mechanism.

To be fair, whether you get location by value or by reference, the validity
issues on the PIDF are the same.  When you have a reference, you do not know
for how long the reference is valid.  When you place the call, you may get a
"fresh" location before you place the call, or you may not.  If you have to
dereference it, at least there is an opportunity for the LIS to say "I have
someone that wants this location now, I can give him my best current
estimate of that location".  I do worry quite a bit about proxy insertion of
location and how the USERS's notion of access control will be honored and
reflected in the ruleset.  I think whether you get the value by location or
by reference, the rules that are supposed to be in there should be the same,
and will mean the same thing.

I agree that you may want to send both a value and a reference for a case
where there may be mobility.

I think you must control both the integrity and the privacy of the location
reference in the SIP header.  The former is important to make sure that the
PIDF, with its ruleset, is delivered intact.  The latter is important to
make sure the user's identity cannot be linked with his location if he
wishes to do so.  Access control at the LIS on a dereference may be another
chance to control access.

Brian

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Tuesday, July 25, 2006 4:30 PM
> To: Brian Rosen; sip@ietf.org
> Subject: Re: [Sip] Consensus on changes to location-conveyance
> 
> Brian,
> 
> Three more remarks:
> 
> 1 - Location by value is much easier to get interoperable than location by
> reference, also because the latter has many semantic issues (in addition
> to
> performance, chance at failure):
> * When dereferencing, for what time instance do I get the location?
> * if { moment of call }, how do you guarantee that the dereferenced source
> will have exactly that information?
> * if { moment of dereference }, for how long is the URI supposed / assumed
> to be usable?
> * how does the user control/limit access to that location information?
> 
> All of the above is clear for location-by-value. It is therefore much
> easier
> to ensure privacy for that, in a way that is easily explained to the user:
> either you give your location at that moment, or you dont. Period
> 
> 2 - In particular for emergency scenarios, I think you would want both (by
> value and by reference), even within the same request: rough location by
> value (eg cell-id) could be used to route the call to the appropriate
> emergency center, then a Location URI (provided in the same request) can
> be
> dereferenced from there, which may give a more precise location after some
> time (say 10s of seconds). Moreover, the reference could be used to track
> the location over time
> 
> 3 Rather than relying on TLS for security of the location-user mapping, I
> always assumed there would be requirements for access control at the
> location server? To me requiring TLS is a definite way to ensure that this
> will not see widespread use...
> 
> Regards,
> 
> Jeroen
> 
> 
> ----- Original Message -----
> From: "Brian Rosen" <br@brianrosen.net>
> To: "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>; <sip@ietf.org>
> Sent: Tuesday, July 25, 2006 8:30 PM
> Subject: RE: [Sip] Consensus on changes to location-conveyance
> 
> 
> > Jeroen
> >
> > I've always disliked location-by-reference because it has delay, and
> > introduces the possibility (in the emergency case especially) where the
> > dereferencer has connectivity to the endpoint but not the LIS.
> >
> > The WG however, has consensus that these issues are not significant, and
> > the
> > benefits outweigh the problems.
> >
> > Having said that using a reference is NOT seen as a performance problem
> > then
> > it's not a performance problem, and you can't argue it's a use case.
> >
> > Agree that if the endpoint wants location to be removed after routing
> > before
> > forwarding, then it has to use by-reference.  Don't see that as a
> problem.
> >
> > Yes, TLS protects the URL, which is important, because anyone observing
> > the
> > SIP messaging can connect the identity of the SIP correspondent with the
> > location (if it is allowed to dereference it), which is bad.
> >
> > Brian
> >
> >> -----Original Message-----
> >> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> >> Sent: Tuesday, July 25, 2006 2:14 PM
> >> To: Brian Rosen; sip@ietf.org
> >> Subject: Re: [Sip] Consensus on changes to location-conveyance
> >>
> >> Brian,
> >>
> >> Your point 4 below is a concrete use case for point 1 (location by
> value
> >> in
> >> header). Using location for routing decisions benefits from quick
> access,
> >> a
> >> lookup by reference introduces delay.
> >>
> >> If you want a proxy to remove the location info, you cannot put the
> >> location
> >> info in the body, because proxies are not allowed to modify that. So
> you
> >> are
> >> left with location by reference. The use of TLS is then only to protect
> >> this
> >> URL?
> >>
> >> Regards,
> >>
> >> Jeroen
> >>
> >> ----- Original Message -----
> >> From: "Brian Rosen" <br@brianrosen.net>
> >> To: <sip@ietf.org>
> >> Sent: Tuesday, July 25, 2006 7:41 PM
> >> Subject: [Sip] Consensus on changes to location-conveyance
> >>
> >>
> >> >I have put out a number of emails on proposed changes to location
> >> > conveyance.  Some of them have sparked a number of comments.  My
> >> > summary
> >> > of
> >> > consensus on these issues is:
> >> >
> >> > 1. We proposed to explicitly disallow use of a Data URI, which would
> >> allow
> >> > specification of a Location-by-value in a header.  There was a lot of
> >> list
> >> > traffic on this, currently in the state where Keith asked for a real
> >> > use
> >> > case for it.  So far, we don't see a use case that cannot be provided
> >> > by
> >> > location-by-reference in a header.  If we don't get a good use case,
> I
> >> > think
> >> > consensus is to disallow location-by-value in a header.
> >> >
> >> > 2. Dereferencing location-by-reference.  We had initially proposed to
> >> > document a process to use a raw HTTP(S) Get to dereference.  There
> was
> >> > a
> >> > lot
> >> > of discussion, and then I made a new proposal, see item 6.
> >> >
> >> > 3. We proposed to allow a Reason header with a 424 Bad Location
> >> response.
> >> > This seems to be acceptable to the WG.  Geopriv will create and
> >> > populate
> >> > the
> >> > registry in a separate draft.
> >> >
> >> > 4. We proposed to explicitly allow hop-by-hop TLS security for
> location
> >> > when
> >> > the endpoint wished to allow routing based on location.  We further
> >> > proposed
> >> > to interpret "Do not Distribute" flags as allowing sending of the
> >> location
> >> > to the routing database (LoST for example).  Hannes pointed out that
> >> there
> >> > was a proposal to allow two Location headers, one of which could be
> >> > used
> >> > for
> >> > routing and another for onward forwarding to the recipient.  Hannes'
> >> > message
> >> > seemed to agree that "Do not distribute" would allow sending of the
> >> > location
> >> > to the routing-by-location database, but would obligate the proxy
> that
> >> did
> >> > that to remove the location before it forwarded the message.  There
> >> > were
> >> > no
> >> > other messages on this thread.  We conclude that allowing hop-by-hop
> >> > security for location based routing is acceptable, and Do Not
> >> > Distribute
> >> > DOES allow forwarding to a routing database, but requires that the
> >> > proxy
> >> > doing the location based route to remove the location it uses for
> that
> >> > purpose.
> >> >
> >> > 5. We proposed to create a parameter and a registry to distinguish
> >> > multiple
> >> > ways to dereference a location where HTTP(S) was the transport.
> >> > Resolution
> >> > of this issue depends on resolution of issue 6.
> >> >
> >> > 6. We proposed to allow sip/sips and pres: in the location header.  I
> >> > later
> >> > speculated that if we allow this, it forms a simple way to do
> >> > location-by-reference, and we could eliminate the raw HTTP(S) Get
> >> > proposal,
> >> > with the parameter and registry entirely.  There was some list
> traffic
> >> in
> >> > favor of this.  Martin Thompson is skeptical that SIP Presence
> >> > Subscribe/Notify is actually allowed as a geopriv "using protocol".
> >> > Several
> >> > of think that it is.  If we don't get any more comments on this
> >> proposal,
> >> > we
> >> > will allow sip/sips and pres, not have any http(s) resolution text,
> and
> >> > not
> >> > propose a parameter and accompanying registry.
> >> >
> >> > If this does not reflect your position, please speak up now.
> >> >
> >> > Brian
> >> >
> >> >
> >> > _______________________________________________
> >> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> > This list is for NEW development of the core SIP Protocol
> >> > Use sip-implementors@cs.columbia.edu for questions on current sip
> >> > Use sipping@ietf.org for new developments on the application of sip
> >


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To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [Sip] Hitchhikers guide - when is it done?
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Sorry for being the voice out on the left field, but I think 1
is better.  Once the 'core' documents have RFC numbers, it makes
sense to go ahed and publish that as an Informational (maybe
name it "A Hitchhikers Guide to SIP - Part 1: Core Specification").

Then deal with the current I-Ds and other work as a separate
Informational, probably named "A Hitchhikers Guide to SIP -
Part 2: Extensions".  The first RFC will have an informative
reference to the second to-be RFC, so readers can be warned
that the document is split into to (plus the naming convention
of the title is a clue).

With Part 2 being in flux for a while, I would presume that it
would be easier from a process point of view to rev a I-D than
it is to obsolete or update a RFC.

Plus there is some poetic justice in the SIP WG closing doors
when the last RFC issued is one that (partially) summarizes what
we have done as a WG.  Too bad RFC number 4242 is already taken.

Thanks.

Jonathan Rosenberg wrote:
> During the IETF meeting, I discussed the one current open issue in the 
> Hitchhikers guide. The issue is: when does the document finish? When do 
> we send to IESG to issue an RFC?
> 
> There were two approaches:
> 
> 1. split the document in two. One part contains only the core specs, and 
> issues relatively soon. The other part has everything else, and 
> basically ends up being the last RFC produced by the SIP WG before it 
> shuts its doors (which has always been 1.5 years from now for the last 5 
> years).
> 
> 2. keep it as one document. Issue an RFC at some point, and then revise 
> it a few times as needed. We need to decide when to issue it.
> 
> 
> I'd like to propose option 2. I think people get more value from it as a 
> consolidated document. Furthermore, I'd propose that we send the 
> document to IESG for consideration as informational at the instant all 
> of the documents listed as 'core' have RFC numbers issued by RFC-ed.
> 
> Comments?

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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Thread-Topic: [Geopriv] Consensus on changes to location-conveyance
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "Ted Hardie" <hardie@qualcomm.com>, <sip@ietf.org>
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Summarizing, you think hop-by-hop security (TLS) is acceptable when the
location is sent for the purposes of location based routing, but you are
not in favor of defining "retransmission=3Dno" to permit sending the
location to the routing database (e.g. LoST).  You want a more explicit
permission.

I don't think a cid: with a ruleset is acceptable, as it would not
permit location by reference added by a proxy to specify this.

If this is the consensus, I think we would have to add a bit to the PIDF
to explicitly control this behavior.

Why do you think we must specify an HTTP dereference procedure in this
document?  What is the purpose when the SIP based method is available?
We had believed we had to have SOME method of dereferencing to make
location-by-reference meaningful in the document.  If we have one sip
based method, we meet the requirement.  Having more options, especially
completely new HTTP based protocols IN THIS DOCUMENT seems unnecessary
and unwise.  For example, if we define such a protocol, it becomes a
geopriv "using protocol" and needs an explicit analysis against the
geopriv criteria.

Brian

> -----Original Message-----
> From: Ted Hardie [mailto:hardie@qualcomm.com]
> Sent: Tuesday, July 25, 2006 2:19 PM
> To: Rosen, Brian; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: Re: [Geopriv] Consensus on changes to location-conveyance
>=20
> At 1:38 PM -0400 7/25/06, Rosen, Brian wrote:
> >4. We proposed to explicitly allow hop-by-hop TLS security for
location
> >when the endpoint wished to allow routing based on location.  We
further
> >proposed to interpret "Do not Distribute" flags as allowing sending
of
> >the location to the routing database (LoST for example).  Hannes
pointed
> >out that there was a proposal to allow two Location headers, one of
> >which could be used for routing and another for onward forwarding to
the
> >recipient.  Hannes' message seemed to agree that "Do not distribute"
> >would allow sending of the location to the routing-by-location
database,
> >but would obligate the proxy that did that to remove the location
before
> >it forwarded the message.  There were no other messages on this
thread.
>=20
> Actually, I commented on the same problem, though it may have been
> connected to one of the related threads; here's the text of that
message:
>=20
> >At 5:23 PM -0400 7/19/06, Brian Rosen wrote:
> >>NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU REPLY, SEND IT
> ONLY TO
> >>GEOPRIV
> >>However, this opens up the issue of geopriv privacy concerns because
the
> >>sender does not know the identity of the element (proxy) that will
route
> the
> >>call using LoST.  It may be that the UAC itself does this, but it
may be
> >>some other intermediary.
> >>
> >>If the identity of the intermediary is not known, then protecting
the
> >>location information becomes more difficult, because we cannot
encrypt
> using
> >>keying material from the intended recipient.
> >
> >Just to mention this, it appears to also rule out the idea of
encrypting
> >with the user's key, since there is no way of knowing whether the
> "intended
> >recipient" has that key material either.  Intended recipient is
actually
> >a bit of a problematic term here, since it's really the on-path
network
> elements
> >which consume the information.  They aren't the intended recipient of
> >the communication, they are the recipients of the signaling (of which
> location
> >is now a part).
> >
> >This is a bit confusing because it seems to be possible to include a
> >"re-transmission allowed"  rule with a value of "no" (intended to be
> >consumed by the "intended recipient" of the underlying
communication),
> >while still wanting to allow the location to be passed along the
path.
> >
> >>In fact, we would need to use
> >>hop-by-hop security, with transitive trust.  This is what the
emergency
> case
> >>does, but it also allows no channel security if necessary (usually
> meaning,
> >>try it with TLS and if that fails, try again without TLS).
> >
> >While I certainly encourage hop-by-hop security, I don't think it
really
> >meets the geopriv privacy concerns.  I think you  have to have some
way
> of
> >expressing "re-transmission for routing permitted" (and likely
retention
> >limited), since that is a separate statement.
> >
> >Just typing aloud here, but could we come up with a way to use an
> >external ruleset to express this?  With an embedded cid, that would
> >not actually require dereferencing.
> >					Ted
> >
>=20
>=20
>=20
> >We conclude that allowing hop-by-hop security for location based
routing
> >is acceptable, and Do Not Distribute DOES allow forwarding to a
routing
> >database, but requires that the proxy doing the location based route
to
> >remove the location it uses for that purpose.
>=20
> To reiterate this, I do not believe that this is sufficient; I think
you
> have to
> have some way of saying "re-transmission for routing permitted" before
> you meet the geopriv privacy concerns.
>=20
> >5. We proposed to create a parameter and a registry to distinguish
> >multiple ways to dereference a location where HTTP(S) was the
transport.
> >Resolution of this issue depends on resolution of issue 6.
>=20
> I believe that an HTTP GET of a PIDF-LO object is a reasonable
baseline.
> Saying that other mechanisms should update this document with their
> dereferencing mechanisms if they could be confused with the baseline
> seems to me to be sufficient.  I'm not sure a registry is needed.  In
> practice,
> I don't think we would hit the interoperability problems often. These
> days,
> extensions to HTTP commonly use different  mime types to indicate the
> special
> handling, and I expect anyone creating a more feature-rich
dereferencing
> mechanism built on HTTP would not end up using pidf-lo.
>=20
> That means the "uri confusion" aspects of the discussion are probably
not
> that severe.  The only issue is if someone supports only the more
feature-
> rich
> alternative and not a baseline GET of the pidf-lo.  I think we should
> discourage that, by establishing a real baseline.
>=20
> Note also that HTTP URIs don't indicate the HTTP Method directly;
often
> they
> support multiple methods (e.g. HEAD and GET).
>=20
> >6. We proposed to allow sip/sips and pres: in the location header.  I
> >later speculated that if we allow this, it forms a simple way to do
> >location-by-reference, and we could eliminate the raw HTTP(S) Get
> >proposal, with the parameter and registry entirely.  There was some
list
> >traffic in favor of this.  Martin Thompson is skeptical that SIP
> >Presence Subscribe/Notify is actually allowed as a geopriv "using
> >protocol".  Several of think that it is.  If we don't get any more
> >comments on this proposal, we will allow sip/sips and pres, not have
any
> >http(s) resolution text, and not propose a parameter and accompanying
> >registry.
>=20
> I share some of Martin's skepticism here, but I won't fall on my sword
> if the WG goes in this direction.
>=20
> Thanks for the clear set of issues and opportunity to comment,
>=20
> 					Ted Hardie

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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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X-OriginalArrivalTime: 25 Jul 2006 22:39:06.0364 (UTC)
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Thread-Topic: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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--===============0748706261==--



From sip-bounces@ietf.org Tue Jul 25 18:53:19 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Marc Linsner'" <mlinsner@cisco.com>,
	"Peterson, Jon" <jon.peterson@neustar.biz>,
	"'Andrew Newton'" <andy@hxr.us>, "Rosen, Brian" <Brian.Rosen@neustar.biz>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Tue, 25 Jul 2006 18:51:21 -0400
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So, you are arguing against proxy insertion of a location-by-reference?

You don't have a problem with UA insertion of location-by-reference in a
Location header?

Agree that Location insertion by proxy could assume an INVITE arriving at a
proxy without any location.

I've always had a problem with proxy insertion of location because I am
worried that proxies will do this without regard to user's wishes.  In an
emergency case, however, there are no such worries.  I don't see any
difference between by-value and by-reference.

Proxies may disregard advice we supply, but Location URIs in Location
headers probably should not have any user identification information in
either the URI itself, or the resulting PIDF.  If such information is
desired, use the AoR.

De-referencing should always be subject to user rulesets, and would not be
any different than by-value except the obvious issue that the owners of the
proxy may not provide the user with the facilities to control the ruleset,
my original worry.  

Would you not agree that if a proxy is allowed to insert location-by-value,
the concerns would be the same?

Brian



> -----Original Message-----
> From: Marc Linsner [mailto:mlinsner@cisco.com]
> Sent: Tuesday, July 25, 2006 6:17 PM
> To: 'Peterson, Jon'; 'Andrew Newton'; 'Rosen, Brian'
> Cc: sip@ietf.org; geopriv@ietf.org
> Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
> 
> I have no problem with pidf/pidf-lo/presence.
> 
> I have no problem with a ua placing a uri pointing to their presence
> service
> in a header of a sip invite.
> 
> Context check:  We're discussing a sip invite (assumed pointed to
> emergency
> service) that arrives at a proxy with zero location information.
> 
> I'm the only one that has a problem with any (visited) proxy/LIS creating
> a
> uri that can be de-referenced by anyone without regard to user
> security/privacy rules?
> 
> -Marc-
> 
> 
> 
> 
> 
> > -----Original Message-----
> > From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> > Sent: Tuesday, July 25, 2006 4:42 PM
> > To: Andrew Newton; Rosen, Brian
> > Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> > Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to
> > location-conveyance
> >
> >
> > As Brian points out, the whole point of using PIDF for
> > location information was the similarity of the early defined
> > use cases to the presence architecture. This certainly
> > included use cases in which one wanted to fetch or subscribe
> > to location information. Those operations require the
> > existence of a reference, and that reference was commonly
> > described as a URI, at the time. Lately, I think the idea of
> > having a URI pointing to location information has become
> > associated with a specific proposal (i.e. HELD), but clearly
> > RFC4079 sketched out an architecture for using presence to
> > acquire location information and it specifically mentions
> > URIs as a way to refer to presence.
> >
> > There is no "hidden magic" in any of this. Subscribing to
> > presence, and using URIs to indicate presence information, is
> > exhaustively specified for SIP. PIDF-LO was designed to reuse
> > all of that work and specifically not to require us to go out
> > and reinvent everything in order to be able to refer to
> > location information out in the network.
> >
> > Basically, I support the direction re: bullet 6. I might
> > further suggest that the CID URI scheme be permitted in the
> > Location header, in order to be able to refer to MIME bodies
> > included in SIP messages. I'm also neutral/favorable on using
> > HTTP GET. But I think either of these URI schemes and their
> > usage in the Location header could just as well be described
> > in subsequent documents.
> >
> > Jon Peterson
> > NeuStar, Inc,
> >
> > > -----Original Message-----
> > > From: Andrew Newton [mailto:andy@hxr.us]
> > > Sent: Tuesday, July 25, 2006 1:05 PM
> > > To: Rosen, Brian
> > > Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> > > Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
> > > location-conveyance
> > >
> > >
> > > For the record, I agree with Brian.
> > >
> > > -andy
> > >
> > > On Jul 25, 2006, at 3:50 PM, Rosen, Brian wrote:
> > >
> > > > When the original PIDF-LO RFC came out, location-by-reference was
> > > > defined.
> > > >
> > > > You have a URI.  You can exchange the URI for location.
> > There is a
> > > > protocol defined for that purpose (SIP SUBSCRIBE).  That
> > is location
> > > > by reference.  It also serves as the location update mechanism,
> > > > clearly something we need.
> > > >
> > > > Some day, there may be other location URI references.  We
> > don't need
> > > > to cover them in this draft.  The basic location by
> > reference horse
> > > > has left the barn.  Were you planning on creating an RFC that
> > > specifically
> > > > disallows a Subscribe on a sip/sips/pres URI to get location?
> > > > THAT WAS
> > > > THE POINT OF MAKING IT PART OF A PIDF.
> > > >
> > > > I do think there is some value in defining a way to express a
> > > > location-by-reference aka presence URI which is not just
> > the AoR.
> > > > That
> > > > is one reason why I want to define the Location URI as
> > being able to
> > > > contain a sip/sips/pres URI.  The purpose is to allow a UA or a
> > > > proxy to insert location where there is no association between the
> > > identity of
> > > > the SIP user and the location.
> > > >
> > > > But that is all I think we need.
> > > >
> > > > We have a requirement for proxy insertion of location.  If
> > > we disallow
> > > > location-by-reference (which I think is a good idea in
> > general, but
> > > > futile in practice), then we would need something like the
> > > Data URI to
> > > > support location insertion by proxy.  If we DO allow
> > > > location-by-reference, then it also serves to meet the location
> > > > insertion by proxy requirement, and we don't need the Data
> > > URI.  We're
> > > > looking for a use case which requires proxy insertion by
> > value, but
> > > > haven't heard one yet.
> > > >
> > > > You can make an argument that you can take the From:,
> > assume it's an
> > > > AoR, and SUBSCRIBE to the presence of that AoR to get
> > location, thus
> > > > not needing a Location header.  I really think letting the
> > > Location header
> > > > specify an alternate presence URI is a good idea.
> > > >
> > > > Brian
> > > >
> > > >> -----Original Message-----
> > > >> From: Marc Linsner [mailto:mlinsner@cisco.com]
> > > >> Sent: Tuesday, July 25, 2006 3:29 PM
> > > >> To: Rosen, Brian; sip@ietf.org
> > > >> Cc: geopriv@ietf.org
> > > >> Subject: RE: [Geopriv] Consensus on changes to
> > location-conveyance
> > > >>
> > > >> In-line.....
> > > >>
> > > >>> -----Original Message-----
> > > >>> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > >>> Sent: Tuesday, July 25, 2006 1:38 PM
> > > >>> To: sip@ietf.org
> > > >>> Cc: geopriv@ietf.org
> > > >>> Subject: [Geopriv] Consensus on changes to location-conveyance
> > > >>>
> > > >>> I have put out a number of emails on proposed changes
> > to location
> > > >>> conveyance.  Some of them have sparked a number of
> > comments.  My
> > > >>> summary of consensus on these issues is:
> > > >>>
> > > >>> 1. We proposed to explicitly disallow use of a Data URI, which
> > > >>> would allow specification of a Location-by-value in a header.
> > > >>> There was a lot of list traffic on this, currently in the state
> > > >>> where Keith asked for a real use case for it.
> > > >>> So far, we don't see a use case that cannot be provided by
> > > >>> location-by-reference in a header.  If we don't get a good use
> > > >>> case, I think consensus is to disallow location-by-value in a
> > > >>> header.
> > > >>>
> > > >>
> > > >> I see a problem here.  Location-by-reference is nothing more than
> > > > 'magic
> > > >> happens here'!  Please point to a wg item documenting the
> > > >> location-by-reference mechanism.  How can you state the
> > > >> location-by-reference will cover all the use cases that
> > > the Data URI
> > > > would
> > > >> when location-by-reference hasn't even been defined
> > > anywhere?  You're
> > > >> loading up the magic with a lot of assumptions.  Location-by-
> > > >> reference
> > > > is
> > > >> an
> > > >> individual submission as a solution to a set of requirements that
> > > > haven't
> > > >> been fleshed-out in geopriv.
> > > >>
> > > >> I believe that Henning/Hannes also recognized this
> > issue, hence the
> > > >> suggestion of a Data URI.  Dismissing this suggestion based on an
> > > > already
> > > >> accepted non-existent mechanism seems like getting the
> > > cart before
> > > >> the
> > > >> horse.
> > > >>
> > > >> ..........snip
> > > >>
> > > >>> 6. We proposed to allow sip/sips and pres: in the
> > location header.
> > > >>> I later speculated that if we allow this, it forms a
> > simple way to
> > > >>> do location-by-reference, and we could eliminate the
> > raw HTTP(S)
> > > >>> Get proposal, with the parameter and registry entirely.
> >  There was
> > > >>> some list traffic in favor of this.  Martin Thompson is
> > skeptical
> > > >>> that SIP Presence Subscribe/Notify is actually allowed as a
> > > >>> geopriv "using protocol".  Several of think that it is.  If we
> > > >>> don't get any more comments on this proposal, we will allow
> > > >>> sip/sips and pres, not have any
> > > >>> http(s) resolution text, and not propose a parameter and
> > > >>> accompanying registry.
> > > >>
> > > >> I certainly think it's more prudent to define the dereferencing
> > > > mechanism
> > > >> prior to making claims about the protocol.
> > > >>
> > > >> Do I believe location-by-reference will happen?  Probably,
> > > but making
> > > >> assumptions about the mechanism now is quite premature.
> > > >>
> > > >> -Marc-
> > > >>
> > > >>>
> > > >>> If this does not reflect your position, please speak up now.
> > > >>>
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use
> > > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > > sipping@ietf.org for new developments on the application of sip
> > >
> > >
> > > _______________________________________________
> > > Geopriv mailing list
> > > Geopriv@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/geopriv
> > >
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


_______________________________________________
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A couple notes on the whole "Do not Distribute" flag and its meaning in =
a SIP context.

Firstly, I think it's obligatory to flex the muscles of location =
by-reference in response to this. When you send only a reference to =
location information in an INVITE, that INVITE is not sharing the =
location information with routing entities, is not a using protocol, and =
is not "distributing" location information in any sense. Any routing =
entity that dereferences the URI, and attempts to acquire location =
information, can be individually authorized by the Location Server, and =
the protocol over which location information is transmitted to those =
entities (be it SIP NOTIFYs or whatever) is the using protocol, and is =
distributing presence information. The entities that receive PIDF-LO =
through that mechanism are bound not to retransmit it by the "Do not =
Distribute" flag, but can use it for their personal consumption to make =
decisions. In short, my contention would be that nothing is broken in =
this case.

So, what if you're using location by-value? Well, if you can use =
encryption, then "Do not Distribute" still has some plausible =
applicability. Of course, encrypting the PIDF-LO object is cumbersome, =
and you have to be certain to whom you are encrypting it, but I don't =
think the PIDF-LO can be said to have been "distributed" to anyone that =
can't even inspect it. They're just relaying a closed envelope to the =
entity that can open it. That entity would be bound by any "Do not =
Distribute" flag. Without speaking to the details of the LoST protocol, =
or making any specific proposal to modify it, I can imagine an approach =
where the PIDF-LO object could only be decrypted by the LoST server, and =
even the proxy that sends the query to LoST could not decrypt the =
object.

And supposing you can't or won't encrypt? Well, in that case I agree =
that sending location information by-value in an INVITE is ambiguous =
with regards to the "Do not Distribute" flag. Many, if not most, SIP =
proxies don't inspect MIME bodies by default, and probably wouldn't even =
be aware that they are distributing location information - we can't =
reasonably hold automata responsible for violating the flag in the =
course of their conventional behavior, anyway. I think back in the day, =
we roughly concluded that if you need to send location information =
by-value in cleartext in something like a SIP INVITE, you should set the =
flag off. There was much talk back at the time that in emergency =
services cases, you had to sacrifice a certain expectation of privacy =
that you might want to retain for other location-based services.

I also would like to stress something that I've said a couple of times =
in the history of this draft - that location conveyance does not exist =
solely for the purposes of ECRIT's use cases. Don't forget that there =
are other location-based services that we want to enable in a SIP =
context, and that even if some approaches wouldn't make sense for ECRIT, =
that doesn't mean that they shouldn't be specified.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Ted Hardie [mailto:hardie@qualcomm.com]
> Sent: Tuesday, July 25, 2006 11:19 AM
> To: Rosen, Brian; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: Re: [Geopriv] Consensus on changes to location-conveyance
>=20
>=20
> At 1:38 PM -0400 7/25/06, Rosen, Brian wrote:
> >4. We proposed to explicitly allow hop-by-hop TLS security=20
> for location
> >when the endpoint wished to allow routing based on location.=20
>  We further
> >proposed to interpret "Do not Distribute" flags as allowing=20
> sending of
> >the location to the routing database (LoST for example). =20
> Hannes pointed
> >out that there was a proposal to allow two Location headers, one of
> >which could be used for routing and another for onward=20
> forwarding to the
> >recipient.  Hannes' message seemed to agree that "Do not distribute"
> >would allow sending of the location to the=20
> routing-by-location database,
> >but would obligate the proxy that did that to remove the=20
> location before
> >it forwarded the message.  There were no other messages on=20
> this thread.
>=20
> Actually, I commented on the same problem, though it may have been
> connected to one of the related threads; here's the text of=20
> that message:
>=20
> >At 5:23 PM -0400 7/19/06, Brian Rosen wrote:
> >>NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU=20
> REPLY, SEND IT ONLY TO
> >>GEOPRIV
> >>However, this opens up the issue of geopriv privacy=20
> concerns because the
> >>sender does not know the identity of the element (proxy)=20
> that will route the
> >>call using LoST.  It may be that the UAC itself does this,=20
> but it may be
> >>some other intermediary.
> >>
> >>If the identity of the intermediary is not known, then=20
> protecting the
> >>location information becomes more difficult, because we=20
> cannot encrypt using
> >>keying material from the intended recipient.
> >
> >Just to mention this, it appears to also rule out the idea=20
> of encrypting
> >with the user's key, since there is no way of knowing=20
> whether the "intended
> >recipient" has that key material either.  Intended recipient=20
> is actually
> >a bit of a problematic term here, since it's really the=20
> on-path network elements
> >which consume the information.  They aren't the intended recipient of
> >the communication, they are the recipients of the signaling=20
> (of which location
> >is now a part).
> >
> >This is a bit confusing because it seems to be possible to include a
> >"re-transmission allowed"  rule with a value of "no" (intended to be
> >consumed by the "intended recipient" of the underlying=20
> communication),
> >while still wanting to allow the location to be passed along=20
> the path.
> >
> >>In fact, we would need to use
> >>hop-by-hop security, with transitive trust.  This is what=20
> the emergency case
> >>does, but it also allows no channel security if necessary=20
> (usually meaning,
> >>try it with TLS and if that fails, try again without TLS).
> >
> >While I certainly encourage hop-by-hop security, I don't=20
> think it really
> >meets the geopriv privacy concerns.  I think you  have to=20
> have some way of
> >expressing "re-transmission for routing permitted" (and=20
> likely retention
> >limited), since that is a separate statement.
> >
> >Just typing aloud here, but could we come up with a way to use an
> >external ruleset to express this?  With an embedded cid, that would
> >not actually require dereferencing.
> >					Ted
> >
>=20
>=20
>=20
> >We conclude that allowing hop-by-hop security for location=20
> based routing
> >is acceptable, and Do Not Distribute DOES allow forwarding=20
> to a routing
> >database, but requires that the proxy doing the location=20
> based route to
> >remove the location it uses for that purpose.
>=20
> To reiterate this, I do not believe that this is sufficient;=20
> I think you have to
> have some way of saying "re-transmission for routing permitted" before
> you meet the geopriv privacy concerns.
>=20
> >5. We proposed to create a parameter and a registry to distinguish
> >multiple ways to dereference a location where HTTP(S) was=20
> the transport.
> >Resolution of this issue depends on resolution of issue 6.
>=20
> I believe that an HTTP GET of a PIDF-LO object is a=20
> reasonable baseline.
> Saying that other mechanisms should update this document with their
> dereferencing mechanisms if they could be confused with the baseline
> seems to me to be sufficient.  I'm not sure a registry is=20
> needed.  In practice,
> I don't think we would hit the interoperability problems=20
> often. These days,
> extensions to HTTP commonly use different  mime types to=20
> indicate the special
> handling, and I expect anyone creating a more feature-rich=20
> dereferencing
> mechanism built on HTTP would not end up using pidf-lo. =20
>=20
> That means the "uri confusion" aspects of the discussion are=20
> probably not
> that severe.  The only issue is if someone supports only the=20
> more feature-rich
> alternative and not a baseline GET of the pidf-lo.  I think we should
> discourage that, by establishing a real baseline.
>=20
> Note also that HTTP URIs don't indicate the HTTP Method=20
> directly; often they
> support multiple methods (e.g. HEAD and GET).
>=20
> >6. We proposed to allow sip/sips and pres: in the location header.  I
> >later speculated that if we allow this, it forms a simple way to do
> >location-by-reference, and we could eliminate the raw HTTP(S) Get
> >proposal, with the parameter and registry entirely.  There=20
> was some list
> >traffic in favor of this.  Martin Thompson is skeptical that SIP
> >Presence Subscribe/Notify is actually allowed as a geopriv "using
> >protocol".  Several of think that it is.  If we don't get any more
> >comments on this proposal, we will allow sip/sips and pres,=20
> not have any
> >http(s) resolution text, and not propose a parameter and accompanying
> >registry.
>=20
> I share some of Martin's skepticism here, but I won't fall on my sword
> if the WG goes in this direction.
>=20
> Thanks for the clear set of issues and opportunity to comment,
> 			=09
> 					Ted Hardie
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

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From sip-bounces@ietf.org Tue Jul 25 19:23:33 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 01:21:42 +0200
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Brian,

> Would you not agree that if a proxy is allowed to insert 
> location-by-value,
> the concerns would be the same?

That would depend whether the URL dereferences the current location, or the 
location at a particular moment in time. The value of a URI with up2date 
location info is much higher and much more privacy invading than a one-time 
snapshot of a user's location. So perhaps same concerns, but much stronger

For location-by-reference, you'd probably want some additional requirements:
- URL must be valid for a limited amount of time
- URL must be cryptographically hard to guess
- URL must not contain any information that identifies the user / device / 
AoR
- for whatever transport protocol is used: response must be marked as 'no 
cache'
- user must be able to remove the information at the URL, ie explicit 
invalidation
- user must be able to verify correctness of the issued information (ie user 
can access the URL himself)
- user must be able to control who accesses the URL, both upfront and 
history of accesses (for a reasonable period)
- there must be explicit consent before a proxy would insert user location

For the latter point: except for emergency scenario's, the UAC should 
include some flag in the INVITE saying "proxy: please append location". 
You'd probably also want some feedback (eg proxy or UAS adding a header to 
the response saying 'this is the URL that I appended/got'

Regards,

Jeroen




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From sip-bounces@ietf.org Tue Jul 25 20:03:45 2006
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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Tue, 25 Jul 2006 20:01:45 -0400
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I don't really understand why you think location by reference is so much of
a concern.  If you look at the discussion in context, PIDF-LO defines
location as part of presence.  All of these concerns are relevant there, and
I think they have been adequately covered there.  We even have some work on
anonymization of the identity (and thus of the URI) in presence.  Generally,
a presence uri:

Is not valid for a limited amount of time
Is not hard to guess, if it uses an AoR and could be with anonymous URI.
Identifies the user if it uses an AoR and doesn't with anonymous URI

The protocol (SIP) for retrieving presence, and thus location has the
location ruleset as we defined for PIDF-LO, but otherwise does not have "no
cache" semantics (albeit caching headers is not very interesting in SIP).

Generally, presence systems do have user control of who is allowed to access
information, including location and identifying information, so I think that
is covered fairly well.

Proxy insertion of location, whether by value or by reference has the threat
of not being as subject to user control, but I think in every case you have
to trust your provider to do as you wish, whether it is your presence
provider or your location provider independent of your presence provider,
and proxy insertion doesn't fundamentally alter that.  I worry more about
it, but I don't objectively think it's much different.

Now, if there are other location dereferencing systems, they may need an
analysis like what you are suggesting, but with respect to SIP Presence, as
a form of location by reference, I think we have been over the ground
sufficiently, and we have RFCs that permit it.  Adding restrictions on the
Location header, which I propose is useful precisely to allow an anonymous
presence uri that does not reveal identity is not so great I think.  If you
don't allow that, you would kind of have to use an AoR, which may be
appropriate in some cases, but would not in others.

I think the proponents of proxy insertion argue that providing it means
endpoints don't have to do anything, so I would think they would object to
explicit permission to do so.  You might get away with a flag that says
"don't do it" in the IETF process, but whether such a flag would be deployed
is another matter.   I think we all agree that the emergency case is
different here; laws govern behavior, and in most cases compel location
assertion.

I kind of like the idea that the proxy inserts the header in the response so
the UAC can see it.  What do others think of that? 

Brian


> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Tuesday, July 25, 2006 7:22 PM
> To: Brian Rosen; 'Marc Linsner'; Peterson, Jon; 'Andrew Newton'; Rosen,
> Brian
> Cc: sip@ietf.org; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
> 
> Brian,
> 
> > Would you not agree that if a proxy is allowed to insert
> > location-by-value,
> > the concerns would be the same?
> 
> That would depend whether the URL dereferences the current location, or
> the
> location at a particular moment in time. The value of a URI with up2date
> location info is much higher and much more privacy invading than a one-
> time
> snapshot of a user's location. So perhaps same concerns, but much stronger
> 
> For location-by-reference, you'd probably want some additional
> requirements:
> - URL must be valid for a limited amount of time
> - URL must be cryptographically hard to guess
> - URL must not contain any information that identifies the user / device /
> AoR
> - for whatever transport protocol is used: response must be marked as 'no
> cache'
> - user must be able to remove the information at the URL, ie explicit
> invalidation
> - user must be able to verify correctness of the issued information (ie
> user
> can access the URL himself)
> - user must be able to control who accesses the URL, both upfront and
> history of accesses (for a reasonable period)
> - there must be explicit consent before a proxy would insert user location
> 
> For the latter point: except for emergency scenario's, the UAC should
> include some flag in the INVITE saying "proxy: please append location".
> You'd probably also want some feedback (eg proxy or UAS adding a header to
> the response saying 'this is the URL that I appended/got'
> 
> Regards,
> 
> Jeroen
> 



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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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On Jul 25, 2006, at 7:21 PM, Jeroen van Bemmel wrote:
> For location-by-reference, you'd probably want some additional  
> requirements:
> - URL must be valid for a limited amount of time
> - URL must be cryptographically hard to guess
> - URL must not contain any information that identifies the user /  
> device / AoR
> - for whatever transport protocol is used: response must be marked  
> as 'no cache'
> - user must be able to remove the information at the URL, ie  
> explicit invalidation
> - user must be able to verify correctness of the issued information  
> (ie user can access the URL himself)
> - user must be able to control who accesses the URL, both upfront  
> and history of accesses (for a reasonable period)
> - there must be explicit consent before a proxy would insert user  
> location

Please tell me that you are merely suggesting guidelines and  
considerations by operators and are not suggesting 2119 language for  
these?  Because just like you'd never get agreement on the intended  
duration or accuracy or confidence of location-by-value, you'd never  
get agreement on these.

> For the latter point: except for emergency scenario's, the UAC  
> should include some flag in the INVITE saying "proxy: please append  
> location". You'd probably also want some feedback (eg proxy or UAS  
> adding a header to the response saying 'this is the URL that I  
> appended/got'

Just thinking out loud here, but maybe the rule should be that  
proxies never append a Location header except for emergencies.

-andy

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From: "Brian Rosen" <br@brianrosen.net>
To: "'Andrew Newton'" <andy@hxr.us>, "'Jeroen van Bemmel'" <jbemmel@zonnet.nl>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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> Please tell me that you are merely suggesting guidelines and
> considerations by operators and are not suggesting 2119 language for
> these?  Because just like you'd never get agreement on the intended
> duration or accuracy or confidence of location-by-value, you'd never
> get agreement on these.
I agree here

> 
> > For the latter point: except for emergency scenario's, the UAC
> > should include some flag in the INVITE saying "proxy: please append
> > location". You'd probably also want some feedback (eg proxy or UAS
> > adding a header to the response saying 'this is the URL that I
> > appended/got'
> 
> Just thinking out loud here, but maybe the rule should be that
> proxies never append a Location header except for emergencies.
Works for me.  Probably wouldn't work for those carriers who do not intend
to support location acquisition by endpoints and do plan to support
commercial location based routing.  They plan to make mucho dineros charging
for location used for location based routing, and don't want those pesky
users to get in the way of their plans.  Of course, they can overcome all
manner of protocol machinery with "As a condition of providing service, user
agrees ...."

Brian


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	"Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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Brian,

The fact that people can make money out of a solution scarcely seems a
good reason not to permit it!



> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]
> Sent: Wednesday, 26 July 2006 10:40 AM
> To: 'Andrew Newton'; 'Jeroen van Bemmel'
> Cc: sip@ietf.org; Rosen, Brian; 'Marc Linsner'; geopriv@ietf.org;
> Peterson,Jon
> Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
>=20
> > Please tell me that you are merely suggesting guidelines and
> > considerations by operators and are not suggesting 2119 language for
> > these?  Because just like you'd never get agreement on the intended
> > duration or accuracy or confidence of location-by-value, you'd never
> > get agreement on these.
> I agree here
>=20
> >
> > > For the latter point: except for emergency scenario's, the UAC
> > > should include some flag in the INVITE saying "proxy: please
append
> > > location". You'd probably also want some feedback (eg proxy or UAS
> > > adding a header to the response saying 'this is the URL that I
> > > appended/got'
> >
> > Just thinking out loud here, but maybe the rule should be that
> > proxies never append a Location header except for emergencies.
> Works for me.  Probably wouldn't work for those carriers who do not
intend
> to support location acquisition by endpoints and do plan to support
> commercial location based routing.  They plan to make mucho dineros
> charging
> for location used for location based routing, and don't want those
pesky
> users to get in the way of their plans.  Of course, they can overcome
all
> manner of protocol machinery with "As a condition of providing
service,
> user
> agrees ...."
>=20
> Brian
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
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[mf2]

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From sip-bounces@ietf.org Tue Jul 25 20:58:38 2006
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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
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Ah, yes, but they can do the same with endpoint assertion of location,
except that the user has more control over it.

Don't object to carriers, or other service providers, making money off
of location based routing.  I just want the user to decide when it is,
and isn't, okay to do that with his location.  Controlling when the
location header is attached to the message is a really hard stop on
not-so-nice folks.  When you let them assert location, then it's harder.


Now, reiterating what I have said several times, I think location
assertion by proxy is a done deal, whether I like it or not, just as I
think location by reference is a done deal.  There is one use case I
like -- emergency call during transition, when the network knows, in
some circumstances, where you are, but the endpoint and/or the access
network is not yet upgraded.  I just hate opening a big loophole for
that one case though.

Brian

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Tuesday, July 25, 2006 8:46 PM
> To: Brian Rosen; Andrew Newton; Jeroen van Bemmel
> Cc: sip@ietf.org; Rosen, Brian; Marc Linsner; geopriv@ietf.org;
Peterson,
> Jon
> Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
>=20
> Brian,
>=20
> The fact that people can make money out of a solution scarcely seems a
> good reason not to permit it!
>=20
>=20
>=20
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Wednesday, 26 July 2006 10:40 AM
> > To: 'Andrew Newton'; 'Jeroen van Bemmel'
> > Cc: sip@ietf.org; Rosen, Brian; 'Marc Linsner'; geopriv@ietf.org;
> > Peterson,Jon
> > Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-
> > conveyance
> >
> > > Please tell me that you are merely suggesting guidelines and
> > > considerations by operators and are not suggesting 2119 language
for
> > > these?  Because just like you'd never get agreement on the
intended
> > > duration or accuracy or confidence of location-by-value, you'd
never
> > > get agreement on these.
> > I agree here
> >
> > >
> > > > For the latter point: except for emergency scenario's, the UAC
> > > > should include some flag in the INVITE saying "proxy: please
> append
> > > > location". You'd probably also want some feedback (eg proxy or
UAS
> > > > adding a header to the response saying 'this is the URL that I
> > > > appended/got'
> > >
> > > Just thinking out loud here, but maybe the rule should be that
> > > proxies never append a Location header except for emergencies.
> > Works for me.  Probably wouldn't work for those carriers who do not
> intend
> > to support location acquisition by endpoints and do plan to support
> > commercial location based routing.  They plan to make mucho dineros
> > charging
> > for location used for location based routing, and don't want those
> pesky
> > users to get in the way of their plans.  Of course, they can
overcome
> all
> > manner of protocol machinery with "As a condition of providing
> service,
> > user
> > agrees ...."
> >
> > Brian
> >
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>
------------------------------------------------------------------------
--
> ----------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
>
------------------------------------------------------------------------
--
> ----------------------
> [mf2]

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From: Andrew Newton <andy@hxr.us>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Tue, 25 Jul 2006 21:00:54 -0400
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Now, now.  Before this degenerates, for non-emergency situations  
having the location flow from the acquisition protocol through the  
UAC on to the call path is completely acceptable and gives the user  
control.  We recognize that emergency cases are different, users  
often are concerned about the emergency and don't need to be bothered  
by interfaces asking "should I include your location".

-andy

On Jul 25, 2006, at 8:45 PM, Winterbottom, James wrote:

> Brian,
>
> The fact that people can make money out of a solution scarcely seems a
> good reason not to permit it!
>
>
>
>> -----Original Message-----
>> From: Brian Rosen [mailto:br@brianrosen.net]
>> Works for me.  Probably wouldn't work for those carriers who do not
> intend
>> to support location acquisition by endpoints and do plan to support
>> commercial location based routing.  They plan to make mucho dineros
>> charging
>> for location used for location based routing, and don't want those
> pesky
>> users to get in the way of their plans.  Of course, they can overcome
> all
>> manner of protocol machinery with "As a condition of providing
> service,
>> user
>> agrees ...."


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Hi Kingston,

 

Timer C. See section 16.8 of rfc3261.

 

BR,

Stephen

  _____  

From: Kingston.Smiler@flextronicssoftware.com
[mailto:Kingston.Smiler@flextronicssoftware.com] 
Sent: Tuesday, July 25, 2006 8:50 PM
To: Stephen Zhou
Cc: 'Manjunath Warad'; netch@portaone.com; sip@ietf.org;
sip-implementors@cs.columbia.edu
Subject: RE: [Sip] RE: [Sip-implementors] Regarding non-INVITE
servertransaction

 


Hi,
       I agree with Stephen because it is not a problem in non - IST; 
This is generic to non-IST as well as IST. 

But consider a scenario like this 

UAC of        UAS of                     UAS of UE2 
      UE1                 proxy                 
 |                |      |                        | 
 |         non-INV        |      |     non-INV                | 
 |--------------------->|      |---------------------------->| 
 |                |      |                2xx        | 
 |                |      |                *<---------------| 
 |                |      |                        | 
                    UAC of 
                                     Proxy 
        

         Here UAS of UE2 is re-transmitting the 2xx message toward proxy. 
But the message that has been sent by UAS of UE2 hasn't reached UAC of proxy

(due to network breakage or some other means). So the proxy core (TU)
doesn't 
send the response to UAS of proxy.  In this case how will UAS txn of proxy
can be released. 

Regards,
S.Kingston Smiler.






Stephen Zhou <zhouhong@huawei.com> 

07/25/2006 03:21 PM 


To

"'Manjunath Warad'" <manjunathwarad@huawei.com>, netch@portaone.com,
Kingston Smiler/BLR/HSS@HSS 


cc

sip@ietf.org, sip-implementors@cs.columbia.edu 


Subject

RE: [Sip] RE: [Sip-implementors] Regarding non-INVITE
servertransaction

 


 

 




Hi,

Manjunath is right.

We MUST assume the communication between TU layer and transation lay in a
sip stack  is reliable. If there have problems, I thinks this is program
problem rather than protocol(transaction layer) problem. 
Additionally, the same issue exist in IST, if the IST cann't receive the
responses from TU.

BR,
Stephen
-----Original Message-----
From: Manjunath Warad [mailto:manjunathwarad@huawei.com] 
Sent: Tuesday, July 25, 2006 5:08 PM
To: netch@portaone.com; Kingston.Smiler@flextronicssoftware.com
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: [Sip] RE: [Sip-implementors] Regarding non-INVITE server
transaction

Hi,
                Below mentioned behaviour is for client transaction, not for
server transaction.

Rgds,
Manju

-----Original Message-----
From: sip-implementors-bounces@cs.columbia.edu
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of Valentin
Nechayev
Sent: Tuesday, July 25, 2006 4:55 PM
To: Kingston.Smiler@flextronicssoftware.com
Cc: sip@ietf.org; sip-implementors@cs.columbia.edu
Subject: Re: [Sip-implementors] Regarding non-INVITE server transaction

Mon, Jul 24, 2006 at 20:08:13, Kingston.Smiler wrote about
"[Sip-implementors] Regarding non-INVITE server transaction": 

>         If the non-IST doesn't receive any RESPONSE message then what
will 
> happen?
> RFC 3261 (in Figure 8: non-INVITE server transaction ) doesn't mention

> properly 
> what will happen if the  non-IST doesn't receive any provisional 
> response/final response.

==={{{
8.1.3.1 Transaction Layer Errors

  In some cases, the response returned by the transaction layer will
  not be a SIP message, but rather a transaction layer error.  When a
  timeout error is received from the transaction layer, it MUST be
  treated as if a 408 (Request Timeout) status code has been received.
  If a fatal transport error is reported by the transport layer
  (generally, due to fatal ICMP errors in UDP or connection failures in
  TCP), the condition MUST be treated as a 503 (Service Unavailable)
  status code.
===}}}

-- 
Valentin Nechayev
PortaOne Inc., Software Engineer
mailto:netch@portaone.com
_______________________________________________
Sip-implementors mailing list
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_______________________________________________
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<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>Hi Kingston,<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>Timer C. See section 16.8
of rfc3261.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>BR,<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>Stephen<o:p></o:p></span></font></p>

</div>

<div>

<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>

<hr size=2 width="100%" align=center tabindex=-1>

</span></font></div>

<p class=MsoNormal><b><font size=2 face=Tahoma><span lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
Kingston.Smiler@flextronicssoftware.com
[mailto:Kingston.Smiler@flextronicssoftware.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, July 25, 2006 8:50
PM<br>
<b><span style='font-weight:bold'>To:</span></b> Stephen Zhou<br>
<b><span style='font-weight:bold'>Cc:</span></b> 'Manjunath Warad';
netch@portaone.com; sip@ietf.org; sip-implementors@cs.columbia.edu<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Sip] RE:
[Sip-implementors] Regarding non-INVITE servertransaction</span></font><span
lang=EN-US><o:p></o:p></span></p>

</div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'><br>
</span></font><font size=2 face=sans-serif><span lang=EN-US style='font-size:
10.0pt;font-family:sans-serif'>Hi,<br>
&nbsp; &nbsp; &nbsp; &nbsp;I agree with </span></font><b><font size=1
face=sans-serif><span lang=EN-US style='font-size:7.5pt;font-family:sans-serif;
font-weight:bold'>Stephen</span></font></b><font size=2 face=sans-serif><span
lang=EN-US style='font-size:10.0pt;font-family:sans-serif'> because it is not a
problem in non - IST; </span></font><span lang=EN-US><br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>This is generic to non-IST as well as IST. </span></font><span
lang=EN-US><br>
<br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>But consider a scenario like this </span></font><span
lang=EN-US><br>
<br>
UAC of &nbsp; &nbsp; &nbsp; &nbsp;UAS of &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; UAS of UE2 <br>
&nbsp; &nbsp; &nbsp; UE1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; proxy &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;|</span></font><span lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; non-INV &nbsp; &nbsp;
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; non-INV &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</span></font><span lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp;|---------------------&gt;| &nbsp; &nbsp; &nbsp;|----------------------------&gt;|</span></font><span
lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;2xx
&nbsp; &nbsp; &nbsp; &nbsp;|</span></font><span lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;*&lt;---------------|</span></font><span
lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;| &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;|</span></font><span lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; UAC of </span></font><span lang=EN-US><br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Proxy</span></font><span
lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp; &nbsp; &nbsp; &nbsp; </span></font><span
lang=EN-US><br>
<br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Here UAS of UE2 is
re-transmitting the 2xx message toward proxy.</span></font><span lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>But the message that has been sent by UAS of UE2 hasn't
reached UAC of proxy</span></font><span lang=EN-US> <br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>(due to network breakage or some other means). So the
proxy core (TU) doesn't </span></font><span lang=EN-US><br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>send the response to UAS of proxy. &nbsp;In this case
how will UAS txn of proxy can be released.</span></font><span lang=EN-US> <br>
<br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'>Regards,<br>
S.Kingston Smiler.<br>
<br>
</span></font><span lang=EN-US><br>
<br>
<o:p></o:p></span></p>

<table class=MsoNormalTable border=0 cellpadding=0 width="100%"
 style='width:100.0%'>
 <tr>
  <td width="40%" valign=top style='width:40.0%;padding:.75pt .75pt .75pt .75pt'>
  <p class=MsoNormal><b><font size=1 face=sans-serif><span lang=EN-US
  style='font-size:7.5pt;font-family:sans-serif;font-weight:bold'>Stephen Zhou
  &lt;zhouhong@huawei.com&gt;</span></font></b><font size=1 face=sans-serif><span
  lang=EN-US style='font-size:7.5pt;font-family:sans-serif'> </span></font><span
  lang=EN-US><o:p></o:p></span></p>
  <p><font size=1 face=sans-serif><span lang=EN-US style='font-size:7.5pt;
  font-family:sans-serif'>07/25/2006 03:21 PM</span></font><span lang=EN-US> <o:p></o:p></span></p>
  </td>
  <td width="59%" valign=top style='width:59.0%;padding:.75pt .75pt .75pt .75pt'>
  <table class=MsoNormalTable border=0 cellpadding=0 width="100%"
   style='width:100.0%'>
   <tr>
    <td style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal align=right style='text-align:right'><font size=1
    face=sans-serif><span lang=EN-US style='font-size:7.5pt;font-family:sans-serif'>To</span></font><span
    lang=EN-US><o:p></o:p></span></p>
    </td>
    <td valign=top style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal><font size=1 face=sans-serif><span lang=EN-US
    style='font-size:7.5pt;font-family:sans-serif'>&quot;'Manjunath
    Warad'&quot; &lt;manjunathwarad@huawei.com&gt;, netch@portaone.com,
    Kingston Smiler/BLR/HSS@HSS</span></font><span lang=EN-US> <o:p></o:p></span></p>
    </td>
   </tr>
   <tr>
    <td style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal align=right style='text-align:right'><font size=1
    face=sans-serif><span lang=EN-US style='font-size:7.5pt;font-family:sans-serif'>cc</span></font><span
    lang=EN-US><o:p></o:p></span></p>
    </td>
    <td valign=top style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal><font size=1 face=sans-serif><span lang=EN-US
    style='font-size:7.5pt;font-family:sans-serif'>sip@ietf.org,
    sip-implementors@cs.columbia.edu</span></font><span lang=EN-US> <o:p></o:p></span></p>
    </td>
   </tr>
   <tr>
    <td style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal align=right style='text-align:right'><font size=1
    face=sans-serif><span lang=EN-US style='font-size:7.5pt;font-family:sans-serif'>Subject</span></font><span
    lang=EN-US><o:p></o:p></span></p>
    </td>
    <td valign=top style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal><font size=1 face=sans-serif><span lang=EN-US
    style='font-size:7.5pt;font-family:sans-serif'>RE: [Sip] RE:
    [Sip-implementors] Regarding non-INVITE &nbsp; &nbsp; &nbsp; &nbsp;
    servertransaction</span></font><span lang=EN-US><o:p></o:p></span></p>
    </td>
   </tr>
  </table>
  <p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
  style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>
  <table class=MsoNormalTable border=0 cellpadding=0>
   <tr>
    <td valign=top style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
    style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>
    </td>
    <td valign=top style='padding:.75pt .75pt .75pt .75pt'>
    <p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
    style='font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>
    </td>
   </tr>
  </table>
  <p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
  style='font-size:12.0pt'><o:p></o:p></span></font></p>
  </td>
 </tr>
</table>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'><br>
<br>
<br>
</span></font><tt><font size=2 face="Courier New"><span lang=EN-US
style='font-size:10.0pt'>Hi,</span></font></tt><font size=2 face="Courier New"><span
lang=EN-US style='font-size:10.0pt;font-family:"Courier New"'><br>
<br>
<tt><font face="Courier New">Manjunath is right.</font></tt><br>
<br>
<tt><font face="Courier New">We MUST assume the communication between TU layer
and transation lay in a</font></tt><br>
<tt><font face="Courier New">sip stack &nbsp;is reliable. If there have
problems, I thinks this is program</font></tt><br>
<tt><font face="Courier New">problem rather than protocol(transaction layer)
problem. </font></tt><br>
<tt><font face="Courier New">Additionally, the same issue exist in IST, if the
IST cann't receive the</font></tt><br>
<tt><font face="Courier New">responses from TU.</font></tt><br>
<br>
<tt><font face="Courier New">BR,</font></tt><br>
<tt><font face="Courier New">Stephen</font></tt><br>
<tt><font face="Courier New">-----Original Message-----</font></tt><br>
<tt><font face="Courier New">From: Manjunath Warad
[mailto:manjunathwarad@huawei.com] </font></tt><br>
<tt><font face="Courier New">Sent: Tuesday, July 25, 2006 5:08 PM</font></tt><br>
<tt><font face="Courier New">To: netch@portaone.com;
Kingston.Smiler@flextronicssoftware.com</font></tt><br>
<tt><font face="Courier New">Cc: sip@ietf.org; sip-implementors@cs.columbia.edu</font></tt><br>
<tt><font face="Courier New">Subject: [Sip] RE: [Sip-implementors] Regarding non-INVITE
server</font></tt><br>
<tt><font face="Courier New">transaction</font></tt><br>
<br>
<tt><font face="Courier New">Hi,</font></tt><br>
<tt><font face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; Below mentioned behaviour is for client transaction, not for</font></tt><br>
<tt><font face="Courier New">server transaction.</font></tt><br>
<br>
<tt><font face="Courier New">Rgds,</font></tt><br>
<tt><font face="Courier New">Manju</font></tt><br>
<br>
<tt><font face="Courier New">-----Original Message-----</font></tt><br>
<tt><font face="Courier New">From: sip-implementors-bounces@cs.columbia.edu</font></tt><br>
<tt><font face="Courier New">[mailto:sip-implementors-bounces@cs.columbia.edu]
On Behalf Of Valentin</font></tt><br>
<tt><font face="Courier New">Nechayev</font></tt><br>
<tt><font face="Courier New">Sent: Tuesday, July 25, 2006 4:55 PM</font></tt><br>
<tt><font face="Courier New">To: Kingston.Smiler@flextronicssoftware.com</font></tt><br>
<tt><font face="Courier New">Cc: sip@ietf.org; sip-implementors@cs.columbia.edu</font></tt><br>
<tt><font face="Courier New">Subject: Re: [Sip-implementors] Regarding non-INVITE
server transaction</font></tt><br>
<br>
<tt><font face="Courier New">Mon, Jul 24, 2006 at 20:08:13, Kingston.Smiler
wrote about</font></tt><br>
<tt><font face="Courier New">&quot;[Sip-implementors] Regarding non-INVITE
server transaction&quot;: </font></tt><br>
<br>
<tt><font face="Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; If the non-IST
doesn't receive any RESPONSE message then what</font></tt><br>
<tt><font face="Courier New">will </font></tt><br>
<tt><font face="Courier New">&gt; happen?</font></tt><br>
<tt><font face="Courier New">&gt; RFC 3261 (in Figure 8: non-INVITE server
transaction ) doesn't mention</font></tt><br>
<br>
<tt><font face="Courier New">&gt; properly </font></tt><br>
<tt><font face="Courier New">&gt; what will happen if the &nbsp;non-IST doesn't
receive any provisional </font></tt><br>
<tt><font face="Courier New">&gt; response/final response.</font></tt><br>
<br>
<tt><font face="Courier New">==={{{</font></tt><br>
<tt><font face="Courier New">8.1.3.1 Transaction Layer Errors</font></tt><br>
<br>
<tt><font face="Courier New">&nbsp; In some cases, the response returned by the
transaction layer will</font></tt><br>
<tt><font face="Courier New">&nbsp; not be a SIP message, but rather a
transaction layer error. &nbsp;When a</font></tt><br>
<tt><font face="Courier New">&nbsp; timeout error is received from the
transaction layer, it MUST be</font></tt><br>
<tt><font face="Courier New">&nbsp; treated as if a 408 (Request Timeout)
status code has been received.</font></tt><br>
<tt><font face="Courier New">&nbsp; If a fatal transport error is reported by
the transport layer</font></tt><br>
<tt><font face="Courier New">&nbsp; (generally, due to fatal ICMP errors in UDP
or connection failures in</font></tt><br>
<tt><font face="Courier New">&nbsp; TCP), the condition MUST be treated as a
503 (Service Unavailable)</font></tt><br>
<tt><font face="Courier New">&nbsp; status code.</font></tt><br>
<tt><font face="Courier New">===}}}</font></tt><br>
<br>
<tt><font face="Courier New">-- </font></tt><br>
<tt><font face="Courier New">Valentin Nechayev</font></tt><br>
<tt><font face="Courier New">PortaOne Inc., Software Engineer</font></tt><br>
<tt><font face="Courier New">mailto:netch@portaone.com</font></tt><br>
<tt><font face="Courier New">_______________________________________________</font></tt><br>
<tt><font face="Courier New">Sip-implementors mailing list</font></tt><br>
<tt><font face="Courier New">Sip-implementors@cs.columbia.edu</font></tt><br>
<tt><font face="Courier New">https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors</font></tt><br>
<br>
<br>
<tt><font face="Courier New">_______________________________________________</font></tt><br>
<tt><font face="Courier New">Sip mailing list
&nbsp;https://www1.ietf.org/mailman/listinfo/sip</font></tt><br>
<tt><font face="Courier New">This list is for NEW development of the core SIP
Protocol</font></tt><br>
<tt><font face="Courier New">Use sip-implementors@cs.columbia.edu for questions
on current sip</font></tt><br>
<tt><font face="Courier New">Use sipping@ietf.org for new developments on the
application of sip</font></tt><br>
<br>
<br>
</span></font><span lang=EN-US><br>
</span><font size=2 face=sans-serif><span lang=EN-US style='font-size:10.0pt;
font-family:sans-serif'><br>
<br>
*********************** &nbsp;FSS-Unclassified &nbsp; ***********************</span></font><span
lang=EN-US><o:p></o:p></span></p>

</div>

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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Tue Jul 25 22:59:13 2006
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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Hitchhikers guide - when is it done?
Date: Tue, 25 Jul 2006 21:57:04 -0500
To: "Vijay K. Gurbani" <vkg@lucent.com>
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On Jul 25, 2006, at 4:43 PM, Vijay K. Gurbani wrote:

>
> Plus there is some poetic justice in the SIP WG closing doors
> when the last RFC issued is one that (partially) summarizes what
> we have done as a WG.  Too bad RFC number 4242 is already taken.
>

I'm aiming for RFC 5060.

--
Dean



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From sip-bounces@ietf.org Wed Jul 26 02:50:54 2006
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Subject: RE: [Sip] SIP Security mechanism?
Date: Wed, 26 Jul 2006 08:47:59 +0200
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From: "Fries, Steffen" <steffen.fries@siemens.com>
To: <fwmiller@cornfed.com>, "Cullen Jennings" <fluffy@cisco.com>
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Hi Frank,

the method you are describing at least requires the usage of TLS to
protect the key exchange between Alice and Bob.  Moreover, you do need
to trust the proxies in between, as they have access to the keys
(assumed you send the key from Alice to Bob in the SDP). You may combine
the SIP_CERTS approach with draft-ono-sipping-smime-keyreuse-01.txt to
have one S/MIME encryption based on certificates and use the key-reuse
for the rest of the session. But as I wrote last time, the draft
regarding key-usage has expired  and I'm not sure if it will be updated.

Ciao
	Steffen

> -----Original Message-----
> From: Frank W. Miller [mailto:fwmiller@cornfed.com]=20
> Sent: Saturday, July 22, 2006 3:58 AM
> To: Cullen Jennings
> Cc: sip@ietf.org
> Subject: Re: [Sip] SIP Security mechanism?
>=20
>=20
> Thanks for the pointer.  I have scanned this draft over.
>=20
> The goal of your draft is similar to what I'm targeting but=20
> slightly different.  Please correct me if I am=20
> misunderstanding something.  The draft (which is now=20
> draft-ietf-sip-certs-01 I believe) has the goal is to provide=20
> a way to make the distribution of certificates more automated=20
> by providing specific infrastructure that is based on simple=20
> user actions.  The draft specifies publication and an event=20
> package that allows for notification to interested User=20
> Agents of certificates.
>=20
> The great benefit to this approach is its preservation of=20
> user authentication.
>=20
> My thought was much simpler.  Let me drop in a quote from the=20
> ZRTP draft, draft-zimmermann-avt-zrtp-01.txt:
>=20
> ZRTP is key agreement protocol which performs Diffie-Hellman key
>    exchange during call setup in-band in the Real-time Transport
>    Protocol (RTP) [1] media stream which has been established=20
> using some
>    other signaling protocol such as Session Initiation Protocol (SIP)
>    [11].  This generates a shared secret which is then used=20
> to generate
>    keys and salt for a Secure RTP (SRTP) [2] session.
>=20
> Although it uses
>    a public key algorithm, it does not rely on a public key
>    infrastructure (PKI).
>=20
> Were we to put in place some sort of simple transaction that=20
> did this key exchange before any other signaling, we could=20
> use the keys to do encryption of the message bodies (similar=20
> to what S/MIME is doing) without having to deal with PKI at=20
> all.  Since the main problem with the use of S/MIME to date=20
> has been the certificate management, this approach is also=20
> targeting the problem that your draft is.
>=20
> However, there are a couple of differences.  First, your=20
> draft approach preserves full S/MIME certificate based=20
> authentication, this approach would likely not do that, it=20
> would be to allow for encryption only.  The ZRTP draft=20
> proposes prompting the user with a short authentication=20
> string for the media but I don't know if that would be useful=20
> for the signaling.  Second, this approach would be much=20
> simpler, likely requiring only a couple of messages between=20
> the UA endpoints.  It would not require the centralized=20
> server elements that would be necessary for the=20
> PUBLISH/SUBSCRIBE/NOTIFY components of your draft.
>=20
> Consider the messages used to do key exchange in ZRTP (again=20
> from the ZRTP draft):
>=20
>  Alice                                      Bob
>      |                                         |
>      | Alice and Bob establish a media session.|
>      |                                         |
>      |                   RTP                   |
>      =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
>      |                                         |
>      | Hello (ver,cid,hash,cipher,pkt,sas,Alice's ZID) F1
>      |---------------------------------------->|
>      |                             HelloACK F2 |
>      |<----------------------------------------|
>      | Hello (ver,cid,hash,cipher,pkt,sas,Bob's ZID) F3
>      |<----------------------------------------|
>      | HelloACK F4                             |
>      |---------------------------------------->|
>      |                                         |
>      |        Bob acts as the initiator        |
>      |                                         |
>      |   Commit (Bob's ZID,hash,cipher,pkt,hvi) F5
>      |<----------------------------------------|
>      | DHPart1 (pvr,rs1IDr,rs2IDr,sigsIDr,srtpsIDr,other_secretIDr) F6
>      |---------------------------------------->|
>      | DHPart2 (pvi,rs1IDi,rs2IDi,sigsIDi,ssrtpIDi,other_secretIDi) F7
>      |<----------------------------------------|
>      |                                         |
>      | Alice and Bob generate SRTP session key.|
>      |                                         |
>      |               SRTP begins               |
>      =
|<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
>      |                                         |
>      | Confirm1 (plaintext,sasflag,hmac) F8    |
>      |---------------------------------------->|
>      |    Confirm2 (plaintext,sasflag,hmac) F9 |
>      |<----------------------------------------|
>      | Confirm2AK F10                          |
>      |---------------------------------------->|
>=20
>=20
> They are generating keys for each of media streams here, i.e.=20
> an inband message exchange is happening in each direction. =20
> Something like this could probably be simplified to just two=20
> messages for the signaling, something like:
>=20
>     Bob                             Alice
>      |           HELLO                |
>      |------------------------------->|
>      |           200 OK               |
>      |<-------------------------------|
>      |                                |
>=20
> This example uses a new method I made up called HELLO but I=20
> suppose this could be folded into the INVITE transaction too,=20
> using a mechanism similar to the 407.  The key that Alice is=20
> supposed to use to encrypt the bodies sent to Bob would be in=20
> the HELLO and the key that Bob is supposed to use encrypt the=20
> bodies sent to Alice would be in the 200 OK.
>=20
> Is this clearer?  Does it make sense?  Its simpler but the=20
> authentication may not be there.  That works for the media=20
> because you can possibly recognize someone's voice.  In the=20
> signaling the authentication thing may be a bigger issue.
>=20
> FM
>=20
>=20
> On Fri, 2006-07-21 at 00:17 -0700, Cullen Jennings wrote:
> > you might want to have a read of=20
> draft-ietf-sipping-certs-04 if your=20
> > goal is to make sure that users can find the keys of other=20
> users for=20
> > use with S/MIME
> >=20
> > On Jul 19, 2006, at 12:17 PM, fwmiller@cornfed.com wrote:
> >=20
> > >
> > >
> > > I've been reading about the ZRTP lately.  I read=20
> Henning's comment=20
> > > at the IETF meeting about how S/MIME relies on the=20
> receiver's public=20
> > > key to be available to allow for encryption/decryption to work.
> > >
> > > Why couldnt we put in place a mechanism that did a key=20
> exchange for=20
> > > SIP signaling similar to whats done in ZRTP (i.e. modified Diffie-
> > > Hellman) prior to any signaling traffic, then use the=20
> exchanged keys=20
> > > as the basis for S/MIME encryption of the bodies?  Or we=20
> could even=20
> > > get really carried away and use it to do encryption of the entire=20
> > > SIP message if the signaling did not traverse proxies? =20
> This could=20
> > > be useful for encrypting UDP signaling, again if no proxy were=20
> > > involved.
> > >
> > > Thoughts?
> > > FM
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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From sip-bounces@ietf.org Wed Jul 26 02:54:59 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Brian Rosen" <br@brianrosen.net>, "'Marc Linsner'" <mlinsner@cisco.com>,
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References: <024e01c6b046$b1b36080$640fa8c0@cis.neustar.com>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 08:46:22 +0200
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>I don't really understand why you think location by reference is so much of
> a concern.

Because location by reference involves storage of personal information on a 
server. That suddenly makes all sorts of regulations apply, the points I 
mentioned address only a small part of that. Having an anonymous URL for 
example often makes a difference, because then the information is no longer 
directly linked to a particular person.

When an endpoint inserts location by value, you can reasonably argue that 
opt-in consent requirements (which many countries have) are fulfilled, 
provided the endpoint gives the user control over whether or not location is 
inserted. Same holds for the endpoint inserting a flag asking a proxy to 
fill in the location (by value).
For location-by-reference this is much more fuzzy. You probably don't need a 
lecture on privacy regulations, but e.g. 
http://www.isoc.org/briefings/015/index.shtml (from 2003 but still 
applicable IMO) illustrates the mess.

A while ago I submitted a draft on Temporarily Routable User Agent URIs 
(http://www.ietf.org/internet-drafts/draft-jbemmel-sip-truu-02.txt), which 
provides anonymous, temporary URIs that can be explicitly invalidated. You 
may want to consider something similar for location-by-reference

Regards,

Jeroen 


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From sip-bounces@ietf.org Wed Jul 26 04:19:02 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "IETF SIP List" <sip@ietf.org>,
	"Robert Sparks" <rjsparks@nostrum.com>
References: <C0B78680-67C7-463E-AE3E-2161E23E46C3@nostrum.com>
Subject: Re: [Sip] Update to fork-loop-fix submitted
Date: Wed, 26 Jul 2006 10:16:11 +0200
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>
> http://www.nostrum.com/~rjsparks/draft-ietf-sip-fork-loop-fix-02.txt
>

Robert,

With respect to Open Issue 3 (Proxy-Authorization and Proxy-Require) it is 
worth noting (restating actually, this has been pointed out before 
somewhere) that including them means that loop detection cannot be used when 
forwarding CANCEL or ACK to non-2xx:

- for both, branch must match the INVITE, so outcome of loop hash must match
- Proxy-Require/Proxy-Authorization won't appear in CANCEL or ACK to non-2xx 
(since these are generated hop-by-hop, and also because both are not allowed 
in CANCEL)

So whatever the reasons were for including them, there are reasons for *not* 
including them

Personally, I would like to separate the loop detection hash from the branch 
parameter. Can we not introduce a separate via parameter for this?

Regards,

Jeroen 


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From sip-bounces@ietf.org Wed Jul 26 04:20:11 2006
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Subject: RE: [Sip] Hitchhikers guide - when is it done?
Date: Wed, 26 Jul 2006 11:17:28 +0300
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Because of the overhead and clerical delays involved in the publication
of RFCs I would suggest a refinement to option 2. Provide a reference in
the RFC to a Web page maintained and refreshed by the SIP WG or the RAI
Area including the 'to-the-best-of-our-knowledge' updates in-between two
versions of the RFC.=20

Dan




=20
=20

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Tuesday, July 25, 2006 8:56 PM
> To: IETF SIP List
> Subject: [Sip] Hitchhikers guide - when is it done?
>=20
> During the IETF meeting, I discussed the one current open=20
> issue in the Hitchhikers guide. The issue is: when does the=20
> document finish? When do we send to IESG to issue an RFC?
>=20
> There were two approaches:
>=20
> 1. split the document in two. One part contains only the core=20
> specs, and issues relatively soon. The other part has=20
> everything else, and basically ends up being the last RFC=20
> produced by the SIP WG before it shuts its doors (which has=20
> always been 1.5 years from now for the last 5 years).
>=20
> 2. keep it as one document. Issue an RFC at some point, and=20
> then revise it a few times as needed. We need to decide when=20
> to issue it.
>=20
>=20
> I'd like to propose option 2. I think people get more value=20
> from it as a=20
> consolidated document. Furthermore, I'd propose that we send the=20
> document to IESG for consideration as informational at the=20
> instant all=20
> of the documents listed as 'core' have RFC numbers issued by RFC-ed.
>=20
> Comments?
>=20
> -Jonathan R.
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From: "Marc Linsner" <mlinsner@cisco.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
	"'Andrew Newton'" <andy@hxr.us>, "'Rosen, Brian'" <Brian.Rosen@neustar.biz>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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I have no problem with pidf/pidf-lo/presence.

I have no problem with a ua placing a uri pointing to their presence service
in a header of a sip invite.

Context check:  We're discussing a sip invite (assumed pointed to emergency
service) that arrives at a proxy with zero location information.

I'm the only one that has a problem with any (visited) proxy/LIS creating a
uri that can be de-referenced by anyone without regard to user
security/privacy rules?

-Marc-





> -----Original Message-----
> From: Peterson, Jon [mailto:jon.peterson@neustar.biz] 
> Sent: Tuesday, July 25, 2006 4:42 PM
> To: Andrew Newton; Rosen, Brian
> Cc: sip@ietf.org; geopriv@ietf.org; From sip-bounces@ietf.org Wed Jul 26 08:04:24 2006
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From: "Marc Linsner" <mlinsner@cisco.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
	"'Andrew Newton'" <andy@hxr.us>, "'Rosen, Brian'" <Brian.Rosen@neustar.biz>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Tue, 25 Jul 2006 18:17:23 -0400
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I have no problem with pidf/pidf-lo/presence.

I have no problem with a ua placing a uri pointing to their presence service
in a header of a sip invite.

Context check:  We're discussing a sip invite (assumed pointed to emergency
service) that arrives at a proxy with zero location information.

I'm the only one that has a problem with any (visited) proxy/LIS creating a
uri that can be de-referenced by anyone without regard to user
security/privacy rules?

-Marc-





> -----Original Message-----
> From: Peterson, Jon [mailto:jon.peterson@neustar.biz] 
> Sent: Tuesday, July 25, 2006 4:42 PM
> To: Andrew Newton; Rosen, Brian
> Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to 
> location-conveyance
> 
> 
> As Brian points out, the whole point of using PIDF for 
> location information was the similarity of the early defined 
> use cases to the presence architecture. This certainly 
> included use cases in which one wanted to fetch or subscribe 
> to location information. Those operations require the 
> existence of a reference, and that reference was commonly 
> described as a URI, at the time. Lately, I think the idea of 
> having a URI pointing to location information has become 
> associated with a specific proposal (i.e. HELD), but clearly 
> RFC4079 sketched out an architecture for using presence to 
> acquire location information and it specifically mentions 
> URIs as a way to refer to presence. 
> 
> There is no "hidden magic" in any of this. Subscribing to 
> presence, and using URIs to indicate presence information, is 
> exhaustively specified for SIP. PIDF-LO was designed to reuse 
> all of that work and specifically not to require us to go out 
> and reinvent everything in order to be able to refer to 
> location information out in the network.
> 
> Basically, I support the direction re: bullet 6. I might 
> further suggest that the CID URI scheme be permitted in the 
> Location header, in order to be able to refer to MIME bodies 
> included in SIP messages. I'm also neutral/favorable on using 
> HTTP GET. But I think either of these URI schemes and their 
> usage in the Location header could just as well be described 
> in subsequent documents.
> 
> Jon Peterson
> NeuStar, Inc,
> 
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Tuesday, July 25, 2006 1:05 PM
> > To: Rosen, Brian
> > Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> > Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to 
> > location-conveyance
> > 
> > 
> > For the record, I agree with Brian.
> > 
> > -andy
> > 
> > On Jul 25, 2006, at 3:50 PM, Rosen, Brian wrote:
> > 
> > > When the original PIDF-LO RFC came out, location-by-reference was 
> > > defined.
> > >
> > > You have a URI.  You can exchange the URI for location.  
> There is a 
> > > protocol defined for that purpose (SIP SUBSCRIBE).  That 
> is location 
> > > by reference.  It also serves as the location update mechanism, 
> > > clearly something we need.
> > >
> > > Some day, there may be other location URI references.  We 
> don't need 
> > > to cover them in this draft.  The basic location by 
> reference horse 
> > > has left the barn.  Were you planning on creating an RFC that
> > specifically
> > > disallows a Subscribe on a sip/sips/pres URI to get location?    
> > > THAT WAS
> > > THE POINT OF MAKING IT PART OF A PIDF.
> > >
> > > I do think there is some value in defining a way to express a
> > > location-by-reference aka presence URI which is not just 
> the AoR.   
> > > That
> > > is one reason why I want to define the Location URI as 
> being able to 
> > > contain a sip/sips/pres URI.  The purpose is to allow a UA or a 
> > > proxy to insert location where there is no association between the
> > identity of
> > > the SIP user and the location.
> > >
> > > But that is all I think we need.
> > >
> > > We have a requirement for proxy insertion of location.  If
> > we disallow
> > > location-by-reference (which I think is a good idea in 
> general, but 
> > > futile in practice), then we would need something like the
> > Data URI to
> > > support location insertion by proxy.  If we DO allow 
> > > location-by-reference, then it also serves to meet the location 
> > > insertion by proxy requirement, and we don't need the Data
> > URI.  We're
> > > looking for a use case which requires proxy insertion by 
> value, but 
> > > haven't heard one yet.
> > >
> > > You can make an argument that you can take the From:, 
> assume it's an 
> > > AoR, and SUBSCRIBE to the presence of that AoR to get 
> location, thus 
> > > not needing a Location header.  I really think letting the
> > Location header
> > > specify an alternate presence URMarc Linsner
> Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to 
> location-conveyance
> 
> 
> As Brian points out, the whole point of using PIDF for 
> location information was the similarity of the early defined 
> use cases to the presence architecture. This certainly 
> included use cases in which one wanted to fetch or subscribe 
> to location information. Those operations require the 
> existence of a reference, and that reference was commonly 
> described as a URI, at the time. Lately, I think the idea of 
> having a URI pointing to location information has become 
> associated with a specific proposal (i.e. HELD), but clearly 
> RFC4079 sketched out an architecture for using presence to 
> acquire location information and it specifically mentions 
> URIs as a way to refer to presence. 
> 
> There is no "hidden magic" in any of this. Subscribing to 
> presence, and using URIs to indicate presence information, is 
> exhaustively specified for SIP. PIDF-LO was designed to reuse 
> all of that work and specifically not to require us to go out 
> and reinvent everything in order to be able to refer to 
> location information out in the network.
> 
> Basically, I support the direction re: bullet 6. I might 
> further suggest that the CID URI scheme be permitted in the 
> Location header, in order to be able to refer to MIME bodies 
> included in SIP messages. I'm also neutral/favorable on using 
> HTTP GET. But I think either of these URI schemes and their 
> usage in the Location header could just as well be described 
> in subsequent documents.
> 
> Jon Peterson
> NeuStar, Inc,
> 
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Tuesday, July 25, 2006 1:05 PM
> > To: Rosen, Brian
> > Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> > Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to 
> > location-conveyance
> > 
> > 
> > For the record, I agree with Brian.
> > 
> > -andy
> > 
> > On Jul 25, 2006, at 3:50 PM, Rosen, Brian wrote:
> > 
> > > When the original PIDF-LO RFC came out, location-by-reference was 
> > > defined.
> > >
> > > You have a URI.  You can exchange the URI for location.  
> There is a 
> > > protocol defined for that purpose (SIP SUBSCRIBE).  That 
> is location 
> > > by reference.  It also serves as the location update mechanism, 
> > > clearly something we need.
> > >
> > > Some day, there may be other location URI references.  We 
> don't need 
> > > to cover them in this draft.  The basic location by 
> reference horse 
> > > has left the barn.  Were you planning on creating an RFC that
> > specifically
> > > disallows a Subscribe on a sip/sips/pres URI to get location?    
> > > THAT WAS
> > > THE POINT OF MAKING IT PART OF A PIDF.
> > >
> > > I do think there is some value in defining a way to express a
> > > location-by-reference aka presence URI which is not just 
> the AoR.   
> > > That
> > > is one reason why I want to define the Location URI as 
> being able to 
> > > contain a sip/sips/pres URI.  The purpose is to allow a UA or a 
> > > proxy to insert location where there is no association between the
> > identity of
> > > the SIP user and the location.
> > >
> > > But that is all I think we need.
> > >
> > > We have a requirement for proxy insertion of location.  If
> > we disallow
> > > location-by-reference (which I think is a good idea in 
> general, but 
> > > futile in practice), then we would need something like the
> > Data URI to
> > > support location insertion by proxy.  If we DO allow 
> > > location-by-reference, then it also serves to meet the location 
> > > insertion by proxy requirement, and we don't need the Data
> > URI.  We're
> > > looking for a use case which requires proxy insertion by 
> value, but 
> > > haven't heard one yet.
> > >
> > > You can make an argument that you can take the From:, 
> assume it's an 
> > > AoR, and SUBSCRIBE to the presence of that AoR to get 
> location, thus 
> > > not needing a Location header.  I really think letting the
> > Location header
> > > specify an alternate presence URI is a good idea.
> > >
> > > Brian
> > >
> > >> -----Original Message-----
> > >> From: Marc Linsner [mailto:mlinsner@cisco.com]
> > >> Sent: Tuesday, July 25, 2006 3:29 PM
> > >> To: Rosen, Brian; sip@ietf.org
> > >> Cc: geopriv@ietf.org
> > >> Subject: RE: [Geopriv] Consensus on changes to 
> location-conveyance
> > >>
> > >> In-line.....
> > >>
> > >>> -----Original Message-----
> > >>> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > >>> Sent: Tuesday, July 25, 2006 1:38 PM
> > >>> To: sip@ietf.org
> > >>> Cc: geopriv@ietf.org
> > >>> Subject: [Geopriv] Consensus on changes to location-conveyance
> > >>>
> > >>> I have put out a number of emails on proposed changes 
> to location 
> > >>> conveyance.  Some of them have sparked a number of 
> comments.  My 
> > >>> summary of consensus on these issues is:
> > >>>
> > >>> 1. We proposed to explicitly disallow use of a Data URI, which 
> > >>> would allow specification of a Location-by-value in a header.  
> > >>> There was a lot of list traffic on this, currently in the state 
> > >>> where Keith asked for a real use case for it.
> > >>> So far, we don't see a use case that cannot be provided by 
> > >>> location-by-reference in a header.  If we don't get a good use 
> > >>> case, I think consensus is to disallow location-by-value in a 
> > >>> header.
> > >>>
> > >>
> > >> I see a problem here.  Location-by-reference is nothing more than
> > > 'magic
> > >> happens here'!  Please point to a wg item documenting the 
> > >> location-by-reference mechanism.  How can you state the 
> > >> location-by-reference will cover all the use cases that
> > the Data URI
> > > would
> > >> when location-by-reference hasn't even been defined
> > anywhere?  You're
> > >> loading up the magic with a lot of assumptions.  Location-by- 
> > >> reference
> > > is
> > >> an
> > >> individual submission as a solution to a set of requirements that
> > > haven't
> > >> been fleshed-out in geopriv.
> > >>
> > >> I believe that Henning/Hannes also recognized this 
> issue, hence the 
> > >> suggestion of a Data URI.  Dismissing this suggestion based on an
> > > already
> > >> accepted non-existent mechanism seems like getting the
> > cart before
> > >> the
> > >> horse.
> > >>
> > >> ..........snip
> > >>
> > >>> 6. We proposed to allow sip/sips and pres: in the 
> location header.  
> > >>> I later speculated that if we allow this, it forms a 
> simple way to 
> > >>> do location-by-reference, and we could eliminate the 
> raw HTTP(S) 
> > >>> Get proposal, with the parameter and registry entirely. 
>  There was 
> > >>> some list traffic in favor of this.  Martin Thompson is 
> skeptical 
> > >>> that SIP Presence Subscribe/Notify is actually allowed as a 
> > >>> geopriv "using protocol".  Several of think that it is.  If we 
> > >>> don't get any more comments on this proposal, we will allow 
> > >>> sip/sips and pres, not have any
> > >>> http(s) resolution text, and not propose a parameter and 
> > >>> accompanying registry.
> > >>
> > >> I certainly think it's more prudent to define the dereferencing
> > > mechanism
> > >> prior to making claims about the protocol.
> > >>
> > >> Do I believe location-by-reference will happen?  Probably,
> > but making
> > >> assumptions about the mechanism now is quite premature.
> > >>
> > >> -Marc-
> > >>
> > >>>
> > >>> If this does not reflect your position, please speak up now.
> > >>>
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sipping@ietf.org for new developments on the application of sip
> > 
> > 
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> > 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP ProtocoI is a good idea.
> > >
> > > Brian
> > >
> > >> -----Original Message-----
> > >> From: Marc Linsner [mailto:mlinsner@cisco.com]
> > >> Sent: Tuesday, July 25, 2006 3:29 PM
> > >> To: Rosen, Brian; sip@ietf.org
> > >> Cc: geopriv@ietf.org
> > >> Subject: RE: [Geopriv] Consensus on changes to 
> location-conveyance
> > >>
> > >> In-line.....
> > >>
> > >>> -----Original Message-----
> > >>> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > >>> Sent: Tuesday, July 25, 2006 1:38 PM
> > >>> To: sip@ietf.org
> > >>> Cc: geopriv@ietf.org
> > >>> Subject: [Geopriv] Consensus on changes to location-conveyance
> > >>>
> > >>> I have put out a number of emails on proposed changes 
> to location 
> > >>> conveyance.  Some of them have sparked a number of 
> comments.  My 
> > >>> summary of consensus on these issues is:
> > >>>
> > >>> 1. We proposed to explicitly disallow use of a Data URI, which 
> > >>> would allow specification of a Location-by-value in a header.  
> > >>> There was a lot of list traffic on this, currently in the state 
> > >>> where Keith asked for a real use case for it.
> > >>> So far, we don't see a use case that cannot be provided by 
> > >>> location-by-reference in a header.  If we don't get a good use 
> > >>> case, I think consensus is to disallow location-by-value in a 
> > >>> header.
> > >>>
> > >>
> > >> I see a problem here.  Location-by-reference is nothing more than
> > > 'magic
> > >> happens here'!  Please point to a wg item documenting the 
> > >> location-by-reference mechanism.  How can you state the 
> > >> location-by-reference will cover all the use cases that
> > the Data URI
> > > would
> > >> when location-by-reference hasn't even been defined
> > anywhere?  You're
> > >> loading up the magic with a lot of assumptions.  Location-by- 
> > >> reference
> > > is
> > >> an
> > >> individual submission as a solution to a set of requirements that
> > > haven't
> > >> been fleshed-out in geopriv.
> > >>
> > >> I believe that Henning/Hannes also recognized this 
> issue, hence the 
> > >> suggestion of a Data URI.  Dismissing this suggestion based on an
> > > already
> > >> accepted non-existent mechanism seems like getting the
> > cart before
> > >> the
> > >> horse.
> > >>
> > >> ..........snip
> > >>
> > >>> 6. We proposed to allow sip/sips and pres: in the 
> location header.  
> > >>> I later speculated that if we allow this, it forms a 
> simple way to 
> > >>> do location-by-reference, and we could eliminate the 
> raw HTTP(S) 
> > >>> Get proposal, with the parameter and registry entirely. 
>  There was 
> > >>> some list traffic in favor of this.  Martin Thompson is 
> skeptical 
> > >>> that SIP Presence Subscribe/Notify is actually allowed as a 
> > >>> geopriv "using protocol".  Several of think that it is.  If we 
> > >>> don't get any more comments on this proposal, we will allow 
> > >>> sip/sips and pres, not have any
> > >>> http(s) resolution text, and not propose a parameter and 
> > >>> accompanying registry.
> > >>
> > >> I certainly think it's more prudent to define the dereferencing
> > > mechanism
> > >> prior to making claims about the protocol.
> > >>
> > >> Do I believe location-by-reference will happen?  Probably,
> > but making
> > >> assumptions about the mechanism now is quite premature.
> > >>
> > >> -Marc-
> > >>
> > >>>
> > >>> If this does not reflect your position, please speak up now.
> > >>>
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sipping@ietf.org for new developments on the application of sip
> > 
> > 
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> > 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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In-line..... 

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz] 
> Sent: Tuesday, July 25, 2006 1:38 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: [Geopriv] Consensus on changes to location-conveyance
> 
> I have put out a number of emails on proposed changes to 
> location conveyance.  Some of them have sparked a number of 
> comments.  My summary of consensus on these issues is:
> 
> 1. We proposed to explicitly disallow use of a Data URI, 
> which would allow specification of a Location-by-value in a 
>l
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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In-line..... 

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz] 
> Sent: Tuesday, July 25, 2006 1:38 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: [Geopriv] Consensus on changes to location-conveyance
> 
> I have put out a number of emails on proposed changes to 
> location conveyance.  Some of them have sparked a number of 
> comments.  My summary of consensus on these issues is:
> 
> 1. We proposed to explicitly disallow use of a Data URI, 
> which would allow specification of a Location-by-value in a 
> header.  There was a lot of list traffic on this, currently 
> in the state where Keith asked for a real use case for it.  
> So far, we don't see a use case that cannot be provided by 
> location-by-reference in a header.  If we don't get a good 
> use case, I think consensus is to disallow location-by-value 
> in a header.
> 

I see a problem here.  Location-by-reference is nothing more than 'magic
happens here'!  Please point to a wg item documenting the
location-by-reference mechanism.  How can you state the
location-by-reference will cover all the use cases that the Data URI would
when location-by-reference hasn't even been defined anywhere?  You're
loading up the magic with a lot of assumptions.  Location-by-reference is an
individual submission as a solution to a set of requirements that haven't
been fleshed-out in geopriv.  

I believe that Henning/Hannes also recognized this issue, hence the
suggestion of a Data URI.  Dismissing this suggestion based on an already
accepted non-existent mechanism seems like getting the cart before the
horse.

..........snip

> 6. We proposed to allow sip/sips and pres: in the location 
> header.  I later speculated that if we allow this, it forms a 
> simple way to do location-by-reference, and we could 
> eliminate the raw HTTP(S) Get proposal, with the parameter 
> and registry entirely.  There was some list traffic in favor 
> of this.  Martin Thompson is skeptical that SIP Presence 
> Subscribe/Notify is actually allowed as a geopriv "using 
> protocol".  Several of think that it is.  If we don't get any 
> more comments on this proposal, we will allow sip/sips and 
> pres, not have any
> http(s) resolution text, and not propose a parameter and 
> accompanying registry.

I certainly think it's more prudent to define the dereferencing mechanism
prior to making claims about the protocol.

Do I believe location-by-reference will happen?  Probably, but making
assumptions about the mechanism now is quite premature.

-Marc-

> 
> If this does not reflect your position, please speak up now.
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



 header.  There was a lot of list traffic on this, currently 
> in the state where Keith asked for a real use case for it.  
> So far, we don't see a use case that cannot be provided by 
> location-by-reference in a header.  If we don't get a good 
> use case, I think consensus is to disallow location-by-value 
> in a header.
> 

I see a problem here.  Location-by-reference is nothing more than 'magic
happens here'!  Please point to a wg item documenting the
location-by-reference mechanism.  How can you state the
location-by-reference will cover all the use cases that the Data URI would
when location-by-reference hasn't even been defined anywhere?  You're
loading up the magic with a lot of assumptions.  Location-by-reference is an
individual submission as a solution to a set of requirements that haven't
been fleshed-out in geopriv.  

I believe that Henning/Hannes also recognized this issue, hence the
suggestion of a Data URI.  Dismissing this suggestion based on an already
accepted non-existent mechanism seems like getting the cart before the
horse.

..........snip

> 6. We proposed to allow sip/sips and pres: in the location 
> header.  I later speculated that if we allow this, it forms a 
> simple way to do location-by-reference, and we could 
> eliminate the raw HTTP(S) Get proposal, with the parameter 
> and registry entirely.  There was some list traffic in favor 
> of this.  Martin Thompson is skeptical that SIP Presence 
> Subscribe/Notify is actually allowed as a geopriv "using 
> protocol".  Several of think that it is.  If we don't get any 
> more comments on this proposal, we will allow sip/sips and 
> pres, not have any
> http(s) resolution text, and not propose a parameter and 
> accompanying registry.

I certainly think it's more prudent to define the dereferencing mechanism
prior to making claims about the protocol.

Do I believe location-by-reference will happen?  Probably, but making
assumptions about the mechanism now is quite premature.

-Marc-

> 
> If this does not reflect your position, please speak up now.
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: Ted Hardie <hardie@qualcomm.com>
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At 1:38 PM -0400 7/25/06, Rosen, Brian wrote:
>4. We proposed to explicitly allow hop-by-hop TLS security for location
>when the endpoint wished to allow routing based on location.  We further
>proposed to interpret "Do not Distribute" flags as allowing sending of
>the location to the routing database (LoST for example).  Hannes pointed
>out that there was a proposal to allow two Location headers, one of
>which could be used for routing and another for onward forwarding to the
>recipient.  Hannes' message seemed to agree that "Do not distribute"
>would allow sending of the location to the routing-by-location database,
>but would obligate the proxy that did that to remove the location before
>it forwarded the message.  There were no other messages on this thread.

Actually, I commented on the same problem, though it may have been
connected to one of the related threads; here's the text of that message:

>At 5:23 PM -0400 7/19/06, Brian Rosen wrote:
>>NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU REPLY, SEND IT ONLY TO
>>GEOPRIV
>>However, this opens up the issue of geopriv privacy concerns because the
>>sender does not know the identity of the element (proxy) that will route the
>>call using LoST.  It may be that the UAC itself does this, but it may be
>>some other intermediary.
>>
>>If the identity of the intermediary is not known, then protecting the
>>location information becomes more difficult, because we cannot encrypt using
>>keying material from the intended recipient.
>
>Just to mention this, it appears to also rule out the idea of encrypting
>with the user's key, since there is no way of knowing whether the "intended
>recipient" has that key material either.  Intended recipient is actually
>a bit of a problematic term here, since it's really the on-path network elements
>which consume the information.  They aren't the intended recipient of
>the communication, they are the recipients of the signaling (of which location
>is now a part).
>
>This is a bit confusing because it seems to be possible to include a
>"re-transmission allowed"  rule with a value of "no" (intended to be
>consumed by the "intended recipient" of the underlying communication),
>while still wanting to allow the location to be passed along the path.
>
>>In fact, we would need to use
>>hop-by-hop security, with transitive trust.  This is what the emergency case
>>does, but it also allows no channel security if necessary (usually meaning,
>>try it with TLS and if that fails, try again without TLS).
>
>While I certainly encourage hop-by-hop security, I don't think it really
>meets the geopriv privacy concerns.  I think you  have to have some way of
>expressing "re-transmission for routing permitted" (and likely retention
>limited), since that is a separate statement.
>
>Just typing aloud here, but could we come up with a way to use an
>external ruleset to express this?  With an embedded cid, that would
>not actually require dereferencing.
>					Ted
>



>We conclude that allowing hop-by-hop security for location based routing
>is acceptable, and Do Not Distribute DOES allow forwarding to a routing
>database, but requires that the proxy doing the location based route to
>remove the location it uses for that purpose.

To reiterate this, I do not believe that this is sufficient; I think you have to
have some way of saying "re-transmission for routing permitted" before
you meet the geopriv privacy concerns.

>5. We proposed to create a parameter and a registry to distinguish
>multiple ways to dereference a location where HTTP(S) was the transport.
>Resolution of this issue depends on resolution of issue 6.

I believe that an HTTP GET of a PIDF-LO object is a reasonable baseline.
Saying that other mechanisms should update this document with their
dereferencing mechanisms if they could be confused with the baseline
seems to me to be sufficient.  I'm not sure a registry is needed.  In practice,
I don't think we would hit the interoperability problems often. These days,
extensions to HTTP commonly use different  mime types to indicate the special
handling, and I expect anyone creating a more feature-rich dereferencing
mechanism built on HTTP would not end up using pidf-lo.  

That means the "uri confusion" aspects of the discussion are probably not
that severe.  The only issue is if someone supports only the more feature-rich
alternative and not a baseline GET of the pidf-lo.  I think we should
discourage that, by establishing a real baseline.

Note also that HTTP URIs don't indicate the HTTP Method directly; often they
support multiple methods (e.g. HEAD and GET).

>6. We proposed to allow sip/sips and pres: in the location header.  I
>later speculated that if we allow this, it forms a simple way to do
>location-by-reference, and we could eliminate the raw HTTP(S) Get
>proposal, with the parameter and registry entirely.  There was some list
>traffic in favor of this.  Martin Thompson is skeptical that SIP
>Presence Subscribe/Notify is actually allowed as a geopriv "using
>protocol".  Several of think that it is.  If we don't get any more
>comments on this proposal, we will allow sip/sips and pres, not have any
>http(s) resolution text, and not propose a parameter and accompanying
>registry.

I share some of Martin's skepticism here, but I won't fall on my sword
if the WG goes in this direction.

Thanks for the clear set of issues and opportunity to comment,
				
					Ted Hardie

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Date: Tue, 25 Jul 2006 11:18:43 -0700
To: "Rosen, Brian" <Brian.Rosen@neustar.biz>, <sip@ietf.org>
From: Ted Hardie <hardie@qualcomm.com>
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Subject: [Sip] Re: [Geopriv] Consensus on changes to location-conveyance
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At 1:38 PM -0400 7/25/06, Rosen, Brian wrote:
>4. We proposed to explicitly allow hop-by-hop TLS security for location
>when the endpoint wished to allow routing based on location.  We further
>proposed to interpret "Do not Distribute" flags as allowing sending of
>the location to the routing database (LoST for example).  Hannes pointed
>out that there was a proposal to allow two Location headers, one of
>which could be used for routing and another for onward forwarding to the
>recipient.  Hannes' message seemed to agree that "Do not distribute"
>would allow sending of the location to the routing-by-location database,
>but would obligate the proxy that did that to remove the location before
>it forwarded the message.  There were no other messages on this thread.

Actually, I commented on the same problem, though it may have been
connected to one of the related threads; here's the text of that message:

>At 5:23 PM -0400 7/19/06, Brian Rosen wrote:
>>NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU REPLY, SEND IT ONLY TO
>>GEOPRIV
>>However, this opens up the issue of geopriv privacy concerns because the
>>sender does not know the identity of the element (proxy) that will route the
>>call using LoST.  It may be that the UAC itself does this, but it may be
>>some other intermediary.
>>
>>If the identity of the intermediary is not known, then protecting the
>>location information becomes more difficult, because we cannot encrypt using
>>keying material from the intended recipient.
>
>Just to mention this, it appears to also rule out the idea of encrypting
>with the user's key, since there is no way of knowing whether the "intended
>recipient" has that key material either.  Intended recipient is actually
>a bit of a problematic term here, since it's really the on-path network elements
>which consume the information.  They aren't the intended recipient of
>the communication, they are the recipients of the signaling (of which location
>is now a part).
>
>This is a bit confusing because it seems to be possibllien.qualcomm.com ([129.46.51.59])
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Date: Tue, 25 Jul 2006 13:33:31 -0700
To: "Rosen, Brian" <Brian.Rosen@neustar.biz>, <sip@ietf.org>
From: Ted Hardie <hardie@qualcomm.com>
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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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At 2:45 PM -0400 7/25/06, Rosen, Brian wrote:
>Summarizing, you think hop-by-hop security (TLS) is acceptable when the
>location is sent for the purposes of location based routing, but you are
>not in favor of defining "retransmission=no" to permit sending the
>location to the routing database (e.g. LoST).  You want a more explicit
>permission.

Yes.

>I don't think a cid: with a ruleset is acceptable, as it would not
>permit location by reference added by a proxy to specify this.

Fair enough.



>If this is the consensus, I think we would have to add a bit to the PIDF
>to explicitly control this behavior.

I think an extension that says "retransmission for routing=yes" is the cleanest
way to go.


>Why do you think we must specify an HTTP dereference procedure in this
>document?  What is the purpose when the SIP based method is available?
>We had believed we had to have SOME method of dereferencing to make
>location-by-reference meaningful in the document.  If we have one sip
>based method, we meet the requirement.  Having more options, especially
>completely new HTTP based protocols IN THIS DOCUMENT seems unnecessary
>and unwise.  For example, if we define such a protocol, it becomes a
>geopriv "using protocol" and needs an explicit analysis against the
>geopriv criteria.

I don't think we have to do it in this document, but I read your previous
message as a statement of consensus about the HTTP dereference mechanisms
that have been discussed.   I don't agree with that statement, since I think the
HTTP discussions somewhat over-thought the potential problems.  In particular, a
lot of the discussion was around the idea that http://dereference.example.org/foo-pidf-lo
might mean lots of different things, depending on what http meant.  There are
already ways to deal with that; none of them are perfect (see, for example,
the discussion in draft-reschke-webdav-mount-05.txt and its appendix A for a similar
problem in a different context), but they can work, especially if you set a baseline
approach.

				regards,
					Ted Hardie



>Brian
>
>> -----Original Message-----
>> From: Ted Hardie [mailto:hardie@qualcomm.com]
>> Sent: Tuesday, July 25, 2006 2:19 PM
>> To: Rosen, Brian; sip@ietf.org
>> Cc: geopriv@ietf.org
>> Subject: Re: [Geopriv] Consensus on changes to location-conveyance
>>
>> At 1:38 PM -0400 7/25/06, Rosen, Brian wrote:
>> >4. We proposed to explicitly allow hop-by-hoe to include a
>"re-transmission allowed"  rule with a value of "no" (intended to be
>consumed by the "intended recipient" of the underlying communication),
>while still wanting to allow the location to be passed along the path.
>
>>In fact, we would need to use
>>hop-by-hop security, with transitive trust.  This is what the emergency case
>>does, but it also allows no channel security if necessary (usually meaning,
>>try it with TLS and if that fails, try again without TLS).
>
>While I certainly encourage hop-by-hop security, I don't think it really
>meets the geopriv privacy concerns.  I think you  have to have some way of
>expressing "re-transmission for routing permitted" (and likely retention
>limited), since that is a separate statement.
>
>Just typing aloud here, but could we come up with a way to use an
>external ruleset to express this?  With an embedded cid, that would
>not actually require dereferencing.
>					Ted
>



>We conclude that allowing hop-by-hop security for location based routing
>is acceptable, and Do Not Distribute DOES allow forwarding to a routing
>database, but requires that the proxy doing the location based route to
>remove the location it uses for that purpose.

To reiterate this, I do not believe that this is sufficient; I think you have to
have some way of saying "re-transmission for routing permitted" before
you meet the geopriv privacy concerns.

>5. We proposed to create a parameter and a registry to distinguish
>multiple ways to dereference a location where HTTP(S) was the transport.
>Resolution of this issue depends on resolution of issue 6.

I believe that an HTTP GET of a PIDF-LO object is a reasonable baseline.
Saying that other mechanisms should update this document with their
dereferencing mechanisms if they could be confused with the baseline
seems to me to be sufficient.  I'm not sure a registry is needed.  In practice,
I don't think we would hit the interoperability problems often. These days,
extensions to HTTP commonly use different  mime types to indicate the special
handling, and I expect anyone creating a more feature-rich dereferencing
mechanism built on HTTP would not end up using pidf-lo.  

That means the "uri confusion" aspects of the discussion are probably not
that severe.  The only issue is if someone supports only the more feature-rich
alternative and not a baseline GET of the pidf-lo.  I think we should
discourage that, by establishing a real baseline.

Note also that HTTP URIs don't indicate the HTTP Method directly; often they
support multiple methods (e.g. HEAD and GET).

>6. We proposed to allow sip/sips and pres: in the location header.  I
>later speculated that if we allow this, it forms a simple way to do
>location-by-reference, and we could eliminate the raw HTTP(S) Get
>proposal, with the parameter and registry entirely.  There was some list
>traffic in favor of this.  Martin Thompson is skeptical that SIP
>Presence Subscribe/Notify is actually allowed as a geopriv "using
>protocol".  Several of think that it is.  If we don't get any more
>comments on this proposal, we will allow sip/sips and pres, not have any
>http(s) resolution text, and not propose a parameter and accompanying
>registry.

I share some of Martin's skepticism here, but I won't fall on my sword
if the WG goes in this direction.

Thanks for the clear set of issues and opportunity to comment,
				
					Ted Hardie

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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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At 2:45 PM -0400 7/25/06, Rosen, Brian wrote:
>Summarizing, you think hop-by-hop security (TLS) is acceptable when the
>location is sent for the purposes of location based routing, but you are
>not in favor of defining "retransmission=no" to permit sending the
>location to the routing database (e.g. LoST).  You want a more explicit
>permission.

Yes.

>I don't think a cid: with a ruleset is acceptable, as it would not
>permit location by reference added by a proxy to specify this.

Fair enough.



>If this is the consensus, I think we would have to add a bit to the PIDF
>to explicitly control this behavior.

I think an extension that says "retransmission for routing=yes" is the cleanest
way to go.


>Why do you think we must specify an HTTP dereference procedure in this
>document?  What is the purpose when the SIP based method is available?
>We had believed we had to have SOME method of dereferencing to make
>location-by-reference meaningful in the document.  If we have one sip
>based method, we meet the requirement.  Having more options, especially
>completely new HTTP based protocols IN THIS DOCUMENT seems unnecessary
>and unwise.  For example, if we define such a protocol, it becomes a
>geopriv "using protocol" and needs an explicit analysis against the
>geopriv criteria.

I don't think we have to do it in this document, but I read your previous
message as a statement of consensus about the HTTP dereference mechanisms
that have been discussed.   I don't agree with that statement, since I think the
HTTP discussions somewhat over-thought the potential problems.  In particular, a
lot of the discussion was around the idea that http://dereference.example.org/foo-pidf-lo
might mean lots of different things, depending on what http meant.  There are
already ways to deal with that; none of them are perfect (see, for example,
the discussion in draft-reschke-webdav-mount-05.txt and its appendix A for a similar
problem in a different context), but they can work, especially if you set a baseline
approach.

				regards,
					Ted Hardie



>Brian
>
>> -----Original Message-----
>> From: Ted Hardie [mailto:hardie@qualcomm.com]
>> Sent: Tuesday, July 25, 2006 2:19 PM
>> To: Rosen, Brian; sip@ietf.org
>> Cc: geopriv@ietf.org
>> Subject: Re: [Geopriv] Consensus on changes to location-conveyance
>>
>> At 1:38 PM -0400 7/25/06, Rosen, Brian wrote:
>> >4. We proposed to explicitly allow hop-by-honeeded.  In
>> practice,
>> I don't think we would hit the interoperability problems often. These
>> days,
>> extensions to HTTP commonly use different  mime types to indicate the
>> special
>> handling, and I expect anyone creating a more feature-rich
>dereferencing
>> mechanism built on HTTP would not end up using pidf-lo.
>>
>> That means the "uri confusion" aspects of the discussion are probably
>not
>> that severe.  The only issue is if someone supports only the more
>feature-
>> rich
>> alternative and not a baseline GET of the pidf-lo.  I think we should
>> discourage that, by establishing a real baseline.
>>
>> Note also that HTTP URIs don't indicate the HTTP Method directly;
>often
>> they
>> support multiple methods (e.g. HEAD and GET).
>>
>> >6. We proposed to allow sip/sips and pres: in the location header.  I
>> >later speculated that if we allow this, it forms a simple way to do
>> >location-by-reference, and we could eliminate the raw HTTP(S) Get
>> >proposal, with the parameter and registry entirely.  There was some
>list
>> >traffic in favor of this.  Martin Thompson is skeptical that SIP
>> >Presence Subscribe/Notify is actually allowed as a geopriv "using
>> >protocol".  Several of think that it is.  If we don't get any more
>> >comments on this proposal, we will allow sip/sips and pres, not have
>any
>> >http(s) resolution text, and not propose a parameter and accompanying
>> >registry.
>>
>> I share some of Martin's skepticism here, but I won't fall on my sword
>> if the WG goes in this direction.
>>
>> Thanks for the clear set of issues and opportunity to comment,
>>
>>					Ted Hardie


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





p TLS security for
>location
>> >when the endpoint wished to allow routing based on location.  We
>further
>> >proposed to interpret "Do not Distribute" flags as allowing sending
>of
>> >the location to the routing database (LoST for example).  Hannes
>pointed
>> >out that there was a proposal to allow two Location headers, one of
>> >which could be used for routing and another for onward forwarding to
>the
>> >recipient.  Hannes' message seemed to agree that "Do not distribute"
>> >would allow sending of the location to the routing-by-location
>database,
>> >but would obligate the proxy that did that to remove the location
>before
>> >it forwarded the message.  There were no other messages on this
>thread.
>>
>> Actually, I commented on the same problem, though it may have been
>> connected to one of the related threads; here's the text of that
>message:
>>
>> >At 5:23 PM -0400 7/19/06, Brian Rosen wrote:
>> >>NOTE: THIS MESSAGE IS COPIED TO SIP AND ECRIT, IF YOU REPLY, SEND IT
>> ONLY TO
>> >>GEOPRIV
>> >>However, this opens up the issue of geopriv privacy concerns because
>the
>> >>sender does not know the identity of the element (proxy) that will
>route
>> the
>> >>call using LoST.  It may be that the UAC itself does this, but it
>may be
>> >>some other intermediary.
>> >>
>> >>If the identity of the intermediary is not known, then protecting
>the
>> >>location information becomes more difficult, because we cannot
>encrypt
>> using
>> >>keying material from the intended recipient.
>> >
>> >Just to mention this, it appears to also rule out the idea of
>encrypting
>> >with the user's key, since there is no way of knowing whether the
>> "intended
>> >recipient" has that key material either.  Intended recipient is
>actually
>> >a bit of a problematic term here, since it's really the on-path
>network
>> elements
>> >which consume the information.  They aren't the intended recipient of
>> >the communication, they are the recipients of the signaling (of which
>> location
>> >is now a part).
>> >
>> >This is a bit confusing because it seems to be possible to include a
>> >"re-transmission allowed"  rule with a value of "no" (intended to be
>> >consumed by the "intended recipient" of the underlying
>communication),
>> >while still wanting to allow the location to be passed along the
>path.
>> >
>> >>In fact, we would need to use
>> >>hop-by-hop security, with transitive trust.  This is what the
>emergency
>> case
>> >>does, but it also allows no channel security if necessary (usually
>> meaning,
>> >>try it with TLS and if that fails, try again without TLS).
>> >
>> >While I certainly encourage hop-by-hop security, I don't think it
>really
>> >meets the geopriv privacy concerns.  I think you  have to have some
>way
>> of
>> >expressing "re-transmission for routing permitted" (and likely
>retention
>> >limited), since that is a separate statement.
>> >
>> >Just typing aloud here, but could we come up with a way to use an
>> >external ruleset to express this?  With an embedded cid, that would
>> >not actually require dereferencing.
>> >					Ted
>> >
>>
>>
>>
>> >We conclude that allowing hop-by-hop security for location based
>routing
>> >is acceptable, and Do Not Distribute DOES allow forwarding to a
>routing
>> >database, but requires that the proxy doing the location based route
>to
>> >remove the location it uses for that purpose.
>>
>> To reiterate this, I do not believe that this is sufficient; I think
>you
>> have to
>> have some way of saying "re-transmission for routing permitted" before
>> you meet the geopriv privacy concerns.
>>
>> >5. We proposed to create a parameter and a registry to distinguish
>> >multiple ways to dereference a location where HTTP(S) was the
>transport.
>> >Resolution of this issue depends on resolution of issue 6.
>>
>> I believe that an HTTP GET of a PIDF-LO object is a reasonable
>baseline.
>> Saying that other mechanisms should update this document with their
>> dereferencing mechanisms if they could be confused with the baseline
>> seems to me to be sufficient.  I'm not sure a registry is needed.  In
>> practice,
>> I don't think we would hit the interoperability problems often. These
>> days,
>> extensions to HTTP commonly use different  mime types to indicate the
>> special
>> handling, and I expect anyone creating a more feature-rich
>dereferencing
>> mechanism built on HTTP would not end up using pidf-lo.
>>
>> That means the "uri confusion" aspects of the discussion are probably
>not
>> that severe.  The only issue is if someone supports only the more
>feature-
>> rich
>> alternative and not a baseline GET of the pidf-lo.  I think we should
>> discourage that, by establishing a real baseline.
>>
>> Note also that HTTP URIs don't indicate the HTTP Method directly;
>often
>> they
>> support multiple methods (e.g. HEAD and GET).
>>
>> >6. We proposed to allow sip/sips and pres: in the location header.  I
>> >later speculated that if we allow this, it forms a simple way to do
>> >location-by-reference, and we could eliminate the raw HTTP(S) Get
>> >proposal, with the parameter and registry entirely.  There was some
>list
>> >traffic in favor of this.  Martin Thompson is skeptical that SIP
>> >Presence Subscribe/Notify is actually allowed as a geopriv "using
>> >protocol".  Several of think that it is.  If we don't get any more
>> >comments on this proposal, we will allow sip/sips and pres, not have
>any
>> >http(s) resolution text, and not propose a parameter and accompanying
>> >registry.
>>
>> I share some of Martin's skepticism here, but I won't fall on my sword
>> if the WG goes in this direction.
>>
>> Thanks for the clear set of issues and opportunity to comment,
>>
>>					Ted Hardie


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Date: Tue, 25 Jul 2006 21:39:57 -0400
To: "Rosen, Brian" <Brian.Rosen@neustar.biz>
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Subject: [Sip] Re: [Geopriv] Consensus on changes to location-conveyance
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> I do think there is some value in defining a way to express a
> location-by-reference aka presence URI which is not just the AoR.   
> That
> is one reason why I want to define the Location URI as being able to
> contain a sip/sips/pres URI.  The purpose is to allow a UA or a  
> proxy to
> insert location where there is no association between the identity of
> the SIP user and the location.

I believe that you have to allow non-AOR SUBSCRIBEs if you want the  
system to be practical. AOR SUBSCRIBEs will require authentication of  
the watcher and we seem to agree that this is a no-go. Thus, the only  
reasonable approach is to have a random-user-ID URL that serves as an  
authenticator.

Henning

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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] Consensus on changes to location-conveyance
Date: Tue, 25 Jul 2006 21:36:57 -0400
To: Brian Rosen <br@brianrosen.net>
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> 1. We proposed to explicitly disallow use of a Data URI, which  
> would allow
> specification of a Location-by-value in a header.  There was a lot  
> of list
> traffic on this, currently in the state where Keith asked for a  
> real use
> case for it.  So far, we don't see a use case that cannot be  
> provided by
> location-by-reference in a header.  If we don't get a good use  
> case, I think
> consensus is to disallow location-by-value in a header.
>

I repeat what I said before: There is a clear advantage to this  
method in terms of simplicity of operation and reliability. Thus,  
prohibiting it has no technical justification.

In addition, when outbound proxies are behind NATs, deploying a  
reference-based mechanism is far more painful and certainly less  
reliable since that proxy now has to put a LIS outside the firewall/ 
proxy. This is likely to be common in small and medium enterprise  
scenarios, where the enterprise, behind a NAT, has a nailed-up  
connection to some SIP service provider. This also happens to be the  
scenario where proxy-based location insertion is most likely since it  
works reasonably well with network-in-box solutions that combine  
access router, DHCP server and SIP server.


> other messages on this thread.  We conclude that allowing hop-by-hop
> security for location based routing is acceptable, and Do Not  
> Distribute
> DOES allow forwarding to a routing database, but requires that the  
> proxy
> doing the location based route to remove the location it uses for that
> purpose.
>

This only applies if there are two locations or a single DND location?


> 6. We proposed to allow sip/sips and pres: in the location header.   
> I later
> speculated that if we allow this, it forms a simple way to do
> location-by-reference, and we could eliminate the raw HTTP(S) Get  
> proposal,
> with the parameter and registry entirely.  There was some list  
> traffic in
> favor of this.  Martin Thompson is skeptical that SIP Presence
> Subscribe/Notify is actually allowed as a geopriv "using  
> protocol".  Several
> of think that it is.  If we don't get any more comments on this  
> proposal, we
> will allow sip/sips and pres, not have any http(s) resolution text,  
> and not
> propose a parameter and accompanying registry.
>

I'm concerned about the implementation complexity of this approach in  
proxies. A SUBSCRIBE/NOTIFY adds another asynchronous operation to  
the time-critical forwarding path. This is also significantly more  
complex for proxies to implement than any of the other mechanisms,  
since they have to correlate incoming NOTIFY arrivals with pending  
messages, maintain multiple timers and worry about denial-of-service  
attacks. (A state exhaustion attack in a proxy would simply have the  
NOTIFY for the SUBSCRIBE take a few minutes, thus requiring the proxy  
to keep state for that duration. An attacker in an HTTP-based system  
can pull a similar trick, but it is somewhat easier to bound the  
permissible delay since it is a one-step operation.)

Needless to say, I believe that the security considerations section  
needs to discuss this.

Based on our measurements, the NOTIFY and SUBSCRIBE probably will  
each take as much effort as handling the INVITE, so that the call  
routing throughput of an emergency service routing proxy will be  
reduced by about 2/3. The same is true, to a somewhat lesser extent,  
for HTTP retrievals and to a larger extent for HTTPS retrievals.  
Thus, either solution is likely to be much more expensive than a data  
URL for the proxy-inserted case.


> If this does not reflect your position, please speak up now.
>
> Brian


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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 09:01:57 -0400
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In-line... 

> 
> So, you are arguing against proxy insertion of a 
> location-by-reference?

No, if done properly....but reading the latest threads, I get the feeling
that www.whereintheworldisbrianrosen.com is an acceptable reference.

> 
> You don't have a problem with UA insertion of 
> location-by-reference in a Location header?

This assumes the user's presence service (contracted by the user) will do
the right things to protect the privacy/security of the location
information.

> 
> Agree that Location insertion by proxy could assume an INVITE 
> arriving at a proxy without any location.

I suspect there will be many cases where proxy will supplant whatever is
supplied by the user with proxy derived data.

> 
> I've always had a problem with proxy insertion of location 
> because I am worried that proxies will do this without regard 
> to user's wishes.  In an emergency case, however, there are 
> no such worries.  I don't see any difference between by-value 
> and by-reference.

Simplicity?  Call/dispatching failures from dereferencing problems?  Since
emergency calling is the use case driving this, why not opt for less
complexity with a better chance of actually working?

What seems odd is to dismiss L-by-V in the header simply because the L-by-R
has been in this draft for so long.  That is the way I read Keith's message.
Yes, if the stars and moon are all aligned, L-by-R will work, hence
obviating the need for an 'overlapping' L-by-V.  But, if it's a cloudy day,
the complexity associated with dereferencing WILL cause failures.

> 
> Proxies may disregard advice we supply, but Location URIs in 
> Location headers probably should not have any user 
> identification information in either the URI itself, or the 
> resulting PIDF.  If such information is desired, use the AoR.
> 
> De-referencing should always be subject to user rulesets, and 
> would not be any different than by-value except the obvious 
> issue that the owners of the proxy may not provide the user 
> with the facilities to control the ruleset, my original worry.

My worry also.  
  
> 
> Would you not agree that if a proxy is allowed to insert 
> location-by-value, the concerns would be the same?

As pointed out by others, by-value is a moment in time, by-reference makes
current information available for a yet-to-be-specified period.


-Marc-

> 
> Brian
> 
> 
> 
> > -----Original Message-----
> > From: Marc Linsner [mailto:mlinsner@cisco.com]
> > Sent: Tuesday, July 25, 2006 6:17 PM
> > To: 'Peterson, Jon'; 'Andrew Newton'; 'Rosen, Brian'
> > Cc: sip@ietf.org; geopriv@ietf.org
> > Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location- 
> > conveyance
> > 
> > I have no problem with pidf/pidf-lo/presence.
> > 
> > I have no problem with a ua placing a uri pointing to their 
> presence 
> > service in a header of a sip invite.
> > 
> > Context check:  We're discussing a sip invite (assumed pointed to 
> > emergency
> > service) that arrives at a proxy with zero location information.
> > 
> > I'm the only one that has a problem with any (visited) proxy/LIS 
> > creating a uri that can be de-referenced by anyone without 
> regard to 
> > user security/privacy rules?
> > 
> > -Marc-
> > 
> > 
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> > > Sent: Tuesday, July 25, 2006 4:42 PM
> > > To: Andrew Newton; Rosen, Brian
> > > Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> > > Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to 
> > > location-conveyance
> > >
> > >
> > > As Brian points out, the whole point of using PIDF for location 
> > > information was the similarity of the early defined use 
> cases to the 
> > > presence architecture. This certainly included use cases in which 
> > > one wanted to fetch or subscribe to location information. Those 
> > > operations require the existence of a reference, and that 
> reference 
> > > was commonly described as a URI, at the time. Lately, I think the 
> > > idea of having a URI pointing to location information has become 
> > > associated with a specific proposal (i.e. HELD), but clearly
> > > RFC4079 sketched out an architecture for using presence 
> to acquire 
> > > location information and it specifically mentions URIs as 
> a way to 
> > > refer to presence.
> > >
> > > There is no "hidden magic" in any of this. Subscribing to 
> presence, 
> > > and using URIs to indicate presence information, is exhaustively 
> > > specified for SIP. PIDF-LO was designed to reuse all of that work 
> > > and specifically not to require us to go out and reinvent 
> everything 
> > > in order to be able to refer to location information out in the 
> > > network.
> > >
> > > Basically, I support the direction re: bullet 6. I might further 
> > > suggest that the CID URI scheme be permitted in the 
> Location header, 
> > > in order to be able to refer to MIME bodies included in SIP 
> > > messages. I'm also neutral/favorable on using HTTP GET. 
> But I think 
> > > either of these URI schemes and their usage in the 
> Location header 
> > > could just as well be described in subsequent documents.
> > >
> > > Jon Peterson
> > > NeuStar, Inc,
> > >
> > > > -----Original Message-----
> > > > From: Andrew Newton [mailto:andy@hxr.us]
> > > > Sent: Tuesday, July 25, 2006 1:05 PM
> > > > To: Rosen, Brian
> > > > Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
> > > > Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to 
> > > > location-conveyance
> > > >
> > > >
> > > > For the record, I agree with Brian.
> > > >
> > > > -andy
> > > >
> > > > On Jul 25, 2006, at 3:50 PM, Rosen, Brian wrote:
> > > >
> > > > > When the original PIDF-LO RFC came out, location-by-reference 
> > > > > was defined.
> > > > >
> > > > > You have a URI.  You can exchange the URI for location.
> > > There is a
> > > > > protocol defined for that purpose (SIP SUBSCRIBE).  That
> > > is location
> > > > > by reference.  It also serves as the location update 
> mechanism, 
> > > > > clearly something we need.
> > > > >
> > > > > Some day, there may be other location URI references.  We
> > > don't need
> > > > > to cover them in this draft.  The basic location by
> > > reference horse
> > > > > has left the barn.  Were you planning on creating an RFC that
> > > > specifically
> > > > > disallows a Subscribe on a sip/sips/pres URI to get location?
> > > > > THAT WAS
> > > > > THE POINT OF MAKING IT PART OF A PIDF.
> > > > >
> > > > > I do think there is some value in defining a way to express a 
> > > > > location-by-reference aka presence URI which is not just
> > > the AoR.
> > > > > That
> > > > > is one reason why I want to define the Location URI as
> > > being able to
> > > > > contain a sip/sips/pres URI.  The purpose is to allow 
> a UA or a 
> > > > > proxy to insert location where there is no 
> association between 
> > > > > the
> > > > identity of
> > > > > the SIP user and the location.
> > > > >
> > > > > But that is all I think we need.
> > > > >
> > > > > We have a requirement for proxy insertion of location.  If
> > > > we disallow
> > > > > location-by-reference (which I think is a good idea in
> > > general, but
> > > > > futile in practice), then we would need something like the
> > > > Data URI to
> > > > > support location insertion by proxy.  If we DO allow 
> > > > > location-by-reference, then it also serves to meet 
> the location 
> > > > > insertion by proxy requirement, and we don't need the Data
> > > > URI.  We're
> > > > > looking for a use case which requires proxy insertion by
> > > value, but
> > > > > haven't heard one yet.
> > > > >
> > > > > You can make an argument that you can take the From:,
> > > assume it's an
> > > > > AoR, and SUBSCRIBE to the presence of that AoR to get
> > > location, thus
> > > > > not needing a Location header.  I really think letting the
> > > > Location header
> > > > > specify an alternate presence URI is a good idea.
> > > > >
> > > > > Brian
> > > > >
> > > > >> -----Original Message-----
> > > > >> From: Marc Linsner [mailto:mlinsner@cisco.com]
> > > > >> Sent: Tuesday, July 25, 2006 3:29 PM
> > > > >> To: Rosen, Brian; sip@ietf.org
> > > > >> Cc: geopriv@ietf.org
> > > > >> Subject: RE: [Geopriv] Consensus on changes to
> > > location-conveyance
> > > > >>
> > > > >> In-line.....
> > > > >>
> > > > >>> -----Original Message-----
> > > > >>> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > >>> Sent: Tuesday, July 25, 2006 1:38 PM
> > > > >>> To: sip@ietf.org
> > > > >>> Cc: geopriv@ietf.org
> > > > >>> Subject: [Geopriv] Consensus on changes to 
> location-conveyance
> > > > >>>
> > > > >>> I have put out a number of emails on proposed changes
> > > to location
> > > > >>> conveyance.  Some of them have sparked a number of
> > > comments.  My
> > > > >>> summary of consensus on these issues is:
> > > > >>>
> > > > >>> 1. We proposed to explicitly disallow use of a Data 
> URI, which 
> > > > >>> would allow specification of a Location-by-value in 
> a header.
> > > > >>> There was a lot of list traffic on this, currently in the 
> > > > >>> state where Keith asked for a real use case for it.
> > > > >>> So far, we don't see a use case that cannot be provided by 
> > > > >>> location-by-reference in a header.  If we don't get 
> a good use 
> > > > >>> case, I think consensus is to disallow 
> location-by-value in a 
> > > > >>> header.
> > > > >>>
> > > > >>
> > > > >> I see a problem here.  Location-by-reference is nothing more 
> > > > >> than
> > > > > 'magic
> > > > >> happens here'!  Please point to a wg item documenting the 
> > > > >> location-by-reference mechanism.  How can you state the 
> > > > >> location-by-reference will cover all the use cases that
> > > > the Data URI
> > > > > would
> > > > >> when location-by-reference hasn't even been defined
> > > > anywhere?  You're
> > > > >> loading up the magic with a lot of assumptions.  
> Location-by- 
> > > > >> reference
> > > > > is
> > > > >> an
> > > > >> individual submission as a solution to a set of requirements 
> > > > >> that
> > > > > haven't
> > > > >> been fleshed-out in geopriv.
> > > > >>
> > > > >> I believe that Henning/Hannes also recognized this
> > > issue, hence the
> > > > >> suggestion of a Data URI.  Dismissing this 
> suggestion based on 
> > > > >> an
> > > > > already
> > > > >> accepted non-existent mechanism seems like getting the
> > > > cart before
> > > > >> the
> > > > >> horse.
> > > > >>
> > > > >> ..........snip
> > > > >>
> > > > >>> 6. We proposed to allow sip/sips and pres: in the
> > > location header.
> > > > >>> I later speculated that if we allow this, it forms a
> > > simple way to
> > > > >>> do location-by-reference, and we could eliminate the
> > > raw HTTP(S)
> > > > >>> Get proposal, with the parameter and registry entirely.
> > >  There was
> > > > >>> some list traffic in favor of this.  Martin Thompson is
> > > skeptical
> > > > >>> that SIP Presence Subscribe/Notify is actually allowed as a 
> > > > >>> geopriv "using protocol".  Several of think that it 
> is.  If we 
> > > > >>> don't get any more comments on this proposal, we will allow 
> > > > >>> sip/sips and pres, not have any
> > > > >>> http(s) resolution text, and not propose a parameter and 
> > > > >>> accompanying registry.
> > > > >>
> > > > >> I certainly think it's more prudent to define the 
> dereferencing
> > > > > mechanism
> > > > >> prior to making claims about the protocol.
> > > > >>
> > > > >> Do I believe location-by-reference will happen?  Probably,
> > > > but making
> > > > >> assumptions about the mechanism now is quite premature.
> > > > >>
> > > > >> -Marc-
> > > > >>
> > > > >>>
> > > > >>> If this does not reflect your position, please speak up now.
> > > > >>>
> > > > >
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol Use 
> > > > > sip-implementors@cs.columbia.edu for questions on current sip 
> > > > > Use sipping@ietf.org for new developments on the 
> application of 
> > > > > sip
> > > >
> > > >
> > > > _______________________________________________
> > > > Geopriv mailing list
> > > > Geopriv@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > >
> > 
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv

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Marc Linsner wrote:
> In-line... 
> 
>> So, you are arguing against proxy insertion of a 
>> location-by-reference?
> 
> No, if done properly....but reading the latest threads, I get the feeling
> that www.whereintheworldisbrianrosen.com is an acceptable reference.

Damn! I was hoping, but www.whereintheworldisbrianrosen.com doesn't seem 
to be working today. :-(

	Paul

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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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I'm in (on?) Mars today, but the ISP doesn't seem to be responding

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Wednesday, July 26, 2006 9:11 AM
> To: Marc Linsner
> Cc: 'Brian Rosen'; Peterson, Jon; 'Andrew Newton'; Rosen, Brian;
> sip@ietf.org; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
>=20
>=20
>=20
> Marc Linsner wrote:
> > In-line...
> >
> >> So, you are arguing against proxy insertion of a
> >> location-by-reference?
> >
> > No, if done properly....but reading the latest threads, I get the
> feeling
> > that www.whereintheworldisbrianrosen.com is an acceptable reference.
>=20
> Damn! I was hoping, but www.whereintheworldisbrianrosen.com doesn't
seem
> to be working today. :-(
>=20
> 	Paul

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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 09:56:14 -0400
To: Andrew Newton <andy@hxr.us>
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I think these are reasonable guidelines unless we're suddenly very  
selective about when privacy applies (or limit l-by-r to emergency  
calling in certain no-consent-required jurisdictions).

Specifying the precise 'limited amount of time' is probably something  
that local regulations need to address.

On Jul 25, 2006, at 8:17 PM, Andrew Newton wrote:

>
> On Jul 25, 2006, at 7:21 PM, Jeroen van Bemmel wrote:
>> For location-by-reference, you'd probably want some additional  
>> requirements:
>> - URL must be valid for a limited amount of time
>> - URL must be cryptographically hard to guess
>> - URL must not contain any information that identifies the user /  
>> device / AoR
>> - for whatever transport protocol is used: response must be marked  
>> as 'no cache'
>> - user must be able to remove the information at the URL, ie  
>> explicit invalidation
>> - user must be able to verify correctness of the issued  
>> information (ie user can access the URL himself)
>> - user must be able to control who accesses the URL, both upfront  
>> and history of accesses (for a reasonable period)
>> - there must be explicit consent before a proxy would insert user  
>> location
>
>
>
> Just thinking out loud here, but maybe the rule should be that  
> proxies never append a Location header except for emergencies.

I think that's desirable unless there's a clear indication by the  
caller to do otherwise, by some consent mechanism. However, by the  
time you implement the consent mechanism, you've probably already  
implemented the better UAC-based mechanism, so it seems hard to argue  
for that complexity. The only technical argument I've heard for proxy  
insertion is legacy equipment, where requiring adding consent flags  
is clearly a non-starter.


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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 09:59:42 -0400
To: Brian Rosen <br@brianrosen.net>
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> Proxy insertion of location, whether by value or by reference has  
> the threat
> of not being as subject to user control, but I think in every case  
> you have
> to trust your provider to do as you wish, whether it is your presence
> provider or your location provider independent of your presence  
> provider,
> and proxy insertion doesn't fundamentally alter that.  I worry more  
> about
> it, but I don't objectively think it's much different.

It is slightly different: For the presence provider, I can fairly  
easily verify whether the presence provider does what I tell him to  
do (unless the presence provider is clearly violating my privacy  
rules, which is probably subject to legal breach-of-contract recourse  
in most jurisdictions). This is much harder to do with proxy insertion.

> sufficiently, and we have RFCs that permit it.  Adding restrictions  
> on the
>

The only one that permits is the presence subscription model, which  
gives the target/rule maker a lot of control of who can do what with  
my location data. I can, on a person-by-person basis, restrict access  
to that data. If my proxy inserts some location object or reference  
to it that it made up, I don't even control the privacy rules in the  
object returned.


>
> I kind of like the idea that the proxy inserts the header in the  
> response so
> the UAC can see it.  What do others think of that?
>

This is probably a good idea, but imposes additional restrictions on  
the longevity of the reference. How long should I be able to "see"  
it? We don't want LIS to keep the location data around any longer  
than is needed, but there are no expectations set of what's  
reasonable. By-value doesn't suffer from these problems.

Henning

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To: Jeroen van Bemmel <jbemmel@zonnet.nl>
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Jeroen van Bemmel wrote:
> Robert,
> 
> With respect to Open Issue 3 (Proxy-Authorization and Proxy-Require)
> it is worth noting (restating actually, this has been pointed out 
> before somewhere) that including them means that loop detection 
> cannot be used when forwarding CANCEL or ACK to non-2xx:
> 
> - for both, branch must match the INVITE, so outcome of loop hash 
> must match
> 
> - Proxy-Require/Proxy-Authorization won't appear in CANCEL or ACK to
> non-2xx (since these are generated hop-by-hop, and also because both
> are not allowed in CANCEL)

Why would you want to do a loop detection for CANCELs and ACKs
to non-2xx?  In the latter case, you can't send a response anyway.
As long as the branch parameter for CANCELs and ACKs to non-2xx
matches a pending INVITE transaction, the state machines can be
updated accordingly.

> So whatever the reasons were for including them, there are reasons 
> for *not* including them

Looking at the notes from the WG, it is not apparent we know now
why we had incldued these in the first place.  FWIW, in our
loop detection code, we do not include these and the interop that
we have done with loop detection has worked without any problems.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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From sip-bounces@ietf.org Wed Jul 26 10:12:29 2006
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Henning Schulzrinne wrote:
> On Jul 25, 2006, at 8:17 PM, Andrew Newton wrote:
>> Just thinking out loud here, but maybe the rule should be that proxies 
>> never append a Location header except for emergencies.
> 
> I think that's desirable unless there's a clear indication by the caller 
> to do otherwise, by some consent mechanism. However, by the time you 
> implement the consent mechanism, you've probably already implemented the 
> better UAC-based mechanism, so it seems hard to argue for that 
> complexity. The only technical argument I've heard for proxy insertion 
> is legacy equipment, where requiring adding consent flags is clearly a 
> non-starter.

I believe there are two technical arguments: one is legacy equipment 
(the issue that I'll be facing for quite some time), and environments 
where the location is generated by the network.  For the latter, 
insertion of the Location header only in emergency situations still puts 
the user in control for non-emergency situations.

-andy


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From sip-bounces@ietf.org Wed Jul 26 11:06:41 2006
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 11:04:20 -0400
To: Andrew Newton <andy@hxr.us>
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> I believe there are two technical arguments: one is legacy  
> equipment (the issue that I'll be facing for quite some time), and  
> environments where the location is generated by the network.  For  
> the latter, insertion of the Location header only in emergency  
> situations still puts the user in control for non-emergency  
> situations.
>
>

For the latter, the combination of the in-progress L7 protocols, DHCP  
and LLDP should presumably allow the UAC to obtain location and  
insert it by value. Is there a use case not covered by that?


> -andy


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From sip-bounces@ietf.org Wed Jul 26 11:09:36 2006
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Subject: [Sip] Re: [Geopriv] Consensus on changes to location-conveyance
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> Why do you think we must specify an HTTP dereference procedure in this
> document?  What is the purpose when the SIP based method is available?

Unless you care about performance. I suspect that the SIP approach  
will take at least twice as many server resources as the HTTP  
approach (which will, in turn, take twice as much server capacity as  
the by-value approach). Strangely, ease of implementation and speed  
don't seem to matter in this discussion, so maybe that's a non-issue.

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Henning Schulzrinne wrote:
>> I believe there are two technical arguments: one is legacy equipment 
>> (the issue that I'll be facing for quite some time), and environments 
>> where the location is generated by the network.  For the latter, 
>> insertion of the Location header only in emergency situations still 
>> puts the user in control for non-emergency situations.
>>
>>
> 
> For the latter, the combination of the in-progress L7 protocols, DHCP 
> and LLDP should presumably allow the UAC to obtain location and insert 
> it by value. Is there a use case not covered by that?

As I said, where location comes from the network.  In this scenario, 
there are two possibilities, both up to the user: 1) the UAC subscribes 
to the location from the network and puts it by value in the call 
flow... meaning all location updates go through the UAC, or 2) the UAC 
hands out a reference so updates can be independent of the UAC or 
updates based on subscription come from the UAC but are independent of 
the call.

-andy


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The problem with any kind of website reference is that they are never 
'official'. There are countless web sites enumerating SIP related 
documents. What makes something an official output of the IETF is that 
its an RFC.

-Jonathan R.

Romascanu, Dan (Dan) wrote:

> Because of the overhead and clerical delays involved in the publication
> of RFCs I would suggest a refinement to option 2. Provide a reference in
> the RFC to a Web page maintained and refreshed by the SIP WG or the RAI
> Area including the 'to-the-best-of-our-knowledge' updates in-between two
> versions of the RFC. 
> 
> Dan
> 
> 
> 
> 
>  
>  
> 
> 
>>-----Original Message-----
>>From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
>>Sent: Tuesday, July 25, 2006 8:56 PM
>>To: IETF SIP List
>>Subject: [Sip] Hitchhikers guide - when is it done?
>>
>>During the IETF meeting, I discussed the one current open 
>>issue in the Hitchhikers guide. The issue is: when does the 
>>document finish? When do we send to IESG to issue an RFC?
>>
>>There were two approaches:
>>
>>1. split the document in two. One part contains only the core 
>>specs, and issues relatively soon. The other part has 
>>everything else, and basically ends up being the last RFC 
>>produced by the SIP WG before it shuts its doors (which has 
>>always been 1.5 years from now for the last 5 years).
>>
>>2. keep it as one document. Issue an RFC at some point, and 
>>then revise it a few times as needed. We need to decide when 
>>to issue it.
>>
>>
>>I'd like to propose option 2. I think people get more value 
>>from it as a 
>>consolidated document. Furthermore, I'd propose that we send the 
>>document to IESG for consideration as informational at the 
>>instant all 
>>of the documents listed as 'core' have RFC numbers issued by RFC-ed.
>>
>>Comments?
>>
>>-Jonathan R.
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>>Cisco Fellow                                   Parsippany, NJ 
>>07054-2711
>>Cisco Systems
>>jdrosen@cisco.com                              FAX:   (973) 952-5050
>>http://www.jdrosen.net                         PHONE: (973) 952-5000
>>http://www.cisco.com
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Wed Jul 26 11:36:41 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>, "Andrew Newton" <andy@hxr.us>
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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 17:34:51 +0200
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>> I believe there are two technical arguments: one is legacy
>> equipment (the issue that I'll be facing for quite some time), and 
>> environments where the location is generated by the network.  For  the 
>> latter, insertion of the Location header only in emergency  situations 
>> still puts the user in control for non-emergency  situations.
>>
>>
>
> For the latter, the combination of the in-progress L7 protocols, DHCP  and 
> LLDP should presumably allow the UAC to obtain location and  insert it by 
> value. Is there a use case not covered by that?

With the current proposals (location by value in body): the use case for a 
location based routing service, where my location does not get passed to the 
selected targets

It is possible to do a redirect-based callflow, but that has several 
drawbacks:
- more complex, requires extensions beyond basic RFC3261
- inefficient signalling
- adds delay
- no parallel forking possible (though some would call this a benefit)

The data: URI in combination with a 'Proxy-Require: remove-location' flag 
would solve all these nicely

Regards,

Jeroen

PS I suppose one could put PIDF-LO in a SAML artifact. At the very least, 
http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-00.txt section 7.1 
and http://docs.oasis-open.org/security/saml/v2.0/saml-bindings-2.0-os.pdf 
may provide some ideas for a HTTP-based location-by-reference profile


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From sip-bounces@ietf.org Wed Jul 26 12:02:27 2006
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Jeroen van Bemmel wrote:
> 
>>> I believe there are two technical arguments: one is legacy
>>> equipment (the issue that I'll be facing for quite some time), and 
>>> environments where the location is generated by the network.  For  
>>> the latter, insertion of the Location header only in emergency  
>>> situations still puts the user in control for non-emergency  situations.
>>>
>>>
>>
>> For the latter, the combination of the in-progress L7 protocols, DHCP  
>> and LLDP should presumably allow the UAC to obtain location and  
>> insert it by value. Is there a use case not covered by that?
> 
> With the current proposals (location by value in body): the use case for 
> a location based routing service, where my location does not get passed 
> to the selected targets
> 
> It is possible to do a redirect-based callflow, but that has several 
> drawbacks:
> - more complex, requires extensions beyond basic RFC3261
> - inefficient signalling
> - adds delay
> - no parallel forking possible (though some would call this a benefit)
> 
> The data: URI in combination with a 'Proxy-Require: remove-location' 
> flag would solve all these nicely

So now we are talking about the data: URI being used in non-emergency 
cases?  Certainly if a UAC can put a data: URI in an INVITE, it can also 
put a PIDF-LO in a MIME part.

With regard to the data: URI, how does that work with a CID URI in the 
ruleset-reference?  I would guess that in the emergency case, there 
wouldn't be one.  But in the non-emergency case....

-andy


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> With regard to the data: URI, how does that work with a CID URI in the 
> ruleset-reference?  I would guess that in the emergency case, there 
> wouldn't be one.  But in the non-emergency case....

I'm afraid I don't understand the issue. The ruleset reference wouldn't 
seem to be affected by where and how it is carried. Can you explain?

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From sip-bounces@ietf.org Wed Jul 26 13:47:41 2006
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Henning Schulzrinne wrote:
> 
>> With regard to the data: URI, how does that work with a CID URI in the 
>> ruleset-reference?  I would guess that in the emergency case, there 
>> wouldn't be one.  But in the non-emergency case....
> 
> I'm afraid I don't understand the issue. The ruleset reference wouldn't 
> seem to be affected by where and how it is carried. Can you explain?

So are saying there will be a likely scenario where a ruleset-reference 
in a PIDF-LO encoded into a data: URI and inserted by a proxy will 
reference a MIME part that already exists?

-andy


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At 10:34 PM 7/24/2006 +0200, Drage, Keith \(Keith\) wrote:
>Well all I can say is that it has been there from the
>draft-ietf-sip-location-conveyance-01 version (dated July 17th 2005)
>which was explicitly presented in the Paris IETF#63 SIP WG meeting in
>August 2005.
>
>Unfortunately the minutes are not particularly helpful in terms of WG
>discussion at that time.

this is because there was a multi-mike free-for-all in the suggestions, 
comments and attacks on the presenter   :-/   I'm not surprised the note 
taker didn't keep up

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> So are saying there will be a likely scenario where a ruleset-reference 
> in a PIDF-LO encoded into a data: URI and inserted by a proxy will 
> reference a MIME part that already exists?

We may be talking past each other. Currently, RFC 4119 has

  <xs:complexType name="locPolicyType">
    <xs:sequence>
      <xs:element name="retransmission-allowed" type="xs:boolean"
         minOccurs="0" maxOccurs="1"/>
      <xs:element name="retention-expiry" type="xs:dateTime"
         minOccurs="0" maxOccurs="1"/>
      <xs:element name="external-ruleset" type="xs:anyURI"
         minOccurs="0" maxOccurs="1"/>
      <xs:element name="note-well" type="tns:notewell"
         minOccurs="0" maxOccurs="1"/>
      <xs:any namespace="##other" processContents="lax" minOccurs="0"
         maxOccurs="unbounded"/>
    </xs:sequence>
   </xs:complexType>

I assume you're talking about the external-ruleset element. Regardless 
of how the rest of the GEOPRIV object is carried, this can point 
anywhere, including some static web object, a cid, or even be another 
data URL if you want to carry the policy along with the actual location 
information.

Clearly, a cid would be rather implausible if the URL was inserted by 
the proxy.

If this is location by reference, the options for the URL would be the 
same, except for cid.

If this is a location object carried in the body, it could be any of the 
  things I mentioned above. The same is true for the Location-by-value 
option.

As you can probably tell, I'm a bit confused as to why this is an issue, 
so I'm probably belaboring what's obvious to you.

> 
> -andy

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=20
I don't agree that there is a consensus to preclude the use of HTTP.=20
I would favor not placing your proposed restriction on the URI scheme
for expressing a location reference in the SIP location conveyance
draft; or at least including HTTP as a valid location-reference URI
type.

Regards,
Nadine Abbott

-----Original Message-----
From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]=20
Sent: Tuesday, July 25, 2006 1:38 PM
To: sip@ietf.org
Cc: geopriv@ietf.org
Subject: [Geopriv] Consensus on changes to location-conveyance

I have put out a number of emails on proposed changes to location
conveyance.  Some of them have sparked a number of comments.  My summary
of consensus on these issues is:

1. We proposed to explicitly disallow use of a Data URI, which would
allow specification of a Location-by-value in a header.  There was a lot
of list traffic on this, currently in the state where Keith asked for a
real use case for it.  So far, we don't see a use case that cannot be
provided by location-by-reference in a header.  If we don't get a good
use case, I think consensus is to disallow location-by-value in a
header.

2. Dereferencing location-by-reference.  We had initially proposed to
document a process to use a raw HTTP(S) Get to dereference.  There was a
lot of discussion, and then I made a new proposal, see item 6.=20

3. We proposed to allow a Reason header with a 424 Bad Location
response.  This seems to be acceptable to the WG.  Geopriv will create
and populate the registry in a separate draft.

4. We proposed to explicitly allow hop-by-hop TLS security for location
when the endpoint wished to allow routing based on location.  We further
proposed to interpret "Do not Distribute" flags as allowing sending of
the location to the routing database (LoST for example).  Hannes pointed
out that there was a proposal to allow two Location headers, one of
which could be used for routing and another for onward forwarding to the
recipient.  Hannes' message seemed to agree that "Do not distribute"
would allow sending of the location to the routing-by-location database,
but would obligate the proxy that did that to remove the location before
it forwarded the message.  There were no other messages on this thread.
We conclude that allowing hop-by-hop security for location based routing
is acceptable, and Do Not Distribute DOES allow forwarding to a routing
database, but requires that the proxy doing the location based route to
remove the location it uses for that purpose.

5. We proposed to create a parameter and a registry to distinguish
multiple ways to dereference a location where HTTP(S) was the transport.
Resolution of this issue depends on resolution of issue 6.

6. We proposed to allow sip/sips and pres: in the location header.  I
later speculated that if we allow this, it forms a simple way to do
location-by-reference, and we could eliminate the raw HTTP(S) Get
proposal, with the parameter and registry entirely.  There was some list
traffic in favor of this.  Martin Thompson is skeptical that SIP
Presence Subscribe/Notify is actually allowed as a geopriv "using
protocol".  Several of think that it is.  If we don't get any more
comments on this proposal, we will allow sip/sips and pres, not have any
http(s) resolution text, and not propose a parameter and accompanying
registry.

If this does not reflect your position, please speak up now.

Brian

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Henning Schulzrinne wrote:
> We may be talking past each other. Currently, RFC 4119 has
> 
>  <xs:complexType name="locPolicyType">
>    <xs:sequence>
>      <xs:element name="retransmission-allowed" type="xs:boolean"
>         minOccurs="0" maxOccurs="1"/>
>      <xs:element name="retention-expiry" type="xs:dateTime"
>         minOccurs="0" maxOccurs="1"/>
>      <xs:element name="external-ruleset" type="xs:anyURI"
>         minOccurs="0" maxOccurs="1"/>
>      <xs:element name="note-well" type="tns:notewell"
>         minOccurs="0" maxOccurs="1"/>
>      <xs:any namespace="##other" processContents="lax" minOccurs="0"
>         maxOccurs="unbounded"/>
>    </xs:sequence>
>   </xs:complexType>
> 
> I assume you're talking about the external-ruleset element. Regardless 
> of how the rest of the GEOPRIV object is carried, this can point 
> anywhere, including some static web object, a cid, or even be another 
> data URL if you want to carry the policy along with the actual location 
> information.

Hmmm... we seem to have caught a bug in 4119.  Because the descriptive 
text calls that element 'ruleset-reference' and not 'external-ruleset'. 
  Anyway, from the text (page 8):

       This URI SHOULD use the HTTPS URI scheme; and if it does, the
       server that holds these rules MUST authenticate any attempt to
       access these rules.  Usage rules themselves may divulge private
       information about a Target or Rule Maker.  The URI MAY,
       alternatively, use the CID URI scheme [7], in which case it MUST
       denote a MIME body carried with the Location Object by the using
       protocol.

In other words, it would appear that the list of URIs allowed is HTTPS 
or CID (though this doc SHOULD be more explicit).  data: is not mentioned.

This means that a PIDF-LO with a CID reference cannot be converted to a 
data: URI.

-andy


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From: "Abbott, Nadine B" <nabbott@telcordia.com>
To: "Brian Rosen" <br@brianrosen.net>, <sip@ietf.org>
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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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Brian,

Actually, I was thinking that in other industry work which is trying to
leverage and provide transitional strategies to migrate to using SIP to
support emergency services, the and long-term solutions that we have
defined assume a capabity to provide location-by-reference.  I think it
would be very useful to have this defined in a standard SIP header
mechanism in such a way that we need not redefine migratory solutions
that have begun to be implemented. If an HTTPS URI can be used as the
location-by-reference, this will be helpful, I think. =20

I don't understand your urgency to make location-by-reference only an
all-sip-all-the-time thing.
And from all the other emails on this topic, it didn't seem to me that
there is a consensus to do this.

Nadine=20

-----Original Message-----
From: Brian Rosen [mailto:br@brianrosen.net]=20
Sent: Wednesday, July 26, 2006 3:15 PM
To: Abbott, Nadine B; sip@ietf.org
Cc: geopriv@ietf.org
Subject: RE: [Geopriv] Consensus on changes to location-conveyance

Nadine

Are you saying that it's your opinion sip-location-conveyance MUST
define an HTTP dereference procedure?

If you do, why?  What is the use-case or requirement?

Would it not be pretty reasonable to require a separate document that
defines a location-by-reference mechanism, protocol, and geopriv "using
protocol" analysis?  If that document specifically extended what was
allowed in the location header, would that not be reasonable?

I think we have to include "A" dereference mechanism in the
location-conveyance document.  It seems just including sip/sips/prez is
a reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.  That
certainly does not preclude another document describing another way.

Doesn't it seem just a little weird to have a sip-location-conveyance
document describing any kind of http based location conveyance
(de-reference) protocol, especially one created whole cloth without any
other parts?

Doesn't it seem in keeping with the geopriv approach that in order to
have another "using protocol" it has to have a geopriv review?  That is
what leads to restricting the URI schemes.  Would you prefer to let it
be wide open, so any protocol anyone thinks about can be used with the
location header?  I can't see geopriv going along with that.  We're
going to make it a one line ABNF to add a new scheme to the list of
allowed schemes.

It seems to me that if it's desired to define some bare-bones HTTP
location by reference protocol, then that should be a separate document,
which can easily extend the allowable schemes in the Location header.
I'm not sure it might be better to define ALL of that in the L7
protocol, but I wouldn't be opposed to a simple "raw GET" RFC as long as
it got a full geopriv review.
I just don't see where it makes any sense to force that into a sip
document.

Brian

> -----Original Message-----
> From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
> Sent: Wednesday, July 26, 2006 2:14 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>=20
>=20
> I don't agree that there is a consensus to preclude the use of HTTP.
> I would favor not placing your proposed restriction on the URI scheme=20
> for expressing a location reference in the SIP location conveyance=20
> draft; or at least including HTTP as a valid location-reference URI=20
> type.
>=20
> Regards,
> Nadine Abbott
>=20
> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Tuesday, July 25, 2006 1:38 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: [Geopriv] Consensus on changes to location-conveyance
>=20
> I have put out a number of emails on proposed changes to location=20
> conveyance.  Some of them have sparked a number of comments.  My=20
> summary of consensus on these issues is:
>=20
> 1. We proposed to explicitly disallow use of a Data URI, which would=20
> allow specification of a Location-by-value in a header.  There was a=20
> lot of list traffic on this, currently in the state where Keith asked=20
> for a real use case for it.  So far, we don't see a use case that=20
> cannot be provided by location-by-reference in a header.  If we don't=20
> get a good use case, I think consensus is to disallow=20
> location-by-value in a header.
>=20
> 2. Dereferencing location-by-reference.  We had initially proposed to=20
> document a process to use a raw HTTP(S) Get to dereference.  There was

> a lot of discussion, and then I made a new proposal, see item 6.
>=20
> 3. We proposed to allow a Reason header with a 424 Bad Location=20
> response.  This seems to be acceptable to the WG.  Geopriv will create

> and populate the registry in a separate draft.
>=20
> 4. We proposed to explicitly allow hop-by-hop TLS security for=20
> location when the endpoint wished to allow routing based on location.

> We further proposed to interpret "Do not Distribute" flags as allowing

> sending of the location to the routing database (LoST for example). =20
> Hannes pointed out that there was a proposal to allow two Location=20
> headers, one of which could be used for routing and another for onward

> forwarding to the recipient.  Hannes' message seemed to agree that "Do
not distribute"
> would allow sending of the location to the routing-by-location=20
> database, but would obligate the proxy that did that to remove the=20
> location before it forwarded the message.  There were no other
messages on this thread.
> We conclude that allowing hop-by-hop security for location based=20
> routing is acceptable, and Do Not Distribute DOES allow forwarding to=20
> a routing database, but requires that the proxy doing the location=20
> based route to remove the location it uses for that purpose.
>=20
> 5. We proposed to create a parameter and a registry to distinguish=20
> multiple ways to dereference a location where HTTP(S) was the
transport.
> Resolution of this issue depends on resolution of issue 6.
>=20
> 6. We proposed to allow sip/sips and pres: in the location header.  I=20
> later speculated that if we allow this, it forms a simple way to do=20
> location-by-reference, and we could eliminate the raw HTTP(S) Get=20
> proposal, with the parameter and registry entirely.  There was some=20
> list traffic in favor of this.  Martin Thompson is skeptical that SIP=20
> Presence Subscribe/Notify is actually allowed as a geopriv "using=20
> protocol".  Several of think that it is.  If we don't get any more=20
> comments on this proposal, we will allow sip/sips and pres, not have=20
> any
> http(s) resolution text, and not propose a parameter and accompanying=20
> registry.
>=20
> If this does not reflect your position, please speak up now.
>=20
> Brian
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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> In other words, it would appear that the list of URIs allowed is HTTPS 
> or CID (though this doc SHOULD be more explicit).  data: is not mentioned.
> 
> This means that a PIDF-LO with a CID reference cannot be converted to a 
> data: URI.

At best, SHOULD NOT. This issue was never discussed during the 4119 
discussion, as far as I can tell, and there doesn't seem to be technical 
reason why it couldn't be.

In any event, for the case at hand, namely proxy insertion, the problem 
is exactly the same: both both (only) use HTTPS. I'm unclear what 
ruleset we can usefully specify today. The common policy set is not 
particularly useful for routing, for example, and not really meant for 
recipients of location information in any event. Thus, at least today, 
this is, at best, some free-text legalese.


> 
> -andy
> 
> 
> _______________________________________________
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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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At 03:50 PM 7/25/2006 -0400, Rosen, Brian wrote:
>When the original PIDF-LO RFC came out, location-by-reference was
>defined.
>
>You have a URI.  You can exchange the URI for location.  There is a
>protocol defined for that purpose (SIP SUBSCRIBE).  That is location by
>reference.  It also serves as the location update mechanism, clearly
>something we need.

Actually, UPDATE serves as the update mechanism for the UAC detecting it 
has changed locations.  This has been in the ID from day 1, and clarified 
in WG meetings over the years.

How a UAC determines/d it has moved obviously not defined in the IETF.

SUBSCRIBE was thought of a a good pull retreaval protocol action for 
updating the location of a target.

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From: "Brian Rosen" <br@brianrosen.net>
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Cc: geopriv@ietf.org
Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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Nadine

Are you saying that it's your opinion sip-location-conveyance MUST define an
HTTP dereference procedure?

If you do, why?  What is the use-case or requirement?

Would it not be pretty reasonable to require a separate document that
defines a location-by-reference mechanism, protocol, and geopriv "using
protocol" analysis?  If that document specifically extended what was allowed
in the location header, would that not be reasonable?

I think we have to include "A" dereference mechanism in the
location-conveyance document.  It seems just including sip/sips/prez is a
reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.  That certainly
does not preclude another document describing another way.

Doesn't it seem just a little weird to have a sip-location-conveyance
document describing any kind of http based location conveyance
(de-reference) protocol, especially one created whole cloth without any
other parts?

Doesn't it seem in keeping with the geopriv approach that in order to have
another "using protocol" it has to have a geopriv review?  That is what
leads to restricting the URI schemes.  Would you prefer to let it be wide
open, so any protocol anyone thinks about can be used with the location
header?  I can't see geopriv going along with that.  We're going to make it
a one line ABNF to add a new scheme to the list of allowed schemes.

It seems to me that if it's desired to define some bare-bones HTTP location
by reference protocol, then that should be a separate document, which can
easily extend the allowable schemes in the Location header.  I'm not sure it
might be better to define ALL of that in the L7 protocol, but I wouldn't be
opposed to a simple "raw GET" RFC as long as it got a full geopriv review.
I just don't see where it makes any sense to force that into a sip document.

Brian

> -----Original Message-----
> From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
> Sent: Wednesday, July 26, 2006 2:14 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
> 
> 
> I don't agree that there is a consensus to preclude the use of HTTP.
> I would favor not placing your proposed restriction on the URI scheme
> for expressing a location reference in the SIP location conveyance
> draft; or at least including HTTP as a valid location-reference URI
> type.
> 
> Regards,
> Nadine Abbott
> 
> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Tuesday, July 25, 2006 1:38 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: [Geopriv] Consensus on changes to location-conveyance
> 
> I have put out a number of emails on proposed changes to location
> conveyance.  Some of them have sparked a number of comments.  My summary
> of consensus on these issues is:
> 
> 1. We proposed to explicitly disallow use of a Data URI, which would
> allow specification of a Location-by-value in a header.  There was a lot
> of list traffic on this, currently in the state where Keith asked for a
> real use case for it.  So far, we don't see a use case that cannot be
> provided by location-by-reference in a header.  If we don't get a good
> use case, I think consensus is to disallow location-by-value in a
> header.
> 
> 2. Dereferencing location-by-reference.  We had initially proposed to
> document a process to use a raw HTTP(S) Get to dereference.  There was a
> lot of discussion, and then I made a new proposal, see item 6.
> 
> 3. We proposed to allow a Reason header with a 424 Bad Location
> response.  This seems to be acceptable to the WG.  Geopriv will create
> and populate the registry in a separate draft.
> 
> 4. We proposed to explicitly allow hop-by-hop TLS security for location
> when the endpoint wished to allow routing based on location.  We further
> proposed to interpret "Do not Distribute" flags as allowing sending of
> the location to the routing database (LoST for example).  Hannes pointed
> out that there was a proposal to allow two Location headers, one of
> which could be used for routing and another for onward forwarding to the
> recipient.  Hannes' message seemed to agree that "Do not distribute"
> would allow sending of the location to the routing-by-location database,
> but would obligate the proxy that did that to remove the location before
> it forwarded the message.  There were no other messages on this thread.
> We conclude that allowing hop-by-hop security for location based routing
> is acceptable, and Do Not Distribute DOES allow forwarding to a routing
> database, but requires that the proxy doing the location based route to
> remove the location it uses for that purpose.
> 
> 5. We proposed to create a parameter and a registry to distinguish
> multiple ways to dereference a location where HTTP(S) was the transport.
> Resolution of this issue depends on resolution of issue 6.
> 
> 6. We proposed to allow sip/sips and pres: in the location header.  I
> later speculated that if we allow this, it forms a simple way to do
> location-by-reference, and we could eliminate the raw HTTP(S) Get
> proposal, with the parameter and registry entirely.  There was some list
> traffic in favor of this.  Martin Thompson is skeptical that SIP
> Presence Subscribe/Notify is actually allowed as a geopriv "using
> protocol".  Several of think that it is.  If we don't get any more
> comments on this proposal, we will allow sip/sips and pres, not have any
> http(s) resolution text, and not propose a parameter and accompanying
> registry.
> 
> If this does not reflect your position, please speak up now.
> 
> Brian
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Abbott, Nadine B" <nabbott@telcordia.com>,
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Brian,

I don't care if you don't mention how to de-reference URI types other
than sip/sips/cid/pres in the document or not. But I do not want the
schema artificially constrained to exclude URI types such as HTTP. There
is clearly not consensus to justify this exclusion, so please leave
schema unchanged and make your recommendations in the text.

Cheers
James

=20

-----Original Message-----
From: Abbott, Nadine B [mailto:nabbott@telcordia.com]=20
Sent: Thursday, 27 July 2006 7:44 AM
To: Brian Rosen; sip@ietf.org
Cc: geopriv@ietf.org
Subject: RE: [Geopriv] Consensus on changes to location-conveyance

Brian,

Actually, I was thinking that in other industry work which is trying to
leverage and provide transitional strategies to migrate to using SIP to
support emergency services, the and long-term solutions that we have
defined assume a capabity to provide location-by-reference.  I think it
would be very useful to have this defined in a standard SIP header
mechanism in such a way that we need not redefine migratory solutions
that have begun to be implemented. If an HTTPS URI can be used as the
location-by-reference, this will be helpful, I think. =20

I don't understand your urgency to make location-by-reference only an
all-sip-all-the-time thing.
And from all the other emails on this topic, it didn't seem to me that
there is a consensus to do this.

Nadine=20

-----Original Message-----
From: Brian Rosen [mailto:br@brianrosen.net]=20
Sent: Wednesday, July 26, 2006 3:15 PM
To: Abbott, Nadine B; sip@ietf.org
Cc: geopriv@ietf.org
Subject: RE: [Geopriv] Consensus on changes to location-conveyance

Nadine

Are you saying that it's your opinion sip-location-conveyance MUST
define an HTTP dereference procedure?

If you do, why?  What is the use-case or requirement?

Would it not be pretty reasonable to require a separate document that
defines a location-by-reference mechanism, protocol, and geopriv "using
protocol" analysis?  If that document specifically extended what was
allowed in the location header, would that not be reasonable?

I think we have to include "A" dereference mechanism in the
location-conveyance document.  It seems just including sip/sips/prez is
a reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.  That
certainly does not preclude another document describing another way.

Doesn't it seem just a little weird to have a sip-location-conveyance
document describing any kind of http based location conveyance
(de-reference) protocol, especially one created whole cloth without any
other parts?

Doesn't it seem in keeping with the geopriv approach that in order to
have another "using protocol" it has to have a geopriv review?  That is
what leads to restricting the URI schemes.  Would you prefer to let it
be wide open, so any protocol anyone thinks about can be used with the
location header?  I can't see geopriv going along with that.  We're
going to make it a one line ABNF to add a new scheme to the list of
allowed schemes.

It seems to me that if it's desired to define some bare-bones HTTP
location by reference protocol, then that should be a separate document,
which can easily extend the allowable schemes in the Location header.
I'm not sure it might be better to define ALL of that in the L7
protocol, but I wouldn't be opposed to a simple "raw GET" RFC as long as
it got a full geopriv review.
I just don't see where it makes any sense to force that into a sip
document.

Brian

> -----Original Message-----
> From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
> Sent: Wednesday, July 26, 2006 2:14 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>=20
>=20
> I don't agree that there is a consensus to preclude the use of HTTP.
> I would favor not placing your proposed restriction on the URI scheme=20
> for expressing a location reference in the SIP location conveyance=20
> draft; or at least including HTTP as a valid location-reference URI=20
> type.
>=20
> Regards,
> Nadine Abbott
>=20
> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Tuesday, July 25, 2006 1:38 PM
> To: sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: [Geopriv] Consensus on changes to location-conveyance
>=20
> I have put out a number of emails on proposed changes to location=20
> conveyance.  Some of them have sparked a number of comments.  My=20
> summary of consensus on these issues is:
>=20
> 1. We proposed to explicitly disallow use of a Data URI, which would=20
> allow specification of a Location-by-value in a header.  There was a=20
> lot of list traffic on this, currently in the state where Keith asked=20
> for a real use case for it.  So far, we don't see a use case that=20
> cannot be provided by location-by-reference in a header.  If we don't=20
> get a good use case, I think consensus is to disallow=20
> location-by-value in a header.
>=20
> 2. Dereferencing location-by-reference.  We had initially proposed to=20
> document a process to use a raw HTTP(S) Get to dereference.  There was

> a lot of discussion, and then I made a new proposal, see item 6.
>=20
> 3. We proposed to allow a Reason header with a 424 Bad Location=20
> response.  This seems to be acceptable to the WG.  Geopriv will create

> and populate the registry in a separate draft.
>=20
> 4. We proposed to explicitly allow hop-by-hop TLS security for=20
> location when the endpoint wished to allow routing based on location.

> We further proposed to interpret "Do not Distribute" flags as allowing

> sending of the location to the routing database (LoST for example). =20
> Hannes pointed out that there was a proposal to allow two Location=20
> headers, one of which could be used for routing and another for onward

> forwarding to the recipient.  Hannes' message seemed to agree that "Do
not distribute"
> would allow sending of the location to the routing-by-location=20
> database, but would obligate the proxy that did that to remove the=20
> location before it forwarded the message.  There were no other
messages on this thread.
> We conclude that allowing hop-by-hop security for location based=20
> routing is acceptable, and Do Not Distribute DOES allow forwarding to=20
> a routing database, but requires that the proxy doing the location=20
> based route to remove the location it uses for that purpose.
>=20
> 5. We proposed to create a parameter and a registry to distinguish=20
> multiple ways to dereference a location where HTTP(S) was the
transport.
> Resolution of this issue depends on resolution of issue 6.
>=20
> 6. We proposed to allow sip/sips and pres: in the location header.  I=20
> later speculated that if we allow this, it forms a simple way to do=20
> location-by-reference, and we could eliminate the raw HTTP(S) Get=20
> proposal, with the parameter and registry entirely.  There was some=20
> list traffic in favor of this.  Martin Thompson is skeptical that SIP=20
> Presence Subscribe/Notify is actually allowed as a geopriv "using=20
> protocol".  Several of think that it is.  If we don't get any more=20
> comments on this proposal, we will allow sip/sips and pres, not have=20
> any
> http(s) resolution text, and not propose a parameter and accompanying=20
> registry.
>=20
> If this does not reflect your position, please speak up now.
>=20
> Brian
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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At 09:11 AM 7/26/2006 -0400, Paul Kyzivat wrote:


>Marc Linsner wrote:
>>In-line...
>>>So, you are arguing against proxy insertion of a location-by-reference?
>>No, if done properly....but reading the latest threads, I get the feeling
>>that www.whereintheworldisbrianrosen.com is an acceptable reference.
>
>Damn! I was hoping, but www.whereintheworldisbrianrosen.com doesn't seem 
>to be working today. :-(

I think he's .biz now...   ;-)


>         Paul
>
>_______________________________________________
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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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On a general note and generalizing a bit on Nadine's message below, I  
find the nanny tone of this discussion a bit peculiar ("No, I don't  
want you to play with that!"). I can see the technical rationale for  
requiring implementation of a limited set of choices for  
interoperability, but proactively ruling out other choices seems  
highly unusual for an IETF protocol spec. It is clear that only a  
relatively small number of people are participating in this  
discussion, so declaring consensus to rule out one URL scheme or  
another as 'evil bits' seems a bit peculiar.

On Jul 26, 2006, at 2:14 PM, Abbott, Nadine B wrote:

>
> I don't agree that there is a consensus to preclude the use of HTTP.
> I would favor not placing your proposed restriction on the URI scheme
> for expressing a location reference in the SIP location conveyance
> draft; or at least including HTTP as a valid location-reference URI
> type.
>
> Regards,
> Nadine Abbott
>


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For location-based routing case, the likelihood of significant updates 
in the roughly one second that a call might take from caller to callee 
seems unlikely. UAS subscription and polling is clearly more valuable, 
so maybe we need to distinguish these cases. (As discussed earlier, 
mid-routing updates may lead to bad outcomes since the call might have 
been sent to a place A that doesn't know what to do with location update 
n+1 since that new location is really handled by place B.)

Andrew Newton wrote:
> Henning Schulzrinne wrote:
>>> I believe there are two technical arguments: one is legacy equipment 
>>> (the issue that I'll be facing for quite some time), and environments 
>>> where the location is generated by the network.  For the latter, 
>>> insertion of the Location header only in emergency situations still 
>>> puts the user in control for non-emergency situations.
>>>
>>>
>>
>> For the latter, the combination of the in-progress L7 protocols, DHCP 
>> and LLDP should presumably allow the UAC to obtain location and insert 
>> it by value. Is there a use case not covered by that?
> 
> As I said, where location comes from the network.  In this scenario, 
> there are two possibilities, both up to the user: 1) the UAC subscribes 
> to the location from the network and puts it by value in the call 
> flow... meaning all location updates go through the UAC, or 2) the UAC 
> hands out a reference so updates can be independent of the UAC or 
> updates based on subscription come from the UAC but are independent of 
> the call.
> 
> -andy

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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>, "Andrew Newton" <andy@hxr.us>
Date: Wed, 26 Jul 2006 20:28:00 -0500
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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In the case you mention below the reality is that if the target has
moved during the course of routing, the target will either be at place B
by the time dispatch occurs, or at point C which may be further away
still.=20
At least in the case of a reference you have the possibility of getting
the current location, which is not possible through any of the
"by-value" means described thus far unless the caller is stationary!


-----Original Message-----
From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]=20
Sent: Thursday, 27 July 2006 3:14 AM
To: Andrew Newton
Cc: geopriv@ietf.org; IETF SIP List
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
location-conveyance

For location-based routing case, the likelihood of significant updates=20
in the roughly one second that a call might take from caller to callee=20
seems unlikely. UAS subscription and polling is clearly more valuable,=20
so maybe we need to distinguish these cases. (As discussed earlier,=20
mid-routing updates may lead to bad outcomes since the call might have=20
been sent to a place A that doesn't know what to do with location update

n+1 since that new location is really handled by place B.)

Andrew Newton wrote:
> Henning Schulzrinne wrote:
>>> I believe there are two technical arguments: one is legacy equipment

>>> (the issue that I'll be facing for quite some time), and
environments=20
>>> where the location is generated by the network.  For the latter,=20
>>> insertion of the Location header only in emergency situations still=20
>>> puts the user in control for non-emergency situations.
>>>
>>>
>>
>> For the latter, the combination of the in-progress L7 protocols, DHCP

>> and LLDP should presumably allow the UAC to obtain location and
insert=20
>> it by value. Is there a use case not covered by that?
>=20
> As I said, where location comes from the network.  In this scenario,=20
> there are two possibilities, both up to the user: 1) the UAC
subscribes=20
> to the location from the network and puts it by value in the call=20
> flow... meaning all location updates go through the UAC, or 2) the UAC

> hands out a reference so updates can be independent of the UAC or=20
> updates based on subscription come from the UAC but are independent of

> the call.
>=20
> -andy

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From sip-bounces@ietf.org Wed Jul 26 21:38:13 2006
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From: Andrew Newton <andy@hxr.us>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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I don't think Brian ever proposed ruling out URI schemes.  What he  
proposed was restricting what can go in it now and a mechanism for  
defining future URI schemes that will be allowed.  We certainly do  
want using protocols passing around PIDF-LOs getting review by GEOPRIV.

-andy

On Jul 26, 2006, at 8:33 PM, Henning Schulzrinne wrote:

> On a general note and generalizing a bit on Nadine's message below,  
> I find the nanny tone of this discussion a bit peculiar ("No, I  
> don't want you to play with that!"). I can see the technical  
> rationale for requiring implementation of a limited set of choices  
> for interoperability, but proactively ruling out other choices  
> seems highly unusual for an IETF protocol spec. It is clear that  
> only a relatively small number of people are participating in this  
> discussion, so declaring consensus to rule out one URL scheme or  
> another as 'evil bits' seems a bit peculiar.
>
> On Jul 26, 2006, at 2:14 PM, Abbott, Nadine B wrote:
>
>>
>> I don't agree that there is a consensus to preclude the use of HTTP.
>> I would favor not placing your proposed restriction on the URI scheme
>> for expressing a location reference in the SIP location conveyance
>> draft; or at least including HTTP as a valid location-reference URI
>> type.
>>
>> Regards,
>> Nadine Abbott
>>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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On Jul 26, 2006, at 5:50 PM, Henning Schulzrinne wrote:

>> In other words, it would appear that the list of URIs allowed is  
>> HTTPS or CID (though this doc SHOULD be more explicit).  data: is  
>> not mentioned.
>> This means that a PIDF-LO with a CID reference cannot be converted  
>> to a data: URI.
>
> At best, SHOULD NOT. This issue was never discussed during the 4119  
> discussion, as far as I can tell, and there doesn't seem to be  
> technical reason why it couldn't be.

It was discussed at length at our interim meeting in DC, and nobody  
ever mentioned anything but HTTPS and CID.

> In any event, for the case at hand, namely proxy insertion, the  
> problem is exactly the same: both both (only) use HTTPS. I'm  
> unclear what ruleset we can usefully specify today. The common  
> policy set is not particularly useful for routing, for example, and  
> not really meant for recipients of location information in any  
> event. Thus, at least today, this is, at best, some free-text  
> legalese.

This is yet another rule, adding more complexity.  You can pass by  
value either via MIME or data: URI.  If you pass by value using the  
data: URI, you have to restrict ruleset references.

-andy

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Andy,

RFC-4119 says "However, the usage of this location object format is not
   limited to presence-using protocols-- any protocol that can carry XML
   or MIME types can carry PIDF."

It seems to me that restricting the allowed schema in what is supposed
to be a location conveyance specification is in opposition to this
statement.


-----Original Message-----
From: Andrew Newton [mailto:andy@hxr.us]=20
Sent: Thursday, 27 July 2006 11:36 AM
To: Henning Schulzrinne
Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
location-conveyance

I don't think Brian ever proposed ruling out URI schemes.  What he =20
proposed was restricting what can go in it now and a mechanism for =20
defining future URI schemes that will be allowed.  We certainly do =20
want using protocols passing around PIDF-LOs getting review by GEOPRIV.

-andy

On Jul 26, 2006, at 8:33 PM, Henning Schulzrinne wrote:

> On a general note and generalizing a bit on Nadine's message below, =20
> I find the nanny tone of this discussion a bit peculiar ("No, I =20
> don't want you to play with that!"). I can see the technical =20
> rationale for requiring implementation of a limited set of choices =20
> for interoperability, but proactively ruling out other choices =20
> seems highly unusual for an IETF protocol spec. It is clear that =20
> only a relatively small number of people are participating in this =20
> discussion, so declaring consensus to rule out one URL scheme or =20
> another as 'evil bits' seems a bit peculiar.
>
> On Jul 26, 2006, at 2:14 PM, Abbott, Nadine B wrote:
>
>>
>> I don't agree that there is a consensus to preclude the use of HTTP.
>> I would favor not placing your proposed restriction on the URI scheme
>> for expressing a location reference in the SIP location conveyance
>> draft; or at least including HTTP as a valid location-reference URI
>> type.
>>
>> Regards,
>> Nadine Abbott
>>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From: Andrew Newton <andy@hxr.us>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 21:47:04 -0400
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On Jul 26, 2006, at 9:28 PM, Winterbottom, James wrote:

> In the case you mention below the reality is that if the target has
> moved during the course of routing, the target will either be at  
> place B
> by the time dispatch occurs, or at point C which may be further away
> still.
> At least in the case of a reference you have the possibility of  
> getting
> the current location, which is not possible through any of the
> "by-value" means described thus far unless the caller is stationary!
>

What about a SIP UPDATE?  Section 5.3 of the draft.

-andy





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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Andrew Newton" <andy@hxr.us>
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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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To what do you send the SIP update and for what do you request the
update?


-----Original Message-----
From: Andrew Newton [mailto:andy@hxr.us]=20
Sent: Thursday, 27 July 2006 11:47 AM
To: Winterbottom, James
Cc: Henning Schulzrinne; geopriv@ietf.org; IETF SIP List
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
location-conveyance


On Jul 26, 2006, at 9:28 PM, Winterbottom, James wrote:

> In the case you mention below the reality is that if the target has
> moved during the course of routing, the target will either be at =20
> place B
> by the time dispatch occurs, or at point C which may be further away
> still.
> At least in the case of a reference you have the possibility of =20
> getting
> the current location, which is not possible through any of the
> "by-value" means described thus far unless the caller is stationary!
>

What about a SIP UPDATE?  Section 5.3 of the draft.

-andy





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On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:

> Andy,
>
> RFC-4119 says "However, the usage of this location object format is  
> not
>    limited to presence-using protocols-- any protocol that can  
> carry XML
>    or MIME types can carry PIDF."
>
> It seems to me that restricting the allowed schema in what is supposed
> to be a location conveyance specification is in opposition to this
> statement.

Did somebody say that HTTP was a presence-using protocol?

-andy

ps.  you can find my location with the following reference URIs:

gopher://fatchance.hxr.us/you-still-have-a-gopher-client
news:alt.locations.people.secretly-being-watched
telnet://screen-scrapers.are.us/
dns:cached-publicly-around-the-world.com?type=HOST



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Andrew Newton'" <andy@hxr.us>,
	"'Winterbottom, James'" <James.Winterbottom@andrew.com>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Wed, 26 Jul 2006 23:08:11 -0400
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At this point, I think we need some chair guidance and by that I think it's
a combo of geopriv and sip chairs.

We seem to have a couple of significant open questions.  I think most of the
other stuff (like the "allow sending to router") will converge.

There seems to be some opposition to allowing sip/sips/pres: in a Location
Header.  The opposition seems to be that despite 4119, somehow, SIP
(presence subscription) is not a "using protocol".  I "think" a ruling from
the bench on that would work.

There is also opposition to defining the allowed schemes as a specific list,
with a defined (easy) extension mechanism.  

Related to that, there are some who want location-conveyance to define an
HTTP based location dereference mechanism, and on the opposite end, we have
people saying we haven't even agreed to support any form of
location-by-reference.  

The authors would like to know what to do now.

Brian

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Wednesday, July 26, 2006 9:58 PM
> To: Winterbottom, James
> Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning Schulzrinne
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
> 
> 
> On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
> 
> > Andy,
> >
> > RFC-4119 says "However, the usage of this location object format is
> > not
> >    limited to presence-using protocols-- any protocol that can
> > carry XML
> >    or MIME types can carry PIDF."
> >
> > It seems to me that restricting the allowed schema in what is supposed
> > to be a location conveyance specification is in opposition to this
> > statement.
> 
> Did somebody say that HTTP was a presence-using protocol?
> 
> -andy
> 
> ps.  you can find my location with the following reference URIs:
> 
> gopher://fatchance.hxr.us/you-still-have-a-gopher-client
> news:alt.locations.people.secretly-being-watched
> telnet://screen-scrapers.are.us/
> dns:cached-publicly-around-the-world.com?type=HOST
> 
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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My suggestion would be

- to focus the draft on just defining the Location header syntax,  
without ruling out any particular mechanism or scheme; this would  
make the draft really short (which I would find helpful).

- and then add one section to the draft that prototypically defines  
what a particular scheme that wants to appear in the Location header  
needs to define, using the non-controversial cid mechanism as the  
example. Other, separate, drafts can then define other modes,  
following the same template. Documents such as the phone-bcp can then  
pick some suitable must-implement subset and make emergency-call- 
specific recommendations, rather than sprinkling these throughout the  
conveyance document and repeating some in the phone-bcp. Most of  
these are, in any event, operational recommendations, not for  
ensuring protocol-level interoperability.

That way, the conveyance draft can be extremely short (even shorter  
if the requirements are omitted...) and can get done, and we avoid  
engineering operational requirements on the fly, commingling  
emergency and non-emergency cases, as seems to be happening now.

I think protocol documents should define what's needed for  
interoperability; BCPs can make more specific, situation-specific  
recommendations. Commingling the two seems to be a recipe for  
deadlock or arbitrary, non-technically-grounded decisions that are  
more lawyer-like than engineering-like ("you did not mention  
technique X in some meeting N years ago and thus you can't bring it  
up now!")

Henning

On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:

> At this point, I think we need some chair guidance and by that I  
> think it's
> a combo of geopriv and sip chairs.
>
> We seem to have a couple of significant open questions.  I think  
> most of the
> other stuff (like the "allow sending to router") will converge.
>
> There seems to be some opposition to allowing sip/sips/pres: in a  
> Location
> Header.  The opposition seems to be that despite 4119, somehow, SIP
> (presence subscription) is not a "using protocol".  I "think" a  
> ruling from
> the bench on that would work.
>
> There is also opposition to defining the allowed schemes as a  
> specific list,
> with a defined (easy) extension mechanism.
>
> Related to that, there are some who want location-conveyance to  
> define an
> HTTP based location dereference mechanism, and on the opposite end,  
> we have
> people saying we haven't even agreed to support any form of
> location-by-reference.
>
> The authors would like to know what to do now.
>
> Brian
>
>> -----Original Message-----
>> From: Andrew Newton [mailto:andy@hxr.us]
>> Sent: Wednesday, July 26, 2006 9:58 PM
>> To: Winterbottom, James
>> Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning  
>> Schulzrinne
>> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
>> conveyance
>>
>>
>> On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
>>
>>> Andy,
>>>
>>> RFC-4119 says "However, the usage of this location object format is
>>> not
>>>    limited to presence-using protocols-- any protocol that can
>>> carry XML
>>>    or MIME types can carry PIDF."
>>>
>>> It seems to me that restricting the allowed schema in what is  
>>> supposed
>>> to be a location conveyance specification is in opposition to this
>>> statement.
>>
>> Did somebody say that HTTP was a presence-using protocol?
>>
>> -andy
>>
>> ps.  you can find my location with the following reference URIs:
>>
>> gopher://fatchance.hxr.us/you-still-have-a-gopher-client
>> news:alt.locations.people.secretly-being-watched
>> telnet://screen-scrapers.are.us/
>> dns:cached-publicly-around-the-world.com?type=HOST
>>
>>
>>
>> _______________________________________________
>> Geopriv mailing list
>> Geopriv@ietf.org
>> https://www1.ietf.org/mailman/listinfo/geopriv
>
>
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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From sip-bounces@ietf.org Wed Jul 26 23:27:09 2006
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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Brian Rosen" <br@brianrosen.net>,
	"Andrew Newton" <andy@hxr.us>
Date: Wed, 26 Jul 2006 22:25:17 -0500
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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Let's clarify this further Brian.

I do not want the schema to put a constraint on the URI type.
This is not the same as saying that I expect the document to describe
how to dereference all URI types.


-----Original Message-----
From: Brian Rosen [mailto:br@brianrosen.net]=20
Sent: Thursday, 27 July 2006 1:08 PM
To: 'Andrew Newton'; Winterbottom, James
Cc: sip@ietf.org; geopriv@ietf.org; 'Abbott, Nadine B'; 'Henning
Schulzrinne'
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to
location-conveyance

At this point, I think we need some chair guidance and by that I think
it's
a combo of geopriv and sip chairs.

We seem to have a couple of significant open questions.  I think most of
the
other stuff (like the "allow sending to router") will converge.

There seems to be some opposition to allowing sip/sips/pres: in a
Location
Header.  The opposition seems to be that despite 4119, somehow, SIP
(presence subscription) is not a "using protocol".  I "think" a ruling
from
the bench on that would work.

There is also opposition to defining the allowed schemes as a specific
list,
with a defined (easy) extension mechanism. =20

Related to that, there are some who want location-conveyance to define
an
HTTP based location dereference mechanism, and on the opposite end, we
have
people saying we haven't even agreed to support any form of
location-by-reference. =20

The authors would like to know what to do now.

Brian

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Wednesday, July 26, 2006 9:58 PM
> To: Winterbottom, James
> Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning
Schulzrinne
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
>=20
>=20
> On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
>=20
> > Andy,
> >
> > RFC-4119 says "However, the usage of this location object format is
> > not
> >    limited to presence-using protocols-- any protocol that can
> > carry XML
> >    or MIME types can carry PIDF."
> >
> > It seems to me that restricting the allowed schema in what is
supposed
> > to be a location conveyance specification is in opposition to this
> > statement.
>=20
> Did somebody say that HTTP was a presence-using protocol?
>=20
> -andy
>=20
> ps.  you can find my location with the following reference URIs:
>=20
> gopher://fatchance.hxr.us/you-still-have-a-gopher-client
> news:alt.locations.people.secretly-being-watched
> telnet://screen-scrapers.are.us/
> dns:cached-publicly-around-the-world.com?type=3DHOST
>=20
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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James

We won't say anything like "MUST only use sip/sips/pres/cid, and MUST NOT 
...".  I don't believe we will take a negative tone with what isn't allowed 
in this respect, but we will state explicitly what is defined for use.

We will likely state something like this:  ' this document defines usage of 
the sip, sips, pres and cid schemas for location conveyance, and here's how...'

I think Geopriv really needs to look at HTTP as a RFC 3693 "using protocol" 
to tease out any unique characteristics with that protocol.  That 
discussion and review shouldn't bog this document down.  Does this sound fair?

At 06:31 PM 7/26/2006 -0500, Winterbottom, James wrote:
>Brian,
>
>I don't care if you don't mention how to de-reference URI types other
>than sip/sips/cid/pres in the document or not. But I do not want the
>schema artificially constrained to exclude URI types such as HTTP. There
>is clearly not consensus to justify this exclusion, so please leave
>schema unchanged and make your recommendations in the text.
>
>Cheers
>James
>
>
>
>-----Original Message-----
>From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
>Sent: Thursday, 27 July 2006 7:44 AM
>To: Brian Rosen; sip@ietf.org
>Cc: geopriv@ietf.org
>Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>
>Brian,
>
>Actually, I was thinking that in other industry work which is trying to
>leverage and provide transitional strategies to migrate to using SIP to
>support emergency services, the and long-term solutions that we have
>defined assume a capabity to provide location-by-reference.  I think it
>would be very useful to have this defined in a standard SIP header
>mechanism in such a way that we need not redefine migratory solutions
>that have begun to be implemented. If an HTTPS URI can be used as the
>location-by-reference, this will be helpful, I think.
>
>I don't understand your urgency to make location-by-reference only an
>all-sip-all-the-time thing.
>And from all the other emails on this topic, it didn't seem to me that
>there is a consensus to do this.
>
>Nadine
>
>-----Original Message-----
>From: Brian Rosen [mailto:br@brianrosen.net]
>Sent: Wednesday, July 26, 2006 3:15 PM
>To: Abbott, Nadine B; sip@ietf.org
>Cc: geopriv@ietf.org
>Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>
>Nadine
>
>Are you saying that it's your opinion sip-location-conveyance MUST
>define an HTTP dereference procedure?
>
>If you do, why?  What is the use-case or requirement?
>
>Would it not be pretty reasonable to require a separate document that
>defines a location-by-reference mechanism, protocol, and geopriv "using
>protocol" analysis?  If that document specifically extended what was
>allowed in the location header, would that not be reasonable?
>
>I think we have to include "A" dereference mechanism in the
>location-conveyance document.  It seems just including sip/sips/prez is
>a reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.  That
>certainly does not preclude another document describing another way.
>
>Doesn't it seem just a little weird to have a sip-location-conveyance
>document describing any kind of http based location conveyance
>(de-reference) protocol, especially one created whole cloth without any
>other parts?
>
>Doesn't it seem in keeping with the geopriv approach that in order to
>have another "using protocol" it has to have a geopriv review?  That is
>what leads to restricting the URI schemes.  Would you prefer to let it
>be wide open, so any protocol anyone thinks about can be used with the
>location header?  I can't see geopriv going along with that.  We're
>going to make it a one line ABNF to add a new scheme to the list of
>allowed schemes.
>
>It seems to me that if it's desired to define some bare-bones HTTP
>location by reference protocol, then that should be a separate document,
>which can easily extend the allowable schemes in the Location header.
>I'm not sure it might be better to define ALL of that in the L7
>protocol, but I wouldn't be opposed to a simple "raw GET" RFC as long as
>it got a full geopriv review.
>I just don't see where it makes any sense to force that into a sip
>document.
>
>Brian
>
> > -----Original Message-----
> > From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
> > Sent: Wednesday, July 26, 2006 2:14 PM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: RE: [Geopriv] Consensus on changes to location-conveyance
> >
> >
> > I don't agree that there is a consensus to preclude the use of HTTP.
> > I would favor not placing your proposed restriction on the URI scheme
> > for expressing a location reference in the SIP location conveyance
> > draft; or at least including HTTP as a valid location-reference URI
> > type.
> >
> > Regards,
> > Nadine Abbott
> >
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Tuesday, July 25, 2006 1:38 PM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: [Geopriv] Consensus on changes to location-conveyance
> >
> > I have put out a number of emails on proposed changes to location
> > conveyance.  Some of them have sparked a number of comments.  My
> > summary of consensus on these issues is:
> >
> > 1. We proposed to explicitly disallow use of a Data URI, which would
> > allow specification of a Location-by-value in a header.  There was a
> > lot of list traffic on this, currently in the state where Keith asked
> > for a real use case for it.  So far, we don't see a use case that
> > cannot be provided by location-by-reference in a header.  If we don't
> > get a good use case, I think consensus is to disallow
> > location-by-value in a header.
> >
> > 2. Dereferencing location-by-reference.  We had initially proposed to
> > document a process to use a raw HTTP(S) Get to dereference.  There was
>
> > a lot of discussion, and then I made a new proposal, see item 6.
> >
> > 3. We proposed to allow a Reason header with a 424 Bad Location
> > response.  This seems to be acceptable to the WG.  Geopriv will create
>
> > and populate the registry in a separate draft.
> >
> > 4. We proposed to explicitly allow hop-by-hop TLS security for
> > location when the endpoint wished to allow routing based on location.
>
> > We further proposed to interpret "Do not Distribute" flags as allowing
>
> > sending of the location to the routing database (LoST for example).
> > Hannes pointed out that there was a proposal to allow two Location
> > headers, one of which could be used for routing and another for onward
>
> > forwarding to the recipient.  Hannes' message seemed to agree that "Do
>not distribute"
> > would allow sending of the location to the routing-by-location
> > database, but would obligate the proxy that did that to remove the
> > location before it forwarded the message.  There were no other
>messages on this thread.
> > We conclude that allowing hop-by-hop security for location based
> > routing is acceptable, and Do Not Distribute DOES allow forwarding to
> > a routing database, but requires that the proxy doing the location
> > based route to remove the location it uses for that purpose.
> >
> > 5. We proposed to create a parameter and a registry to distinguish
> > multiple ways to dereference a location where HTTP(S) was the
>transport.
> > Resolution of this issue depends on resolution of issue 6.
> >
> > 6. We proposed to allow sip/sips and pres: in the location header.  I
> > later speculated that if we allow this, it forms a simple way to do
> > location-by-reference, and we could eliminate the raw HTTP(S) Get
> > proposal, with the parameter and registry entirely.  There was some
> > list traffic in favor of this.  Martin Thompson is skeptical that SIP
> > Presence Subscribe/Notify is actually allowed as a geopriv "using
> > protocol".  Several of think that it is.  If we don't get any more
> > comments on this proposal, we will allow sip/sips and pres, not have
> > any
> > http(s) resolution text, and not propose a parameter and accompanying
> > registry.
> >
> > If this does not reflect your position, please speak up now.
> >
> > Brian
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv
>
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>This message is for the designated recipient only and may
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>------------------------------------------------------------------------------------------------
>[mf2]
>
>_______________________________________________
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>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv

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Date: Wed, 26 Jul 2006 22:34:00 -0500
To: Henning Schulzrinne <hgs@cs.columbia.edu>,
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At 08:33 PM 7/26/2006 -0400, Henning Schulzrinne wrote:
>On a general note and generalizing a bit on Nadine's message below, I
>find the nanny tone of this discussion a bit peculiar ("No, I don't
>want you to play with that!"). I can see the technical rationale for
>requiring implementation of a limited set of choices for
>interoperability, but proactively ruling out other choices seems
>highly unusual for an IETF protocol spec. It is clear that only a
>relatively small number of people are participating in this
>discussion, so declaring consensus to rule out one URL scheme or
>another as 'evil bits' seems a bit peculiar.

A few folks have pointed out unique issues with HTTP and HTTPS wrt privacy 
issues. If those are true, do they need to be addressed in this document to 
satisfy Geopriv?  If yes, should a SIP doc discuss the uniqueness of HTTP's 
security stuff?  Or should that discussion be in an HTTP document?  Brian 
is suggesting that be in a HTTP focused document (likely in Geopriv), with 
the fairly unique ability to extend this document without going through the 
SIP Change process (and that's a bonus).


>On Jul 26, 2006, at 2:14 PM, Abbott, Nadine B wrote:
>
>>
>>I don't agree that there is a consensus to preclude the use of HTTP.
>>I would favor not placing your proposed restriction on the URI scheme
>>for expressing a location reference in the SIP location conveyance
>>draft; or at least including HTTP as a valid location-reference URI
>>type.
>>
>>Regards,
>>Nadine Abbott
>
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv

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Thanks for the comments, Cullen. Responses below.

Cullen Jennings wrote:

> 
> This looks good - a few comments - all very trivial
> 
> Does this update 3261? If so, nice if header on first page had that

I got asked that privately by others as well. I don't think it does. It 
uses the defined extension behaviors in RFC 3261 to add new 
functionality. Its not required functionality either; if a client can 
construct a contact that has the gruu property this spec is not needed 
(and indeed in a world without nat and firewalls, it could construct one).

> 
> Section 3 - Long lived definition. I'm really not sure what it means  to 
> say that a GRUU has a long life than any of the registrations or  that 
> it is viable for every. I'm not sure it changes behavior of  anything. 
> Is the important property that if the same AOR/Instance  registers two 
> different times it gets the same AOR? Anyways, I can  live with the text 
> exactly as is but I was not getting why it was  important.

Its important so implementors know that they shouldn't hand out a 
different gruu on each registration. If I turn on my softphone today, 
register, and then log out, and tomorrow do the same thing, I should get 
back the same gruu. This property is important in applications where 
people store the contact and use it later for a callback or something.

> 
> Figure 1: On the diagonal arrow between GRUU and contact, I think the  
> end near contact should have "0..1" and the end near the GRUU should  
> have "0..n" - right now this is swapped.

Hmm, I think it is correct as written. A GRUU has 0..n contacts and a 
contact is bound to a single gruu.


> 
> Section 6, page 12, 4th para or so ... you are talking about mid- dialog 
> requests. Might be nice to mention how a transaction stateful  proxy 
> knows if something is a middialog request or not.

These days I recommend topmost route URI over tags, since its easy to 
fool a proxy into thinking something is a mid-dialog request by faking a 
tag. Its harder to fake a route URI.

Text now reads:

<t> Mid-dialog requests will also be sent to GRUUs, as they are
included as the remote-target in dialog-forming and target refresh
requests and responses. However, in those cases, a proxy SHOULD only
apply services that are meaningful for mid-dialog requests, generally
speaking. This excludes screening functions, as well as forwarding
ones. A proxy can determine that a request is a mid-dialog request
based on the Route header field in the request it receives. If the
topmost URI matches one that the proxy placed into the Record-Route
header field of a dialog-forming request, then the request is a
mid-dialog request. </t>


> 
> Section 6, last para before section 7. I'm not sure I get how  anything 
> behaves differently from an unbound GRUU that exists vs the  GRUU does 
> not exist. Again, i don't have some huge problem with  current text but 
> when I read it, it makes me wonder if there is some  difference which I 
> should understand.

404 vs. 480. 404 if the GRUU simply doesn't exist; 480 if no contacts 
are bound to it. This is described later on.


> 
> Section 7.1.1.1 - There has been some problems with the way 3261  
> allowed a user to install forwarding rules using a REGISTER request.  
> Part of the problem is that it allows an amplification attack on  
> someone else that did not provide any consent. It makes me wonder if  
> now that we have the consent framework moving forward, do we really  
> want to continue down this path. I realize this is pretty orthogonal  
> problem to GRUU.

Yes, it is. I tend to agree that third party registrations are generally 
a bad idea these days. However, they exist in rfc3261 and we've not 
outlawed them. Thus I don't think this is an issue for GRUU to tackle. 
GRUU merely specifies how they interact.

> 
> Section 8.1.1 - 2nd para. You have "However, that AOR will be  rewritten 
> ..." - I found this hard to follow. Is it the request URI?  I'm not sure 
> how to fix but you might want to have a read of the end  of that para 
> and see if it is all clear to you.

Yes. The text is confusing. Hopefully this is clearer:

<t>If the UA instance has obtained multiple GRUUs for different AORs
as a result of a registration, it SHOULD use one corresponding to the
AOR used to send or receive the request. For sending a request, this
means that the GRUU corresponds to the AOR present in the From header
field. Furthermore, this means that the credentials used for
authentication of the request correspond to the ones associated with
that AOR. When a peer sends a request to a UA, that request will
arrive at the home proxy, and be forwarded to the UA. The GRUU placed
into the Contact header field of the response SHOULD correspond to the
AOR to which the request was targeted by the peer. Unfortunately, this
AOR is not directly present in the request when it is received by the
UA. It would have been placed into the Request-URI of the request when
sent by the peer. However, upon reaching the home proxy of the UA, the
Request-URI is rewritten with the registered contact. It is for this
reason that different contacts are needed for each AOR that an
instance registers against. When an incoming request
arrives, the Request URI can be examined. It will be equal to a
registered contact. That contact can be used to map directly to the
AOR, and from there, the correct GRUU can be selected.  </t>


Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Jeroen van Bemmel wrote:

> In http://www.jdrosen.net/papers/draft-ietf-sip-gruu-09.txt section 6 
> figure 1, a GRUU is shown to be associated with 0..n Contacts. Does this 
> include callee capabilities that were indicated in the registration of 
> that contact?

Yes.

> In other words, should the GRUU proxy be able to 
> reconstruct those capabilities, e.g. for caller preference processing 
> purposes?

Yes. Here is what I have done to clarify. In the section on register 
processing:

<t> Registration processing then continues as defined in RFC 3261. If,
after that processing, that contact is bound to the AOR, it also
becomes bound to the GRUU associated with that instance ID/AOR
pair. In addition, if that contact had callee capabilities associated
with it <xref target="RFC3840"/>, those capabilities continue to be
associated with the contact as it is bound to the GRUU. If, after that
processing, the contact was not bound to the AOR (due, for example, to
an expiration of zero), the contact is not bound to the GRUU either.
</t>


and in the section on proxy processing:

<t> Otherwise, handling of the GRUU proceeds as specified in RFC 3261
Section 16. For GRUUs, the abstract location service described in
Section 16.5 is utilized, producing a set of zero or more contacts,
each of which is associated with the same instance ID. If the proxy
supports the caller preferences specification <xref
target="RFC3841"/>, and the request contained Accept-Contact or
Reject-Contact header fields, caller preferences are applied to the
contact set.  If there are more than one contact with the same
instance ID, and those contact were registered using the procedures of
<xref target="I-D.ietf-sip-outbound"/>, those procedures are used to
select one. Otherwise, the most recently updated contact is used.
This produces zero or one contacts. The server MUST copy the "grid"
parameter from the Request-URI (if present) into the new target URI
obtained from the registered contact. If there was no "grid" parameter
in the Request-URI, the proxy MUST insert a "grid" parameter into the
new target URI obtained from the registered contact, and MUST omit a
value for the parameter. Note that the "gruu" URI parameter is not
copied. If the grid was already present in the contact bound to the
GRUU, it is overwritten in this process. If no contacts were bound to
the GRUU, the lookup of the GRUU in the abstract location service will
result in zero target URIs, eventually causing the proxy to reject the
request with a 480 (Temporarily Unavailable) response.  </t>



Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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At 08:28 PM 7/26/2006 -0500, Winterbottom, James wrote:
>In the case you mention below the reality is that if the target has
>moved during the course of routing, the target will either be at place B
>by the time dispatch occurs, or at point C which may be further away
>still.

Henning mentioned above that this is a routing timeframe (of maybe 1 
second), not a dispatch timeframe - and that it seems unlikely that routing 
would be different during that one second.  Regardless, the call get 
completed where the request was sent.

>At least in the case of a reference you have the possibility of getting
>the current location, which is not possible through any of the
>"by-value" means described thus far unless the caller is stationary!

Henning and Brian have discussed this already, and that is where SUB/NOT 
are useful for the ability to subscribe to that calling endpoint to find 
out where they are at any time during the call.  I mentioned that if the UA 
detects it has moved, it should send a SIP agreed upon UPDATE request (with 
a new l-by-v) to the PSAP informing it that the caller has moved.  This is 
something that belongs in the ECRIT phonebcp, and perhaps a more general 
non-emergency sipping doc.



>-----Original Message-----
>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>Sent: Thursday, 27 July 2006 3:14 AM
>To: Andrew Newton
>Cc: geopriv@ietf.org; IETF SIP List
>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
>location-conveyance
>
>For location-based routing case, the likelihood of significant updates
>in the roughly one second that a call might take from caller to callee
>seems unlikely. UAS subscription and polling is clearly more valuable,
>so maybe we need to distinguish these cases. (As discussed earlier,
>mid-routing updates may lead to bad outcomes since the call might have
>been sent to a place A that doesn't know what to do with location update
>
>n+1 since that new location is really handled by place B.)
>
>Andrew Newton wrote:
> > Henning Schulzrinne wrote:
> >>> I believe there are two technical arguments: one is legacy equipment
>
> >>> (the issue that I'll be facing for quite some time), and
>environments
> >>> where the location is generated by the network.  For the latter,
> >>> insertion of the Location header only in emergency situations still
> >>> puts the user in control for non-emergency situations.
> >>>
> >>>
> >>
> >> For the latter, the combination of the in-progress L7 protocols, DHCP
>
> >> and LLDP should presumably allow the UAC to obtain location and
>insert
> >> it by value. Is there a use case not covered by that?
> >
> > As I said, where location comes from the network.  In this scenario,
> > there are two possibilities, both up to the user: 1) the UAC
>subscribes
> > to the location from the network and puts it by value in the call
> > flow... meaning all location updates go through the UAC, or 2) the UAC
>
> > hands out a reference so updates can be independent of the UAC or
> > updates based on subscription come from the UAC but are independent of
>
> > the call.
> >
> > -andy
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv
>
>------------------------------------------------------------------------------------------------
>This message is for the designated recipient only and may
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>------------------------------------------------------------------------------------------------
>[mf2]
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On Jul 26, 2006, at 11:27 PM, James M. Polk wrote:

> I think Geopriv really needs to look at HTTP as a RFC 3693 "using  
> protocol" to tease out any unique characteristics with that  
> protocol.  That discussion and review shouldn't bog this document  
> down.  Does this sound fair?

Most definitely.  We've had discussions on HTTP as a using protocol  
for 3 years in GEOPRIV, HELD being the latest incarnation.  The first  
proposals ran into serious privacy concerns.  Anything that simply  
says "use HTTP" is an attempt to work around answering the serious  
privacy concerns raised in GEOPRIV.

-andy

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From: "Dawson, Martin" <Martin.Dawson@andrew.com>
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Sounds like a very pragmatic approach Henning - I agree.

Cheers,
Martin

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Thursday, 27 July 2006 1:25 PM
> To: Brian Rosen
> Cc: IETF SIP List; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
>=20
> My suggestion would be
>=20
> - to focus the draft on just defining the Location header syntax,
> without ruling out any particular mechanism or scheme; this would
> make the draft really short (which I would find helpful).
>=20
> - and then add one section to the draft that prototypically defines
> what a particular scheme that wants to appear in the Location header
> needs to define, using the non-controversial cid mechanism as the
> example. Other, separate, drafts can then define other modes,
> following the same template. Documents such as the phone-bcp can then
> pick some suitable must-implement subset and make emergency-call-
> specific recommendations, rather than sprinkling these throughout the
> conveyance document and repeating some in the phone-bcp. Most of
> these are, in any event, operational recommendations, not for
> ensuring protocol-level interoperability.
>=20
> That way, the conveyance draft can be extremely short (even shorter
> if the requirements are omitted...) and can get done, and we avoid
> engineering operational requirements on the fly, commingling
> emergency and non-emergency cases, as seems to be happening now.
>=20
> I think protocol documents should define what's needed for
> interoperability; BCPs can make more specific, situation-specific
> recommendations. Commingling the two seems to be a recipe for
> deadlock or arbitrary, non-technically-grounded decisions that are
> more lawyer-like than engineering-like ("you did not mention
> technique X in some meeting N years ago and thus you can't bring it
> up now!")
>=20
> Henning
>=20
> On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
>=20
> > At this point, I think we need some chair guidance and by that I
> > think it's
> > a combo of geopriv and sip chairs.
> >
> > We seem to have a couple of significant open questions.  I think
> > most of the
> > other stuff (like the "allow sending to router") will converge.
> >
> > There seems to be some opposition to allowing sip/sips/pres: in a
> > Location
> > Header.  The opposition seems to be that despite 4119, somehow, SIP
> > (presence subscription) is not a "using protocol".  I "think" a
> > ruling from
> > the bench on that would work.
> >
> > There is also opposition to defining the allowed schemes as a
> > specific list,
> > with a defined (easy) extension mechanism.
> >
> > Related to that, there are some who want location-conveyance to
> > define an
> > HTTP based location dereference mechanism, and on the opposite end,
> > we have
> > people saying we haven't even agreed to support any form of
> > location-by-reference.
> >
> > The authors would like to know what to do now.
> >
> > Brian
> >
> >> -----Original Message-----
> >> From: Andrew Newton [mailto:andy@hxr.us]
> >> Sent: Wednesday, July 26, 2006 9:58 PM
> >> To: Winterbottom, James
> >> Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning
> >> Schulzrinne
> >> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> >> conveyance
> >>
> >>
> >> On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
> >>
> >>> Andy,
> >>>
> >>> RFC-4119 says "However, the usage of this location object format
is
> >>> not
> >>>    limited to presence-using protocols-- any protocol that can
> >>> carry XML
> >>>    or MIME types can carry PIDF."
> >>>
> >>> It seems to me that restricting the allowed schema in what is
> >>> supposed
> >>> to be a location conveyance specification is in opposition to this
> >>> statement.
> >>
> >> Did somebody say that HTTP was a presence-using protocol?
> >>
> >> -andy
> >>
> >> ps.  you can find my location with the following reference URIs:
> >>
> >> gopher://fatchance.hxr.us/you-still-have-a-gopher-client
> >> news:alt.locations.people.secretly-being-watched
> >> telnet://screen-scrapers.are.us/
> >> dns:cached-publicly-around-the-world.com?type=3DHOST
> >>
> >>
> >>
> >> _______________________________________________
> >> Geopriv mailing list
> >> Geopriv@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/geopriv
> >
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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At 09:42 PM 7/26/2006 -0400, Andrew Newton wrote:
>This is yet another rule,

that is if data URI is mentioned...

>adding more complexity.  You can pass by
>value either via MIME or data: URI.  If you pass by value using the
>data: URI, you have to restrict ruleset references.
>
>-andy
>
>_______________________________________________
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>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv

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Subject: Teasing apart positions (was Re: [Sip] RE: [Geopriv] Consensus on
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On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
> The authors would like to know what to do now.

Let's start by trying to tease apart certain positions.

Given Jon Peterson's message:
http://www1.ietf.org/mail-archive/web/sip/current/msg15537.html

Does anybody still believe that subscriptions to PIDF-LOs with pres:  
requires more definition?

-andy

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At 08:48 PM 7/26/2006 -0500, Winterbottom, James wrote:
>To what do you send the SIP update and for what do you request the
>update?

to the latter first: UPDATE is not requested, it is a request message just 
like INVITE.

If the PSAP wants to subscribe to the caller, that's a separate transaction 
in SIP.

to the former, are you asking what's the destination address the UPDATE 
goes to, or what triggers the UPDATE message?  Your phrasing could be to 
either the way I read it. I'll answer both below as best I can:

To a destination address, if before the UAC receives a 200 OK with a 
Contact header (indicating the PSAP URI), the UPDATE needs to be built just 
like the original INVITE (with the new location) and sent to the 
urn:service:sos and get routed to the PSAP.  If this results in a new PSAP 
due to movement, the INVITE times out and the new INVITE is to the 
urn:service:sos with a new location and the call goes through.  We can't 
build this to the corner case of someone moving that fast, as much as that 
won't make some happy.

If the 200 OK has been received by the UAC, the UPDATE will go at the PSAP 
URI in the Contact header of that 200 OK, and therefore go straight at the 
PSAP, the fastest way there (barring any route headers that are forced into 
the request).

Due to a trigger, this is UAC device specific as it is the device that 
senses a sufficient change in its location to send a new location.  This 
will in all likelihood be after the 200 OK has been received, even after 
the call has been established, and should follow what is in the paragraph 
above this one to the PSAP.

It has been suggested that a reINVITE can be sent with the new location 
*after* the call is eestablished, and that's fine, but not what the SIP WG 
agreed to several IETFs ago, but it would work in either case.



>-----Original Message-----
>From: Andrew Newton [mailto:andy@hxr.us]
>Sent: Thursday, 27 July 2006 11:47 AM
>To: Winterbottom, James
>Cc: Henning Schulzrinne; geopriv@ietf.org; IETF SIP List
>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
>location-conveyance
>
>
>On Jul 26, 2006, at 9:28 PM, Winterbottom, James wrote:
>
> > In the case you mention below the reality is that if the target has
> > moved during the course of routing, the target will either be at
> > place B
> > by the time dispatch occurs, or at point C which may be further away
> > still.
> > At least in the case of a reference you have the possibility of
> > getting
> > the current location, which is not possible through any of the
> > "by-value" means described thus far unless the caller is stationary!
> >
>
>What about a SIP UPDATE?  Section 5.3 of the draft.
>
>-andy
>
>
>
>
>
>------------------------------------------------------------------------------------------------
>This message is for the designated recipient only and may
>contain privileged, proprietary, or otherwise private information.
>If you have received it in error, please notify the sender
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>this email is prohibited.
>------------------------------------------------------------------------------------------------
>[mf2]
>
>_______________________________________________
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>Geopriv@ietf.org
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At 11:37 PM 7/26/2006 -0400, Andrew Newton wrote:

>On Jul 26, 2006, at 11:27 PM, James M. Polk wrote:
>
>>I think Geopriv really needs to look at HTTP as a RFC 3693 "using
>>protocol" to tease out any unique characteristics with that
>>protocol.  That discussion and review shouldn't bog this document
>>down.  Does this sound fair?
>
>Most definitely.  We've had discussions on HTTP as a using protocol
>for 3 years in GEOPRIV, HELD being the latest incarnation.  The first
>proposals ran into serious privacy concerns.  Anything that simply
>says "use HTTP" is an attempt to work around answering the serious
>privacy concerns raised in GEOPRIV.

is this said with your WG chair hat on?

It sounds pretty definitive to these ears...


>-andy
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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James,

By the restricting of the schema you are saying MUST NOT as that is what
is defined in the ABNF, so if I implement the specification anything
else is invalid. I agree wholeheartedly with Henning's last suggestion,
leave the schema open but only describe the methods you are going to
specifically recommend.

Cheers
The other James

-----Original Message-----
From: James M. Polk [mailto:jmpolk@cisco.com]=20
Sent: Thursday, 27 July 2006 1:28 PM
To: Winterbottom, James; Abbott, Nadine B; Brian Rosen; sip@ietf.org
Cc: geopriv@ietf.org
Subject: RE: [Geopriv] Consensus on changes to location-conveyance

James

We won't say anything like "MUST only use sip/sips/pres/cid, and MUST
NOT=20
=2E..".  I don't believe we will take a negative tone with what isn't
allowed=20
in this respect, but we will state explicitly what is defined for use.

We will likely state something like this:  ' this document defines usage
of=20
the sip, sips, pres and cid schemas for location conveyance, and here's
how...'

I think Geopriv really needs to look at HTTP as a RFC 3693 "using
protocol"=20
to tease out any unique characteristics with that protocol.  That=20
discussion and review shouldn't bog this document down.  Does this sound
fair?

At 06:31 PM 7/26/2006 -0500, Winterbottom, James wrote:
>Brian,
>
>I don't care if you don't mention how to de-reference URI types other
>than sip/sips/cid/pres in the document or not. But I do not want the
>schema artificially constrained to exclude URI types such as HTTP.
There
>is clearly not consensus to justify this exclusion, so please leave
>schema unchanged and make your recommendations in the text.
>
>Cheers
>James
>
>
>
>-----Original Message-----
>From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
>Sent: Thursday, 27 July 2006 7:44 AM
>To: Brian Rosen; sip@ietf.org
>Cc: geopriv@ietf.org
>Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>
>Brian,
>
>Actually, I was thinking that in other industry work which is trying to
>leverage and provide transitional strategies to migrate to using SIP to
>support emergency services, the and long-term solutions that we have
>defined assume a capabity to provide location-by-reference.  I think it
>would be very useful to have this defined in a standard SIP header
>mechanism in such a way that we need not redefine migratory solutions
>that have begun to be implemented. If an HTTPS URI can be used as the
>location-by-reference, this will be helpful, I think.
>
>I don't understand your urgency to make location-by-reference only an
>all-sip-all-the-time thing.
>And from all the other emails on this topic, it didn't seem to me that
>there is a consensus to do this.
>
>Nadine
>
>-----Original Message-----
>From: Brian Rosen [mailto:br@brianrosen.net]
>Sent: Wednesday, July 26, 2006 3:15 PM
>To: Abbott, Nadine B; sip@ietf.org
>Cc: geopriv@ietf.org
>Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>
>Nadine
>
>Are you saying that it's your opinion sip-location-conveyance MUST
>define an HTTP dereference procedure?
>
>If you do, why?  What is the use-case or requirement?
>
>Would it not be pretty reasonable to require a separate document that
>defines a location-by-reference mechanism, protocol, and geopriv "using
>protocol" analysis?  If that document specifically extended what was
>allowed in the location header, would that not be reasonable?
>
>I think we have to include "A" dereference mechanism in the
>location-conveyance document.  It seems just including sip/sips/prez is
>a reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.  That
>certainly does not preclude another document describing another way.
>
>Doesn't it seem just a little weird to have a sip-location-conveyance
>document describing any kind of http based location conveyance
>(de-reference) protocol, especially one created whole cloth without any
>other parts?
>
>Doesn't it seem in keeping with the geopriv approach that in order to
>have another "using protocol" it has to have a geopriv review?  That is
>what leads to restricting the URI schemes.  Would you prefer to let it
>be wide open, so any protocol anyone thinks about can be used with the
>location header?  I can't see geopriv going along with that.  We're
>going to make it a one line ABNF to add a new scheme to the list of
>allowed schemes.
>
>It seems to me that if it's desired to define some bare-bones HTTP
>location by reference protocol, then that should be a separate
document,
>which can easily extend the allowable schemes in the Location header.
>I'm not sure it might be better to define ALL of that in the L7
>protocol, but I wouldn't be opposed to a simple "raw GET" RFC as long
as
>it got a full geopriv review.
>I just don't see where it makes any sense to force that into a sip
>document.
>
>Brian
>
> > -----Original Message-----
> > From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
> > Sent: Wednesday, July 26, 2006 2:14 PM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: RE: [Geopriv] Consensus on changes to location-conveyance
> >
> >
> > I don't agree that there is a consensus to preclude the use of HTTP.
> > I would favor not placing your proposed restriction on the URI
scheme
> > for expressing a location reference in the SIP location conveyance
> > draft; or at least including HTTP as a valid location-reference URI
> > type.
> >
> > Regards,
> > Nadine Abbott
> >
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Tuesday, July 25, 2006 1:38 PM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: [Geopriv] Consensus on changes to location-conveyance
> >
> > I have put out a number of emails on proposed changes to location
> > conveyance.  Some of them have sparked a number of comments.  My
> > summary of consensus on these issues is:
> >
> > 1. We proposed to explicitly disallow use of a Data URI, which would
> > allow specification of a Location-by-value in a header.  There was a
> > lot of list traffic on this, currently in the state where Keith
asked
> > for a real use case for it.  So far, we don't see a use case that
> > cannot be provided by location-by-reference in a header.  If we
don't
> > get a good use case, I think consensus is to disallow
> > location-by-value in a header.
> >
> > 2. Dereferencing location-by-reference.  We had initially proposed
to
> > document a process to use a raw HTTP(S) Get to dereference.  There
was
>
> > a lot of discussion, and then I made a new proposal, see item 6.
> >
> > 3. We proposed to allow a Reason header with a 424 Bad Location
> > response.  This seems to be acceptable to the WG.  Geopriv will
create
>
> > and populate the registry in a separate draft.
> >
> > 4. We proposed to explicitly allow hop-by-hop TLS security for
> > location when the endpoint wished to allow routing based on
location.
>
> > We further proposed to interpret "Do not Distribute" flags as
allowing
>
> > sending of the location to the routing database (LoST for example).
> > Hannes pointed out that there was a proposal to allow two Location
> > headers, one of which could be used for routing and another for
onward
>
> > forwarding to the recipient.  Hannes' message seemed to agree that
"Do
>not distribute"
> > would allow sending of the location to the routing-by-location
> > database, but would obligate the proxy that did that to remove the
> > location before it forwarded the message.  There were no other
>messages on this thread.
> > We conclude that allowing hop-by-hop security for location based
> > routing is acceptable, and Do Not Distribute DOES allow forwarding
to
> > a routing database, but requires that the proxy doing the location
> > based route to remove the location it uses for that purpose.
> >
> > 5. We proposed to create a parameter and a registry to distinguish
> > multiple ways to dereference a location where HTTP(S) was the
>transport.
> > Resolution of this issue depends on resolution of issue 6.
> >
> > 6. We proposed to allow sip/sips and pres: in the location header.
I
> > later speculated that if we allow this, it forms a simple way to do
> > location-by-reference, and we could eliminate the raw HTTP(S) Get
> > proposal, with the parameter and registry entirely.  There was some
> > list traffic in favor of this.  Martin Thompson is skeptical that
SIP
> > Presence Subscribe/Notify is actually allowed as a geopriv "using
> > protocol".  Several of think that it is.  If we don't get any more
> > comments on this proposal, we will allow sip/sips and pres, not have
> > any
> > http(s) resolution text, and not propose a parameter and
accompanying
> > registry.
> >
> > If this does not reflect your position, please speak up now.
> >
> > Brian
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv
>
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Subject: Re: [Sip] draft-ietf-sip-gruu-09.txt ready for IESG?
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inline.

Dale.Worley@comcast.net wrote:

> I fear this is an issue that was disposed of before and I don't
> realize it:
> 
> I think the I-D needs to make clearer that when sending a request with
> a GRUU that the request must be sent via an outbound proxy for the
> GRUU's domain.  This is needed to make the machinery work, but isn't
> stated explicitly as a UA requirement, it's implied in a section of
> the I-D that appears to be a proxy requirement.

Actually, GRUU works just fine if there is NO outbound proxy at all. 
Consider the case of a call between Alice and Bob. Alice sends the 
INVITE directly to Bob's proxy, which record-routes and forwards to Bob. 
Now, when Bob sends a mid-dialog request, the r-uri has the gruu and the 
Route points to the proxy in Bob's domain. The request goes to that 
proxy. From there, as there are no more route headers, the r-uri is 
looked up, which goes to the home proxy in Alice's domain. The gruu is 
translated by the home proxy to the registered contact and the request 
is routed there.

What doesn't work so well is if a user has multiple outbound proxies for 
multiple providers, and it ends up sending a request to provider X with 
a Contact whose GRUU is from provider Y. In this case we will end up 
with the nasty spiral that was in early versions of GRUU. It might still 
work, but there are issues. I consider this a pretty far fetched corner 
case. However, the assumptions here are worth mentioning.

I've added the following text in the section on using GRUU:

<t> In addition, if the UA is making use of an outbound proxy, the
GRUU that is placed into the Contact header field of a request MUST be
one that was obtained from the same domain as the outbound proxy. This
assumes that a user agent is aware of the domain associated with each
outbound proxy. This specification does not require the use of an
outbound proxy; it merely introduces a requirement in cases where one
is used.  </t>



> 
> There's also the question of how the UA discovers the correct outbound
> proxy address.  Strictly speaking, without a standardized mechanism,
> different vendors could have incompatible implementations.  I admit
> that this isn't much worse than the UA configuration problem as it now
> stands.  Perhaps a good dodge is to allow (absent other configuration)
> the UA to use the proxy for GRUU's domain as an outgoing proxy for
> requests using that GRUU as a Contact?

Per above, you don't need an outbound proxy. There is still this issue 
of, if you use one, how did you discover it. Thats a recognized open 
issue right now with sip-outbound, and not one that I think we need to 
address in this specification.

Thanks,
Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Subject: [Sip] Re: draft-ietf-sip-gruu-09
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Jonathan Rosenberg writes:

 > If the
 > topmost URI matches one that the proxy placed into the Record-Route
 > header field of a dialog-forming request, then the request is a
 > mid-dialog request.

jonathan,

how can proxy tell apart a pre-loaded Route header from proxy placed?

-- juha

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inline.

Dale.Worley@comcast.net wrote:

> I've gone over draft-ietf-sip-gruu-09, and it's a fine piece of work,
> I must say.  As far as I can tell, every doubt I've had about the GRUU
> machinery has been resolved (except for the "edge router" problem,
> which has been dealt with via sip-outbound). 

Thanks!

> I've written up some
> editorial changes to the I-D.  I don't think that any of these change
> the protocol, but this is such an important I-D that it ought to be
> described as rigorously as possible.
> 
> ----------------------------------------------------------------------
> 
> There seems to be disagreement as to what the plural of GRUU is --
> GRUU or GRUUs.  E.g., section 13, page 31 uses both "Furthermore,
> GRUUs do not address" and "These GRUU will be passed".  There is a
> similar inconsistency regarding the plural of AOR.

Well, this topic has come up for the term URI, which is the last U in 
GRUU. So it would stand to reason that if the plural of URI is URIs, 
then the plural of GRUU is GRUUs. I found some email threads on this 
topic, for example:

http://lists.w3.org/Archives/Public/www-tag/2002Aug/0124.html

and apparently this was an open issue in the preparation of the revision 
to 2396:

http://www.gbiv.com/protocols/uri/rev-2002/issues.html#023-URI-plural

the conclusion appears to be that the plural of URI is URIs. So, I 
suggest we follow suit and have the plural of GRUU be GRUUs.


> 
> Section 3, page 5:  "the following is true" should be "the following
> three properties are true", in order to ensure there is no ambiguity
> about the definition.

fixed.

> 
> Section 4.3, page 7:  "Each tuple also has" s.b. "Each tuple has", as
> there is no preceding discussion of anything that "each tuple has".
> (This actually caused me some confusion the first time I read it!)

fixed.

> 
> Section 5, page 9:  ", and to easily recognize that the URI has the
> GRUU property" s.b. omitted, as it is the "gruu" parameter that does
> that, not the "opaque" parameter.

Well, thats debatable. The owner of the domain could key off just the 
opaque parameter in fact. However, I just deleted this bit as you 
suggest since it can be misinterpreted.

> 
> Section 5, page 9:  "that results from stripping out the "opaque" URI
> parameter" s.b. "that results from stripping out the "opaque" and
> "gruu" URI parameters".

good catch.

> 
> But that isn't really the statement we want either, since URI
> "equivalence" is not affected by a URI parameter present in one URI
> and not in the other.  See RFC 3261, section 19.1.4.  That is,
> 
>    sip:carol@chicago.com
>    sip:carol@chicago.com;opaque="dRfHgbHjGhImBv";gruu
> 
> are equivalent already.  But unfortunately so are:
> 
>    sip:carol@chicago.com
>    sip:carol@chicago.com;opaque="dRfHgbHjGhImBv";gruu;hop-sip-and-jump
> 
> and we want to rule out the latter pair.

Interesting. What this means is that, based on URI equivalence, a GRUU 
will in fact equal its AOR. Hmm. Its not obvious to me that this is a 
bug or a feature. It really depends on what is a reasonable basis for 
URI equality. One reasonable definition is that two URI should be 
equivalent only if requests targeted towards them would reach the same 
user, where 'same' refers to the ability to authenticate with a specific 
credential. With that definition, the GRUU should be equal to the AOR 
based on URI equality.

Since I don't see any real issue with the AOR and GRUU being equal, I 
think all we need to do is state that this is the fact. I changed the 
paragraph we're talking about to read:

<t>
If the GRUU contains an "opaque" URI parameter, the URI that results
from stripping out the "opaque" and "gruu" URI parameters MUST be
equivalent to the AOR associated with the GRUU. Indeed, the GRUU
itself will also be equivalent to the AOR based on URI comparison,
since the "gruu" and "opaque" parameters will be ignored during
comparison, being present in one URI (the GRUU) and not the other (the
AOR).
</t>



> 
> Section 7.1.1.1, page 13:  "characteristic [8].  The instance ID MUST
> identify the UA that is performing the registration."
> s.b. "characteristic [8], whose value MUST be the instance ID that
> identifies the UA being registered."  This is needed to explicitly
> state what the value of the feature tag is, as well as to be correct
> for third-party registrations.

fixed.

> 
> Section 8.1.1, page 18:  "the grammar for the parameter" s.b. "the
> grammar for the parameter given in section 10".

fixed.

> 
> Section 8.1.1, page 18:  "(which is the case for administratively
> constructed GRUUs)" s.b. omitted -- as far as I can tell, this isn't
> always true.

I'd like to give an example of what I'm talking about here. I'll say, 
"which can be the case for administratively constructed GRUUs, for example)

> 
> Section 8.2.2, page 21:  "location service for that user"
> s.b. "location service for that AOR", since "user" is not used
> elsewhere in that paragraph.

ok.

> 
> Section 10, page 23:  "The grammar for string-value is obtained from
> [8]." s.b. omitted, since "string-value" is not mentioned elsewhere.

Good catch. Leftovers for when the instance ID was defined here.

> 
> Section 10, page 23:  ""gruu" EQUAL LDQUOT (SIP-URI / SIPS-URI)
> RDQUOT" is not what we want, I think.  I think the semantics we are
> looking for is ""gruu" EQUAL quoted-string, where the value
> represented by the quoted-string is a SIP-URI or SIPS-URI".  (That
> latter could be represented in BNF, but that would make it less
> clear.)

Thats what the grammar says now. I don't see why its not clear.

> 
> Due to the details of the grammar, the new formulation differs from
> the original only in some unusual cases, such as allowing:
> 
>      Contact: <...>;gruu="sip:gruu\~123456789@example.com;gruu"
> 
> (The GRUU is <sip:gruu~123456789@example.com;gruu>.)
> 
> But it seems to me that it is non-orthogonal (and inviting trouble) to
> have the grammar for c-p-gruu forbid backslash-escapes of characters
> that do not require them, when the grammar for generic-param allows
> them.

Egads. I disagree here. The SIP URI itself cannot contain a quote, so 
there is no need to ever use backslash escaping. I think folks might get 
confused by backslash escaping here. If you just put the unescaped URI 
in, you'll never have a problem. Why allow escaping?

> 
> Section 12, page 29:  Messages 13, 14, 15, and 16 (from figure 2, page
> 26) are not shown.  This may be intentional, but probably should be
> noted explicitly, as the other requests seem to be shown.

Its intentional. Not all messages are shown, but most are at least 
mentioned. I now mention these.

Thanks,
Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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That's right - and those mechanisms by which the URI remains anonymous
and temporary and access rights to the anonymous form of the location
are controlled are quite extensive considerations.

I understand the conveyance draft just says how to get the location
object and/or reference from one signaling peer to another. It's asking
to much of this specification to anticipate all the possible mechanisms
for applying security to privacy in the by-reference form and thereby
restricting the form of the reference to a particular protocol. This
should be out of scope for the conveyance protocol.

Cheers,
Martin

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Wednesday, 26 July 2006 4:46 PM
> To: Brian Rosen; 'Marc Linsner'; Peterson, Jon; 'Andrew Newton';
Rosen,
> Brian
> Cc: sip@ietf.org; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
>=20
> >I don't really understand why you think location by reference is so
much
> of
> > a concern.
>=20
> Because location by reference involves storage of personal information
on
> a
> server. That suddenly makes all sorts of regulations apply, the points
I
> mentioned address only a small part of that. Having an anonymous URL
for
> example often makes a difference, because then the information is no
> longer
> directly linked to a particular person.
>=20
> When an endpoint inserts location by value, you can reasonably argue
that
> opt-in consent requirements (which many countries have) are fulfilled,
> provided the endpoint gives the user control over whether or not
location
> is
> inserted. Same holds for the endpoint inserting a flag asking a proxy
to
> fill in the location (by value).
> For location-by-reference this is much more fuzzy. You probably don't
need
> a
> lecture on privacy regulations, but e.g.
> http://www.isoc.org/briefings/015/index.shtml (from 2003 but still
> applicable IMO) illustrates the mess.
>=20
> A while ago I submitted a draft on Temporarily Routable User Agent
URIs
> (http://www.ietf.org/internet-drafts/draft-jbemmel-sip-truu-02.txt),
which
> provides anonymous, temporary URIs that can be explicitly invalidated.
You
> may want to consider something similar for location-by-reference
>=20
> Regards,
>=20
> Jeroen
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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From sip-bounces@ietf.org Thu Jul 27 03:33:29 2006
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Would be interested to know how proxies treat the following request:

CANCEL sip:user@example.com
Via: {valid data};branch=3D{cookie}random
Route: <sip:127.0.0.1;lr>
{rest valid data}

Same for "localhost", "[::1]" and other IPv6 local addresses, and other req=
uests than CANCEL

Regards,

Jeroen=


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>> - Proxy-Require/Proxy-Authorization won't appear in CANCEL or ACK to
>> non-2xx (since these are generated hop-by-hop, and also because both
>> are not allowed in CANCEL)
>
> Why would you want to do a loop detection for CANCELs and ACKs
> to non-2xx?  In the latter case, you can't send a response anyway.
> As long as the branch parameter for CANCELs and ACKs to non-2xx
> matches a pending INVITE transaction, the state machines can be
> updated accordingly.

Because CANCEL and ACK can loop like any other request. The concern mostly 
applies to *stateless* proxies (stateful proxies mostly dont "forward" but 
"generate" CANCEL and ACK to non-2xx), but it matters also to a stateful 
proxy that receives a CANCEL with no matching INVITE ST (RFC3261 says "MUST 
forward statelessly").

For the latter case, I think it might be a good idea to create a 
servertransaction for the CANCEL, and forward statefully (using the same 
branch parameter as received) to catch any loops in an easy way

For the fork-loop-fix case I guess most concern goes to INVITEs, since they 
are the only requests that fork and thus cause amplification. But given that 
also a forking (=>stateful) proxy can receive an ACK to non-2xx or CANCEL 
with no matching INVITE servertransaction, it should be mentioned.

A looping ACK should obviously not be responded to, but it should be 
dropped.

Regards,

Jeroen 


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At 05:34 27/07/2006, Jonathan Rosenberg wrote:
>>Section 6, page 12, 4th para or so ... you are talking about mid- dialog requests. Might be nice to mention how a transaction stateful  proxy knows if something is a middialog request or not.
>
>These days I recommend topmost route URI over tags, since its easy to fool a proxy into thinking something is a mid-dialog request by faking a tag. Its harder to fake a route URI.

Faking tags is easy, but is there a way to exploit that? I mean a UAS that received
a request with to-tag which matches no dialog just discards it.

Faking Routes may not be that difficult either, unless implementors choose to append
some cryptographically secured "this-was-me" tags to it.

Anyway, I think there can be quite some ways to do the same thing (some folks I am sure
will prefer a dialog state silo too), and if we show a way to do this test, it should
be properly labeled as an example. 

[...]


>>Section 7.1.1.1 - There has been some problems with the way 3261  
>>allowed a user to install forwarding rules using a REGISTER request.  
>>Part of the problem is that it allows an amplification attack on  
>>someone else that did not provide any consent. It makes me wonder if  
>>now that we have the consent framework moving forward, do we really  
>>want to continue down this path. I realize this is pretty orthogonal  
>>problem to GRUU.
>
>Yes, it is. I tend to agree that third party registrations are generally a bad idea these days. 

I think there are actually three orthogonal things:

a) uploading forwarding rules using REGISTER. (which is not IMO 3rd-party registration, as long
   as identity of the registering party equals that of URI owner). I don't know if this is worth
   documenting. IMO, this is an aspect to be addressed by registrar's policy. A registrar may
   for example decide to deny domain names in contacts, it may deny contact addresses that are
   IP-wise different from transport addresses from which REGISTER came from, etc. It could also
   accept contacts in REGISTER that includes P-ISwearThisIsNotForwarding: Yes :-)) There are
   subscribers and consequently provides who consider this a feature. We specify protocols and 
   not policies, don't we. 

b) 3rd-party registration, which is IMO the case, when the identity of registering party (i.e.,
   username as in digest) is not the same as owner of URI. E.g., if jiri@iptel.org with valid
   credentials puts jdr@iptel.org in From/To. This is IMO a policy issue too, and thus I think 
   it is appropriate to leave it as is. Already the fact that there is no SIP data model as part
   of the standard would make possible fixes kind of difficult; i.e., there is actually no clear 
   definition for 3rd party registration without precise notion of URI ownership. 

c) GRUU. (no particular comments on this other than it is different from a and b)

-jiri

--
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From sip-bounces@ietf.org Thu Jul 27 05:28:18 2006
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That's not a protocol aspect. That's an aspect of server's policy. That's because use
of localhost in Route may be perfectly valid in some cases. My suggestion for users 
of SIP servers is then to ask their favorite vendors about what types of policies they 
can set to deal with this case, if it is not desired.


-jiri

At 09:31 27/07/2006, Jeroen Van Bemmel wrote:
>Would be interested to know how proxies treat the following request:
>
>CANCEL sip:user@example.com
>Via: {valid data};branch={cookie}random
>Route: <sip:127.0.0.1;lr>
>{rest valid data}
>
>Same for "localhost", "[::1]" and other IPv6 local addresses, and other requests than CANCEL
>
>Regards,
>
>Jeroen
>
>_______________________________________________
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>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

--
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At 08:31 27/07/2006, Juha Heinanen wrote:
>Jonathan Rosenberg writes:
>
> > If the
> > topmost URI matches one that the proxy placed into the Record-Route
> > header field of a dialog-forming request, then the request is a
> > mid-dialog request.
>
>jonathan,
>
>how can proxy tell apart a pre-loaded Route header from proxy placed?

I would say there are far too many ways to do it. One of those is the
proxy puts in the Record-Route header-fields its signature "I put it here".
This can have many various forms, up to proxy vendor or configurator to choose
one. We typically configure what we call "avp-cookies" in Record-route header field 
to capture some fragments of dialog state. If such a piece of information 
appears later in Route, we know it is our "own" route. If not, it is not
(or there is a crappy UA which strips some parts of the HF :-))

This is more robust than looking at hostname/IP, because that may be too
amibguous (it is hard to differentiate between preloaded/record-routed,
if IP address is used in multidomain environments it is difficult to
assigne the request to a domain, etc.) Sometimes for similar cases,
you don't actually the signature to be an IP address -- you may have 
a server farm, and all you want to know is whether any of farm members 
did record-route or not.

The record-routing information may include various things to provide
better value: name of initially "dialed" domain, cryptographic checksum
to avoid forged record-routing information, and once that begins to bee
all too much, just index referring to database full of call state.

-jiri


--
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From: Andrew Newton <andy@hxr.us>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Thu, 27 Jul 2006 08:32:54 -0400
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On Jul 27, 2006, at 12:08 AM, James M. Polk wrote:

> At 11:37 PM 7/26/2006 -0400, Andrew Newton wrote:
>
>> On Jul 26, 2006, at 11:27 PM, James M. Polk wrote:
>>
>>> I think Geopriv really needs to look at HTTP as a RFC 3693 "using
>>> protocol" to tease out any unique characteristics with that
>>> protocol.  That discussion and review shouldn't bog this document
>>> down.  Does this sound fair?
>>
>> Most definitely.  We've had discussions on HTTP as a using protocol
>> for 3 years in GEOPRIV, HELD being the latest incarnation.  The first
>> proposals ran into serious privacy concerns.  Anything that simply
>> says "use HTTP" is an attempt to work around answering the serious
>> privacy concerns raised in GEOPRIV.
>
> is this said with your WG chair hat on?
>
> It sounds pretty definitive to these ears...

I don't believe working group chairs are given any special powers  
that such a decision is in their hands.  Any individual participant  
can approach an AD or the IESG either informally or via appeal about  
attempts to work around working group consensus.  Such things are  
taken seriously by the IESG, to the point that they can stop the  
publication of an RFC sent directly to the RFC Editor on this basis.

-andy

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From sip-bounces@ietf.org Thu Jul 27 08:38:15 2006
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From: Andrew Newton <andy@hxr.us>
Subject: Re: Teasing apart positions (was Re: [Sip] RE: [Geopriv] Consensus
	onchanges to location-conveyance)
Date: Thu, 27 Jul 2006 08:36:27 -0400
To: "Thomson, Martin" <Martin.Thomson@andrew.com>
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Martin, you did not answer my question.  You are right that HTTP as a  
using protocol is a separate question, one which I will ask about.

-andy

On Jul 27, 2006, at 1:00 AM, Thomson, Martin wrote:

> Since I raised the question, I'll answer.
>
> We have a set of requirements on a using protocol in RFC 3693.
>
> I believe that HTTP with proper constraints (TLS, etc...) can meet,  
> or already meets, those requirements.  That is irrespective of any  
> decisions that need to be made about secondary concerns like  
> whether HTTP is desirable, optimal, deployable etc...  However, it  
> is clear that people aren't happy with this position and are  
> requesting that this be documented, as an RFC.  Just to get it all  
> out in the open.  Presumably this would be a standards track RFC  
> that, like location-conveyance, highlights how the RFC 3693  
> requirements are addressed.
>
> That's a sound argument, and obviously location-conveyance had to  
> meet that standard for us to be happy.  Now, I ask that you go  
> back, take the above paragraph and perform one simple operation and  
> read it again: s/HTTP/SIP/g.  Is the argument any different in the  
> presence of RFC 4079?  Is an informational RFC that doesn't address  
> specific requirements adequate?  I'd suggest that it wouldn't make  
> the grade for HTTP.
>
>> -----Original Message-----
>> From: Andrew Newton [mailto:andy@hxr.us]
>> Sent: Thursday, 27 July 2006 1:55 PM
>> To: IETF SIP List; GEOPRIV
>> Subject: Teasing apart positions (was Re: [Sip] RE: [Geopriv]  
>> Consensus
>> onchanges to location-conveyance)
>>
>>
>> On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
>>> The authors would like to know what to do now.
>>
>> Let's start by trying to tease apart certain positions.
>>
>> Given Jon Peterson's message:
>> http://www1.ietf.org/mail-archive/web/sip/current/msg15537.html
>>
>> Does anybody still believe that subscriptions to PIDF-LOs with pres:
>> requires more definition?
>>
>> -andy
>>
>> _______________________________________________
>> Geopriv mailing list
>> Geopriv@ietf.org
>> https://www1.ietf.org/mailman/listinfo/geopriv
>
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From sip-bounces@ietf.org Thu Jul 27 08:46:23 2006
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There have been a number of opinions regarding HTTP as a using protocol.

I'd like to know, how many people believe that HTTP requires no or  
very little specification to meet the requirements in RFC 3693 and  
RFC 3694.  If you believe this, what basis do you have for this  
conclusion?  Or do you believe more specification needs to be done,  
but it is possible for HTTP to be a GEOPRIV using protocol?

Finally, how many people believe that HTTP should not be a using  
protocol?

-andy

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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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Works for me...

-Marc- 

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu] 
> Sent: Wednesday, July 26, 2006 11:25 PM
> To: Brian Rosen
> Cc: IETF SIP List; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to 
> location-conveyance
> 
> My suggestion would be
> 
> - to focus the draft on just defining the Location header 
> syntax, without ruling out any particular mechanism or 
> scheme; this would make the draft really short (which I would 
> find helpful).
> 
> - and then add one section to the draft that prototypically 
> defines what a particular scheme that wants to appear in the 
> Location header needs to define, using the non-controversial 
> cid mechanism as the example. Other, separate, drafts can 
> then define other modes, following the same template. 
> Documents such as the phone-bcp can then pick some suitable 
> must-implement subset and make emergency-call- specific 
> recommendations, rather than sprinkling these throughout the 
> conveyance document and repeating some in the phone-bcp. Most 
> of these are, in any event, operational recommendations, not 
> for ensuring protocol-level interoperability.
> 
> That way, the conveyance draft can be extremely short (even 
> shorter if the requirements are omitted...) and can get done, 
> and we avoid engineering operational requirements on the fly, 
> commingling emergency and non-emergency cases, as seems to be 
> happening now.
> 
> I think protocol documents should define what's needed for 
> interoperability; BCPs can make more specific, 
> situation-specific recommendations. Commingling the two seems 
> to be a recipe for deadlock or arbitrary, 
> non-technically-grounded decisions that are more lawyer-like 
> than engineering-like ("you did not mention technique X in 
> some meeting N years ago and thus you can't bring it up now!")
> 
> Henning
> 
> On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
> 
> > At this point, I think we need some chair guidance and by 
> that I think 
> > it's a combo of geopriv and sip chairs.
> >
> > We seem to have a couple of significant open questions.  I 
> think most 
> > of the other stuff (like the "allow sending to router") 
> will converge.
> >
> > There seems to be some opposition to allowing sip/sips/pres: in a 
> > Location Header.  The opposition seems to be that despite 4119, 
> > somehow, SIP (presence subscription) is not a "using protocol".  I 
> > "think" a ruling from the bench on that would work.
> >
> > There is also opposition to defining the allowed schemes as 
> a specific 
> > list, with a defined (easy) extension mechanism.
> >
> > Related to that, there are some who want 
> location-conveyance to define 
> > an HTTP based location dereference mechanism, and on the 
> opposite end, 
> > we have people saying we haven't even agreed to support any form of 
> > location-by-reference.
> >
> > The authors would like to know what to do now.
> >
> > Brian
> >
> >> -----Original Message-----
> >> From: Andrew Newton [mailto:andy@hxr.us]
> >> Sent: Wednesday, July 26, 2006 9:58 PM
> >> To: Winterbottom, James
> >> Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning 
> >> Schulzrinne
> >> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location- 
> >> conveyance
> >>
> >>
> >> On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
> >>
> >>> Andy,
> >>>
> >>> RFC-4119 says "However, the usage of this location object 
> format is 
> >>> not
> >>>    limited to presence-using protocols-- any protocol 
> that can carry 
> >>> XML
> >>>    or MIME types can carry PIDF."
> >>>
> >>> It seems to me that restricting the allowed schema in what is 
> >>> supposed to be a location conveyance specification is in 
> opposition 
> >>> to this statement.
> >>
> >> Did somebody say that HTTP was a presence-using protocol?
> >>
> >> -andy
> >>
> >> ps.  you can find my location with the following reference URIs:
> >>
> >> gopher://fatchance.hxr.us/you-still-have-a-gopher-client
> >> news:alt.locations.people.secretly-being-watched
> >> telnet://screen-scrapers.are.us/
> >> dns:cached-publicly-around-the-world.com?type=HOST
> >>
> >>
> >>
> >> _______________________________________________
> >> Geopriv mailing list
> >> Geopriv@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/geopriv
> >
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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Thread-Topic: [Geopriv] Consensus on changes to location-conveyance
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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
To: "Winterbottom, James" <James.Winterbottom@andrew.com>,
	"Abbott, Nadine B" <nabbott@telcordia.com>,
	"Brian Rosen" <br@brianrosen.net>, <sip@ietf.org>
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Subject: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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(as individual)

I am not sure I understand your argument here.

As far as I understand, and in addition to any other constraints, any
protocol used as a result of the reference to carrt location must be a
location using protocol in and pass the IETF (as policed by GEOPRIV)
privacy provisions on location.

In the absence of any other normative document by GEOPRIV that
normatively lists those protocols, the dereferencing protocol
specification (currently specified in sip-location-conveyance) must
normatively indicate that list. Whether it does it by ABNF or text is
academic - they are both normative specifications, and they both need an
RFC to update them.=20

I believe we do have a requirement that the dereferencing protocol
entity MUST NOT use a protocol that is not a location using protocol.

Regards

Keith

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]=20
> Sent: 27 July 2006 00:31
> To: Abbott, Nadine B; Brian Rosen; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>=20
> Brian,
>=20
> I don't care if you don't mention how to de-reference URI=20
> types other than sip/sips/cid/pres in the document or not.=20
> But I do not want the schema artificially constrained to=20
> exclude URI types such as HTTP. There is clearly not=20
> consensus to justify this exclusion, so please leave schema=20
> unchanged and make your recommendations in the text.
>=20
> Cheers
> James
>=20
> =20
>=20
> -----Original Message-----
> From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
> Sent: Thursday, 27 July 2006 7:44 AM
> To: Brian Rosen; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>=20
> Brian,
>=20
> Actually, I was thinking that in other industry work which is=20
> trying to leverage and provide transitional strategies to=20
> migrate to using SIP to support emergency services, the and=20
> long-term solutions that we have defined assume a capabity to=20
> provide location-by-reference.  I think it would be very=20
> useful to have this defined in a standard SIP header=20
> mechanism in such a way that we need not redefine migratory=20
> solutions that have begun to be implemented. If an HTTPS URI=20
> can be used as the location-by-reference, this will be=20
> helpful, I think. =20
>=20
> I don't understand your urgency to make location-by-reference=20
> only an all-sip-all-the-time thing.
> And from all the other emails on this topic, it didn't seem=20
> to me that there is a consensus to do this.
>=20
> Nadine=20
>=20
> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]
> Sent: Wednesday, July 26, 2006 3:15 PM
> To: Abbott, Nadine B; sip@ietf.org
> Cc: geopriv@ietf.org
> Subject: RE: [Geopriv] Consensus on changes to location-conveyance
>=20
> Nadine
>=20
> Are you saying that it's your opinion sip-location-conveyance=20
> MUST define an HTTP dereference procedure?
>=20
> If you do, why?  What is the use-case or requirement?
>=20
> Would it not be pretty reasonable to require a separate=20
> document that defines a location-by-reference mechanism,=20
> protocol, and geopriv "using protocol" analysis?  If that=20
> document specifically extended what was allowed in the=20
> location header, would that not be reasonable?
>=20
> I think we have to include "A" dereference mechanism in the=20
> location-conveyance document.  It seems just including=20
> sip/sips/prez is a reasonable, all-sip-all-the-time approach=20
> IN THIS DOCUMENT.  That certainly does not preclude another=20
> document describing another way.
>=20
> Doesn't it seem just a little weird to have a=20
> sip-location-conveyance document describing any kind of http=20
> based location conveyance
> (de-reference) protocol, especially one created whole cloth=20
> without any other parts?
>=20
> Doesn't it seem in keeping with the geopriv approach that in=20
> order to have another "using protocol" it has to have a=20
> geopriv review?  That is what leads to restricting the URI=20
> schemes.  Would you prefer to let it be wide open, so any=20
> protocol anyone thinks about can be used with the location=20
> header?  I can't see geopriv going along with that.  We're=20
> going to make it a one line ABNF to add a new scheme to the=20
> list of allowed schemes.
>=20
> It seems to me that if it's desired to define some bare-bones=20
> HTTP location by reference protocol, then that should be a=20
> separate document, which can easily extend the allowable=20
> schemes in the Location header.
> I'm not sure it might be better to define ALL of that in the=20
> L7 protocol, but I wouldn't be opposed to a simple "raw GET"=20
> RFC as long as it got a full geopriv review.
> I just don't see where it makes any sense to force that into=20
> a sip document.
>=20
> Brian
>=20
> > -----Original Message-----
> > From: Abbott, Nadine B [mailto:nabbott@telcordia.com]
> > Sent: Wednesday, July 26, 2006 2:14 PM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: RE: [Geopriv] Consensus on changes to location-conveyance
> >=20
> >=20
> > I don't agree that there is a consensus to preclude the use of HTTP.
> > I would favor not placing your proposed restriction on the=20
> URI scheme=20
> > for expressing a location reference in the SIP location conveyance=20
> > draft; or at least including HTTP as a valid location-reference URI=20
> > type.
> >=20
> > Regards,
> > Nadine Abbott
> >=20
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Tuesday, July 25, 2006 1:38 PM
> > To: sip@ietf.org
> > Cc: geopriv@ietf.org
> > Subject: [Geopriv] Consensus on changes to location-conveyance
> >=20
> > I have put out a number of emails on proposed changes to location=20
> > conveyance.  Some of them have sparked a number of comments.  My=20
> > summary of consensus on these issues is:
> >=20
> > 1. We proposed to explicitly disallow use of a Data URI,=20
> which would=20
> > allow specification of a Location-by-value in a header. =20
> There was a=20
> > lot of list traffic on this, currently in the state where=20
> Keith asked=20
> > for a real use case for it.  So far, we don't see a use case that=20
> > cannot be provided by location-by-reference in a header. =20
> If we don't=20
> > get a good use case, I think consensus is to disallow=20
> > location-by-value in a header.
> >=20
> > 2. Dereferencing location-by-reference.  We had initially=20
> proposed to=20
> > document a process to use a raw HTTP(S) Get to dereference.=20
>  There was
>=20
> > a lot of discussion, and then I made a new proposal, see item 6.
> >=20
> > 3. We proposed to allow a Reason header with a 424 Bad Location=20
> > response.  This seems to be acceptable to the WG.  Geopriv=20
> will create
>=20
> > and populate the registry in a separate draft.
> >=20
> > 4. We proposed to explicitly allow hop-by-hop TLS security for=20
> > location when the endpoint wished to allow routing based on=20
> location.
>=20
> > We further proposed to interpret "Do not Distribute" flags=20
> as allowing
>=20
> > sending of the location to the routing database (LoST for=20
> example). =20
> > Hannes pointed out that there was a proposal to allow two Location=20
> > headers, one of which could be used for routing and another=20
> for onward
>=20
> > forwarding to the recipient.  Hannes' message seemed to=20
> agree that "Do
> not distribute"
> > would allow sending of the location to the routing-by-location=20
> > database, but would obligate the proxy that did that to remove the=20
> > location before it forwarded the message.  There were no other
> messages on this thread.
> > We conclude that allowing hop-by-hop security for location based=20
> > routing is acceptable, and Do Not Distribute DOES allow=20
> forwarding to=20
> > a routing database, but requires that the proxy doing the location=20
> > based route to remove the location it uses for that purpose.
> >=20
> > 5. We proposed to create a parameter and a registry to distinguish=20
> > multiple ways to dereference a location where HTTP(S) was the
> transport.
> > Resolution of this issue depends on resolution of issue 6.
> >=20
> > 6. We proposed to allow sip/sips and pres: in the location=20
> header.  I=20
> > later speculated that if we allow this, it forms a simple way to do=20
> > location-by-reference, and we could eliminate the raw HTTP(S) Get=20
> > proposal, with the parameter and registry entirely.  There was some=20
> > list traffic in favor of this.  Martin Thompson is=20
> skeptical that SIP=20
> > Presence Subscribe/Notify is actually allowed as a geopriv "using=20
> > protocol".  Several of think that it is.  If we don't get any more=20
> > comments on this proposal, we will allow sip/sips and pres,=20
> not have=20
> > any
> > http(s) resolution text, and not propose a parameter and=20
> accompanying=20
> > registry.
> >=20
> > If this does not reflect your position, please speak up now.
> >=20
> > Brian
> >=20
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> >=20
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20
> --------------------------------------------------------------
> ----------------------------------
> This message is for the designated recipient only and may=20
> contain privileged, proprietary, or otherwise private information. =20
> If you have received it in error, please notify the sender=20
> immediately and delete the original.  Any unauthorized use of=20
> this email is prohibited.
> --------------------------------------------------------------
> ----------------------------------
> [mf2]
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

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Jonathan Rosenberg wrote:
> Cullen Jennings wrote:

>> Section 6, page 12, 4th para or so ... you are talking about mid- 
>> dialog requests. Might be nice to mention how a transaction stateful  
>> proxy knows if something is a middialog request or not.
> 
> These days I recommend topmost route URI over tags, since its easy to 
> fool a proxy into thinking something is a mid-dialog request by faking a 
> tag. Its harder to fake a route URI.
> 
> Text now reads:
> 
> <t> Mid-dialog requests will also be sent to GRUUs, as they are
> included as the remote-target in dialog-forming and target refresh
> requests and responses. However, in those cases, a proxy SHOULD only
> apply services that are meaningful for mid-dialog requests, generally
> speaking. This excludes screening functions, as well as forwarding
> ones. A proxy can determine that a request is a mid-dialog request
> based on the Route header field in the request it receives. If the
> topmost URI matches one that the proxy placed into the Record-Route
> header field of a dialog-forming request, then the request is a
> mid-dialog request. </t>

Is it really harder to fake?

Suppose Alice is permitted to call Bob, but not to subscribe to the reg 
event package. So Alice first calls Bob, and notes the Record-Route that 
results. Then in a separate dialog, Alice sends a SUBSCRIBE to the reg 
event package, including a Route header with a URI of Bob's home proxy 
extracted from the old R-R.

Seems like this strategy also requires some protection against a "replay 
attack" like this.

But that isn't really an issue for this draft.

	Paul

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Jonathan Rosenberg wrote:
> Dale.Worley@comcast.net wrote:

>> There seems to be disagreement as to what the plural of GRUU is --
>> GRUU or GRUUs.  E.g., section 13, page 31 uses both "Furthermore,
>> GRUUs do not address" and "These GRUU will be passed".  There is a
>> similar inconsistency regarding the plural of AOR.
> 
> Well, this topic has come up for the term URI, which is the last U in 
> GRUU. So it would stand to reason that if the plural of URI is URIs, 
> then the plural of GRUU is GRUUs. I found some email threads on this 
> topic, for example:
> 
> http://lists.w3.org/Archives/Public/www-tag/2002Aug/0124.html
> 
> and apparently this was an open issue in the preparation of the revision 
> to 2396:
> 
> http://www.gbiv.com/protocols/uri/rev-2002/issues.html#023-URI-plural
> 
> the conclusion appears to be that the plural of URI is URIs. So, I 
> suggest we follow suit and have the plural of GRUU be GRUUs.

Hey, that's not fair!

I was looking for a precedent when cleaning up 
draft-ietf-sipping-gruu-reg-event-06, and noticed there was 
inconsistency in draft-ietf-sip-gruu. We discussed this, and I thought 
we decided that the plural is GRUU. I made my draft consistent with that.

If you decide this way then I should respin 
draft-ietf-sipping-gruu-reg-event.

	Paul

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From sip-bounces@ietf.org Thu Jul 27 09:29:28 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Andrew Newton'" <andy@hxr.us>, "'IETF SIP List'" <sip@ietf.org>,
	"'GEOPRIV'" <geopriv@ietf.org>
Subject: RE: [Sip] teasing apart: http as a GEOPRIV using protocol
Date: Thu, 27 Jul 2006 09:27:35 -0400
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I think HTTP requires some specification.  A raw GET based retrieval of a
PIDF would be a fairly short RFC, but I think we need that RFC.

While I don't think it's necessary (SIP SUBSCRIBE is sufficient), I don't
object to defining HTTP as a using protocol in a suitable RFC.

Brian

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Thursday, July 27, 2006 8:45 AM
> To: IETF SIP List; GEOPRIV
> Subject: [Sip] teasing apart: http as a GEOPRIV using protocol
> 
> There have been a number of opinions regarding HTTP as a using protocol.
> 
> I'd like to know, how many people believe that HTTP requires no or
> very little specification to meet the requirements in RFC 3693 and
> RFC 3694.  If you believe this, what basis do you have for this
> conclusion?  Or do you believe more specification needs to be done,
> but it is possible for HTTP to be a GEOPRIV using protocol?
> 
> Finally, how many people believe that HTTP should not be a using
> protocol?
> 
> -andy
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Thu Jul 27 09:35:57 2006
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] teasing apart: http as a GEOPRIV using protocol
Date: Thu, 27 Jul 2006 09:33:48 -0400
To: "Brian Rosen" <br@brianrosen.net>
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For all protocols, the hardest part is specifying how authentication  
works in conjunction with the privacy rules. This is far from obvious  
for HTTP, for example. (Is it the Digest user identifier? Can Basic  
over TLS be used instead? What about crypto-random request URIs? Is  
HTTP ok in that case or is HTTPS needed - the answer isn't obvious  
since an observer would only see a location, not a location-identity  
pair.)

For SIP, this is largely done, except for the crypto-random request  
URI case and variations, such as user + password.

On Jul 27, 2006, at 9:27 AM, Brian Rosen wrote:

> I think HTTP requires some specification.  A raw GET based  
> retrieval of a
> PIDF would be a fairly short RFC, but I think we need that RFC.
>
> While I don't think it's necessary (SIP SUBSCRIBE is sufficient), I  
> don't
> object to defining HTTP as a using protocol in a suitable RFC.
>
> Brian


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From sip-bounces@ietf.org Thu Jul 27 10:24:32 2006
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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Thu, 27 Jul 2006 16:22:32 +0200
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Thread-Topic: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
To: "Brian Rosen" <br@brianrosen.net>
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(as SIP WG chair)
(Warning - degree of rant is directly proportional to the value of the
item number, I decided that 6 through 10 were not publishable)

1)	I think we do have a significant amount of discussion and
agreement already that does need to be reflected in a revised draft. You
promised that revised draft in early August, and whatever the state of
the open issues, you should still provide that revised draft, and merely
indicate in the text that open issues exist in particular areas if they
do not get resolved before you hit the send button. We are certainly
getting repeats of comments that relate to stuff that will already be
addressed in the revised version.

2)	This document is making SIP a "using protocol" for location. Any
normative requirements in this document must be consistent with GEOPRIV
documentation in RFC 3693, RFC 3694, RFC 4079 and RFC 4119. Therefore if
comments are being made that are inconsistent with any of these
documents, then the commentors should in the first instance be
addressing their comments to GEOPRIV in terms of corrections to these
RFCs. I don't want a SIP document held hostage because someone either
refused to read a GEOPRIV document before it was approved, or at the
very least failed to understand what was written at that time. If it is
a question of interpretation of the GEOPRIV documents, then I guess
there is room for some discussion, but at least Jon Peterson, who played
a key role in this work in GEOPRIV, has indicated that the presence work
is a GEOPRIV using protocol. On that basis, I would allow him to hum
directly into the microphone if a hum was in progress on this one.

3)	Using protocols are sanctioned by GEOPRIV. If GEOPRIV has not
made a decision, then it does not exist as a using protocol!

4)	We are past the design stage on this document. The document has
had its current technical content for 12 months. In my view that raises
the bar somewhat. If something new is required in this document, then we
are not in the process of just adding it as yet another variant - if
only because that hits interoperability. I want to see clear statements
about why the current stuff does not work, and what is improved by the
new addition. In many cases, I am failing to see reasoned arguments
based on the GEOPRIV documents, or other IETF work, as to why something
is right or wrong - I do not want to see proposals based on general
feelings, or, "that is what I want to implement therefore it has got to
be included".=20

5)	We still have to make decisions by concensus. We need consensus
to take something out, we need consensus to put something in. While the
editors do need to represent that consensus, they are also entitled to
their views on the discussion as individuals. At the end of the day, if
we fail to reach consensus about the document, then we kill either the
charter item, or the SIP WG itself.

Regards

Keith

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]=20
> Sent: 27 July 2006 04:08
> To: 'Andrew Newton'; 'Winterbottom, James'
> Cc: sip@ietf.org; geopriv@ietf.org; 'Henning Schulzrinne';=20
> 'Abbott, Nadine B'
> Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to=20
> location-conveyance
>=20
> At this point, I think we need some chair guidance and by=20
> that I think it's a combo of geopriv and sip chairs.
>=20
> We seem to have a couple of significant open questions.  I=20
> think most of the other stuff (like the "allow sending to=20
> router") will converge.
>=20
> There seems to be some opposition to allowing sip/sips/pres:=20
> in a Location Header.  The opposition seems to be that=20
> despite 4119, somehow, SIP (presence subscription) is not a=20
> "using protocol".  I "think" a ruling from the bench on that=20
> would work.
>=20
> There is also opposition to defining the allowed schemes as a=20
> specific list, with a defined (easy) extension mechanism. =20
>=20
> Related to that, there are some who want location-conveyance=20
> to define an HTTP based location dereference mechanism, and=20
> on the opposite end, we have people saying we haven't even=20
> agreed to support any form of location-by-reference. =20
>=20
> The authors would like to know what to do now.
>=20
> Brian
>=20
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Wednesday, July 26, 2006 9:58 PM
> > To: Winterbottom, James
> > Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning=20
> > Schulzrinne
> > Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-=20
> > conveyance
> >=20
> >=20
> > On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
> >=20
> > > Andy,
> > >
> > > RFC-4119 says "However, the usage of this location object=20
> format is=20
> > > not
> > >    limited to presence-using protocols-- any protocol=20
> that can carry=20
> > > XML
> > >    or MIME types can carry PIDF."
> > >
> > > It seems to me that restricting the allowed schema in what is=20
> > > supposed to be a location conveyance specification is in=20
> opposition=20
> > > to this statement.
> >=20
> > Did somebody say that HTTP was a presence-using protocol?
> >=20
> > -andy
> >=20
> > ps.  you can find my location with the following reference URIs:
> >=20
> > gopher://fatchance.hxr.us/you-still-have-a-gopher-client
> > news:alt.locations.people.secretly-being-watched
> > telnet://screen-scrapers.are.us/
> > dns:cached-publicly-around-the-world.com?type=3DHOST
> >=20
> >=20
> >=20
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

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From sip-bounces@ietf.org Thu Jul 27 10:27:52 2006
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Subject: [Sip] Teasing apart: constraints on dereference URI
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There have been many conversations regarding constraining the URI in the 
Location header.  There have been many positions taken on this subject, 
and this email is intended to understand the various opinions.

Is there anybody who believes that the type of URI in the Location 
header should not be constrained in any way?  In other words, any URI 
scheme may be valid and there should be no attempt to specify the URI 
scheme or associated protocol for dereferencing location.  And if so, do 
you believe location will only be dereferenced via a user clicking on a 
link or by automatic software action?  If the latter, how do you see 
software adapting to multiple URI schemes and their associated protocols?

Is there anybody who believes that the dereference URI should be 
restricted to a specific set of schemes, enforced via ABNF, XML Schema, 
or other grammar specification and re-enforced with MUST/SHOULD 
language?  The intent here being that the set would be definitive or at 
the very least require major effort to update.

Finally, how many people desire a middle ground?  Perhaps where the 
ABNF, XML Schema or other grammar specification allows for any URI 
scheme but there is an IANA registry listing the allowed dereference URI 
schemes and associated protocols?

-andy


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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
References: <022b01c6b03c$dbc169d0$640fa8c0@cis.neustar.com>	<001101c6b041$171edfe0$31713b51@BEMBUSTER>	<B74032F3-6BF6-4564-A971-D474F04FE620@hxr.us><8FF48E53-9E83-40AB-B1DC-B159931233F6@cs.columbia.edu><44C777D8.9030008@hxr.us>	<46C30BB8-1D03-44D4-BF50-3C38FE0C23A3@cs.columbia.edu>	<011501c6b0c9$09681910$31713b51@BEMBUSTER>	<44C7919D.1010203@hxr.us>	<44C79CA6.5040604@cs.columbia.edu>
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Hi Andy,

in the use case where the proxy either adds a location-by-value or 
location-by-reference where would the proxy get the privacy policies 
from in the first place?

If it has the rule and wants to convey location-by-value then why 
wouldn't it just include the rules instead of putting a reference to the 
ruleset in there that has to be retrieved first.

In the Geopriv-L7 design team we were puzzled about the case where a LIS 
adds privacy rules to the PIDF-LO before signing it. In this case the 
LIS is the SIP proxy and the challenges remain the same.

Ciao
Hannes


Andrew Newton wrote:
> Henning Schulzrinne wrote:
> 
>> We may be talking past each other. Currently, RFC 4119 has
>>
>>  <xs:complexType name="locPolicyType">
>>    <xs:sequence>
>>      <xs:element name="retransmission-allowed" type="xs:boolean"
>>         minOccurs="0" maxOccurs="1"/>
>>      <xs:element name="retention-expiry" type="xs:dateTime"
>>         minOccurs="0" maxOccurs="1"/>
>>      <xs:element name="external-ruleset" type="xs:anyURI"
>>         minOccurs="0" maxOccurs="1"/>
>>      <xs:element name="note-well" type="tns:notewell"
>>         minOccurs="0" maxOccurs="1"/>
>>      <xs:any namespace="##other" processContents="lax" minOccurs="0"
>>         maxOccurs="unbounded"/>
>>    </xs:sequence>
>>   </xs:complexType>
>>
>> I assume you're talking about the external-ruleset element. Regardless 
>> of how the rest of the GEOPRIV object is carried, this can point 
>> anywhere, including some static web object, a cid, or even be another 
>> data URL if you want to carry the policy along with the actual 
>> location information.
> 
> 
> Hmmm... we seem to have caught a bug in 4119.  Because the descriptive 
> text calls that element 'ruleset-reference' and not 'external-ruleset'. 
>  Anyway, from the text (page 8):
> 
>       This URI SHOULD use the HTTPS URI scheme; and if it does, the
>       server that holds these rules MUST authenticate any attempt to
>       access these rules.  Usage rules themselves may divulge private
>       information about a Target or Rule Maker.  The URI MAY,
>       alternatively, use the CID URI scheme [7], in which case it MUST
>       denote a MIME body carried with the Location Object by the using
>       protocol.
> 
> In other words, it would appear that the list of URIs allowed is HTTPS 
> or CID (though this doc SHOULD be more explicit).  data: is not mentioned.
> 
> This means that a PIDF-LO with a CID reference cannot be converted to a 
> data: URI.
> 
> -andy
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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Hi Brian,

are you talking about the end host adding a location-by-reference into 
the SIP header or a proxy doing so?

In any case I don't understand your objection against the usage of HTTP 
and your preference for SIP/SIPS/PREZ?

Ciao
Hannes

Brian Rosen wrote:
> Nadine
> 
> Are you saying that it's your opinion sip-location-conveyance MUST
> define an HTTP dereference procedure?
> 
> If you do, why?  What is the use-case or requirement?
> 
> Would it not be pretty reasonable to require a separate document that
>  defines a location-by-reference mechanism, protocol, and geopriv
> "using protocol" analysis?  If that document specifically extended
> what was allowed in the location header, would that not be
> reasonable?
> 
> I think we have to include "A" dereference mechanism in the 
> location-conveyance document.  It seems just including sip/sips/prez
> is a reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.
> That certainly does not preclude another document describing another
> way.
> 
> Doesn't it seem just a little weird to have a sip-location-conveyance
>  document describing any kind of http based location conveyance 
> (de-reference) protocol, especially one created whole cloth without
> any other parts?
> 
> Doesn't it seem in keeping with the geopriv approach that in order to
> have another "using protocol" it has to have a geopriv review?  That
> is what leads to restricting the URI schemes.  Would you prefer to
> let it be wide open, so any protocol anyone thinks about can be used
> with the location header?  I can't see geopriv going along with that.
> We're going to make it a one line ABNF to add a new scheme to the
> list of allowed schemes.
> 
> It seems to me that if it's desired to define some bare-bones HTTP
> location by reference protocol, then that should be a separate
> document, which can easily extend the allowable schemes in the
> Location header.  I'm not sure it might be better to define ALL of
> that in the L7 protocol, but I wouldn't be opposed to a simple "raw
> GET" RFC as long as it got a full geopriv review. I just don't see
> where it makes any sense to force that into a sip document.
> 
> Brian
> 
> 
>> -----Original Message----- From: Abbott, Nadine B
>> [mailto:nabbott@telcordia.com] Sent: Wednesday, July 26, 2006 2:14
>> PM To: sip@ietf.org Cc: geopriv@ietf.org Subject: RE: [Geopriv]
>> Consensus on changes to location-conveyance
>> 
>> 
>> I don't agree that there is a consensus to preclude the use of
>> HTTP. I would favor not placing your proposed restriction on the
>> URI scheme for expressing a location reference in the SIP location
>> conveyance draft; or at least including HTTP as a valid
>> location-reference URI type.
>> 
>> Regards, Nadine Abbott
>> 
>> -----Original Message----- From: Rosen, Brian
>> [mailto:Brian.Rosen@neustar.biz] Sent: Tuesday, July 25, 2006 1:38
>> PM To: sip@ietf.org Cc: geopriv@ietf.org Subject: [Geopriv]
>> Consensus on changes to location-conveyance
>> 
>> I have put out a number of emails on proposed changes to location 
>> conveyance.  Some of them have sparked a number of comments.  My
>> summary of consensus on these issues is:
>> 
>> 1. We proposed to explicitly disallow use of a Data URI, which
>> would allow specification of a Location-by-value in a header.
>> There was a lot of list traffic on this, currently in the state
>> where Keith asked for a real use case for it.  So far, we don't see
>> a use case that cannot be provided by location-by-reference in a
>> header.  If we don't get a good use case, I think consensus is to
>> disallow location-by-value in a header.
>> 
>> 2. Dereferencing location-by-reference.  We had initially proposed
>> to document a process to use a raw HTTP(S) Get to dereference.
>> There was a lot of discussion, and then I made a new proposal, see
>> item 6.
>> 
>> 3. We proposed to allow a Reason header with a 424 Bad Location 
>> response.  This seems to be acceptable to the WG.  Geopriv will
>> create and populate the registry in a separate draft.
>> 
>> 4. We proposed to explicitly allow hop-by-hop TLS security for
>> location when the endpoint wished to allow routing based on
>> location.  We further proposed to interpret "Do not Distribute"
>> flags as allowing sending of the location to the routing database
>> (LoST for example).  Hannes pointed out that there was a proposal
>> to allow two Location headers, one of which could be used for
>> routing and another for onward forwarding to the recipient.
>> Hannes' message seemed to agree that "Do not distribute" would
>> allow sending of the location to the routing-by-location database, 
>> but would obligate the proxy that did that to remove the location
>> before it forwarded the message.  There were no other messages on
>> this thread. We conclude that allowing hop-by-hop security for
>> location based routing is acceptable, and Do Not Distribute DOES
>> allow forwarding to a routing database, but requires that the proxy
>> doing the location based route to remove the location it uses for
>> that purpose.
>> 
>> 5. We proposed to create a parameter and a registry to distinguish 
>> multiple ways to dereference a location where HTTP(S) was the
>> transport. Resolution of this issue depends on resolution of issue
>> 6.
>> 
>> 6. We proposed to allow sip/sips and pres: in the location header.
>> I later speculated that if we allow this, it forms a simple way to
>> do location-by-reference, and we could eliminate the raw HTTP(S)
>> Get proposal, with the parameter and registry entirely.  There was
>> some list traffic in favor of this.  Martin Thompson is skeptical
>> that SIP Presence Subscribe/Notify is actually allowed as a geopriv
>> "using protocol".  Several of think that it is.  If we don't get
>> any more comments on this proposal, we will allow sip/sips and
>> pres, not have any http(s) resolution text, and not propose a
>> parameter and accompanying registry.
>> 
>> If this does not reflect your position, please speak up now.
>> 
>> Brian
>> 
>> _______________________________________________ Geopriv mailing
>> list Geopriv@ietf.org 
>> https://www1.ietf.org/mailman/listinfo/geopriv
>> 
>> _______________________________________________ Geopriv mailing
>> list Geopriv@ietf.org 
>> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 
> 
> _______________________________________________ Sip mailing list
> https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
> development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
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> 
> 


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From sip-bounces@ietf.org Thu Jul 27 10:36:05 2006
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Hi Marc,

I agree with you.

I also think we should split the work into two parts as Henning 
suggested: One document covers the end host adds location-by-value (and 
potentially location-by-referenc) and another one that deals with the 
proxy doing something. The proxy case is far less understood as we know.

Ciao
Hannes

Marc Linsner wrote:
> I have no problem with pidf/pidf-lo/presence.
> 
> I have no problem with a ua placing a uri pointing to their presence service
> in a header of a sip invite.
> 
> Context check:  We're discussing a sip invite (assumed pointed to emergency
> service) that arrives at a proxy with zero location information.
> 
> I'm the only one that has a problem with any (visited) proxy/LIS creating a
> uri that can be de-referenced by anyone without regard to user
> security/privacy rules?
> 
> -Marc-
> 
> 
> 
> 
> 
> 
>>-----Original Message-----
>>From: Peterson, Jon [mailto:jon.peterson@neustar.biz] 
>>Sent: Tuesday, July 25, 2006 4:42 PM
>>To: Andrew Newton; Rosen, Brian
>>Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
>>Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to 
>>location-conveyance
>>
>>
>>As Brian points out, the whole point of using PIDF for 
>>location information was the similarity of the early defined 
>>use cases to the presence architecture. This certainly 
>>included use cases in which one wanted to fetch or subscribe 
>>to location information. Those operations require the 
>>existence of a reference, and that reference was commonly 
>>described as a URI, at the time. Lately, I think the idea of 
>>having a URI pointing to location information has become 
>>associated with a specific proposal (i.e. HELD), but clearly 
>>RFC4079 sketched out an architecture for using presence to 
>>acquire location information and it specifically mentions 
>>URIs as a way to refer to presence. 
>>
>>There is no "hidden magic" in any of this. Subscribing to 
>>presence, and using URIs to indicate presence information, is 
>>exhaustively specified for SIP. PIDF-LO was designed to reuse 
>>all of that work and specifically not to require us to go out 
>>and reinvent everything in order to be able to refer to 
>>location information out in the network.
>>
>>Basically, I support the direction re: bullet 6. I might 
>>further suggest that the CID URI scheme be permitted in the 
>>Location header, in order to be able to refer to MIME bodies 
>>included in SIP messages. I'm also neutral/favorable on using 
>>HTTP GET. But I think either of these URI schemes and their 
>>usage in the Location header could just as well be described 
>>in subsequent documents.
>>
>>Jon Peterson
>>NeuStar, Inc,
>>
>>
>>>-----Original Message-----
>>>From: Andrew Newton [mailto:andy@hxr.us]
>>>Sent: Tuesday, July 25, 2006 1:05 PM
>>>To: Rosen, Brian
>>>Cc: sip@ietf.org; geopriv@ietf.org; Marc Linsner
>>>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to 
>>>location-conveyance
>>>
>>>
>>>For the record, I agree with Brian.
>>>
>>>-andy
>>>
>>>On Jul 25, 2006, at 3:50 PM, Rosen, Brian wrote:
>>>
>>>
>>>>When the original PIDF-LO RFC came out, location-by-reference was 
>>>>defined.
>>>>
>>>>You have a URI.  You can exchange the URI for location.  
>>
>>There is a 
>>
>>>>protocol defined for that purpose (SIP SUBSCRIBE).  That 
>>
>>is location 
>>
>>>>by reference.  It also serves as the location update mechanism, 
>>>>clearly something we need.
>>>>
>>>>Some day, there may be other location URI references.  We 
>>
>>don't need 
>>
>>>>to cover them in this draft.  The basic location by 
>>
>>reference horse 
>>
>>>>has left the barn.  Were you planning on creating an RFC that
>>>
>>>specifically
>>>
>>>>disallows a Subscribe on a sip/sips/pres URI to get location?    
>>>>THAT WAS
>>>>THE POINT OF MAKING IT PART OF A PIDF.
>>>>
>>>>I do think there is some value in defining a way to express a
>>>>location-by-reference aka presence URI which is not just 
>>
>>the AoR.   
>>
>>>>That
>>>>is one reason why I want to define the Location URI as 
>>
>>being able to 
>>
>>>>contain a sip/sips/pres URI.  The purpose is to allow a UA or a 
>>>>proxy to insert location where there is no association between the
>>>
>>>identity of
>>>
>>>>the SIP user and the location.
>>>>
>>>>But that is all I think we need.
>>>>
>>>>We have a requirement for proxy insertion of location.  If
>>>
>>>we disallow
>>>
>>>>location-by-reference (which I think is a good idea in 
>>
>>general, but 
>>
>>>>futile in practice), then we would need something like the
>>>
>>>Data URI to
>>>
>>>>support location insertion by proxy.  If we DO allow 
>>>>location-by-reference, then it also serves to meet the location 
>>>>insertion by proxy requirement, and we don't need the Data
>>>
>>>URI.  We're
>>>
>>>>looking for a use case which requires proxy insertion by 
>>
>>value, but 
>>
>>>>haven't heard one yet.
>>>>
>>>>You can make an argument that you can take the From:, 
>>
>>assume it's an 
>>
>>>>AoR, and SUBSCRIBE to the presence of that AoR to get 
>>
>>location, thus 
>>
>>>>not needing a Location header.  I really think letting the
>>>
>>>Location header
>>>
>>>>specify an alternate presence URI is a good idea.
>>>>
>>>>Brian
>>>>
>>>>
>>>>>-----Original Message-----
>>>>>From: Marc Linsner [mailto:mlinsner@cisco.com]
>>>>>Sent: Tuesday, July 25, 2006 3:29 PM
>>>>>To: Rosen, Brian; sip@ietf.org
>>>>>Cc: geopriv@ietf.org
>>>>>Subject: RE: [Geopriv] Consensus on changes to 
>>
>>location-conveyance
>>
>>>>>In-line.....
>>>>>
>>>>>
>>>>>>-----Original Message-----
>>>>>>From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
>>>>>>Sent: Tuesday, July 25, 2006 1:38 PM
>>>>>>To: sip@ietf.org
>>>>>>Cc: geopriv@ietf.org
>>>>>>Subject: [Geopriv] Consensus on changes to location-conveyance
>>>>>>
>>>>>>I have put out a number of emails on proposed changes 
>>
>>to location 
>>
>>>>>>conveyance.  Some of them have sparked a number of 
>>
>>comments.  My 
>>
>>>>>>summary of consensus on these issues is:
>>>>>>
>>>>>>1. We proposed to explicitly disallow use of a Data URI, which 
>>>>>>would allow specification of a Location-by-value in a header.  
>>>>>>There was a lot of list traffic on this, currently in the state 
>>>>>>where Keith asked for a real use case for it.
>>>>>>So far, we don't see a use case that cannot be provided by 
>>>>>>location-by-reference in a header.  If we don't get a good use 
>>>>>>case, I think consensus is to disallow location-by-value in a 
>>>>>>header.
>>>>>>
>>>>>
>>>>>I see a problem here.  Location-by-reference is nothing more than
>>>>
>>>>'magic
>>>>
>>>>>happens here'!  Please point to a wg item documenting the 
>>>>>location-by-reference mechanism.  How can you state the 
>>>>>location-by-reference will cover all the use cases that
>>>
>>>the Data URI
>>>
>>>>would
>>>>
>>>>>when location-by-reference hasn't even been defined
>>>
>>>anywhere?  You're
>>>
>>>>>loading up the magic with a lot of assumptions.  Location-by- 
>>>>>reference
>>>>
>>>>is
>>>>
>>>>>an
>>>>>individual submission as a solution to a set of requirements that
>>>>
>>>>haven't
>>>>
>>>>>been fleshed-out in geopriv.
>>>>>
>>>>>I believe that Henning/Hannes also recognized this 
>>
>>issue, hence the 
>>
>>>>>suggestion of a Data URI.  Dismissing this suggestion based on an
>>>>
>>>>already
>>>>
>>>>>accepted non-existent mechanism seems like getting the
>>>
>>>cart before
>>>
>>>>>the
>>>>>horse.
>>>>>
>>>>>..........snip
>>>>>
>>>>>
>>>>>>6. We proposed to allow sip/sips and pres: in the 
>>
>>location header.  
>>
>>>>>>I later speculated that if we allow this, it forms a 
>>
>>simple way to 
>>
>>>>>>do location-by-reference, and we could eliminate the 
>>
>>raw HTTP(S) 
>>
>>>>>>Get proposal, with the parameter and registry entirely. 
>>
>> There was 
>>
>>>>>>some list traffic in favor of this.  Martin Thompson is 
>>
>>skeptical 
>>
>>>>>>that SIP Presence Subscribe/Notify is actually allowed as a 
>>>>>>geopriv "using protocol".  Several of think that it is.  If we 
>>>>>>don't get any more comments on this proposal, we will allow 
>>>>>>sip/sips and pres, not have any
>>>>>>http(s) resolution text, and not propose a parameter and 
>>>>>>accompanying registry.
>>>>>
>>>>>I certainly think it's more prudent to define the dereferencing
>>>>
>>>>mechanism
>>>>
>>>>>prior to making claims about the protocol.
>>>>>
>>>>>Do I believe location-by-reference will happen?  Probably,
>>>
>>>but making
>>>
>>>>>assumptions about the mechanism now is quite premature.
>>>>>
>>>>>-Marc-
>>>>>
>>>>>
>>>>>>If this does not reflect your position, please speak up now.
>>>>>>
>>>>
>>>>_______________________________________________
>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>>sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>>_______________________________________________
>>>Geopriv mailing list
>>>Geopriv@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>>
> 
> 
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From sip-bounces@ietf.org Thu Jul 27 10:36:05 2006
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Subject: [Sip] Re: [Geopriv] Consensus on changes to location-conveyance
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Hi Brian,

I guess you are only talking about the current Location Conveyance 
document.

Rosen, Brian wrote:
> I have put out a number of emails on proposed changes to location
> conveyance.  Some of them have sparked a number of comments.  My summary
> of consensus on these issues is:
> 
> 1. We proposed to explicitly disallow use of a Data URI, which would
> allow specification of a Location-by-value in a header.  There was a lot
> of list traffic on this, currently in the state where Keith asked for a
> real use case for it.  So far, we don't see a use case that cannot be
> provided by location-by-reference in a header.  If we don't get a good
> use case, I think consensus is to disallow location-by-value in a
> header.

I don't agree with your statements but I am (in the meanwhile) convinced 
that we need to cover this in a separate document.

> 
> 2. Dereferencing location-by-reference.  We had initially proposed to
> document a process to use a raw HTTP(S) Get to dereference.  There was a
> lot of discussion, and then I made a new proposal, see item 6. 
> 
> 3. We proposed to allow a Reason header with a 424 Bad Location
> response.  This seems to be acceptable to the WG.  Geopriv will create
> and populate the registry in a separate draft.
> 
> 4. We proposed to explicitly allow hop-by-hop TLS security for location
> when the endpoint wished to allow routing based on location.  We further
> proposed to interpret "Do not Distribute" flags as allowing sending of
> the location to the routing database (LoST for example).  Hannes pointed
> out that there was a proposal to allow two Location headers, one of
> which could be used for routing and another for onward forwarding to the
> recipient.  Hannes' message seemed to agree that "Do not distribute"
> would allow sending of the location to the routing-by-location database,
> but would obligate the proxy that did that to remove the location before
> it forwarded the message.  There were no other messages on this thread.
> We conclude that allowing hop-by-hop security for location based routing
> is acceptable, and Do Not Distribute DOES allow forwarding to a routing
> database, but requires that the proxy doing the location based route to
> remove the location it uses for that purpose.

Ok.

> 
> 5. We proposed to create a parameter and a registry to distinguish
> multiple ways to dereference a location where HTTP(S) was the transport.
> Resolution of this issue depends on resolution of issue 6.
> 
> 6. We proposed to allow sip/sips and pres: in the location header.  I
> later speculated that if we allow this, it forms a simple way to do
> location-by-reference, and we could eliminate the raw HTTP(S) Get
> proposal, with the parameter and registry entirely.  There was some list
> traffic in favor of this.  Martin Thompson is skeptical that SIP
> Presence Subscribe/Notify is actually allowed as a geopriv "using
> protocol".  Several of think that it is.  If we don't get any more
> comments on this proposal, we will allow sip/sips and pres, not have any
> http(s) resolution text, and not propose a parameter and accompanying
> registry.

I disagree with location-by-reference being added to the document (or 
suggest to get it removed from the document).

Ciao
Hannes

> 
> If this does not reflect your position, please speak up now.
> 
> Brian
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From sip-bounces@ietf.org Thu Jul 27 10:36:10 2006
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Hi Brian,

Brian Rosen wrote:
> I don't really understand why you think location by reference is so much of
> a concern. 

It is interesting that you say that. In the Geopriv L7 design team you
are the only person that wants to see an applicability statement being
written.

Having a SIP proxy adding something (either location-by-value or
location-by-reference) is not different to the stuff we discuss in the
design team. There, you see a lot of privacy, security, etc. problems.
Here you don't. How come?

If you go through my mail with the large number of scenarios then you
find a number of open issues that need to be answered. Alternatively,
you can also go through draft-tschofenig-geopriv-l7-lcp-ps-00.txt

  If you look at the discussion in context, PIDF-LO defines
> location as part of presence.  All of these concerns are relevant there, and
> I think they have been adequately covered there.  We even have some work on
> anonymization of the identity (and thus of the URI) in presence.  Generally,
> a presence uri:
> 
> Is not valid for a limited amount of time
> Is not hard to guess, if it uses an AoR and could be with anonymous URI.
> Identifies the user if it uses an AoR and doesn't with anonymous URI
> 
> The protocol (SIP) for retrieving presence, and thus location has the
> location ruleset as we defined for PIDF-LO, but otherwise does not have "no
> cache" semantics (albeit caching headers is not very interesting in SIP).

You cannot compare presence witht he location-by-reference solution
since the deployment case is different.

> 
> Generally, presence systems do have user control of who is allowed to access
> information, including location and identifying information, so I think that
> is covered fairly well.

Not true in this case.

Here we have a SIP proxy in the access network doing all the work. You
can compare it with a SIP-based LIS from the Geopriv-L7 work. This node
has no clue about your privacy preferrences and does not know your
identity. This is what we discussed in the past and several proposals
where made, namely:

* end host provides privacy preferences (see HELD)
* end host does not provide privacy preferences since there is no
PIDF-LO (see RELO).

> 
> Proxy insertion of location, whether by value or by reference has the threat
> of not being as subject to user control, but I think in every case you have
> to trust your provider to do as you wish, whether it is your presence
> provider or your location provider independent of your presence provider,
> and proxy insertion doesn't fundamentally alter that.  I worry more about
> it, but I don't objectively think it's much different.

In this case you don't really know the provider and it might not be your
  provider.

> 
> Now, if there are other location dereferencing systems, they may need an
> analysis like what you are suggesting, but with respect to SIP Presence, as
> a form of location by reference, I think we have been over the ground
> sufficiently, and we have RFCs that permit it.  Adding restrictions on the
> Location header, which I propose is useful precisely to allow an anonymous
> presence uri that does not reveal identity is not so great I think.  If you
> don't allow that, you would kind of have to use an AoR, which may be
> appropriate in some cases, but would not in others.

The reference might not reveal the identity but what about the identity
in the PIDF-LO. If you argue that there does not need to be one (or it
is a faked one) then I am puzzled why it has been a source of major
confusion in the Geopriv-L7 design team work (and in previous
discussions about the same subject in Geopriv).

> 
> I think the proponents of proxy insertion argue that providing it means
> endpoints don't have to do anything, so I would think they would object to
> explicit permission to do so.  You might get away with a flag that says
> "don't do it" in the IETF process, but whether such a flag would be deployed
> is another matter.   I think we all agree that the emergency case is
> different here; laws govern behavior, and in most cases compel location
> assertion.
This is indeed an interesting case where the end host indicates
something whether it wants the proxy to add location (by reference).

> 
> I kind of like the idea that the proxy inserts the header in the response so
> the UAC can see it.  What do others think of that? 
> 

Ciao
Hannes


> Brian
> 
> 
> 
>>-----Original Message-----
>>From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>Sent: Tuesday, July 25, 2006 7:22 PM
>>To: Brian Rosen; 'Marc Linsner'; Peterson, Jon; 'Andrew Newton'; Rosen,
>>Brian
>>Cc: sip@ietf.org; geopriv@ietf.org
>>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
>>conveyance
>>
>>Brian,
>>
>>
>>>Would you not agree that if a proxy is allowed to insert
>>>location-by-value,
>>>the concerns would be the same?
>>
>>That would depend whether the URL dereferences the current location, or
>>the
>>location at a particular moment in time. The value of a URI with up2date
>>location info is much higher and much more privacy invading than a one-
>>time
>>snapshot of a user's location. So perhaps same concerns, but much stronger
>>
>>For location-by-reference, you'd probably want some additional
>>requirements:
>>- URL must be valid for a limited amount of time
>>- URL must be cryptographically hard to guess
>>- URL must not contain any information that identifies the user / device /
>>AoR
>>- for whatever transport protocol is used: response must be marked as 'no
>>cache'
>>- user must be able to remove the information at the URL, ie explicit
>>invalidation
>>- user must be able to verify correctness of the issued information (ie
>>user
>>can access the URL himself)
>>- user must be able to control who accesses the URL, both upfront and
>>history of accesses (for a reasonable period)
>>- there must be explicit consent before a proxy would insert user location
>>
>>For the latter point: except for emergency scenario's, the UAC should
>>include some flag in the INVITE saying "proxy: please append location".
>>You'd probably also want some feedback (eg proxy or UAS adding a header to
>>the response saying 'this is the URL that I appended/got'
>>
>>Regards,
>>
>>Jeroen
>>
> 
> 
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 



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From sip-bounces@ietf.org Thu Jul 27 10:36:37 2006
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Hi Henning,

I like your suggestion.

In fact I would even go a step further. The SIP Location Conveyance 
document just defines a mechanism to carry a PIDF-LO in the body of the 
SIP message. Done.
Adding location-by-reference to the SIP header by the end host sounds OK 
but there are still some loose ends where this reference comes from.

The other stuff can be dealt with once we concluded our Geopriv-L7 work 
where all the architectural pieces should are created. The 
subscription-URI is an example where we recently proposed something and 
haven't finished our technical discussions.

I think the draft got fuzzy when it came to the area where a SIP proxy 
adds a location-by-reference. I must admit that I proposed to add 
location-by-value into the SIP header as well but in the meanwhile I 
don't think that any of the SIP proxy stories belong in there.

Ciao
Hannes

>>-----Original Message-----
>>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>>Sent: Thursday, 27 July 2006 1:25 PM
>>To: Brian Rosen
>>Cc: IETF SIP List; geopriv@ietf.org
>>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
>>conveyance
>>
>>My suggestion would be
>>
>>- to focus the draft on just defining the Location header syntax,
>>without ruling out any particular mechanism or scheme; this would
>>make the draft really short (which I would find helpful).
>>
>>- and then add one section to the draft that prototypically defines
>>what a particular scheme that wants to appear in the Location header
>>needs to define, using the non-controversial cid mechanism as the
>>example. Other, separate, drafts can then define other modes,
>>following the same template. Documents such as the phone-bcp can then
>>pick some suitable must-implement subset and make emergency-call-
>>specific recommendations, rather than sprinkling these throughout the
>>conveyance document and repeating some in the phone-bcp. Most of
>>these are, in any event, operational recommendations, not for
>>ensuring protocol-level interoperability.
>>
>>That way, the conveyance draft can be extremely short (even shorter
>>if the requirements are omitted...) and can get done, and we avoid
>>engineering operational requirements on the fly, commingling
>>emergency and non-emergency cases, as seems to be happening now.
>>
>>I think protocol documents should define what's needed for
>>interoperability; BCPs can make more specific, situation-specific
>>recommendations. Commingling the two seems to be a recipe for
>>deadlock or arbitrary, non-technically-grounded decisions that are
>>more lawyer-like than engineering-like ("you did not mention
>>technique X in some meeting N years ago and thus you can't bring it
>>up now!")
>>
>>Henning
>>
>>On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
>>
>>
>>>At this point, I think we need some chair guidance and by that I
>>>think it's
>>>a combo of geopriv and sip chairs.
>>>
>>>We seem to have a couple of significant open questions.  I think
>>>most of the
>>>other stuff (like the "allow sending to router") will converge.
>>>
>>>There seems to be some opposition to allowing sip/sips/pres: in a
>>>Location
>>>Header.  The opposition seems to be that despite 4119, somehow, SIP
>>>(presence subscription) is not a "using protocol".  I "think" a
>>>ruling from
>>>the bench on that would work.
>>>
>>>There is also opposition to defining the allowed schemes as a
>>>specific list,
>>>with a defined (easy) extension mechanism.
>>>
>>>Related to that, there are some who want location-conveyance to
>>>define an
>>>HTTP based location dereference mechanism, and on the opposite end,
>>>we have
>>>people saying we haven't even agreed to support any form of
>>>location-by-reference.
>>>
>>>The authors would like to know what to do now.
>>>
>>>Brian
>>>
>>>
>>>>-----Original Message-----
>>>>From: Andrew Newton [mailto:andy@hxr.us]
>>>>Sent: Wednesday, July 26, 2006 9:58 PM
>>>>To: Winterbottom, James
>>>>Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning
>>>>Schulzrinne
>>>>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
>>>>conveyance
>>>>
>>>>
>>>>On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
>>>>
>>>>
>>>>>Andy,
>>>>>
>>>>>RFC-4119 says "However, the usage of this location object format
> 
> is
> 
>>>>>not
>>>>>   limited to presence-using protocols-- any protocol that can
>>>>>carry XML
>>>>>   or MIME types can carry PIDF."
>>>>>
>>>>>It seems to me that restricting the allowed schema in what is
>>>>>supposed
>>>>>to be a location conveyance specification is in opposition to this
>>>>>statement.
>>>>
>>>>Did somebody say that HTTP was a presence-using protocol?
>>>>
>>>>-andy
>>>>
>>>>ps.  you can find my location with the following reference URIs:
>>>>
>>>>gopher://fatchance.hxr.us/you-still-have-a-gopher-client
>>>>news:alt.locations.people.secretly-being-watched
>>>>telnet://screen-scrapers.are.us/
>>>>dns:cached-publicly-around-the-world.com?type=HOST
>>>>
>>>>
>>>>
>>>>_______________________________________________
>>>>Geopriv mailing list
>>>>Geopriv@ietf.org
>>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>>
>>>
>>>_______________________________________________
>>>Geopriv mailing list
>>>Geopriv@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>
>>
>>_______________________________________________
>>Geopriv mailing list
>>Geopriv@ietf.org
>>https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 
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Subject: RE: Teasing apart positions (was Re: [Sip] RE: [Geopriv]
	Consensusonchanges to location-conveyance)
Date: Thu, 27 Jul 2006 16:39:34 +0200
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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
To: "Thomson, Martin" <Martin.Thomson@andrew.com>,
	"Andrew Newton" <andy@hxr.us>
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(as individual)

It's a GEOPRIV problem to identify what height the bar is. At least my
understanding is that Presence was a location using protocol and the
level of documentation was agreed by GEOPRIV. That makes the bar around
at least informational RFC required.

For use in location by reference I believe we have two criteria to meet:

i)	the dereferencing protocol must be a geopriv using protocol

ii)	if we have more than one option, we either need some means of
determining that the dereferencer supports the one the client chooses to
use, or require support of all options. As we seem to be lacking any
mechanism for the former, then it would imply that all options have to
be supported at the dereferencer, and therefore that the list needs to
be a small list.

Regards

Keith

> -----Original Message-----
> From: Thomson, Martin [mailto:Martin.Thomson@andrew.com]=20
> Sent: 27 July 2006 06:00
> To: Andrew Newton
> Cc: IETF SIP List; GEOPRIV
> Subject: RE: Teasing apart positions (was Re: [Sip] RE:=20
> [Geopriv] Consensusonchanges to location-conveyance)
>=20
> Since I raised the question, I'll answer.
>=20
> We have a set of requirements on a using protocol in RFC 3693.
>=20
> I believe that HTTP with proper constraints (TLS, etc...) can=20
> meet, or already meets, those requirements.  That is=20
> irrespective of any decisions that need to be made about=20
> secondary concerns like whether HTTP is desirable, optimal,=20
> deployable etc...  However, it is clear that people aren't=20
> happy with this position and are requesting that this be=20
> documented, as an RFC.  Just to get it all out in the open. =20
> Presumably this would be a standards track RFC that, like=20
> location-conveyance, highlights how the RFC 3693 requirements=20
> are addressed.
>=20
> That's a sound argument, and obviously location-conveyance=20
> had to meet that standard for us to be happy.  Now, I ask=20
> that you go back, take the above paragraph and perform one=20
> simple operation and read it again: s/HTTP/SIP/g.  Is the=20
> argument any different in the presence of RFC 4079?  Is an=20
> informational RFC that doesn't address specific requirements=20
> adequate?  I'd suggest that it wouldn't make the grade for HTTP.
>=20
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Thursday, 27 July 2006 1:55 PM
> > To: IETF SIP List; GEOPRIV
> > Subject: Teasing apart positions (was Re: [Sip] RE: [Geopriv]=20
> > Consensus onchanges to location-conveyance)
> >=20
> >=20
> > On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
> > > The authors would like to know what to do now.
> >=20
> > Let's start by trying to tease apart certain positions.
> >=20
> > Given Jon Peterson's message:
> > http://www1.ietf.org/mail-archive/web/sip/current/msg15537.html
> >=20
> > Does anybody still believe that subscriptions to PIDF-LOs with pres:
> > requires more definition?
> >=20
> > -andy
> >=20
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20
> --------------------------------------------------------------
> ----------------------------------
> This message is for the designated recipient only and may=20
> contain privileged, proprietary, or otherwise private information. =20
> If you have received it in error, please notify the sender=20
> immediately and delete the original.  Any unauthorized use of=20
> this email is prohibited.
> --------------------------------------------------------------
> ----------------------------------
> [mf2]
>=20

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From sip-bounces@ietf.org Thu Jul 27 10:42:29 2006
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Subject: [Sip] Location by Reference Requirements
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Hi Andy,
Hi Jeroen,

Andrew Newton wrote:
> 
> On Jul 25, 2006, at 7:21 PM, Jeroen van Bemmel wrote:
> 
>> For location-by-reference, you'd probably want some additional  
>> requirements:
>> - URL must be valid for a limited amount of time
Ok.

Should it be end point controlled or should the end point be informed 
about the lifetime?

>> - URL must be cryptographically hard to guess

must contain a random,non-guessable component.

>> - URL must not contain any information that identifies the user /  
>> device / AoR

more difficult if you consider that a PIDF-LO contains identity 
information.

>> - for whatever transport protocol is used: response must be marked  as 
>> 'no cache'
Sounds reasonble.


>> - user must be able to remove the information at the URL, ie  explicit 
>> invalidation

Difficult with the SIP based approach we are discussing.
Simpler with the Geopriv-L7 approach being discussed.

>> - user must be able to verify correctness of the issued information  
>> (ie user can access the URL himself)


Difficult with the discussed SIP-based mechanism since the user does not 
ever see the URI.

>> - user must be able to control who accesses the URL, both upfront  and 
>> history of accesses (for a reasonable period)

This has been subject for a long discussion with no conclusion.

>> - there must be explicit consent before a proxy would insert user  
>> location
That's not what some of the use cases indicate.

Nice requirements; maybe something for the Geopriv L7 design team.

> 
> 
> Please tell me that you are merely suggesting guidelines and  
> considerations by operators and are not suggesting 2119 language for  
> these?  Because just like you'd never get agreement on the intended  
> duration or accuracy or confidence of location-by-value, you'd never  
> get agreement on these.

Well, some of these requirements are actually protocol capabilities. 
With some other stuff I agree with you, Andy.

> 
>> For the latter point: except for emergency scenario's, the UAC  should 
>> include some flag in the INVITE saying "proxy: please append  
>> location". You'd probably also want some feedback (eg proxy or UAS  
>> adding a header to the response saying 'this is the URL that I  
>> appended/got'

This is in fact interesting. I haven't seen these requirements so far.

> 
> 
> Just thinking out loud here, but maybe the rule should be that  proxies 
> never append a Location header except for emergencies.

The problem is that the IETF cannot enforce this. Once you have a 
solution for adding location-by-reference by a proxy it can do it.

Ciao
Hannes

> 
> -andy
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 


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From sip-bounces@ietf.org Thu Jul 27 10:42:29 2006
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Hi all,

I would really appreciate if you guys make a difference between an end 
host adding a location-by-reference and a proxy adding a 
location-by-reference to a SIP message. If the end host adds something 
you can further differentiate whether it is added to the header or the body.

This differentiation is really important and that was the reason why I 
wrote a long mail (and it took some time) to elaborate on the different 
scenarios and the challenges.

Jeroen van Bemmel wrote:
>> I don't really understand why you think location by reference is so 
>> much of
>> a concern.
> 
> 
> Because location by reference involves storage of personal information 
> on a server.

In general it does not.

  That suddenly makes all sorts of regulations apply, the
> points I mentioned address only a small part of that. Having an 
> anonymous URL for example often makes a difference, because then the 
> information is no longer directly linked to a particular person.
> 
> When an endpoint inserts location by value, you can reasonably argue 
> that opt-in consent requirements (which many countries have) are 
> fulfilled, provided the endpoint gives the user control over whether or 
> not location is inserted. Same holds for the endpoint inserting a flag 
> asking a proxy to fill in the location (by value).

One of the use cases I have been told regarding the 
location-by-reference addition by the SIP proxy is that providers want 
to deal with legacy devices that do not learn location information from 
the network.


> For location-by-reference this is much more fuzzy. You probably don't 
> need a lecture on privacy regulations, but e.g. 
> http://www.isoc.org/briefings/015/index.shtml (from 2003 but still 
> applicable IMO) illustrates the mess.
> 
> A while ago I submitted a draft on Temporarily Routable User Agent URIs 
> (http://www.ietf.org/internet-drafts/draft-jbemmel-sip-truu-02.txt), 
> which provides anonymous, temporary URIs that can be explicitly 
> invalidated. You may want to consider something similar for 
> location-by-reference

Ciao
Hannes

> 
> Regards,
> 
> Jeroen
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
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(as individual)

I am not supporting or objecting, but I do believe that if it is to be a
using protocol, then there must be some form of RFC that specifies how
it meets the GEOPRIV requirements.

Regards

Keith=20

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]=20
> Sent: 27 July 2006 13:45
> To: IETF SIP List; GEOPRIV
> Subject: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>=20
> There have been a number of opinions regarding HTTP as a=20
> using protocol.
>=20
> I'd like to know, how many people believe that HTTP requires=20
> no or very little specification to meet the requirements in=20
> RFC 3693 and RFC 3694.  If you believe this, what basis do=20
> you have for this conclusion?  Or do you believe more=20
> specification needs to be done, but it is possible for HTTP=20
> to be a GEOPRIV using protocol?
>=20
> Finally, how many people believe that HTTP should not be a=20
> using protocol?
>=20
> -andy
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

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> ii)	if we have more than one option, we either need some means of
> determining that the dereferencer supports the one the client chooses to
> use, or require support of all options. As we seem to be lacking any
> mechanism for the former, then it would imply that all options have to
> be supported at the dereferencer, and therefore that the list needs to
> be a small list.

This isn't quite true. It is quite possible to include multiple options, 
as long as one of them is universally supported. For example, you can 
support a "universal & slow" and a "sometimes & fast" solution, and 
everybody wins.

This approach is not new: MIME email does exactly that, where messages 
with HTML or rich text often also contain an alternate plain text rendering.

Henning

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From sip-bounces@ietf.org Thu Jul 27 11:37:46 2006
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Hi Andy,

I would like to see the architectural questions being solved first. I 
don't think that it should go into the location conveyance document if 
there is interest to get it done soon.

Finally, I would like to see some strawman proposals first. I don't want 
to sign a blank check.

Ciao
Hannes

Andrew Newton wrote:
> There have been many conversations regarding constraining the URI in the 
> Location header.  There have been many positions taken on this subject, 
> and this email is intended to understand the various opinions.
> 
> Is there anybody who believes that the type of URI in the Location 
> header should not be constrained in any way?  In other words, any URI 
> scheme may be valid and there should be no attempt to specify the URI 
> scheme or associated protocol for dereferencing location.  And if so, do 
> you believe location will only be dereferenced via a user clicking on a 
> link or by automatic software action?  If the latter, how do you see 
> software adapting to multiple URI schemes and their associated protocols?
> 
> Is there anybody who believes that the dereference URI should be 
> restricted to a specific set of schemes, enforced via ABNF, XML Schema, 
> or other grammar specification and re-enforced with MUST/SHOULD 
> language?  The intent here being that the set would be definitive or at 
> the very least require major effort to update.
> 
> Finally, how many people desire a middle ground?  Perhaps where the 
> ABNF, XML Schema or other grammar specification allows for any URI 
> scheme but there is an IANA registry listing the allowed dereference URI 
> schemes and associated protocols?
> 
> -andy
> 
> 
> _______________________________________________
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> 


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From sip-bounces@ietf.org Thu Jul 27 11:37:52 2006
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Hi Keith,

Drage, Keith (Keith) wrote:
> (as individual)
> 
> It's a GEOPRIV problem to identify what height the bar is. At least my
> understanding is that Presence was a location using protocol and the
> level of documentation was agreed by GEOPRIV. That makes the bar around
> at least informational RFC required.

The presence based using protocol we did was meant to be used in a 
different environment. Let me point you to one main difference: The Rule 
Maker had a relationship with the Location Server. The Rule Maker might, 
in many cases, be the Target. In the discussions about a 
location-by-reference being added by a SIP proxy this is not the case.

I strongly believe that there is not a single 1-to-1 translation between 
the presence work and the work we do here when it concerns 
location-based routing as argued in other mails. You can convince me 
that this is not true (and you would probably also solve our Geopriv-L7 
  challenge as a side effect).

> 
> For use in location by reference I believe we have two criteria to meet:

Where do these requirements come from?

> 
> i)	the dereferencing protocol must be a geopriv using protocol

I think that there is a terminology problem here.

> 
> ii)	if we have more than one option, we either need some means of
> determining that the dereferencer supports the one the client chooses to
> use, or require support of all options. As we seem to be lacking any
> mechanism for the former, then it would imply that all options have to
> be supported at the dereferencer, and therefore that the list needs to
> be a small list.
This is indeed an issue but seems to be similar to the topics we deal 
with in the phone BCP.

In fact we already have this problem today if I understood 
draft-hdesinen-geopriv-pidflo-extn-00.txt correctly.

Ciao
Hannes


> 
> Regards
> 
> Keith
> 
> 
>>-----Original Message-----
>>From: Thomson, Martin [mailto:Martin.Thomson@andrew.com] 
>>Sent: 27 July 2006 06:00
>>To: Andrew Newton
>>Cc: IETF SIP List; GEOPRIV
>>Subject: RE: Teasing apart positions (was Re: [Sip] RE: 
>>[Geopriv] Consensusonchanges to location-conveyance)
>>
>>Since I raised the question, I'll answer.
>>
>>We have a set of requirements on a using protocol in RFC 3693.
>>
>>I believe that HTTP with proper constraints (TLS, etc...) can 
>>meet, or already meets, those requirements.  That is 
>>irrespective of any decisions that need to be made about 
>>secondary concerns like whether HTTP is desirable, optimal, 
>>deployable etc...  However, it is clear that people aren't 
>>happy with this position and are requesting that this be 
>>documented, as an RFC.  Just to get it all out in the open.  
>>Presumably this would be a standards track RFC that, like 
>>location-conveyance, highlights how the RFC 3693 requirements 
>>are addressed.
>>
>>That's a sound argument, and obviously location-conveyance 
>>had to meet that standard for us to be happy.  Now, I ask 
>>that you go back, take the above paragraph and perform one 
>>simple operation and read it again: s/HTTP/SIP/g.  Is the 
>>argument any different in the presence of RFC 4079?  Is an 
>>informational RFC that doesn't address specific requirements 
>>adequate?  I'd suggest that it wouldn't make the grade for HTTP.
>>
>>
>>>-----Original Message-----
>>>From: Andrew Newton [mailto:andy@hxr.us]
>>>Sent: Thursday, 27 July 2006 1:55 PM
>>>To: IETF SIP List; GEOPRIV
>>>Subject: Teasing apart positions (was Re: [Sip] RE: [Geopriv] 
>>>Consensus onchanges to location-conveyance)
>>>
>>>
>>>On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
>>>
>>>>The authors would like to know what to do now.
>>>
>>>Let's start by trying to tease apart certain positions.
>>>
>>>Given Jon Peterson's message:
>>>http://www1.ietf.org/mail-archive/web/sip/current/msg15537.html
>>>
>>>Does anybody still believe that subscriptions to PIDF-LOs with pres:
>>>requires more definition?
>>>
>>>-andy
>>>
>>>_______________________________________________
>>>Geopriv mailing list
>>>Geopriv@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>
>>--------------------------------------------------------------
>>----------------------------------
>>This message is for the designated recipient only and may 
>>contain privileged, proprietary, or otherwise private information.  
>>If you have received it in error, please notify the sender 
>>immediately and delete the original.  Any unauthorized use of 
>>this email is prohibited.
>>--------------------------------------------------------------
>>----------------------------------
>>[mf2]
>>
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From sip-bounces@ietf.org Thu Jul 27 11:38:15 2006
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Hi Henning,

here is the reason why I think we are not talking about a 'using 
protocol' here: All we want is to perform dereferencing.

We don't want something that does all the fancy features of a presence 
solution.

I am not even sure where the privacy rules should come from. In the 
Geopriv L7 discussion we had at least a proposal that some of us didn't 
like (namely the end host provides the privacy rules to the LIS). I 
haven't seen such a proposal here.

Here is my question (I raised it already with my scenario list): Where 
does the SIP proxy get the information to construct a PIDF-LO?

* We assume that it gets the location information somehow. Not further 
specified and might depend on the deployment environment.

* What is the "entity" attribute of the 'presence' element?

* Where does the content for the usage-rules (e.g, 
'retransmission-allowed', 'retention-expires', 'ruleset-reference', 
'note-well') come from?

Note that these questions are independent of HTTP. They also apply to SIP.

You might also recall the discussion regarding the difference between 
RELO and HELD when it comes to the threat model.

In RELO you assume that the entity that learns the reference is allowed 
to resolve it. In HELD there is the assumption that the end host 
additionally uploads the Geopriv authorization policies to control the 
access. We never came to a conclusion about the threat model that is 
applicable in the Geopriv L7 case.

I strongly believe that the questions are similar here and there are no 
answers to them. We are just exchanging one protocol (HTTP of RELO and 
HELD) with another one: SIP and route the call via this entity in the 
access network. If the SIP proxy that adds location information (by 
reference or value) is not in the access network then it cannot know the 
  location information of the end host....

Ciao
Hannes

  Henning Schulzrinne wrote:
> For all protocols, the hardest part is specifying how authentication  
> works in conjunction with the privacy rules. This is far from obvious  
> for HTTP, for example. (Is it the Digest user identifier? Can Basic  
> over TLS be used instead? What about crypto-random request URIs? Is  
> HTTP ok in that case or is HTTPS needed - the answer isn't obvious  
> since an observer would only see a location, not a location-identity  
> pair.)
> 
> For SIP, this is largely done, except for the crypto-random request  URI 
> case and variations, such as user + password.
> 
> On Jul 27, 2006, at 9:27 AM, Brian Rosen wrote:
> 
>> I think HTTP requires some specification.  A raw GET based  retrieval 
>> of a
>> PIDF would be a fairly short RFC, but I think we need that RFC.
>>
>> While I don't think it's necessary (SIP SUBSCRIBE is sufficient), I  
>> don't
>> object to defining HTTP as a using protocol in a suitable RFC.
>>
>> Brian
> 
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From sip-bounces@ietf.org Thu Jul 27 11:44:46 2006
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Hi Andy,

if we talk about location-by-reference then I think all solutions need 
more specification work.

If we talk about end host adding a reference then we need to have a 
story about the source of the reference. This is subject to the current 
Geopriv-L7 DT work that is not yet finished.

If we are talking about the case where the SIP proxy adds a reference 
(no matter what type of reference it is) then also further work is needed.

I think we should develop an HTTP based solution since it is very simple.

Btw, I am not sure that the name 'using protocol' is actually 
appropriate in this case. We haven't used the term using protocol for 
the Geopriv L7 work either. Terminology is hell in this area.

Ciao
Hannes

Andrew Newton wrote:
> There have been a number of opinions regarding HTTP as a using protocol.
> 
> I'd like to know, how many people believe that HTTP requires no or  very 
> little specification to meet the requirements in RFC 3693 and  RFC 
> 3694.  If you believe this, what basis do you have for this  
> conclusion?  Or do you believe more specification needs to be done,  but 
> it is possible for HTTP to be a GEOPRIV using protocol?
> 
> Finally, how many people believe that HTTP should not be a using  protocol?
> 
> -andy
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From: "Brian Rosen" <br@brianrosen.net>
To: "'Hannes Tschofenig'" <Hannes.Tschofenig@gmx.net>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Thu, 27 Jul 2006 12:17:30 -0400
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	Nadine B'" <nabbott@telcordia.com>
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Either.

I object to HTTP because there is no RFC I can reference that describes an
HTTP based geopriv using protocol.  I prefer SIP/SIPS/PRES because I can.

I don't think sip-location-conveyance should define an HTTP based geopriv
using protocol.

I don't object to a separate document that defines one.  I expect to see the
L7 protocol do that, for example.

Brian

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> Sent: Thursday, July 27, 2006 10:34 AM
> To: Brian Rosen
> Cc: 'Abbott, Nadine B'; sip@ietf.org; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
> 
> Hi Brian,
> 
> are you talking about the end host adding a location-by-reference into
> the SIP header or a proxy doing so?
> 
> In any case I don't understand your objection against the usage of HTTP
> and your preference for SIP/SIPS/PREZ?
> 
> Ciao
> Hannes
> 
> Brian Rosen wrote:
> > Nadine
> >
> > Are you saying that it's your opinion sip-location-conveyance MUST
> > define an HTTP dereference procedure?
> >
> > If you do, why?  What is the use-case or requirement?
> >
> > Would it not be pretty reasonable to require a separate document that
> >  defines a location-by-reference mechanism, protocol, and geopriv
> > "using protocol" analysis?  If that document specifically extended
> > what was allowed in the location header, would that not be
> > reasonable?
> >
> > I think we have to include "A" dereference mechanism in the
> > location-conveyance document.  It seems just including sip/sips/prez
> > is a reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.
> > That certainly does not preclude another document describing another
> > way.
> >
> > Doesn't it seem just a little weird to have a sip-location-conveyance
> >  document describing any kind of http based location conveyance
> > (de-reference) protocol, especially one created whole cloth without
> > any other parts?
> >
> > Doesn't it seem in keeping with the geopriv approach that in order to
> > have another "using protocol" it has to have a geopriv review?  That
> > is what leads to restricting the URI schemes.  Would you prefer to
> > let it be wide open, so any protocol anyone thinks about can be used
> > with the location header?  I can't see geopriv going along with that.
> > We're going to make it a one line ABNF to add a new scheme to the
> > list of allowed schemes.
> >
> > It seems to me that if it's desired to define some bare-bones HTTP
> > location by reference protocol, then that should be a separate
> > document, which can easily extend the allowable schemes in the
> > Location header.  I'm not sure it might be better to define ALL of
> > that in the L7 protocol, but I wouldn't be opposed to a simple "raw
> > GET" RFC as long as it got a full geopriv review. I just don't see
> > where it makes any sense to force that into a sip document.
> >
> > Brian
> >
> >
> >> -----Original Message----- From: Abbott, Nadine B
> >> [mailto:nabbott@telcordia.com] Sent: Wednesday, July 26, 2006 2:14
> >> PM To: sip@ietf.org Cc: geopriv@ietf.org Subject: RE: [Geopriv]
> >> Consensus on changes to location-conveyance
> >>
> >>
> >> I don't agree that there is a consensus to preclude the use of
> >> HTTP. I would favor not placing your proposed restriction on the
> >> URI scheme for expressing a location reference in the SIP location
> >> conveyance draft; or at least including HTTP as a valid
> >> location-reference URI type.
> >>
> >> Regards, Nadine Abbott
> >>
> >> -----Original Message----- From: Rosen, Brian
> >> [mailto:Brian.Rosen@neustar.biz] Sent: Tuesday, July 25, 2006 1:38
> >> PM To: sip@ietf.org Cc: geopriv@ietf.org Subject: [Geopriv]
> >> Consensus on changes to location-conveyance
> >>
> >> I have put out a number of emails on proposed changes to location
> >> conveyance.  Some of them have sparked a number of comments.  My
> >> summary of consensus on these issues is:
> >>
> >> 1. We proposed to explicitly disallow use of a Data URI, which
> >> would allow specification of a Location-by-value in a header.
> >> There was a lot of list traffic on this, currently in the state
> >> where Keith asked for a real use case for it.  So far, we don't see
> >> a use case that cannot be provided by location-by-reference in a
> >> header.  If we don't get a good use case, I think consensus is to
> >> disallow location-by-value in a header.
> >>
> >> 2. Dereferencing location-by-reference.  We had initially proposed
> >> to document a process to use a raw HTTP(S) Get to dereference.
> >> There was a lot of discussion, and then I made a new proposal, see
> >> item 6.
> >>
> >> 3. We proposed to allow a Reason header with a 424 Bad Location
> >> response.  This seems to be acceptable to the WG.  Geopriv will
> >> create and populate the registry in a separate draft.
> >>
> >> 4. We proposed to explicitly allow hop-by-hop TLS security for
> >> location when the endpoint wished to allow routing based on
> >> location.  We further proposed to interpret "Do not Distribute"
> >> flags as allowing sending of the location to the routing database
> >> (LoST for example).  Hannes pointed out that there was a proposal
> >> to allow two Location headers, one of which could be used for
> >> routing and another for onward forwarding to the recipient.
> >> Hannes' message seemed to agree that "Do not distribute" would
> >> allow sending of the location to the routing-by-location database,
> >> but would obligate the proxy that did that to remove the location
> >> before it forwarded the message.  There were no other messages on
> >> this thread. We conclude that allowing hop-by-hop security for
> >> location based routing is acceptable, and Do Not Distribute DOES
> >> allow forwarding to a routing database, but requires that the proxy
> >> doing the location based route to remove the location it uses for
> >> that purpose.
> >>
> >> 5. We proposed to create a parameter and a registry to distinguish
> >> multiple ways to dereference a location where HTTP(S) was the
> >> transport. Resolution of this issue depends on resolution of issue
> >> 6.
> >>
> >> 6. We proposed to allow sip/sips and pres: in the location header.
> >> I later speculated that if we allow this, it forms a simple way to
> >> do location-by-reference, and we could eliminate the raw HTTP(S)
> >> Get proposal, with the parameter and registry entirely.  There was
> >> some list traffic in favor of this.  Martin Thompson is skeptical
> >> that SIP Presence Subscribe/Notify is actually allowed as a geopriv
> >> "using protocol".  Several of think that it is.  If we don't get
> >> any more comments on this proposal, we will allow sip/sips and
> >> pres, not have any http(s) resolution text, and not propose a
> >> parameter and accompanying registry.
> >>
> >> If this does not reflect your position, please speak up now.
> >>
> >> Brian
> >>
> >> _______________________________________________ Geopriv mailing
> >> list Geopriv@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/geopriv
> >>
> >> _______________________________________________ Geopriv mailing
> >> list Geopriv@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/geopriv
> >
> >
> >
> > _______________________________________________ Sip mailing list
> > https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
> > development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip Use
> > sipping@ietf.org for new developments on the application of sip
> >
> >


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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
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Subject: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV using protocol
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(As individual)

At the moment I'm afraid I don't buy the argument.

Starting from RFC 3693, section 3 defines:

	Using Protocol: A protocol that carries a Location Object.

For which we also have:

      Location Object (LO): An object conveying location information
         (and possibly privacy rules) to which Geopriv security
         mechanisms and privacy rules are to be applied.

Section 4 defines the primary GEOPRIV entities and=20
this defines a model:

(Now for SIP location by value, the using protocol is SIP, and the SIP
UAC is the location server, and the SIP UAS (or possibly proxy) is the
location recipient.)

For SIP location by reference, the using protocol is the dereferencing
protocol, i.e. the reader of the Location header is the location
recipient, and the server referenced by the URI is the location server.
The SIP protocol is this case is not a using protocol, because it does
not contain a location object. Therefore it falls outside the rules of
using protocols specified by GEOPRIV.

I see nothing in RFC 3693 that prevents a third party entity holding
location on behalf of a user, and in fact some of the examples seem to
show exactly that.

Now presumably all the security and privacy constraints relating to the
user are covered by the using protocol. The fact that I tell you that X
has my location doesn't mean you are entitled to receive it. Further,
while there may be some other protocol (or bilateral agreement or
whatever) that covers the relationship between X and the user, I do not
see any requirement in RFC 3963 that mandates that.

At the moment you seem to be arguing that information about who has my
location is subject to some form of consent framework over and above
that required by the using protocol itself. I'm afraid I do not see that
argument at all. Agreed the URI of the server may say something about
the user's location, but to me this is covered by the following
statement in RFC 3693:

   Excluded from this definition is the determination of location
   information wholly without the knowledge or consent of the Target (or
   the Target's network or access service provider), based on generally
   available information such as an IP or e-mail address.  In some
   cases, information like IP address can enable someone to estimate (at
   least roughly) a location.  Commercial services exist that provide
   rough location information based on IP addresses.  Currently, this
   type of location information is typically less precise than the type
   of location information addressed in this document.  Although this
   type of location computation still raises significant potential
   privacy and public privacy concerns, such scenarios are generally
   outside the scope of this document.

We have specified lots of things in SIP and many other protocols that
need lots more rules if this is going to become an issue.

Furthermore, if I follow your argument to its logical conclusion, then I
would seem to need lots more work before I can put a URI that points at
a location server in an ordinary email, yet I seem to be able to do that
today.

So in conclusion, I see location by reference as two independent
protocols:

-	a location using protocol which is the deferencing protocol, and
which obviously has to meet the using protocol requirements.

-	a URI carried by SIP which is subject to no normative
constraints about who can read or whatever, which identifies the
location server.

I agree that there may be arguments for giving more information to the
location recipient so that the location recipient knows how to populate
the dereferencing protocol. Some of this may however already be covered
by other SIP protocol work, and to me is not directly necessary to make
the protocol work. If we agree any information in time, we can include
it, otherwise it can be added at a later stage.=20



Regards

Keith

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: 27 July 2006 16:36
> To: Andrew Newton
> Cc: IETF SIP List; GEOPRIV
> Subject: Re: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>=20
> Hi Andy,
>=20
> if we talk about location-by-reference then I think all=20
> solutions need more specification work.
>=20
> If we talk about end host adding a reference then we need to=20
> have a story about the source of the reference. This is=20
> subject to the current
> Geopriv-L7 DT work that is not yet finished.
>=20
> If we are talking about the case where the SIP proxy adds a=20
> reference (no matter what type of reference it is) then also=20
> further work is needed.
>=20
> I think we should develop an HTTP based solution since it is=20
> very simple.
>=20
> Btw, I am not sure that the name 'using protocol' is actually=20
> appropriate in this case. We haven't used the term using=20
> protocol for the Geopriv L7 work either. Terminology is hell=20
> in this area.
>=20
> Ciao
> Hannes
>=20
> Andrew Newton wrote:
> > There have been a number of opinions regarding HTTP as a=20
> using protocol.
> >=20
> > I'd like to know, how many people believe that HTTP requires no or =20
> > very little specification to meet the requirements in RFC 3693 and =20
> > RFC 3694.  If you believe this, what basis do you have for this=20
> > conclusion?  Or do you believe more specification needs to=20
> be done, =20
> > but it is possible for HTTP to be a GEOPRIV using protocol?
> >=20
> > Finally, how many people believe that HTTP should not be a=20
> using  protocol?
> >=20
> > -andy
> >=20
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

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From sip-bounces@ietf.org Thu Jul 27 13:27:41 2006
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Inline...

Hannes Tschofenig wrote:
> Hi Henning,
> 
> here is the reason why I think we are not talking about a 'using 
> protocol' here: All we want is to perform dereferencing.

Since the place where the reference is made also includes an identity, 
it is very easy to get the location of a user... and hence the line is 
crossed with regard to privacy.

> We don't want something that does all the fancy features of a presence 
> solution.

So why did GEOPRIV use PIDF as its basis?  Has as been quoted on this 
list recently, 4119 does not require the use of presence solution, but 
presence solutions were very much a target platform.

> I am not even sure where the privacy rules should come from. In the 
> Geopriv L7 discussion we had at least a proposal that some of us didn't 
> like (namely the end host provides the privacy rules to the LIS). I 
> haven't seen such a proposal here.

An L7-LCP does not have to be a using protocol primarily because the 
usage is for a much narrower purpose.  It can be a GEOPRIV using 
protocol, but I believe one of the complaints against HELD (which 
attempted to be an L7-LCP and GEOPRIV using protocol) is that it was 
doing too much.

> Here is my question (I raised it already with my scenario list): Where 
> does the SIP proxy get the information to construct a PIDF-LO?
> 
> * We assume that it gets the location information somehow. Not further 
> specified and might depend on the deployment environment.
> 
> * What is the "entity" attribute of the 'presence' element?
> 
> * Where does the content for the usage-rules (e.g, 
> 'retransmission-allowed', 'retention-expires', 'ruleset-reference', 
> 'note-well') come from?

It is possible that a network operator does have enough information to 
formulate a PIDF-LO, at least what is necessary for emergencies.

-andy


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Subject: [Sip] RFC4320 and 408 to non-INVITE
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Robert,

RFC4320 bans 408 responses to NIT, but I think there are situations in whic=
h it is still useful and needed. In particular, consider when the UAC is us=
ing T1=3D2000 and the proxy uses T1=3D500 (eg because the client is on a sl=
ow wireless link)

The CT at the proxy times out after 64*T1=3D32 sec, while the client is sti=
ll retransmitting. In this case, 408 does help

Regards,
Jeroen=


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From: "Marc Berryman" <MBerryman@911.org>
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Cc: IETF SIP List <sip@ietf.org>, geopriv@ietf.org,
	Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: [Sip] RE: [Geopriv] Henning's Suggestion
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Sounds like Hannes and Henning are moving down the right track. Focusing
on the location header syntax for the draft.



Marc B

-----Original Message-----
From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
Sent: Thursday, July 27, 2006 9:35 AM
To: Dawson, Martin
Cc: IETF SIP List; geopriv@ietf.org; Henning Schulzrinne
Subject: [Geopriv] Henning's Suggestion

Hi Henning,

I like your suggestion.

In fact I would even go a step further. The SIP Location Conveyance=20
document just defines a mechanism to carry a PIDF-LO in the body of the=20
SIP message. Done.
Adding location-by-reference to the SIP header by the end host sounds OK

but there are still some loose ends where this reference comes from.

The other stuff can be dealt with once we concluded our Geopriv-L7 work=20
where all the architectural pieces should are created. The=20
subscription-URI is an example where we recently proposed something and=20
haven't finished our technical discussions.

I think the draft got fuzzy when it came to the area where a SIP proxy=20
adds a location-by-reference. I must admit that I proposed to add=20
location-by-value into the SIP header as well but in the meanwhile I=20
don't think that any of the SIP proxy stories belong in there.

Ciao
Hannes

>>-----Original Message-----
>>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>>Sent: Thursday, 27 July 2006 1:25 PM
>>To: Brian Rosen
>>Cc: IETF SIP List; geopriv@ietf.org
>>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
>>conveyance
>>
>>My suggestion would be
>>
>>- to focus the draft on just defining the Location header syntax,
>>without ruling out any particular mechanism or scheme; this would
>>make the draft really short (which I would find helpful).
>>
>>- and then add one section to the draft that prototypically defines
>>what a particular scheme that wants to appear in the Location header
>>needs to define, using the non-controversial cid mechanism as the
>>example. Other, separate, drafts can then define other modes,
>>following the same template. Documents such as the phone-bcp can then
>>pick some suitable must-implement subset and make emergency-call-
>>specific recommendations, rather than sprinkling these throughout the
>>conveyance document and repeating some in the phone-bcp. Most of
>>these are, in any event, operational recommendations, not for
>>ensuring protocol-level interoperability.
>>
>>That way, the conveyance draft can be extremely short (even shorter
>>if the requirements are omitted...) and can get done, and we avoid
>>engineering operational requirements on the fly, commingling
>>emergency and non-emergency cases, as seems to be happening now.
>>
>>I think protocol documents should define what's needed for
>>interoperability; BCPs can make more specific, situation-specific
>>recommendations. Commingling the two seems to be a recipe for
>>deadlock or arbitrary, non-technically-grounded decisions that are
>>more lawyer-like than engineering-like ("you did not mention
>>technique X in some meeting N years ago and thus you can't bring it
>>up now!")
>>
>>Henning
>>
>>On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
>>
>>
>>>At this point, I think we need some chair guidance and by that I
>>>think it's
>>>a combo of geopriv and sip chairs.
>>>
>>>We seem to have a couple of significant open questions.  I think
>>>most of the
>>>other stuff (like the "allow sending to router") will converge.
>>>
>>>There seems to be some opposition to allowing sip/sips/pres: in a
>>>Location
>>>Header.  The opposition seems to be that despite 4119, somehow, SIP
>>>(presence subscription) is not a "using protocol".  I "think" a
>>>ruling from
>>>the bench on that would work.
>>>
>>>There is also opposition to defining the allowed schemes as a
>>>specific list,
>>>with a defined (easy) extension mechanism.
>>>
>>>Related to that, there are some who want location-conveyance to
>>>define an
>>>HTTP based location dereference mechanism, and on the opposite end,
>>>we have
>>>people saying we haven't even agreed to support any form of
>>>location-by-reference.
>>>
>>>The authors would like to know what to do now.
>>>
>>>Brian
>>>
>>>
>>>>-----Original Message-----
>>>>From: Andrew Newton [mailto:andy@hxr.us]
>>>>Sent: Wednesday, July 26, 2006 9:58 PM
>>>>To: Winterbottom, James
>>>>Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning
>>>>Schulzrinne
>>>>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
>>>>conveyance
>>>>
>>>>
>>>>On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
>>>>
>>>>
>>>>>Andy,
>>>>>
>>>>>RFC-4119 says "However, the usage of this location object format
>=20
> is
>=20
>>>>>not
>>>>>   limited to presence-using protocols-- any protocol that can
>>>>>carry XML
>>>>>   or MIME types can carry PIDF."
>>>>>
>>>>>It seems to me that restricting the allowed schema in what is
>>>>>supposed
>>>>>to be a location conveyance specification is in opposition to this
>>>>>statement.
>>>>
>>>>Did somebody say that HTTP was a presence-using protocol?
>>>>
>>>>-andy
>>>>
>>>>ps.  you can find my location with the following reference URIs:
>>>>
>>>>gopher://fatchance.hxr.us/you-still-have-a-gopher-client
>>>>news:alt.locations.people.secretly-being-watched
>>>>telnet://screen-scrapers.are.us/
>>>>dns:cached-publicly-around-the-world.com?type=3DHOST
>>>>
>>>>
>>>>
>>>>_______________________________________________
>>>>Geopriv mailing list
>>>>Geopriv@ietf.org
>>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>>
>>>
>>>_______________________________________________
>>>Geopriv mailing list
>>>Geopriv@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>
>>
>>_______________________________________________
>>Geopriv mailing list
>>Geopriv@ietf.org
>>https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>=20
>
------------------------------------------------------------------------
------------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information. =20
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
>
------------------------------------------------------------------------
------------------------
> [mf2]
>=20
> _______________________________________________
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>=20
>=20


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From sip-bounces@ietf.org Thu Jul 27 13:40:53 2006
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Hi Andy,

Andrew Newton wrote:
> Inline...
> 
> Hannes Tschofenig wrote:
> 
>> Hi Henning,
>>
>> here is the reason why I think we are not talking about a 'using 
>> protocol' here: All we want is to perform dereferencing.
> 
> 
> Since the place where the reference is made also includes an identity, 
> it is very easy to get the location of a user... and hence the line is 
> crossed with regard to privacy.

What identity is used where?

> 
>> We don't want something that does all the fancy features of a presence 
>> solution.
> 
> 
> So why did GEOPRIV use PIDF as its basis? 

You misunderstood my statement. For dereferencing a 
location-by-reference (not a subscription URI) we don't need a lot of 
functionality.

  Has as been quoted on this
> list recently, 4119 does not require the use of presence solution, but 
> presence solutions were very much a target platform.
> 
>> I am not even sure where the privacy rules should come from. In the 
>> Geopriv L7 discussion we had at least a proposal that some of us 
>> didn't like (namely the end host provides the privacy rules to the 
>> LIS). I haven't seen such a proposal here.
> 
> 
> An L7-LCP does not have to be a using protocol primarily because the 
> usage is for a much narrower purpose.  It can be a GEOPRIV using 
> protocol, but I believe one of the complaints against HELD (which 
> attempted to be an L7-LCP and GEOPRIV using protocol) is that it was 
> doing too much.

I like to discuss about requirements and functionality rather than 
terminology. I think that statements like "has to be a using protocol" 
help to make a lot of progress.

That's my only objection.

I am the last one to say that we do not need to deal with privacy aspects.

> 
>> Here is my question (I raised it already with my scenario list): Where 
>> does the SIP proxy get the information to construct a PIDF-LO?
>>
>> * We assume that it gets the location information somehow. Not further 
>> specified and might depend on the deployment environment.
>>
>> * What is the "entity" attribute of the 'presence' element?
>>
>> * Where does the content for the usage-rules (e.g, 
>> 'retransmission-allowed', 'retention-expires', 'ruleset-reference', 
>> 'note-well') come from?
> 
> 
> It is possible that a network operator does have enough information to 
> formulate a PIDF-LO, at least what is necessary for emergencies.
Would you be happy with this level of detail?
... and: We are not only focusing on emergency services.


Ciao
Hannes

> 
> -andy
> 
> 


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With hindsight, I think the term 'using protocol' has been the source of 
  more word-lawyering than engineering insight. I personally find the 
distinctions

   - first person vs. third person (i.e., find out location about myself 
vs. find out location about some third party)

   - location vs. location + identity

   - token-based authorization vs. identity-based authorization

more helpful.

Token-based authorization hands the recipient a key that allows him to 
get a location object, without caring as to who is retrieving the 
object. ("Anybody seeing this SIP request can convert the token into an 
LO.") Identity-based naturally uses some authenticated identity to 
release that information. Strictly speaking, this is only a question of 
a level of indirection, since proving identity usually involves proving 
possession of a secret.

We've been trying to get at some of these distinctions with the sighting 
vs. using protocol terminology, but that distinction also seems to 
degenerate into discussions more suited for a law school or theology 
department than an engineering organization.

For example, by strict definition, LoST would be a 'Using Protocol' 
since it "carries a Location Object", but I hope we all agree that this 
is silly. We had earlier agreed that the location-by-DHCP also were not 
Using Protocol, even though they obviously carry a location object of 
some form.

Privacy issues apply iff
   - third person
   - location + identity (not just location)

The L7 discussion was about 'first person' and 'location only' (although 
the latter is subject to debate, depending on whether one considers an 
IP address of identifying an individual or not).

Depending on the circumstances, SIP-based retrieval can either involve 
privacy concerns (subscribe to alice@example.com) or not (subscribe to 
location of one-time random unlinkable identifier 
jdfa8f9348@example.com). There can also be different privacy concerns 
depending on the perspective. For example, if a SIP request contains 
both an identity and the location retrieval reference, this raises 
privacy concerns for the proxy, but the LIS may not have any such 
concerns since it may only be the repository of these randomly-named 
objects, without tie to an identity.


Henning





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From: Robert Sparks <rjsparks@estacado.net>
Date: Thu, 27 Jul 2006 12:51:28 -0500
To: Jeroen Van Bemmel <jbemmel@zonnet.nl>
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Subject: [Sip] Re: RFC4320 and 408 to non-INVITE
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This is not the way the timers are meant to be changed.

Both of the transaction-stateful things on either end of a given  
transaction must share the same
values for the transaction timers or the results are not defined.

What you describe is not unlike throwing away the timing belt in an  
engine and spinning
the head at some rate unrelated to the crankshaft. You might get  
lucky, but usually very bad things
are going to happen.

RjS

On Jul 27, 2006, at 12:29 PM, Jeroen Van Bemmel wrote:

> Robert,
>
> RFC4320 bans 408 responses to NIT, but I think there are situations  
> in which it is still useful and needed. In particular, consider  
> when the UAC is using T1=2000 and the proxy uses T1=500 (eg because  
> the client is on a slow wireless link)
>
> The CT at the proxy times out after 64*T1=32 sec, while the client  
> is still retransmitting. In this case, 408 does help
>
> Regards,
> Jeroen


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From: "Brian Rosen" <br@brianrosen.net>
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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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> > I don't really understand why you think location by reference is so much
> of
> > a concern.
> 
> It is interesting that you say that. In the Geopriv L7 design team you
> are the only person that wants to see an applicability statement being
> written.
We haven't gotten to that yet, but it's a problem of proxy assertion, not
location-by-reference.  Basically, coming up with an identifier to determine
what location to assert (by value or by reference) is much worse than the L7
case, because the SIP signaling plane definitely is subject to being carried
over VPNs and other IP address obscuring things.  You can't possibly use the
IP address as the identifier.  You have to use something else.  What else?
I have no idea.  I can imagine some kinds of wireless systems being able to
know that the device is on their own network, and have some id that works on
that network.

In case I have not been clear enough, I personally DO NOT LIKE proxy
assertion of location.  I do not really like location by reference, although
a SIP presence URI is location-by-reference no matter how you slice and dice
it.

However, I consider both to be requirements in location-conveyance, and
think limiting proxy insertion to be by reference only to be the least
objectionable answer.

> 
> Having a SIP proxy adding something (either location-by-value or
> location-by-reference) is not different to the stuff we discuss in the
> design team. There, you see a lot of privacy, security, etc. problems.
> Here you don't. How come?
I see plenty of problems, but in a protocol sense I don't think anything is
different other than this identifier problem.

> 
> If you go through my mail with the large number of scenarios then you
> find a number of open issues that need to be answered. Alternatively,
> you can also go through draft-tschofenig-geopriv-l7-lcp-ps-00.txt
> 
>   If you look at the discussion in context, PIDF-LO defines
> > location as part of presence.  All of these concerns are relevant there,
> and
> > I think they have been adequately covered there.  We even have some work
> on
> > anonymization of the identity (and thus of the URI) in presence.
> Generally,
> > a presence uri:
> >
> > Is not valid for a limited amount of time
> > Is not hard to guess, if it uses an AoR and could be with anonymous URI.
> > Identifies the user if it uses an AoR and doesn't with anonymous URI
> >
> > The protocol (SIP) for retrieving presence, and thus location has the
> > location ruleset as we defined for PIDF-LO, but otherwise does not have
> "no
> > cache" semantics (albeit caching headers is not very interesting in
> SIP).
> 
> You cannot compare presence witht he location-by-reference solution
> since the deployment case is different.
I believe this to be false on its face.  If I give you a sip URI and tell
you a presence server will respond to it including location, then you can
exchange the URI for location, and that is location-by-reference.

You cannot put that cat back in the bag.  A presence URI that reports
location in the PIDF results in location-by-reference.

The idea that you could have a presence URI where the only information
returned in the PIDF that was interesting was location does not need an RFC.
It's obvious, it's allowed, and the anonymized user stuff was created for
this purpose.

Presence is location by reference (it can be identification-by-reference, it
can be happiness-by-reference).  You have a URI, you exchange it for a PIDF.
You are done, stop arguing.


> 
> >
> > Generally, presence systems do have user control of who is allowed to
> access
> > information, including location and identifying information, so I think
> that
> > is covered fairly well.
> 
> Not true in this case.
> 
> Here we have a SIP proxy in the access network doing all the work. You
> can compare it with a SIP-based LIS from the Geopriv-L7 work. This node
> has no clue about your privacy preferrences and does not know your
> identity. This is what we discussed in the past and several proposals
> where made, namely:
> 
> * end host provides privacy preferences (see HELD)
> * end host does not provide privacy preferences since there is no
> PIDF-LO (see RELO).

I don't agree.  A proxy can assert a presence URI in a number of places.
I'm pretty sure, for example, that many carriers intend the P-A-I to be a
presence URI, or (if a tel uri) convertible trivially to a presence URI.

I don't like it, but whether a proxy asserts your presence URI or your
endpoint does, there is no functional difference.  I worry about the issue
of whether the proxy's URI resolves to a PIDF that has the policies you want
asserted.  But that is not a protocol issue.  That is a contract and local
law issue.  The URI and the PIDF have the same capacity to express the users
wishes with respect to identity, location and all the other parts of the
PIDF.

Your carrier's outgoing proxy is perfectly capable of asserting a presence
URI that resolves to a PIDF that represents your policies on your location.

I prefer the endpoint to be the one doing this, because I think there is a
better chance it will get it done right, but that's a pretty soft thing,
having nothing to do with protocols.  The proxy is in the same domain as the
user, and is being trusted by the user to correctly handle its calls.
Putting his location (via presence) is one of the things it is reasonable to
trust the proxy to do.

You could look at this a different way.  If the network determines the user
location, that information belongs to the network.  It would be up to the
network to tell the user, if it wants to.  The user doesn't own the data,
even if it represents his location.  If I take a picture of you, I own the
picture.  Some laws may limit what I can do with it, but it's my data.
Proxy assertion of location is the network telling someone the data it owns.
It's not yours unless the network decides to give it to you, or gives you
control over it.  Again, laws may limit what it can do with the data.
> 
> >
> > Proxy insertion of location, whether by value or by reference has the
> threat
> > of not being as subject to user control, but I think in every case you
> have
> > to trust your provider to do as you wish, whether it is your presence
> > provider or your location provider independent of your presence
> provider,
> > and proxy insertion doesn't fundamentally alter that.  I worry more
> about
> > it, but I don't objectively think it's much different.
> 
> In this case you don't really know the provider and it might not be your
>   provider.
Actually, it has to be (or it wouldn't know your location).  You have the
case of the visited network not being your carrier, but effectively, if you
are roaming on their network, they are your carrier of choice.

> 
> >
> > Now, if there are other location dereferencing systems, they may need an
> > analysis like what you are suggesting, but with respect to SIP Presence,
> as
> > a form of location by reference, I think we have been over the ground
> > sufficiently, and we have RFCs that permit it.  Adding restrictions on
> the
> > Location header, which I propose is useful precisely to allow an
> anonymous
> > presence uri that does not reveal identity is not so great I think.  If
> you
> > don't allow that, you would kind of have to use an AoR, which may be
> > appropriate in some cases, but would not in others.
> 
> The reference might not reveal the identity but what about the identity
> in the PIDF-LO. If you argue that there does not need to be one (or it
> is a faked one) then I am puzzled why it has been a source of major
> confusion in the Geopriv-L7 design team work (and in previous
> discussions about the same subject in Geopriv).
See above.  It's easy to generate a completely random identity.  The
difficulty in the L7 case is how the endpoint identifies itself to the LIS
in order to receive the right location or reference.  Proxy insertion has
the same problem of how it knows which location to assert.  You can't do
proxy insertion of location unless you know FOR SURE where the device is.
We do need text on that, as above.

> 
> >


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Subject: RE: Teasing apart positions (was Re: [Sip] RE:
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True, but I haven't yet seen any proposals to do that. That would
presumably require two URIs in the Location header or some other means
of identifying the two protocols with the same URI.

Keith=20

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]=20
> Sent: 27 July 2006 16:34
> To: Drage, Keith (Keith)
> Cc: IETF SIP List; GEOPRIV
> Subject: Re: Teasing apart positions (was Re: [Sip] RE:=20
> [Geopriv] Consensusonchanges to location-conveyance)
>=20
> > ii)	if we have more than one option, we either need some means of
> > determining that the dereferencer supports the one the=20
> client chooses=20
> > to use, or require support of all options. As we seem to be lacking=20
> > any mechanism for the former, then it would imply that all options=20
> > have to be supported at the dereferencer, and therefore=20
> that the list=20
> > needs to be a small list.
>=20
> This isn't quite true. It is quite possible to include=20
> multiple options, as long as one of them is universally=20
> supported. For example, you can support a "universal & slow"=20
> and a "sometimes & fast" solution, and everybody wins.
>=20
> This approach is not new: MIME email does exactly that, where=20
> messages with HTML or rich text often also contain an=20
> alternate plain text rendering.
>=20
> Henning
>=20

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Subject: [Sip] Re: [Geopriv] teasing apart: http as a GEOPRIV using protocol
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Hi Keith,

please find my response below:

Drage, Keith (Keith) wrote:
> (As individual)
> 
> At the moment I'm afraid I don't buy the argument.


> 
> Starting from RFC 3693, section 3 defines:
> 
> 	Using Protocol: A protocol that carries a Location Object.
> 
> For which we also have:
> 
>       Location Object (LO): An object conveying location information
>          (and possibly privacy rules) to which Geopriv security
>          mechanisms and privacy rules are to be applied.
> 
> Section 4 defines the primary GEOPRIV entities and 
> this defines a model:
> 
> (Now for SIP location by value, the using protocol is SIP, and the SIP
> UAC is the location server, and the SIP UAS (or possibly proxy) is the
> location recipient.)
> 
> For SIP location by reference, the using protocol is the dereferencing
> protocol, i.e. the reader of the Location header is the location
> recipient, and the server referenced by the URI is the location server.
> The SIP protocol is this case is not a using protocol, because it does
> not contain a location object. Therefore it falls outside the rules of
> using protocols specified by GEOPRIV.

The problem is that this document was written some time ago. We haven't 
thought about many things at that time. Hence, it is fine to say that it 
has to be a using protocol but what does this tell us exactly? Nothing 
since we have to worry about privacy/security aspects with every 
document we develop.

The entire mailing list thread started with a few security related 
questions that Henning and myself raised. I do not recall that we made 
progress with any of these security questions. Instead we go along the 
road of terminology discussions.

> 
> I see nothing in RFC 3693 that prevents a third party entity holding
> location on behalf of a user, and in fact some of the examples seem to
> show exactly that.

That's explicitly supported.

> 
> Now presumably all the security and privacy constraints relating to the
> user are covered by the using protocol. The fact that I tell you that X
> has my location doesn't mean you are entitled to receive it. Further,
> while there may be some other protocol (or bilateral agreement or
> whatever) that covers the relationship between X and the user, I do not
> see any requirement in RFC 3963 that mandates that.

I believe that security is part of the architecture and in this case 
several protocols play together. I don't want to get the security work 
being pushed to the HTTP using protocol.

> 
> At the moment you seem to be arguing that information about who has my
> location is subject to some form of consent framework over and above
> that required by the using protocol itself. I'm afraid I do not see that
> argument at all. Agreed the URI of the server may say something about
> the user's location, but to me this is covered by the following
> statement in RFC 3693:

I don't think I said that.

> 
>    Excluded from this definition is the determination of location
>    information wholly without the knowledge or consent of the Target (or
>    the Target's network or access service provider), based on generally
>    available information such as an IP or e-mail address.  In some
>    cases, information like IP address can enable someone to estimate (at
>    least roughly) a location.  Commercial services exist that provide
>    rough location information based on IP addresses.  Currently, this
>    type of location information is typically less precise than the type
>    of location information addressed in this document.  Although this
>    type of location computation still raises significant potential
>    privacy and public privacy concerns, such scenarios are generally
>    outside the scope of this document.
> 
> We have specified lots of things in SIP and many other protocols that
> need lots more rules if this is going to become an issue.
> 
> Furthermore, if I follow your argument to its logical conclusion, then I
> would seem to need lots more work before I can put a URI that points at
> a location server in an ordinary email, yet I seem to be able to do that
> today.

If I just put a location server URI into the mail then there is still 
the question how you would obtain my location information.

If my email address is used for this purpose then you would have just 
created a very nice privacy leaking system. Everyone that knows the 
location server and my email address could learn my location.

Furthermore, mail relays would be able to see that information and could 
happily use it.

Btw, you can also put location information into a mail today and without 
signing and privacy rules it would be bad as well.


> 
> So in conclusion, I see location by reference as two independent
> protocols:
> 
> -	a location using protocol which is the deferencing protocol, and
> which obviously has to meet the using protocol requirements.
> 
> -	a URI carried by SIP which is subject to no normative
> constraints about who can read or whatever, which identifies the
> location server.

I would particuarly like to make anyone aware that these are not just 
two independent protocols. I strongly believe that you can make the 
security/privacy independent for each of them. You have to consider the 
big picture.

We do this all the time. Take for example, the SAML work. Two 
"independent things": SAML and SIP
Still, there are many aspects to consider.
The SAML community calls these interworking between two (or more 
protocols) 'profile'. What we do here is essentially the same.

A recent comment that the end host should indicate whether it wants the 
SIP proxy to add something is also a topic for SIP-SAML. If an entity 
acts on behalf of the end host then it needs to be aware of the context 
this entity is in (Henning would call it sphere).

> I agree that there may be arguments for giving more information to the
> location recipient so that the location recipient knows how to populate
> the dereferencing protocol. Some of this may however already be covered
> by other SIP protocol work, and to me is not directly necessary to make
> the protocol work. If we agree any information in time, we can include
> it, otherwise it can be added at a later stage. 

Ciao
Hannes



> 
> 
> Regards
> 
> Keith
> 
> 
>>-----Original Message-----
>>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>>Sent: 27 July 2006 16:36
>>To: Andrew Newton
>>Cc: IETF SIP List; GEOPRIV
>>Subject: Re: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>>
>>Hi Andy,
>>
>>if we talk about location-by-reference then I think all 
>>solutions need more specification work.
>>
>>If we talk about end host adding a reference then we need to 
>>have a story about the source of the reference. This is 
>>subject to the current
>>Geopriv-L7 DT work that is not yet finished.
>>
>>If we are talking about the case where the SIP proxy adds a 
>>reference (no matter what type of reference it is) then also 
>>further work is needed.
>>
>>I think we should develop an HTTP based solution since it is 
>>very simple.
>>
>>Btw, I am not sure that the name 'using protocol' is actually 
>>appropriate in this case. We haven't used the term using 
>>protocol for the Geopriv L7 work either. Terminology is hell 
>>in this area.
>>
>>Ciao
>>Hannes
>>
>>Andrew Newton wrote:
>>
>>>There have been a number of opinions regarding HTTP as a 
>>
>>using protocol.
>>
>>>I'd like to know, how many people believe that HTTP requires no or  
>>>very little specification to meet the requirements in RFC 3693 and  
>>>RFC 3694.  If you believe this, what basis do you have for this 
>>>conclusion?  Or do you believe more specification needs to 
>>
>>be done,  
>>
>>>but it is possible for HTTP to be a GEOPRIV using protocol?
>>>
>>>Finally, how many people believe that HTTP should not be a 
>>
>>using  protocol?
>>
>>>-andy
>>>
>>>_______________________________________________
>>>Geopriv mailing list
>>>Geopriv@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/geopriv
>>>
>>>
>>
>>
>>_______________________________________________
>>Geopriv mailing list
>>Geopriv@ietf.org
>>https://www1.ietf.org/mailman/listinfo/geopriv
>>
> 
> 
> 


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From sip-bounces@ietf.org Thu Jul 27 14:07:04 2006
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Subject: RE: [Sip] Hitchhikers guide - when is it done?
Date: Thu, 27 Jul 2006 21:05:14 +0300
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
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I am aware about this. On the other hand, the problem with RFCs is that
they tend to be updated at a slow pace. How often will the Hitchhiker
Guide be updated? - every two, three years? I am looking at a
semi-official solution to be used in-between, maintained at a known well
place by a registrar nominated by the SIP WG or RAI Area.=20

 Dan

=20

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Wednesday, July 26, 2006 6:33 PM
> To: Romascanu, Dan (Dan)
> Cc: IETF SIP List
> Subject: Re: [Sip] Hitchhikers guide - when is it done?
>=20
> The problem with any kind of website reference is that they=20
> are never 'official'. There are countless web sites=20
> enumerating SIP related documents. What makes something an=20
> official output of the IETF is that its an RFC.
>=20
> -Jonathan R.
>=20
> Romascanu, Dan (Dan) wrote:
>=20
> > Because of the overhead and clerical delays involved in the=20
> > publication of RFCs I would suggest a refinement to option=20
> 2. Provide=20
> > a reference in the RFC to a Web page maintained and=20
> refreshed by the=20
> > SIP WG or the RAI Area including the 'to-the-best-of-our-knowledge'=20
> > updates in-between two versions of the RFC.
> >=20
> > Dan
> >=20
> >=20
> >=20
> >=20
> > =20
> > =20
> >=20
> >=20
> >>-----Original Message-----
> >>From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> >>Sent: Tuesday, July 25, 2006 8:56 PM
> >>To: IETF SIP List
> >>Subject: [Sip] Hitchhikers guide - when is it done?
> >>
> >>During the IETF meeting, I discussed the one current open=20
> issue in the=20
> >>Hitchhikers guide. The issue is: when does the document=20
> finish? When=20
> >>do we send to IESG to issue an RFC?
> >>
> >>There were two approaches:
> >>
> >>1. split the document in two. One part contains only the=20
> core specs,=20
> >>and issues relatively soon. The other part has everything else, and=20
> >>basically ends up being the last RFC produced by the SIP WG=20
> before it=20
> >>shuts its doors (which has always been 1.5 years from now=20
> for the last=20
> >>5 years).
> >>
> >>2. keep it as one document. Issue an RFC at some point, and then=20
> >>revise it a few times as needed. We need to decide when to issue it.
> >>
> >>
> >>I'd like to propose option 2. I think people get more value=20
> >>from it as a=20
> >>consolidated document. Furthermore, I'd propose that we send the=20
> >>document to IESG for consideration as informational at the=20
> >>instant all=20
> >>of the documents listed as 'core' have RFC numbers issued by RFC-ed.
> >>
> >>Comments?
> >>
> >>-Jonathan R.
> >>--=20
> >>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> >>Cisco Fellow                                   Parsippany, NJ=20
> >>07054-2711
> >>Cisco Systems
> >>jdrosen@cisco.com                              FAX:   (973) 952-5050
> >>http://www.jdrosen.net                         PHONE: (973) 952-5000
> >>http://www.cisco.com
> >>
> >>_______________________________________________
> >>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>This list is for NEW development of the core SIP Protocol
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sipping@ietf.org for new developments on the application of sip
> >>
> >=20
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20

_______________________________________________
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At 08:32 AM 7/27/2006 -0400, Andrew Newton wrote:

>On Jul 27, 2006, at 12:08 AM, James M. Polk wrote:
>
>>At 11:37 PM 7/26/2006 -0400, Andrew Newton wrote:
>>
>>>On Jul 26, 2006, at 11:27 PM, James M. Polk wrote:
>>>
>>>>I think Geopriv really needs to look at HTTP as a RFC 3693 "using
>>>>protocol" to tease out any unique characteristics with that
>>>>protocol.  That discussion and review shouldn't bog this document
>>>>down.  Does this sound fair?
>>>
>>>Most definitely.  We've had discussions on HTTP as a using protocol
>>>for 3 years in GEOPRIV, HELD being the latest incarnation.  The first
>>>proposals ran into serious privacy concerns.  Anything that simply
>>>says "use HTTP" is an attempt to work around answering the serious
>>>privacy concerns raised in GEOPRIV.
>>
>>is this said with your WG chair hat on?
>>
>>It sounds pretty definitive to these ears...
>
>I don't believe working group chairs are given any special powers
>that such a decision is in their hands.

fair - but Brian asked for WG chair guidance because several points were 
churning over and over again (perhaps at Layer 8 - Religious), and I was 
asking if this view could be taken as WG chair guidance?

>Any individual participant
>can approach an AD or the IESG either informally or via appeal about
>attempts to work around working group consensus.  Such things are
>taken seriously by the IESG, to the point that they can stop the
>publication of an RFC sent directly to the RFC Editor on this basis.
>
>-andy

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At 09:27 AM 7/27/2006 -0400, Brian Rosen wrote:
>I think HTTP requires some specification.  A raw GET based retrieval of a
>PIDF would be a fairly short RFC, but I think we need that RFC.

I agree with the above now, based on the comments that HTTP has privacy 
concerns raised, and not addressed, in Geopriv.


>While I don't think it's necessary (SIP SUBSCRIBE is sufficient), I don't
>object to defining HTTP as a using protocol in a suitable RFC.
>
>Brian
>
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Thursday, July 27, 2006 8:45 AM
> > To: IETF SIP List; GEOPRIV
> > Subject: [Sip] teasing apart: http as a GEOPRIV using protocol
> >
> > There have been a number of opinions regarding HTTP as a using protocol.
> >
> > I'd like to know, how many people believe that HTTP requires no or
> > very little specification to meet the requirements in RFC 3693 and
> > RFC 3694.  If you believe this, what basis do you have for this
> > conclusion?  Or do you believe more specification needs to be done,
> > but it is possible for HTTP to be a GEOPRIV using protocol?
> >
> > Finally, how many people believe that HTTP should not be a using
> > protocol?
> >
> > -andy
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv

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Romascanu, Dan (Dan) wrote:
> I am aware about this. On the other hand, the problem with RFCs is that
> they tend to be updated at a slow pace. How often will the Hitchhiker
> Guide be updated? - every two, three years?

That is one reason why I was partial towards (1) from Jonathan's
original email.  If we could get the core specification guide
out now as an RFC, we could tackle the extensions as an I-D,
which is much easier to rev than a RFC would be.  Plus, the I-D
would be a WG agenda item, so that gives it some semblance
of permanence (as opposed to an individual I-D).

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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From sip-bounces@ietf.org Thu Jul 27 14:17:15 2006
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A using protocol carries location between a server and a watcher.  That is
PRECISELY what a dereference is.  Dereferencing protocols MUST be using
protocols.

However, I think that if the reference is a presence URI (sip/sips/pres)
then these issues are addressed.  If there is to be some other protocol used
to dereference, then it has to be a using protocol, and these issues must be
addressed.  If the protocol always returns a PIDF, many of these issues can
be answered by using the PIDF mechanisms to express user rules.

Now, for Presence (asserted by a proxy):
> 
> * We assume that it gets the location information somehow. Not further
> specified and might depend on the deployment environment.
Yeah, with the identifier mess

> 
> * What is the "entity" attribute of the 'presence' element?
The user, just as it would be if the endpoint asserted it, possibly
anonymized as we discussed.  I think the latter is more appropriate when
used in a Location header.  
> 
> * Where does the content for the usage-rules (e.g,
> 'retransmission-allowed', 'retention-expires', 'ruleset-reference',
> 'note-well') come from?
>From the user, just as it would be if the user asserted it.  It's just a
different mechanism for sending the same data.  Probably actually has the
same user interface, it's just the difference between whether the prototype
PIDF that the location is laid into gets sent to the phone or the proxy.
The UI to do this is a probably a web browser.
> 
> >


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At 10:25 AM 7/27/2006 -0400, Andrew Newton wrote:
>There have been many conversations regarding constraining the URI in the 
>Location header.  There have been many positions taken on this subject, 
>and this email is intended to understand the various opinions.
>
>Is there anybody who believes that the type of URI in the Location header 
>should not be constrained in any way?  In other words, any URI scheme may 
>be valid and there should be no attempt to specify the URI scheme or 
>associated protocol for dereferencing location.  And if so, do you believe 
>location will only be dereferenced via a user clicking on a link or by 
>automatic software action?  If the latter, how do you see software 
>adapting to multiple URI schemes and their associated protocols?
>
>Is there anybody who believes that the dereference URI should be 
>restricted to a specific set of schemes, enforced via ABNF, XML Schema, or 
>other grammar specification and re-enforced with MUST/SHOULD 
>language?  The intent here being that the set would be definitive or at 
>the very least require major effort to update.

Andy - I vote for the scheme be defined within the using protocol's 
(general) area.

For example, if a SIP document defines SIP and a Geopriv using protocol, 
then it should define one or more schemas near SIP (SIP, SIMPLE, 
MMUSIC).  This does result in sip/sips/pres/cid being in the SIP using 
protocol doc.

If there were an HTTP document promoting HTTP being a Geopriv using 
protocol, that document should be the one defining HTTP area schemas, 
meaning http/https.

and so on for other protocols


>Finally, how many people desire a middle ground?  Perhaps where the ABNF, 
>XML Schema or other grammar specification allows for any URI scheme but 
>there is an IANA registry listing the allowed dereference URI schemes and 
>associated protocols?

I think any IANA Registry here will point back to a Standards Track RFC 
that defined the usage and semantics, which your question is not 
allowing.  Therefore, for what you have immediately above, I disagree.


>-andy
>
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv

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At 04:33 PM 7/27/2006 +0200, Hannes Tschofenig wrote:
>Hi Brian,
>
>are you talking about the end host adding a location-by-reference into the 
>SIP header or a proxy doing so?
>
>In any case I don't understand your objection against the usage of HTTP 
>and your preference for SIP/SIPS/PREZ?

Hannes - I think Brian has a problem with a SIP doc defining how HTTP 
works.  He has stated an HTTP doc should do this.  We think this HTTP doc 
should be very, very short.  This SIP doc is allowing an extension to the 
URI schemes without going though the SIP WG as a SIP WG item (which is 
normally required - and takes a lot of time, because it must first start in 
SIPPING).


>Ciao
>Hannes
>
>Brian Rosen wrote:
>>Nadine
>>Are you saying that it's your opinion sip-location-conveyance MUST
>>define an HTTP dereference procedure?
>>If you do, why?  What is the use-case or requirement?
>>Would it not be pretty reasonable to require a separate document that
>>  defines a location-by-reference mechanism, protocol, and geopriv
>>"using protocol" analysis?  If that document specifically extended
>>what was allowed in the location header, would that not be
>>reasonable?
>>I think we have to include "A" dereference mechanism in the 
>>location-conveyance document.  It seems just including sip/sips/prez
>>is a reasonable, all-sip-all-the-time approach IN THIS DOCUMENT.
>>That certainly does not preclude another document describing another
>>way.
>>Doesn't it seem just a little weird to have a sip-location-conveyance
>>  document describing any kind of http based location conveyance 
>> (de-reference) protocol, especially one created whole cloth without
>>any other parts?
>>Doesn't it seem in keeping with the geopriv approach that in order to
>>have another "using protocol" it has to have a geopriv review?  That
>>is what leads to restricting the URI schemes.  Would you prefer to
>>let it be wide open, so any protocol anyone thinks about can be used
>>with the location header?  I can't see geopriv going along with that.
>>We're going to make it a one line ABNF to add a new scheme to the
>>list of allowed schemes.
>>It seems to me that if it's desired to define some bare-bones HTTP
>>location by reference protocol, then that should be a separate
>>document, which can easily extend the allowable schemes in the
>>Location header.  I'm not sure it might be better to define ALL of
>>that in the L7 protocol, but I wouldn't be opposed to a simple "raw
>>GET" RFC as long as it got a full geopriv review. I just don't see
>>where it makes any sense to force that into a sip document.
>>Brian
>>
>>>-----Original Message----- From: Abbott, Nadine B
>>>[mailto:nabbott@telcordia.com] Sent: Wednesday, July 26, 2006 2:14
>>>PM To: sip@ietf.org Cc: geopriv@ietf.org Subject: RE: [Geopriv]
>>>Consensus on changes to location-conveyance
>>>
>>>I don't agree that there is a consensus to preclude the use of
>>>HTTP. I would favor not placing your proposed restriction on the
>>>URI scheme for expressing a location reference in the SIP location
>>>conveyance draft; or at least including HTTP as a valid
>>>location-reference URI type.
>>>Regards, Nadine Abbott
>>>-----Original Message----- From: Rosen, Brian
>>>[mailto:Brian.Rosen@neustar.biz] Sent: Tuesday, July 25, 2006 1:38
>>>PM To: sip@ietf.org Cc: geopriv@ietf.org Subject: [Geopriv]
>>>Consensus on changes to location-conveyance
>>>I have put out a number of emails on proposed changes to location 
>>>conveyance.  Some of them have sparked a number of comments.  My
>>>summary of consensus on these issues is:
>>>1. We proposed to explicitly disallow use of a Data URI, which
>>>would allow specification of a Location-by-value in a header.
>>>There was a lot of list traffic on this, currently in the state
>>>where Keith asked for a real use case for it.  So far, we don't see
>>>a use case that cannot be provided by location-by-reference in a
>>>header.  If we don't get a good use case, I think consensus is to
>>>disallow location-by-value in a header.
>>>2. Dereferencing location-by-reference.  We had initially proposed
>>>to document a process to use a raw HTTP(S) Get to dereference.
>>>There was a lot of discussion, and then I made a new proposal, see
>>>item 6.
>>>3. We proposed to allow a Reason header with a 424 Bad Location 
>>>response.  This seems to be acceptable to the WG.  Geopriv will
>>>create and populate the registry in a separate draft.
>>>4. We proposed to explicitly allow hop-by-hop TLS security for
>>>location when the endpoint wished to allow routing based on
>>>location.  We further proposed to interpret "Do not Distribute"
>>>flags as allowing sending of the location to the routing database
>>>(LoST for example).  Hannes pointed out that there was a proposal
>>>to allow two Location headers, one of which could be used for
>>>routing and another for onward forwarding to the recipient.
>>>Hannes' message seemed to agree that "Do not distribute" would
>>>allow sending of the location to the routing-by-location database, but 
>>>would obligate the proxy that did that to remove the location
>>>before it forwarded the message.  There were no other messages on
>>>this thread. We conclude that allowing hop-by-hop security for
>>>location based routing is acceptable, and Do Not Distribute DOES
>>>allow forwarding to a routing database, but requires that the proxy
>>>doing the location based route to remove the location it uses for
>>>that purpose.
>>>5. We proposed to create a parameter and a registry to distinguish 
>>>multiple ways to dereference a location where HTTP(S) was the
>>>transport. Resolution of this issue depends on resolution of issue
>>>6.
>>>6. We proposed to allow sip/sips and pres: in the location header.
>>>I later speculated that if we allow this, it forms a simple way to
>>>do location-by-reference, and we could eliminate the raw HTTP(S)
>>>Get proposal, with the parameter and registry entirely.  There was
>>>some list traffic in favor of this.  Martin Thompson is skeptical
>>>that SIP Presence Subscribe/Notify is actually allowed as a geopriv
>>>"using protocol".  Several of think that it is.  If we don't get
>>>any more comments on this proposal, we will allow sip/sips and
>>>pres, not have any http(s) resolution text, and not propose a
>>>parameter and accompanying registry.
>>>If this does not reflect your position, please speak up now.
>>>Brian
>>>_______________________________________________ Geopriv mailing
>>>list Geopriv@ietf.org https://www1.ietf.org/mailman/listinfo/geopriv
>>>_______________________________________________ Geopriv mailing
>>>list Geopriv@ietf.org https://www1.ietf.org/mailman/listinfo/geopriv
>>
>>_______________________________________________ Sip mailing list
>>https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
>>development of the core SIP Protocol Use
>>sip-implementors@cs.columbia.edu for questions on current sip Use
>>sipping@ietf.org for new developments on the application of sip
>
>
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Subject: [Sip] Re: [Geopriv] Teasing apart: constraints on dereference URI
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James M. Polk wrote:
>> Finally, how many people desire a middle ground?  Perhaps where the 
>> ABNF, XML Schema or other grammar specification allows for any URI 
>> scheme but there is an IANA registry listing the allowed dereference 
>> URI schemes and associated protocols?
> 
> I think any IANA Registry here will point back to a Standards Track RFC 
> that defined the usage and semantics, which your question is not 
> allowing.  Therefore, for what you have immediately above, I disagree.

I guess that may have been too much hand waving on my part.  So let me 
ask you this question:  if the grammar were to only specify "URI" and no 
particular kind of URI, yet the kinds of URIs allowed must have to pass 
some sort of review before their use in the Location header is 
considered acceptable, would you find favor in this approach?

And if so, how would you implement it?

-andy


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At 09:44 AM 7/27/2006 -0500, Marc Berryman wrote:
>Sounds like Hannes and Henning are moving down the right track. Focusing
>on the location header syntax for the draft.

Mark

Defining syntax alone doesn't satisfy the SIP requirements for any SIP 
extension within RFC 4485, which require per element behaviors be described 
too (section 4.7 I believe), which are not exactly the same per SIP request 
type, or per call intent type (e2e vs. proxy-routing) which gets use back 
to a better version of the -03 ID we have, that the authors have committed 
to submitting in the first half of August.


>Marc B
>
>-----Original Message-----
>From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
>Sent: Thursday, July 27, 2006 9:35 AM
>To: Dawson, Martin
>Cc: IETF SIP List; geopriv@ietf.org; Henning Schulzrinne
>Subject: [Geopriv] Henning's Suggestion
>
>Hi Henning,
>
>I like your suggestion.
>
>In fact I would even go a step further. The SIP Location Conveyance
>document just defines a mechanism to carry a PIDF-LO in the body of the
>SIP message. Done.
>Adding location-by-reference to the SIP header by the end host sounds OK
>
>but there are still some loose ends where this reference comes from.
>
>The other stuff can be dealt with once we concluded our Geopriv-L7 work
>where all the architectural pieces should are created. The
>subscription-URI is an example where we recently proposed something and
>haven't finished our technical discussions.
>
>I think the draft got fuzzy when it came to the area where a SIP proxy
>adds a location-by-reference. I must admit that I proposed to add
>location-by-value into the SIP header as well but in the meanwhile I
>don't think that any of the SIP proxy stories belong in there.
>
>Ciao
>Hannes
>
> >>-----Original Message-----
> >>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> >>Sent: Thursday, 27 July 2006 1:25 PM
> >>To: Brian Rosen
> >>Cc: IETF SIP List; geopriv@ietf.org
> >>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> >>conveyance
> >>
> >>My suggestion would be
> >>
> >>- to focus the draft on just defining the Location header syntax,
> >>without ruling out any particular mechanism or scheme; this would
> >>make the draft really short (which I would find helpful).
> >>
> >>- and then add one section to the draft that prototypically defines
> >>what a particular scheme that wants to appear in the Location header
> >>needs to define, using the non-controversial cid mechanism as the
> >>example. Other, separate, drafts can then define other modes,
> >>following the same template. Documents such as the phone-bcp can then
> >>pick some suitable must-implement subset and make emergency-call-
> >>specific recommendations, rather than sprinkling these throughout the
> >>conveyance document and repeating some in the phone-bcp. Most of
> >>these are, in any event, operational recommendations, not for
> >>ensuring protocol-level interoperability.
> >>
> >>That way, the conveyance draft can be extremely short (even shorter
> >>if the requirements are omitted...) and can get done, and we avoid
> >>engineering operational requirements on the fly, commingling
> >>emergency and non-emergency cases, as seems to be happening now.
> >>
> >>I think protocol documents should define what's needed for
> >>interoperability; BCPs can make more specific, situation-specific
> >>recommendations. Commingling the two seems to be a recipe for
> >>deadlock or arbitrary, non-technically-grounded decisions that are
> >>more lawyer-like than engineering-like ("you did not mention
> >>technique X in some meeting N years ago and thus you can't bring it
> >>up now!")
> >>
> >>Henning
> >>
> >>On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
> >>
> >>
> >>>At this point, I think we need some chair guidance and by that I
> >>>think it's
> >>>a combo of geopriv and sip chairs.
> >>>
> >>>We seem to have a couple of significant open questions.  I think
> >>>most of the
> >>>other stuff (like the "allow sending to router") will converge.
> >>>
> >>>There seems to be some opposition to allowing sip/sips/pres: in a
> >>>Location
> >>>Header.  The opposition seems to be that despite 4119, somehow, SIP
> >>>(presence subscription) is not a "using protocol".  I "think" a
> >>>ruling from
> >>>the bench on that would work.
> >>>
> >>>There is also opposition to defining the allowed schemes as a
> >>>specific list,
> >>>with a defined (easy) extension mechanism.
> >>>
> >>>Related to that, there are some who want location-conveyance to
> >>>define an
> >>>HTTP based location dereference mechanism, and on the opposite end,
> >>>we have
> >>>people saying we haven't even agreed to support any form of
> >>>location-by-reference.
> >>>
> >>>The authors would like to know what to do now.
> >>>
> >>>Brian
> >>>
> >>>
> >>>>-----Original Message-----
> >>>>From: Andrew Newton [mailto:andy@hxr.us]
> >>>>Sent: Wednesday, July 26, 2006 9:58 PM
> >>>>To: Winterbottom, James
> >>>>Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning
> >>>>Schulzrinne
> >>>>Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> >>>>conveyance
> >>>>
> >>>>
> >>>>On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
> >>>>
> >>>>
> >>>>>Andy,
> >>>>>
> >>>>>RFC-4119 says "However, the usage of this location object format
> >
> > is
> >
> >>>>>not
> >>>>>   limited to presence-using protocols-- any protocol that can
> >>>>>carry XML
> >>>>>   or MIME types can carry PIDF."
> >>>>>
> >>>>>It seems to me that restricting the allowed schema in what is
> >>>>>supposed
> >>>>>to be a location conveyance specification is in opposition to this
> >>>>>statement.
> >>>>
> >>>>Did somebody say that HTTP was a presence-using protocol?
> >>>>
> >>>>-andy
> >>>>
> >>>>ps.  you can find my location with the following reference URIs:
> >>>>
> >>>>gopher://fatchance.hxr.us/you-still-have-a-gopher-client
> >>>>news:alt.locations.people.secretly-being-watched
> >>>>telnet://screen-scrapers.are.us/
> >>>>dns:cached-publicly-around-the-world.com?type=HOST
> >>>>
> >>>>
> >>>>
> >>>>_______________________________________________
> >>>>Geopriv mailing list
> >>>>Geopriv@ietf.org
> >>>>https://www1.ietf.org/mailman/listinfo/geopriv
> >>>
> >>>
> >>>_______________________________________________
> >>>Geopriv mailing list
> >>>Geopriv@ietf.org
> >>>https://www1.ietf.org/mailman/listinfo/geopriv
> >>
> >>
> >>_______________________________________________
> >>Geopriv mailing list
> >>Geopriv@ietf.org
> >>https://www1.ietf.org/mailman/listinfo/geopriv
> >
> >
> >
>------------------------------------------------------------------------
>------------------------
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> >
>------------------------------------------------------------------------
>------------------------
> > [mf2]
> >
> > _______________________________________________
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> >
> >
>
>
>_______________________________________________
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>
>
>
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At 11:33 AM 7/27/2006 -0400, Henning Schulzrinne wrote:
>>ii)     if we have more than one option, we either need some means of
>>determining that the dereferencer supports the one the client chooses to
>>use, or require support of all options. As we seem to be lacking any
>>mechanism for the former, then it would imply that all options have to
>>be supported at the dereferencer, and therefore that the list needs to
>>be a small list.
>
>This isn't quite true. It is quite possible to include multiple options, 
>as long as one of them is universally supported. For example, you can 
>support a "universal & slow" and a "sometimes & fast" solution, and 
>everybody wins.

Are you saying that the doc now has to state that more than one schema has 
to be supported within the same Location header?


>This approach is not new: MIME email does exactly that, where messages 
>with HTML or rich text often also contain an alternate plain text rendering.
>
>Henning
>
>_______________________________________________
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>Geopriv@ietf.org
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Subject: RE: [Sip] Hitchhikers guide - when is it done?
Date: Thu, 27 Jul 2006 21:12:12 +0200
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Thread-Topic: [Sip] Hitchhikers guide - when is it done?
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From: "Drage, Keith \(Keith\)" <drage@lucent.com>
To: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>,
	"Jonathan Rosenberg" <jdrosen@cisco.com>
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Lets indicate an expectation of once every 12 months. Within that period
there ought to be at least 10/20 references to either add or change.

Regards

Keith=20

> -----Original Message-----
> From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]=20
> Sent: 27 July 2006 19:05
> To: Jonathan Rosenberg
> Cc: IETF SIP List
> Subject: RE: [Sip] Hitchhikers guide - when is it done?
>=20
> I am aware about this. On the other hand, the problem with=20
> RFCs is that they tend to be updated at a slow pace. How=20
> often will the Hitchhiker Guide be updated? - every two,=20
> three years? I am looking at a semi-official solution to be=20
> used in-between, maintained at a known well place by a=20
> registrar nominated by the SIP WG or RAI Area.=20
>=20
>  Dan
>=20
> =20
>=20
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> > Sent: Wednesday, July 26, 2006 6:33 PM
> > To: Romascanu, Dan (Dan)
> > Cc: IETF SIP List
> > Subject: Re: [Sip] Hitchhikers guide - when is it done?
> >=20
> > The problem with any kind of website reference is that they=20
> are never=20
> > 'official'. There are countless web sites enumerating SIP related=20
> > documents. What makes something an official output of the=20
> IETF is that=20
> > its an RFC.
> >=20
> > -Jonathan R.
> >=20
> > Romascanu, Dan (Dan) wrote:
> >=20
> > > Because of the overhead and clerical delays involved in the=20
> > > publication of RFCs I would suggest a refinement to option
> > 2. Provide
> > > a reference in the RFC to a Web page maintained and
> > refreshed by the
> > > SIP WG or the RAI Area including the=20
> 'to-the-best-of-our-knowledge'=20
> > > updates in-between two versions of the RFC.
> > >=20
> > > Dan
> > >=20
> > >=20
> > >=20
> > >=20
> > > =20
> > > =20
> > >=20
> > >=20
> > >>-----Original Message-----
> > >>From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> > >>Sent: Tuesday, July 25, 2006 8:56 PM
> > >>To: IETF SIP List
> > >>Subject: [Sip] Hitchhikers guide - when is it done?
> > >>
> > >>During the IETF meeting, I discussed the one current open
> > issue in the
> > >>Hitchhikers guide. The issue is: when does the document
> > finish? When
> > >>do we send to IESG to issue an RFC?
> > >>
> > >>There were two approaches:
> > >>
> > >>1. split the document in two. One part contains only the
> > core specs,
> > >>and issues relatively soon. The other part has everything=20
> else, and=20
> > >>basically ends up being the last RFC produced by the SIP WG
> > before it
> > >>shuts its doors (which has always been 1.5 years from now
> > for the last
> > >>5 years).
> > >>
> > >>2. keep it as one document. Issue an RFC at some point, and then=20
> > >>revise it a few times as needed. We need to decide when=20
> to issue it.
> > >>
> > >>
> > >>I'd like to propose option 2. I think people get more=20
> value from it=20
> > >>as a consolidated document. Furthermore, I'd propose that we send=20
> > >>the document to IESG for consideration as informational at the=20
> > >>instant all of the documents listed as 'core' have RFC numbers=20
> > >>issued by RFC-ed.
> > >>
> > >>Comments?
> > >>
> > >>-Jonathan R.
> > >>--=20
> > >>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > >>Cisco Fellow                                   Parsippany, NJ=20
> > >>07054-2711
> > >>Cisco Systems
> > >>jdrosen@cisco.com                              FAX:  =20
> (973) 952-5050
> > >>http://www.jdrosen.net                         PHONE:=20
> (973) 952-5000
> > >>http://www.cisco.com
> > >>
> > >>_______________________________________________
> > >>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >>This list is for NEW development of the core SIP Protocol Use=20
> > >>sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > >>sipping@ietf.org for new developments on the application of sip
> > >>
> > >=20
> > >=20
> >=20
> > --=20
> > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > Cisco Fellow                                   Parsippany, NJ=20
> > 07054-2711
> > Cisco Systems
> > jdrosen@cisco.com                              FAX:   (973) 952-5050
> > http://www.jdrosen.net                         PHONE: (973) 952-5000
> > http://www.cisco.com
> >=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From sip-bounces@ietf.org Thu Jul 27 15:23:42 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: [Sip] Re: RFC4320 and 408 to non-INVITE
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> This is not the way the timers are meant to be changed.
>
> Both of the transaction-stateful things on either end of a given 
> transaction must share the same
> values for the transaction timers or the results are not defined.
>
> What you describe is not unlike throwing away the timing belt in an 
> engine and spinning
> the head at some rate unrelated to the crankshaft. You might get  lucky, 
> but usually very bad things
> are going to happen.

That's interesting, because it is exactly what is prescribed for IMS

Refer to TS 24.229 release 7 table 7.8 "SIP Timers" 
(http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.zip, page 
132 for me)

It says:
Value to be applied at the UE: T1 = 2s default
Value to be applied between IM CN subsystem elements: T1=500ms default

So you are saying that in IMS configured this way, very bad things will 
happen?

Regards,

jeroen

>
> RjS
>
> On Jul 27, 2006, at 12:29 PM, Jeroen Van Bemmel wrote:
>
>> Robert,
>>
>> RFC4320 bans 408 responses to NIT, but I think there are situations  in 
>> which it is still useful and needed. In particular, consider  when the 
>> UAC is using T1=2000 and the proxy uses T1=500 (eg because  the client is 
>> on a slow wireless link)
>>
>> The CT at the proxy times out after 64*T1=32 sec, while the client  is 
>> still retransmitting. In this case, 408 does help
>>
>> Regards,
>> Jeroen
> 


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From sip-bounces@ietf.org Thu Jul 27 15:35:59 2006
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Hi Henning,

I would very much appreciate to treat this topic in the way you proposed 
it below.

A few minor comments:

Henning Schulzrinne wrote:
> With hindsight, I think the term 'using protocol' has been the source of 
>  more word-lawyering than engineering insight. I personally find the 
> distinctions
> 
>   - first person vs. third person (i.e., find out location about myself 
> vs. find out location about some third party)
> 
>   - location vs. location + identity
> 
>   - token-based authorization vs. identity-based authorization
> 
> more helpful.
> 
> Token-based authorization hands the recipient a key that allows him to 
> get a location object, without caring as to who is retrieving the 
> object. ("Anybody seeing this SIP request can convert the token into an 
> LO.") Identity-based naturally uses some authenticated identity to 
> release that information.


In past common policy discussions we also differentiated between 
limited use addresses and trait-based authorization. A token would match 
the latter.

In some sense we seem to be using limited use addresses in most of the 
cases when we talk about the location-by-references. The difference 
between limited use addresses and trait-based authorization is small in 
this particular usage environment.


  Strictly speaking, this is only a question of
> a level of indirection, since proving identity usually involves proving 
> possession of a secret.

I guess there are some further differences with regard to the lifetime 
of the different identifiers, the ability to revoke rights, the number 
of entities that one can hide behind these identifiers and the storage 
implications.

> We've been trying to get at some of these distinctions with the sighting 
> vs. using protocol terminology, but that distinction also seems to 
> degenerate into discussions more suited for a law school or theology 
> department than an engineering organization.
> 
> For example, by strict definition, LoST would be a 'Using Protocol' 
> since it "carries a Location Object", but I hope we all agree that this 
> is silly. We had earlier agreed that the location-by-DHCP also were not 
> Using Protocol, even though they obviously carry a location object of 
> some form.

Currently LoST is not a using protocol since it does not carry a PIDF-LO 
although some folks would like to use a PIDF-LO; HELD was a using 
protocol before we turned the problem into a Location Configuration 
problem; RELO never was a using protocol;

Isn't this confusing?

> 
> Privacy issues apply iff
>   - third person
>   - location + identity (not just location)
> 
> The L7 discussion was about 'first person' and 'location only' (although 
> the latter is subject to debate, depending on whether one considers an 
> IP address of identifying an individual or not).

That was before we added the Location-by-Reference and subscription-URI 
concept ....

> 
> Depending on the circumstances, SIP-based retrieval can either involve 
> privacy concerns (subscribe to alice@example.com) or not (subscribe to 
> location of one-time random unlinkable identifier 
> jdfa8f9348@example.com). There can also be different privacy concerns 
> depending on the perspective. For example, if a SIP request contains 
> both an identity and the location retrieval reference, this raises 
> privacy concerns for the proxy, but the LIS may not have any such 
> concerns since it may only be the repository of these randomly-named 
> objects, without tie to an identity.

Ciao
Hannes

> Henning
> 
> 
> 
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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At 01:27 PM 7/27/2006 -0500, James M. Polk wrote:
>At 10:25 AM 7/27/2006 -0400, Andrew Newton wrote:
>>There have been many conversations regarding constraining the URI in the 
>>Location header.  There have been many positions taken on this subject, 
>>and this email is intended to understand the various opinions.
>>
>>Is there anybody who believes that the type of URI in the Location header 
>>should not be constrained in any way?  In other words, any URI scheme may 
>>be valid and there should be no attempt to specify the URI scheme or 
>>associated protocol for dereferencing location.  And if so, do you 
>>believe location will only be dereferenced via a user clicking on a link 
>>or by automatic software action?  If the latter, how do you see software 
>>adapting to multiple URI schemes and their associated protocols?
>>
>>Is there anybody who believes that the dereference URI should be 
>>restricted to a specific set of schemes, enforced via ABNF, XML Schema, 
>>or other grammar specification and re-enforced with MUST/SHOULD 
>>language?  The intent here being that the set would be definitive or at 
>>the very least require major effort to update.
>
>Andy - I vote for the scheme be defined within the using protocol's 
>(general) area.
>
>For example, if a SIP document defines SIP and

I meant "as" here instead of "and"

>  a Geopriv using protocol, then it should define one or more schemas near 
> SIP (SIP, SIMPLE, MMUSIC).  This does result in sip/sips/pres/cid being 
> in the SIP using protocol doc.
>
>If there were an HTTP document promoting HTTP being a Geopriv using 
>protocol, that document should be the one defining HTTP area schemas, 
>meaning http/https.
>
>and so on for other protocols
>
>
>>Finally, how many people desire a middle ground?  Perhaps where the ABNF, 
>>XML Schema or other grammar specification allows for any URI scheme but 
>>there is an IANA registry listing the allowed dereference URI schemes and 
>>associated protocols?
>
>I think any IANA Registry here will point back to a Standards Track RFC 
>that defined the usage and semantics, which your question is not 
>allowing.  Therefore, for what you have immediately above, I disagree.
>
>
>>-andy
>>
>>
>>_______________________________________________
>>Geopriv mailing list
>>Geopriv@ietf.org
>>https://www1.ietf.org/mailman/listinfo/geopriv
>
>_______________________________________________
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>https://www1.ietf.org/mailman/listinfo/geopriv

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> In past common policy discussions we also differentiated between limited 
> use addresses and trait-based authorization. A token would match the 
> latter.

I'm not so sure. I see this as another level of indirection:

secret -> access

and

secret -> identity ("alice") -> access

and

secret -> trait ("contractor") -> access


> 
> I guess there are some further differences with regard to the lifetime 
> of the different identifiers, the ability to revoke rights, the number 
> of entities that one can hide behind these identifiers and the storage 
> implications.

Right - the more layers of indirections, the more means to revoke. 
However, I don't see a fundamental difference between the three 
mechanisms in this case. In each case, you can revoke access.



> Isn't this confusing?

Unless you spent three years in Advanced GEOPRIV Language & Philosophy 
class, yes.


>>
>> The L7 discussion was about 'first person' and 'location only' 
>> (although the latter is subject to debate, depending on whether one 
>> considers an IP address of identifying an individual or not).
> 
> That was before we added the Location-by-Reference and subscription-URI 
> concept ....

As long as the recipient of the reference and the URI is the target, it 
doesn't much matter. We get into trouble if we use the same mechanism 
for first-person and third-person retrieval.

Henning

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So, specifically, that table says:
SIP Timer
Value to be applied between IM CN subsystem elements
Value to be applied at the UE
Value to be applied at the P-CSCF toward a UE
Meaning
T1
500ms default
2s default
2s default
RTT estimate


So the P-CSCF and the UE are both using 2s for T1.
When the P-CSCF talks to other things (that are not the UE), it uses  
500ms for T1.

What you were describing would be the P-CSCF using 500ms on its side  
of a transaction it was in with a
UA that was using 2s, which is not what this table says to do. In  
that case, yes, undefined things can happen.

RjS

On Jul 27, 2006, at 2:21 PM, Jeroen van Bemmel wrote:

>> This is not the way the timers are meant to be changed.
>>
>> Both of the transaction-stateful things on either end of a given  
>> transaction must share the same
>> values for the transaction timers or the results are not defined.
>>
>> What you describe is not unlike throwing away the timing belt in  
>> an engine and spinning
>> the head at some rate unrelated to the crankshaft. You might get   
>> lucky, but usually very bad things
>> are going to happen.
>
> That's interesting, because it is exactly what is prescribed for IMS
>
> Refer to TS 24.229 release 7 table 7.8 "SIP Timers" (http://www. 
> 3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.zip, page 132  
> for me)
>
> It says:
> Value to be applied at the UE: T1 = 2s default
> Value to be applied between IM CN subsystem elements: T1=500ms default
>
> So you are saying that in IMS configured this way, very bad things  
> will happen?
>
> Regards,
>
> jeroen
>
>>
>> RjS
>>
>> On Jul 27, 2006, at 12:29 PM, Jeroen Van Bemmel wrote:
>>
>>> Robert,
>>>
>>> RFC4320 bans 408 responses to NIT, but I think there are  
>>> situations  in which it is still useful and needed. In  
>>> particular, consider  when the UAC is using T1=2000 and the proxy  
>>> uses T1=500 (eg because  the client is on a slow wireless link)
>>>
>>> The CT at the proxy times out after 64*T1=32 sec, while the  
>>> client  is still retransmitting. In this case, 408 does help
>>>
>>> Regards,
>>> Jeroen
>


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; ">So, specifically, that table =
says:<DIV><TABLE border=3D"1" cellspacing=3D"0" cellpadding=3D"0" =
style=3D"margin-left:-1.0pt; =
border-collapse:collapse;border:none;mso-border-alt:solid black 1.0pt; =
mso-padding-alt:0in 0in 0in 0in"><TBODY><TR style=3D"height:14.2pt"><TD =
width=3D"44" valign=3D"top" =
style=3D"width:44.25pt;border-top:1.0pt;border-left:  =
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class=3D"TAH"><SPAN lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: collapse;">SIP Timer </SPAN><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD width=3D"99" =
valign=3D"top" style=3D"width:99.0pt;border-top:solid black 1.0pt;  =
border-left:none;border-bottom:solid black .75pt;border-right:solid =
black .75pt;  mso-border-left-alt:solid black .75pt;padding:.7pt .7pt =
0in .7pt;height:14.2pt"><DIV class=3D"TAH"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse;">Value to =
be applied between IM CN subsystem  elements </SPAN><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD valign=3D"top" =
style=3D"border-top:solid black 1.0pt;border-left:none;  =
border-bottom:solid black .75pt;border-right:solid black =
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.7pt;height:14.2pt"><DIV class=3D"TAH"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse;">Value to =
be applied at the UE</SPAN><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD valign=3D"top" =
style=3D"border-top:solid black 1.0pt;border-left:none;  =
border-bottom:solid black .75pt;border-right:solid black =
.75pt;mso-border-left-alt:  solid black .75pt;padding:0in 0in 0in =
0in;height:14.2pt"><DIV class=3D"TAH"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse;">Value to =
be applied at the P-CSCF toward a UE</SPAN><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD valign=3D"top" =
style=3D"border-top:solid black 1.0pt;border-left:none;  =
border-bottom:solid black .75pt;border-right:solid black =
1.0pt;mso-border-left-alt:  solid black .75pt;padding:.7pt .7pt 0in =
.7pt;height:14.2pt"><DIV class=3D"TAH"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;">Meaning</SPAN><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD></TR><TR><TD width=3D"44" =
valign=3D"top" style=3D"width:44.25pt;border-top:none;border-left:  =
solid black 1.0pt;border-bottom:solid black 1.0pt;border-right:solid =
black .75pt;  mso-border-top-alt:solid black .75pt;padding:.7pt .7pt 0in =
.7pt"><DIV class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;">T1</SPAN><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD width=3D"99" =
valign=3D"top" style=3D"width:99.0pt;border-top:none;border-left:none;  =
border-bottom:solid black 1.0pt;border-right:solid black =
.75pt;mso-border-top-alt:  solid black .75pt;mso-border-left-alt:solid =
black .75pt;padding:.7pt .7pt 0in .7pt"><DIV class=3D"TAL"><SPAN =
lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;">500ms default</SPAN><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD valign=3D"top" =
style=3D"border-top:none;border-left:none;border-bottom:solid black =
1.0pt;  border-right:solid black .75pt;mso-border-top-alt:solid black =
.75pt;  mso-border-left-alt:solid black .75pt;padding:.7pt .7pt 0in =
.7pt"><DIV class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse;">2s =
default</SPAN><SPAN class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD valign=3D"top" =
style=3D"border-top:none;border-left:none;border-bottom:solid black =
1.0pt;  border-right:solid black .75pt;mso-border-top-alt:solid black =
.75pt;  mso-border-left-alt:solid black .75pt;padding:0in 0in 0in =
0in"><DIV class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse;">2s =
default</SPAN><SPAN class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD><TD valign=3D"top" =
style=3D"border-top:none;border-left:none;border-bottom:solid black =
1.0pt;  border-right:solid black 1.0pt;mso-border-top-alt:solid black =
.75pt;  mso-border-left-alt:solid black .75pt;padding:.7pt .7pt 0in =
.7pt"><DIV class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse;">RTT =
estimate</SPAN><SPAN class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;"><O:P></O:P></SPAN></SPAN></DIV></TD></TR></TBODY></TABLE><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: =
collapse;"><DIV><SPAN class=3D"Apple-style-span" style=3D"border-collapse:=
 separate;"><BR class=3D"khtml-block-placeholder"></SPAN></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV>So the P-CSCF and the UE are =
both using 2s for T1.</SPAN></DIV><DIV>When the P-CSCF talks to other =
things (that are not the UE), it uses 500ms for T1.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>What you were describing =
would be the P-CSCF using 500ms on its side of a transaction it was in =
with a</DIV><DIV>UA that was using 2s, which is not what this table says =
to do. In that case, yes, undefined things can happen.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>RjS</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><DIV><DIV><DIV><DIV><DIV>On =
Jul 27, 2006, at 2:21 PM, Jeroen van Bemmel wrote:</DIV><BR =
class=3D"Apple-interchange-newline"><BLOCKQUOTE type=3D"cite"><BLOCKQUOTE =
type=3D"cite"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">This is not the way the timers =
are meant to be changed.</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Both of the transaction-stateful =
things on either end of a given transaction must share the =
same</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">values for the transaction =
timers or the results are not defined.</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">What you =
describe is not unlike throwing away the timing belt in an engine and =
spinning</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">the head at some rate unrelated =
to the crankshaft. You might get<SPAN class=3D"Apple-converted-space">=A0 =
</SPAN>lucky, but usually very bad things</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">are =
going to happen.</DIV> </BLOCKQUOTE><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">That's interesting, because it =
is exactly what is prescribed for IMS</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Refer to TS =
24.229 release 7 table 7.8 "SIP Timers" (<A =
href=3D"http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.z=
ip">http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.zip</=
A>, page 132 for me)</DIV><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; =
"><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">It says:</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Value to be applied at the UE: T1 =3D 2s =
default</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Value to be applied between IM =
CN subsystem elements: T1=3D500ms default</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">So you are =
saying that in IMS configured this way, very bad things will =
happen?</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Regards,</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">jeroen</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV> <BLOCKQUOTE =
type=3D"cite"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">RjS</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">On Jul 27, 2006, at 12:29 PM, =
Jeroen Van Bemmel wrote:</DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><BR></DIV> <BLOCKQUOTE type=3D"cite"><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">Robert,</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">RFC4320 bans 408 responses to NIT, but I think there =
are situations<SPAN class=3D"Apple-converted-space">=A0 </SPAN>in which =
it is still useful and needed. In particular, consider<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN>when the UAC is using T1=3D2000=
 and the proxy uses T1=3D500 (eg because<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN>the client is on a slow =
wireless link)</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">The CT at the proxy times out after 64*T1=3D32 sec, =
while the client<SPAN class=3D"Apple-converted-space">=A0 </SPAN>is =
still retransmitting. In this case, 408 does help</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">Regards,</DIV><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Jeroen</DIV> =
</BLOCKQUOTE></BLOCKQUOTE><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; =
"><BR></DIV> =
</BLOCKQUOTE></DIV><BR></DIV></DIV></DIV></DIV></BODY></HTML>=

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_______________________________________________
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Date: Thu, 27 Jul 2006 14:52:15 -0500
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Cc: IETF SIP List <sip@ietf.org>, GEOPRIV <geopriv@ietf.org>
Subject: [Sip] Re: [Geopriv] Teasing apart: constraints on dereference URI
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At 02:42 PM 7/27/2006 -0400, Andrew Newton wrote:
>James M. Polk wrote:
>>>Finally, how many people desire a middle ground?  Perhaps where the 
>>>ABNF, XML Schema or other grammar specification allows for any URI 
>>>scheme but there is an IANA registry listing the allowed dereference URI 
>>>schemes and associated protocols?
>>I think any IANA Registry here will point back to a Standards Track RFC 
>>that defined the usage and semantics, which your question is not 
>>allowing.  Therefore, for what you have immediately above, I disagree.
>
>I guess that may have been too much hand waving on my part.  So let me ask 
>you this question:  if the grammar were to only specify "URI" and no 
>particular kind of URI, yet the kinds of URIs allowed must have to pass 
>some sort of review before their use in the Location header is considered 
>acceptable, would you find favor in this approach?

This is generally what the authors here are proposing, with Geopriv being 
the reviewers.  In SIP Location Conveyance, since it is a SIP doc (and SIP 
is the transport protocol of the Location header where the URIs will go), 
this doc will define SIP-based schemas (sip/sips/pres/cid).  Extensions to 
this RFC (if it ever gets there) can ADD schemas, once those specifications 
pass Geopriv consensus.


>And if so, how would you implement it?
>
>-andy

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From sip-bounces@ietf.org Thu Jul 27 15:56:51 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Robert Sparks" <rjsparks@estacado.net>
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Subject: [Sip] Re: RFC4320 and 408 to non-INVITE
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inline
  ----- Original Message -----=20
  From: Robert Sparks=20
  To: Jeroen van Bemmel=20
  Cc: sip@ietf.org=20
  Sent: Thursday, July 27, 2006 9:48 PM
  Subject: Re: RFC4320 and 408 to non-INVITE


  So, specifically, that table says:
        SIP Timer  Value to be applied between IM CN subsystem elements  =
Value to be applied at the UE Value to be applied at the P-CSCF toward a =
UE Meaning=20
        T1 500ms default 2s default 2s default RTT estimate=20





  So the P-CSCF and the UE are both using 2s for T1.
  When the P-CSCF talks to other things (that are not the UE), it uses =
500ms for T1.


  What you were describing would be the P-CSCF using 500ms on its side =
of a transaction it was in with a
  UA that was using 2s, which is not what this table says to do. In that =
case, yes, undefined things can happen.

Then perhaps I was not clear enough. What I meant was the UA using =
T1=3D2000 towards the P-CSCF, and the P-CSCF using T1=3D500 towards the =
S-CSCF. If the latter transaction times out (after 64*500=3D32 seconds), =
it seems to me that the P-CSCF should send a 408 response towards the =
UA. If not, it will keep retransmitting until 64*2000 =3D 128 seconds =
(all assuming UDP of course)

Regards,

Jeroen



  RjS


  On Jul 27, 2006, at 2:21 PM, Jeroen van Bemmel wrote:


      This is not the way the timers are meant to be changed.


      Both of the transaction-stateful things on either end of a given =
transaction must share the same
      values for the transaction timers or the results are not defined.


      What you describe is not unlike throwing away the timing belt in =
an engine and spinning
      the head at some rate unrelated to the crankshaft. You might get  =
lucky, but usually very bad things
      are going to happen.


    That's interesting, because it is exactly what is prescribed for IMS


    Refer to TS 24.229 release 7 table 7.8 "SIP Timers" =
(http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.zip, =
page 132 for me)


    It says:
    Value to be applied at the UE: T1 =3D 2s default
    Value to be applied between IM CN subsystem elements: T1=3D500ms =
default


    So you are saying that in IMS configured this way, very bad things =
will happen?


    Regards,


    jeroen




      RjS


      On Jul 27, 2006, at 12:29 PM, Jeroen Van Bemmel wrote:


        Robert,


        RFC4320 bans 408 responses to NIT, but I think there are =
situations  in which it is still useful and needed. In particular, =
consider  when the UAC is using T1=3D2000 and the proxy uses T1=3D500 =
(eg because  the client is on a slow wireless link)


        The CT at the proxy times out after 64*T1=3D32 sec, while the =
client  is still retransmitting. In this case, 408 does help


        Regards,
        Jeroen




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BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Drjsparks@estacado.net =
href=3D"mailto:rjsparks@estacado.net">Robert=20
  Sparks</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Djbemmel@zonnet.nl=20
  href=3D"mailto:jbemmel@zonnet.nl">Jeroen van Bemmel</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Dsip@ietf.org=20
  href=3D"mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Thursday, July 27, 2006 =
9:48=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: RFC4320 and 408 to =

  non-INVITE</DIV>
  <DIV><BR></DIV>So, specifically, that table says:
  <DIV>
  <TABLE=20
  style=3D"BORDER-RIGHT: medium none; BORDER-TOP: medium none; =
MARGIN-LEFT: -1pt; BORDER-LEFT: medium none; BORDER-BOTTOM: medium none; =
BORDER-COLLAPSE: collapse; mso-border-alt: solid black 1.0pt; =
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        <DIV class=3DTAH><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
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        class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: black 1pt solid; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
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solid black .75pt"=20
      vAlign=3Dtop width=3D99>
        <DIV class=3DTAH><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">Value to be applied between =
IM CN=20
        subsystem elements </SPAN><SPAN class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: black 1pt solid; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
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      vAlign=3Dtop>
        <DIV class=3DTAH><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">Value to be applied at the=20
        UE</SPAN><SPAN class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
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solid; HEIGHT: 14.2pt; mso-border-left-alt: solid black .75pt"=20
      vAlign=3Dtop>
        <DIV class=3DTAH><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">Value to be applied at the =
P-CSCF=20
        toward a UE</SPAN><SPAN class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: black 1pt solid; PADDING-RIGHT: 0.7pt; =
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        <DIV class=3DTAH><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">Meaning</SPAN><SPAN=20
        class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
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      style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: medium none; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: black 1pt solid; WIDTH: 44.25pt; PADDING-TOP: 0.7pt; =
BORDER-BOTTOM: black 1pt solid; mso-border-top-alt: solid black .75pt"=20
      vAlign=3Dtop width=3D44>
        <DIV class=3DTAL><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">T1</SPAN><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: medium none; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; WIDTH: 99pt; PADDING-TOP: 0.7pt; =
BORDER-BOTTOM: black 1pt solid; mso-border-left-alt: solid black .75pt; =
mso-border-top-alt: solid black .75pt"=20
      vAlign=3Dtop width=3D99>
        <DIV class=3DTAL><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">500ms default</SPAN><SPAN=20
        class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: medium none; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 1pt =
solid; mso-border-left-alt: solid black .75pt; mso-border-top-alt: solid =
black .75pt"=20
      vAlign=3Dtop>
        <DIV class=3DTAL><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">2s default</SPAN><SPAN=20
        class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0in; =
BORDER-TOP: medium none; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; PADDING-TOP: 0in; BORDER-BOTTOM: black 1pt =
solid; mso-border-left-alt: solid black .75pt; mso-border-top-alt: solid =
black .75pt"=20
      vAlign=3Dtop>
        <DIV class=3DTAL><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">2s default</SPAN><SPAN=20
        class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: black 1pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: medium none; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 1pt =
solid; mso-border-left-alt: solid black .75pt; mso-border-top-alt: solid =
black .75pt"=20
      vAlign=3Dtop>
        <DIV class=3DTAL><SPAN lang=3DEN-GB><SPAN =
class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: collapse">RTT estimate</SPAN><SPAN=20
        class=3DApple-style-span=20
        style=3D"BORDER-COLLAPSE: =
collapse"><O:P></O:P></SPAN></SPAN></DIV></TD></TR></TBODY></TABLE><SPAN =

  class=3DApple-style-span style=3D"BORDER-COLLAPSE: collapse">
  <DIV><SPAN class=3DApple-style-span style=3D"BORDER-COLLAPSE: =
separate"><BR=20
  class=3Dkhtml-block-placeholder></SPAN></DIV>
  <DIV><BR class=3Dkhtml-block-placeholder></DIV>So the P-CSCF and the =
UE are both=20
  using 2s for T1.</SPAN></DIV>
  <DIV>When the P-CSCF talks to other things (that are not the UE), it =
uses=20
  500ms for T1.</DIV>
  <DIV><BR class=3Dkhtml-block-placeholder></DIV>
  <DIV>What you were describing would be the P-CSCF using 500ms on its =
side of a=20
  transaction it was in with a</DIV>
  <DIV>UA that was using 2s, which is not what this table says to do. In =
that=20
  case, yes, undefined things can happen.</DIV><FONT face=3DArial=20
size=3D2></FONT></BLOCKQUOTE>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BLOCKQUOTE>
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Then perhaps I was not clear=20
enough.&nbsp;What I meant was the UA using T1=3D2000 towards the P-CSCF, =
and the=20
P-CSCF using T1=3D500 towards the S-CSCF. If the latter transaction =
times out=20
(after 64*500=3D32 seconds), it seems to me that the P-CSCF should send =
a 408=20
response towards the UA. If not, it will keep retransmitting until =
64*2000 =3D 128=20
seconds (all assuming UDP of course)</FONT></DIV>
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Regards,</FONT></DIV>
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Jeroen</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
  <DIV><BR class=3Dkhtml-block-placeholder></DIV>
  <DIV>RjS</DIV>
  <DIV><BR class=3Dkhtml-block-placeholder></DIV>
  <DIV>
  <DIV>
  <DIV>
  <DIV>
  <DIV>
  <DIV>On Jul 27, 2006, at 2:21 PM, Jeroen van Bemmel wrote:</DIV><BR=20
  class=3DApple-interchange-newline>
  <BLOCKQUOTE type=3D"cite">
    <BLOCKQUOTE type=3D"cite">
      <DIV style=3D"MARGIN: 0px">This is not the way the timers are =
meant to be=20
      changed.</DIV>
      <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
      <DIV style=3D"MARGIN: 0px">Both of the transaction-stateful things =
on either=20
      end of a given transaction must share the same</DIV>
      <DIV style=3D"MARGIN: 0px">values for the transaction timers or =
the results=20
      are not defined.</DIV>
      <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
      <DIV style=3D"MARGIN: 0px">What you describe is not unlike =
throwing away the=20
      timing belt in an engine and spinning</DIV>
      <DIV style=3D"MARGIN: 0px">the head at some rate unrelated to the=20
      crankshaft. You might get<SPAN =
class=3DApple-converted-space>&nbsp;=20
      </SPAN>lucky, but usually very bad things</DIV>
      <DIV style=3D"MARGIN: 0px">are going to happen.</DIV></BLOCKQUOTE>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
    <DIV style=3D"MARGIN: 0px">That's interesting, because it is exactly =
what is=20
    prescribed for IMS</DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
    <DIV style=3D"MARGIN: 0px">Refer to TS 24.229 release 7 table 7.8 =
"SIP Timers"=20
    (<A=20
    =
href=3D"http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.=
zip">http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.zip=
</A>,=20
    page 132 for me)</DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
    <DIV style=3D"MARGIN: 0px">It says:</DIV>
    <DIV style=3D"MARGIN: 0px">Value to be applied at the UE: T1 =3D 2s=20
default</DIV>
    <DIV style=3D"MARGIN: 0px">Value to be applied between IM CN =
subsystem=20
    elements: T1=3D500ms default</DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
    <DIV style=3D"MARGIN: 0px">So you are saying that in IMS configured =
this way,=20
    very bad things will happen?</DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
    <DIV style=3D"MARGIN: 0px">Regards,</DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
    <DIV style=3D"MARGIN: 0px">jeroen</DIV>
    <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
    <BLOCKQUOTE type=3D"cite">
      <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
      <DIV style=3D"MARGIN: 0px">RjS</DIV>
      <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
      <DIV style=3D"MARGIN: 0px">On Jul 27, 2006, at 12:29 PM, Jeroen =
Van Bemmel=20
      wrote:</DIV>
      <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
      <BLOCKQUOTE type=3D"cite">
        <DIV style=3D"MARGIN: 0px">Robert,</DIV>
        <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
        <DIV style=3D"MARGIN: 0px">RFC4320 bans 408 responses to NIT, =
but I think=20
        there are situations<SPAN class=3DApple-converted-space>&nbsp; =
</SPAN>in=20
        which it is still useful and needed. In particular, =
consider<SPAN=20
        class=3DApple-converted-space>&nbsp; </SPAN>when the UAC is =
using T1=3D2000=20
        and the proxy uses T1=3D500 (eg because<SPAN=20
        class=3DApple-converted-space>&nbsp; </SPAN>the client is on a =
slow=20
        wireless link)</DIV>
        <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
        <DIV style=3D"MARGIN: 0px">The CT at the proxy times out after =
64*T1=3D32=20
        sec, while the client<SPAN class=3DApple-converted-space>&nbsp; =
</SPAN>is=20
        still retransmitting. In this case, 408 does help</DIV>
        <DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px"><BR></DIV>
        <DIV style=3D"MARGIN: 0px">Regards,</DIV>
        <DIV style=3D"MARGIN: =
0px">Jeroen</DIV></BLOCKQUOTE></BLOCKQUOTE>
    <DIV=20
  style=3D"MIN-HEIGHT: 14px; MARGIN: =
0px"><BR></DIV></BLOCKQUOTE></DIV><BR></DIV></DIV></DIV></DIV></BLOCKQUOT=
E></BODY></HTML>

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From sip-bounces@ietf.org Thu Jul 27 16:03:13 2006
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Date: Thu, 27 Jul 2006 15:01:13 -0500
To: "Brian Rosen" <br@brianrosen.net>, "'Andrew Newton'" <andy@hxr.us>
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Cc: 'IETF SIP List' <sip@ietf.org>, 'GEOPRIV' <geopriv@ietf.org>
Subject: [Sip] RE: [Geopriv] Teasing apart: constraints on dereference URI
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At 02:57 PM 7/27/2006 -0400, Brian Rosen wrote:
>It just seems so unlikely to me that geopriv would sanction any URI
>dereference mechanism that didn't have an RFC it could inject its review
>into.  It's got to be limited to things that are RFCs.  We can argue about
>whether it has to be standards track, we can argue about if the limit is
>enforced with ABNF, IANA registration or normative wording, but it's gotta
>be an RFC.

I vote for Standards Track, with Geopriv review, this takes care of all of 
the above.  If extensions to this doc are in the Geopriv WG, it really 
takes care of all of it.


>Brian
>
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Thursday, July 27, 2006 2:42 PM
> > To: James M. Polk
> > Cc: IETF SIP List; GEOPRIV
> > Subject: Re: [Geopriv] Teasing apart: constraints on dereference URI
> >
> > James M. Polk wrote:
> > >> Finally, how many people desire a middle ground?  Perhaps where the
> > >> ABNF, XML Schema or other grammar specification allows for any URI
> > >> scheme but there is an IANA registry listing the allowed dereference
> > >> URI schemes and associated protocols?
> > >
> > > I think any IANA Registry here will point back to a Standards Track RFC
> > > that defined the usage and semantics, which your question is not
> > > allowing.  Therefore, for what you have immediately above, I disagree.
> >
> > I guess that may have been too much hand waving on my part.  So let me
> > ask you this question:  if the grammar were to only specify "URI" and no
> > particular kind of URI, yet the kinds of URIs allowed must have to pass
> > some sort of review before their use in the Location header is
> > considered acceptable, would you find favor in this approach?
> >
> > And if so, how would you implement it?
> >
> > -andy
> >
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv

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Subject: RE: [Sip] Hitchhikers guide - when is it done?
Date: Thu, 27 Jul 2006 16:05:13 -0400
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Thread-Topic: [Sip] Hitchhikers guide - when is it done?
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Vijay K. Gurbani" <vkg@lucent.com>,
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Then we will need an RFC to point to the index RFCs that point to the
base RFCs?

A single RFC pointing document would be best.

Mike
=20

> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]=20
> Sent: Tuesday, July 25, 2006 5:43 PM
> To: Jonathan Rosenberg (jdrosen)
> Cc: IETF SIP List
> Subject: Re: [Sip] Hitchhikers guide - when is it done?
>=20
> Sorry for being the voice out on the left field, but I think=20
> 1 is better.  Once the 'core' documents have RFC numbers, it=20
> makes sense to go ahed and publish that as an Informational=20
> (maybe name it "A Hitchhikers Guide to SIP - Part 1: Core=20
> Specification").
>=20
> Then deal with the current I-Ds and other work as a separate=20
> Informational, probably named "A Hitchhikers Guide to SIP -=20
> Part 2: Extensions".  The first RFC will have an informative=20
> reference to the second to-be RFC, so readers can be warned=20
> that the document is split into to (plus the naming=20
> convention of the title is a clue).
>=20
> With Part 2 being in flux for a while, I would presume that=20
> it would be easier from a process point of view to rev a I-D=20
> than it is to obsolete or update a RFC.
>=20
> Plus there is some poetic justice in the SIP WG closing doors=20
> when the last RFC issued is one that (partially) summarizes=20
> what we have done as a WG.  Too bad RFC number 4242 is already taken.
>=20
> Thanks.
>=20
> Jonathan Rosenberg wrote:
> > During the IETF meeting, I discussed the one current open=20
> issue in the=20
> > Hitchhikers guide. The issue is: when does the document=20
> finish? When=20
> > do we send to IESG to issue an RFC?
> >=20
> > There were two approaches:
> >=20
> > 1. split the document in two. One part contains only the=20
> core specs,=20
> > and issues relatively soon. The other part has everything else, and=20
> > basically ends up being the last RFC produced by the SIP WG=20
> before it=20
> > shuts its doors (which has always been 1.5 years from now=20
> for the last=20
> > 5 years).
> >=20
> > 2. keep it as one document. Issue an RFC at some point, and then=20
> > revise it a few times as needed. We need to decide when to issue it.
> >=20
> >=20
> > I'd like to propose option 2. I think people get more value=20
> from it as=20
> > a consolidated document. Furthermore, I'd propose that we send the=20
> > document to IESG for consideration as informational at the=20
> instant all=20
> > of the documents listed as 'core' have RFC numbers issued by RFC-ed.
> >=20
> > Comments?
>=20
> - vijay
> --
> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> Bell Laboratories, Lucent Technologies, Inc.
> 2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Thu Jul 27 16:18:41 2006
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Thread-Topic: [Geopriv] Re: [Sip] teasing apart: http
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
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While I tend to agree with the sentiment that the term "using protocol" =
has been lawyered into obsolence in this discussion, I do think the term =
remains material to the scope of the location conveyance draft, and the =
question of how restrictive or permissive the URI schemes defined for =
the Location header might be.

Strictly speaking, when a SIP message carries only a reference to =
location information, it isn't really "conveying" location at all - the =
protocol that is used to follow that reference and retrieve location =
information is the "using protocol" as understood in RFC3693. Placing a =
CID, or SIP/SIPS, and for the most part PRES URI scheme in the Location =
header are all cases where SIP does or will do the work of conveyance, =
and will serve as a using protocol. That seems to me to be as good a =
reason as any to draw a line in the sand and say that these schemes are =
what the baseline SIP location conveyance draft will cover. I.e., When a =
SIP/SIPS URI appears in the Location header, the location will end up =
being conveyed by SIP; it seems pretty reasonable for the SIP location =
conveyance draft to cover this case.

I don't think that the syntax of the Location header should forbid the =
use of any particular URI schemes. Moving forward there's no obstacle to =
a separate draft defining HTTP conveyance or whatever else. The early =
failures of the GEOPRIV working group to arrive at a consensus on an =
approach to HTTP were mostly caused by the use cases which were being =
proposed, which were out of parity with much of what the rest of the =
group was trying to accomplish (and by "early" here I mean pre-HELD by a =
long shot).

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> Sent: Thursday, July 27, 2006 12:27 PM
> To: Henning Schulzrinne
> Cc: IETF SIP List; GEOPRIV; Andrew Newton
> Subject: Re: [Geopriv] Re: [Sip] teasing apart: http asa GEOPRIV using
> protocol
>=20
>=20
> Hi Henning,
>=20
> I would very much appreciate to treat this topic in the way=20
> you proposed=20
> it below.
>=20
> A few minor comments:
>=20
> Henning Schulzrinne wrote:
> > With hindsight, I think the term 'using protocol' has been=20
> the source of=20
> >  more word-lawyering than engineering insight. I personally=20
> find the=20
> > distinctions
> >=20
> >   - first person vs. third person (i.e., find out location=20
> about myself=20
> > vs. find out location about some third party)
> >=20
> >   - location vs. location + identity
> >=20
> >   - token-based authorization vs. identity-based authorization
> >=20
> > more helpful.
> >=20
> > Token-based authorization hands the recipient a key that=20
> allows him to=20
> > get a location object, without caring as to who is retrieving the=20
> > object. ("Anybody seeing this SIP request can convert the=20
> token into an=20
> > LO.") Identity-based naturally uses some authenticated identity to=20
> > release that information.
>=20
>=20
> In past common policy discussions we also differentiated between=20
> limited use addresses and trait-based authorization. A token=20
> would match=20
> the latter.
>=20
> In some sense we seem to be using limited use addresses in=20
> most of the=20
> cases when we talk about the location-by-references. The difference=20
> between limited use addresses and trait-based authorization=20
> is small in=20
> this particular usage environment.
>=20
>=20
>   Strictly speaking, this is only a question of
> > a level of indirection, since proving identity usually=20
> involves proving=20
> > possession of a secret.
>=20
> I guess there are some further differences with regard to the=20
> lifetime=20
> of the different identifiers, the ability to revoke rights,=20
> the number=20
> of entities that one can hide behind these identifiers and=20
> the storage=20
> implications.
>=20
> > We've been trying to get at some of these distinctions with=20
> the sighting=20
> > vs. using protocol terminology, but that distinction also seems to=20
> > degenerate into discussions more suited for a law school or=20
> theology=20
> > department than an engineering organization.
> >=20
> > For example, by strict definition, LoST would be a 'Using Protocol'=20
> > since it "carries a Location Object", but I hope we all=20
> agree that this=20
> > is silly. We had earlier agreed that the location-by-DHCP=20
> also were not=20
> > Using Protocol, even though they obviously carry a location=20
> object of=20
> > some form.
>=20
> Currently LoST is not a using protocol since it does not=20
> carry a PIDF-LO=20
> although some folks would like to use a PIDF-LO; HELD was a using=20
> protocol before we turned the problem into a Location Configuration=20
> problem; RELO never was a using protocol;
>=20
> Isn't this confusing?
>=20
> >=20
> > Privacy issues apply iff
> >   - third person
> >   - location + identity (not just location)
> >=20
> > The L7 discussion was about 'first person' and 'location=20
> only' (although=20
> > the latter is subject to debate, depending on whether one=20
> considers an=20
> > IP address of identifying an individual or not).
>=20
> That was before we added the Location-by-Reference and=20
> subscription-URI=20
> concept ....
>=20
> >=20
> > Depending on the circumstances, SIP-based retrieval can=20
> either involve=20
> > privacy concerns (subscribe to alice@example.com) or not=20
> (subscribe to=20
> > location of one-time random unlinkable identifier=20
> > jdfa8f9348@example.com). There can also be different=20
> privacy concerns=20
> > depending on the perspective. For example, if a SIP request=20
> contains=20
> > both an identity and the location retrieval reference, this raises=20
> > privacy concerns for the proxy, but the LIS may not have any such=20
> > concerns since it may only be the repository of these=20
> randomly-named=20
> > objects, without tie to an identity.
>=20
> Ciao
> Hannes
>=20
> > Henning
> >=20
> >=20
> >=20
> >=20
> >=20
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Thu Jul 27 16:46:01 2006
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From: Robert Sparks <rjsparks@estacado.net>
Date: Thu, 27 Jul 2006 15:42:46 -0500
To: Jeroen van Bemmel <jbemmel@zonnet.nl>
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Yeah - I definitely needed that clarification for the problem you are  
describing.

What you are describing now is a special case of something that is  
not timer
specific. The general case is that of a proxy where all possible  
downstream
legs have failed before the original request times out, but none of  
those legs
provided a SIP response that was suitable for forwarding as the  
"best" response.

For instance, each of the downstream legs could have produced an ICMP  
error
immediately.

The response to send in this situation is 480.

RjS


On Jul 27, 2006, at 2:54 PM, Jeroen van Bemmel wrote:

> inline
> ----- Original Message -----
> From: Robert Sparks
> To: Jeroen van Bemmel
> Cc: sip@ietf.org
> Sent: Thursday, July 27, 2006 9:48 PM
> Subject: Re: RFC4320 and 408 to non-INVITE
>
> So, specifically, that table says:
> SIP Timer
> Value to be applied between IM CN subsystem elements
> Value to be applied at the UE
> Value to be applied at the P-CSCF toward a UE
> Meaning
> T1
> 500ms default
> 2s default
> 2s default
> RTT estimate
>
>
> So the P-CSCF and the UE are both using 2s for T1.
> When the P-CSCF talks to other things (that are not the UE), it  
> uses 500ms for T1.
>
> What you were describing would be the P-CSCF using 500ms on its  
> side of a transaction it was in with a
> UA that was using 2s, which is not what this table says to do. In  
> that case, yes, undefined things can happen.
>
> Then perhaps I was not clear enough. What I meant was the UA using  
> T1=2000 towards the P-CSCF, and the P-CSCF using T1=500 towards the  
> S-CSCF. If the latter transaction times out (after 64*500=32  
> seconds), it seems to me that the P-CSCF should send a 408 response  
> towards the UA. If not, it will keep retransmitting until 64*2000 =  
> 128 seconds (all assuming UDP of course)
>
> Regards,
>
> Jeroen
>
>
> RjS
>
> On Jul 27, 2006, at 2:21 PM, Jeroen van Bemmel wrote:
>
>>> This is not the way the timers are meant to be changed.
>>>
>>> Both of the transaction-stateful things on either end of a given  
>>> transaction must share the same
>>> values for the transaction timers or the results are not defined.
>>>
>>> What you describe is not unlike throwing away the timing belt in  
>>> an engine and spinning
>>> the head at some rate unrelated to the crankshaft. You might get   
>>> lucky, but usually very bad things
>>> are going to happen.
>>
>> That's interesting, because it is exactly what is prescribed for IMS
>>
>> Refer to TS 24.229 release 7 table 7.8 "SIP Timers" (http://www. 
>> 3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.zip, page  
>> 132 for me)
>>
>> It says:
>> Value to be applied at the UE: T1 = 2s default
>> Value to be applied between IM CN subsystem elements: T1=500ms  
>> default
>>
>> So you are saying that in IMS configured this way, very bad things  
>> will happen?
>>
>> Regards,
>>
>> jeroen
>>
>>>
>>> RjS
>>>
>>> On Jul 27, 2006, at 12:29 PM, Jeroen Van Bemmel wrote:
>>>
>>>> Robert,
>>>>
>>>> RFC4320 bans 408 responses to NIT, but I think there are  
>>>> situations  in which it is still useful and needed. In  
>>>> particular, consider  when the UAC is using T1=2000 and the  
>>>> proxy uses T1=500 (eg because  the client is on a slow wireless  
>>>> link)
>>>>
>>>> The CT at the proxy times out after 64*T1=32 sec, while the  
>>>> client  is still retransmitting. In this case, 408 does help
>>>>
>>>> Regards,
>>>> Jeroen
>>
>
>


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; ">Yeah - I definitely needed that =
clarification for the problem you are describing.<DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>What you are describing now =
is a special case of something that is not timer</DIV><DIV>specific. The =
general case is that of a proxy where all possible =
downstream</DIV><DIV>legs have failed before the original request times =
out, but none of those legs</DIV><DIV>provided a SIP response that was =
suitable for forwarding as the "best" response.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>For instance, each of the =
downstream legs could have produced an ICMP =
error</DIV><DIV>immediately.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>The response to send in =
this situation is 480.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>RjS</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><DIV><DIV>On Jul 27, 2006, =
at 2:54 PM, Jeroen van Bemmel wrote:</DIV><BR =
class=3D"Apple-interchange-newline"><BLOCKQUOTE type=3D"cite"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px 0px; color: rgb(0, 0, 0); font-family: Helvetica; =
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white-space: normal; widows: 2; word-spacing: 0px; "><DIV><FONT =
face=3D"Arial" size=3D"2"><SPAN class=3D"Apple-style-span" =
style=3D"font-family: Arial; font-size: 10px; =
">inline</SPAN></FONT></DIV><BLOCKQUOTE dir=3D"ltr" =
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px"><DIV style=3D"FONT: =
10pt arial; font-family: arial; font-size: 13.3333px; "><SPAN =
class=3D"Apple-style-span" style=3D"font-family: arial; font-size: =
13.3333px; ">----- Original Message -----</SPAN></DIV><DIV =
style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black; =
font-family: arial; font-size: 13.3333px; "><B style=3D"font-family: =
arial; font-size: 13.3333px; font-weight: bold; "><SPAN =
class=3D"Apple-style-span" style=3D"font-family: arial; font-size: =
13.3333px; font-weight: bold; ">From:</SPAN></B><SPAN =
class=3D"Apple-style-span" style=3D"font-family: arial; font-size: =
13.3333px; "> </SPAN><A title=3D"rjsparks@estacado.net" =
href=3D"mailto:rjsparks@estacado.net"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(0, 0, 238); font-family: arial; font-size: =
13.3333px; -khtml-text-decorations-in-effect: underline; ">Robert =
Sparks</SPAN></A><SPAN class=3D"Apple-style-span" style=3D"font-family: =
arial; font-size: 13.3333px; "></SPAN></DIV><DIV style=3D"FONT: 10pt =
arial; font-family: arial; font-size: 13.3333px; "><B =
style=3D"font-family: arial; font-size: 13.3333px; font-weight: bold; =
"><SPAN class=3D"Apple-style-span" style=3D"font-family: arial; =
font-size: 13.3333px; font-weight: bold; ">To:</SPAN></B><SPAN =
class=3D"Apple-style-span" style=3D"font-family: arial; font-size: =
13.3333px; "> </SPAN><A title=3D"jbemmel@zonnet.nl" =
href=3D"mailto:jbemmel@zonnet.nl"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(0, 0, 238); font-family: arial; font-size: =
13.3333px; -khtml-text-decorations-in-effect: underline; ">Jeroen van =
Bemmel</SPAN></A><SPAN class=3D"Apple-style-span" style=3D"font-family: =
arial; font-size: 13.3333px; "></SPAN></DIV><DIV style=3D"FONT: 10pt =
arial; font-family: arial; font-size: 13.3333px; "><B =
style=3D"font-family: arial; font-size: 13.3333px; font-weight: bold; =
"><SPAN class=3D"Apple-style-span" style=3D"font-family: arial; =
font-size: 13.3333px; font-weight: bold; ">Cc:</SPAN></B><SPAN =
class=3D"Apple-style-span" style=3D"font-family: arial; font-size: =
13.3333px; "> </SPAN><A title=3D"sip@ietf.org" =
href=3D"mailto:sip@ietf.org"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(0, 0, 238); font-family: arial; font-size: =
13.3333px; -khtml-text-decorations-in-effect: underline; =
">sip@ietf.org</SPAN></A><SPAN class=3D"Apple-style-span" =
style=3D"font-family: arial; font-size: 13.3333px; "></SPAN></DIV><DIV =
style=3D"FONT: 10pt arial; font-family: arial; font-size: 13.3333px; =
"><B style=3D"font-family: arial; font-size: 13.3333px; font-weight: =
bold; "><SPAN class=3D"Apple-style-span" style=3D"font-family: arial; =
font-size: 13.3333px; font-weight: bold; ">Sent:</SPAN></B><SPAN =
class=3D"Apple-style-span" style=3D"font-family: arial; font-size: =
13.3333px; "> Thursday, July 27, 2006 9:48 PM</SPAN></DIV><DIV =
style=3D"FONT: 10pt arial; font-family: arial; font-size: 13.3333px; =
"><B style=3D"font-family: arial; font-size: 13.3333px; font-weight: =
bold; "><SPAN class=3D"Apple-style-span" style=3D"font-family: arial; =
font-size: 13.3333px; font-weight: bold; ">Subject:</SPAN></B><SPAN =
class=3D"Apple-style-span" style=3D"font-family: arial; font-size: =
13.3333px; "> Re: RFC4320 and 408 to =
non-INVITE</SPAN></DIV><DIV><BR></DIV>So, specifically, that table =
says:<DIV><TABLE style=3D"BORDER-RIGHT: medium none; BORDER-TOP: medium =
none; MARGIN-LEFT: -1pt; BORDER-LEFT: medium none; BORDER-BOTTOM: medium =
none; BORDER-COLLAPSE: collapse; mso-border-alt: solid black 1.0pt; =
mso-padding-alt: 0in 0in 0in 0in" cellspacing=3D"0" cellpadding=3D"0" =
border=3D"1"><TBODY style=3D"border-collapse: collapse; "><TR =
style=3D"HEIGHT: 14.2pt; border-collapse: collapse; "><TD =
style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: black 1pt solid; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: black 1pt solid; WIDTH: 44.25pt; PADDING-TOP: 0.7pt; =
BORDER-BOTTOM: black 0.75pt solid; HEIGHT: 14.2pt; border-collapse: =
collapse; " valign=3D"top" width=3D"44"><DIV class=3D"TAH"><SPAN =
lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: =
collapse"><SPAN class=3D"Apple-style-span" style=3D"border-collapse: =
collapse; ">SIP Timer</SPAN></SPAN><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><O:P style=3D"border-collapse: =
collapse; "></O:P></SPAN></SPAN></DIV></TD><TD style=3D"BORDER-RIGHT: =
black 0.75pt solid; PADDING-RIGHT: 0.7pt; BORDER-TOP: black 1pt solid; =
PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; =
WIDTH: 99pt; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 0.75pt solid; =
HEIGHT: 14.2pt; mso-border-left-alt: solid black .75pt; border-collapse: =
collapse; " valign=3D"top" width=3D"99"><DIV class=3D"TAH"><SPAN =
lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: =
collapse"><SPAN class=3D"Apple-style-span" style=3D"border-collapse: =
collapse; ">Value to be applied between IM CN subsystem =
elements</SPAN></SPAN><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><O:P style=3D"border-collapse: =
collapse; "></O:P></SPAN></SPAN></DIV></TD><TD style=3D"BORDER-RIGHT: =
black 0.75pt solid; PADDING-RIGHT: 0.7pt; BORDER-TOP: black 1pt solid; =
PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; =
PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 0.75pt solid; HEIGHT: 14.2pt; =
mso-border-left-alt: solid black .75pt; border-collapse: collapse; " =
valign=3D"top"><DIV class=3D"TAH"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse; ">Value =
to be applied at the UE</SPAN></SPAN><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><O:P style=3D"border-collapse: =
collapse; "></O:P></SPAN></SPAN></DIV></TD><TD style=3D"BORDER-RIGHT: =
black 0.75pt solid; PADDING-RIGHT: 0in; BORDER-TOP: black 1pt solid; =
PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; =
PADDING-TOP: 0in; BORDER-BOTTOM: black 0.75pt solid; HEIGHT: 14.2pt; =
mso-border-left-alt: solid black .75pt; border-collapse: collapse; " =
valign=3D"top"><DIV class=3D"TAH"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse; ">Value =
to be applied at the P-CSCF toward a UE</SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><O:P =
style=3D"border-collapse: collapse; "></O:P></SPAN></SPAN></DIV></TD><TD =
style=3D"BORDER-RIGHT: black 1pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: black 1pt solid; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black =
0.75pt solid; HEIGHT: 14.2pt; mso-border-left-alt: solid black .75pt; =
border-collapse: collapse; " valign=3D"top"><DIV class=3D"TAH"><SPAN =
lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: =
collapse"><SPAN class=3D"Apple-style-span" style=3D"border-collapse: =
collapse; ">Meaning</SPAN></SPAN><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><O:P style=3D"border-collapse: =
collapse; "></O:P></SPAN></SPAN></DIV></TD></TR><TR =
style=3D"border-collapse: collapse; "><TD style=3D"BORDER-RIGHT: black =
0.75pt solid; PADDING-RIGHT: 0.7pt; BORDER-TOP: medium none; =
PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; BORDER-LEFT: black 1pt solid; =
WIDTH: 44.25pt; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 1pt solid; =
mso-border-top-alt: solid black .75pt; border-collapse: collapse; " =
valign=3D"top" width=3D"44"><DIV class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse; =
">T1</SPAN></SPAN><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><O:P style=3D"border-collapse: =
collapse; "></O:P></SPAN></SPAN></DIV></TD><TD style=3D"BORDER-RIGHT: =
black 0.75pt solid; PADDING-RIGHT: 0.7pt; BORDER-TOP: medium none; =
PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; =
WIDTH: 99pt; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 1pt solid; =
mso-border-left-alt: solid black .75pt; mso-border-top-alt: solid black =
.75pt; border-collapse: collapse; " valign=3D"top" width=3D"99"><DIV =
class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: collapse; ">500ms default</SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><O:P =
style=3D"border-collapse: collapse; "></O:P></SPAN></SPAN></DIV></TD><TD =
style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: medium none; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 1pt =
solid; mso-border-left-alt: solid black .75pt; mso-border-top-alt: solid =
black .75pt; border-collapse: collapse; " valign=3D"top"><DIV =
class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: collapse; ">2s default</SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><O:P =
style=3D"border-collapse: collapse; "></O:P></SPAN></SPAN></DIV></TD><TD =
style=3D"BORDER-RIGHT: black 0.75pt solid; PADDING-RIGHT: 0in; =
BORDER-TOP: medium none; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; PADDING-TOP: 0in; BORDER-BOTTOM: black 1pt =
solid; mso-border-left-alt: solid black .75pt; mso-border-top-alt: solid =
black .75pt; border-collapse: collapse; " valign=3D"top"><DIV =
class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: collapse; ">2s default</SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><O:P =
style=3D"border-collapse: collapse; "></O:P></SPAN></SPAN></DIV></TD><TD =
style=3D"BORDER-RIGHT: black 1pt solid; PADDING-RIGHT: 0.7pt; =
BORDER-TOP: medium none; PADDING-LEFT: 0.7pt; PADDING-BOTTOM: 0in; =
BORDER-LEFT: medium none; PADDING-TOP: 0.7pt; BORDER-BOTTOM: black 1pt =
solid; mso-border-left-alt: solid black .75pt; mso-border-top-alt: solid =
black .75pt; border-collapse: collapse; " valign=3D"top"><DIV =
class=3D"TAL"><SPAN lang=3D"EN-GB"><SPAN class=3D"Apple-style-span" =
style=3D"BORDER-COLLAPSE: collapse"><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: collapse; ">RTT estimate</SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><O:P =
style=3D"border-collapse: collapse; =
"></O:P></SPAN></SPAN></DIV></TD></TR></TBODY></TABLE><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: collapse"><SPAN =
class=3D"Apple-style-span" style=3D"border-collapse: collapse; =
"></SPAN><DIV style=3D"border-collapse: collapse; "><SPAN =
class=3D"Apple-style-span" style=3D"BORDER-COLLAPSE: separate"><BR =
class=3D"khtml-block-placeholder"></SPAN></DIV><DIV =
style=3D"border-collapse: collapse; "><BR =
class=3D"khtml-block-placeholder"></DIV><SPAN class=3D"Apple-style-span" =
style=3D"border-collapse: collapse; ">So the P-CSCF and the UE are both =
using 2s for T1.</SPAN></SPAN></DIV><DIV>When the P-CSCF talks to other =
things (that are not the UE), it uses 500ms for T1.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>What you were describing =
would be the P-CSCF using 500ms on its side of a transaction it was in =
with a</DIV><DIV>UA that was using 2s, which is not what this table says =
to do. In that case, yes, undefined things can happen.</DIV><FONT =
face=3D"Arial" size=3D"2"></FONT></BLOCKQUOTE><BLOCKQUOTE dir=3D"ltr" =
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px"><DIV><FONT =
face=3D"Arial" size=3D"2"></FONT>=A0</DIV></BLOCKQUOTE><DIV =
dir=3D"ltr"><FONT face=3D"Arial" size=3D"2"><SPAN =
class=3D"Apple-style-span" style=3D"font-family: Arial; font-size: 10px; =
">Then perhaps I was not clear enough.=A0What I meant was the UA using =
T1=3D2000 towards the P-CSCF, and the P-CSCF using T1=3D500 towards the =
S-CSCF. If the latter transaction times out (after 64*500=3D32 seconds), =
it seems to me that the P-CSCF should send a 408 response towards the =
UA. If not, it will keep retransmitting until 64*2000 =3D 128 seconds =
(all assuming UDP of course)</SPAN></FONT></DIV><DIV dir=3D"ltr"><FONT =
face=3D"Arial" size=3D"2"></FONT>=A0</DIV><DIV dir=3D"ltr"><FONT =
face=3D"Arial" size=3D"2"><SPAN class=3D"Apple-style-span" =
style=3D"font-family: Arial; font-size: 10px; =
">Regards,</SPAN></FONT></DIV><DIV dir=3D"ltr"><FONT face=3D"Arial" =
size=3D"2"></FONT>=A0</DIV><DIV dir=3D"ltr"><FONT face=3D"Arial" =
size=3D"2"><SPAN class=3D"Apple-style-span" style=3D"font-family: Arial; =
font-size: 10px; ">Jeroen</SPAN></FONT></DIV><BLOCKQUOTE dir=3D"ltr" =
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px"><DIV><FONT =
face=3D"Arial" size=3D"2"></FONT>=A0</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>RjS</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><DIV><DIV><DIV><DIV><DIV>On =
Jul 27, 2006, at 2:21 PM, Jeroen van Bemmel wrote:</DIV><BR =
class=3D"Apple-interchange-newline"><BLOCKQUOTE type=3D"cite"><BLOCKQUOTE =
type=3D"cite"><DIV style=3D"MARGIN: 0px; color: rgb(0, 99, 18); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 99, 18); ">This is not =
the way the timers are meant to be changed.</SPAN></DIV><DIV =
style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: rgb(0, 99, 18); "><BR =
style=3D"color: rgb(0, 99, 18); "></DIV><DIV style=3D"MARGIN: 0px; =
color: rgb(0, 99, 18); "><SPAN class=3D"Apple-style-span" style=3D"color: =
rgb(0, 99, 18); ">Both of the transaction-stateful things on either end =
of a given transaction must share the same</SPAN></DIV><DIV =
style=3D"MARGIN: 0px; color: rgb(0, 99, 18); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 99, 18); ">values for =
the transaction timers or the results are not defined.</SPAN></DIV><DIV =
style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: rgb(0, 99, 18); "><BR =
style=3D"color: rgb(0, 99, 18); "></DIV><DIV style=3D"MARGIN: 0px; =
color: rgb(0, 99, 18); "><SPAN class=3D"Apple-style-span" style=3D"color: =
rgb(0, 99, 18); ">What you describe is not unlike throwing away the =
timing belt in an engine and spinning</SPAN></DIV><DIV style=3D"MARGIN: =
0px; color: rgb(0, 99, 18); "><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(0, 99, 18); ">the head at some rate unrelated to the =
crankshaft. You might get</SPAN><SPAN =
class=3D"Apple-converted-space"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(0, 99, 18); ">=A0 </SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 99, 18); ">lucky, but =
usually very bad things</SPAN></DIV><DIV style=3D"MARGIN: 0px; color: =
rgb(0, 99, 18); "><SPAN class=3D"Apple-style-span" style=3D"color: =
rgb(0, 99, 18); ">are going to happen.</SPAN></DIV></BLOCKQUOTE><DIV =
style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: rgb(0, 0, 221); "><BR =
style=3D"color: rgb(0, 0, 221); "></DIV><DIV style=3D"MARGIN: 0px; =
color: rgb(0, 0, 221); "><SPAN class=3D"Apple-style-span" style=3D"color: =
rgb(0, 0, 221); ">That's interesting, because it is exactly what is =
prescribed for IMS</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: =
0px; color: rgb(0, 0, 221); "><BR style=3D"color: rgb(0, 0, 221); =
"></DIV><DIV style=3D"MARGIN: 0px; color: rgb(0, 0, 221); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); ">Refer to TS =
24.229 release 7 table 7.8 "SIP Timers" (</SPAN><A =
href=3D"http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.z=
ip"><SPAN class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 238); =
-khtml-text-decorations-in-effect: underline; =
">http://www.3gpp.org/ftp/Specs/archive/24_series/24.229/24229-740.zip</SP=
AN></A><SPAN class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); =
">, page 132 for me)</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: =
0px; color: rgb(0, 0, 221); "><BR style=3D"color: rgb(0, 0, 221); =
"></DIV><DIV style=3D"MARGIN: 0px; color: rgb(0, 0, 221); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); ">It =
says:</SPAN></DIV><DIV style=3D"MARGIN: 0px; color: rgb(0, 0, 221); =
"><SPAN class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); =
">Value to be applied at the UE: T1 =3D 2s default</SPAN></DIV><DIV =
style=3D"MARGIN: 0px; color: rgb(0, 0, 221); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); ">Value to be =
applied between IM CN subsystem elements: T1=3D500ms =
default</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: =
rgb(0, 0, 221); "><BR style=3D"color: rgb(0, 0, 221); "></DIV><DIV =
style=3D"MARGIN: 0px; color: rgb(0, 0, 221); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); ">So you are =
saying that in IMS configured this way, very bad things will =
happen?</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: =
rgb(0, 0, 221); "><BR style=3D"color: rgb(0, 0, 221); "></DIV><DIV =
style=3D"MARGIN: 0px; color: rgb(0, 0, 221); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); =
">Regards,</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; =
color: rgb(0, 0, 221); "><BR style=3D"color: rgb(0, 0, 221); =
"></DIV><DIV style=3D"MARGIN: 0px; color: rgb(0, 0, 221); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 221); =
">jeroen</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: =
rgb(0, 0, 221); "><BR style=3D"color: rgb(0, 0, 221); =
"></DIV><BLOCKQUOTE type=3D"cite"><DIV style=3D"MIN-HEIGHT: 14px; =
MARGIN: 0px; color: rgb(0, 99, 18); "><BR style=3D"color: rgb(0, 99, =
18); "></DIV><DIV style=3D"MARGIN: 0px; color: rgb(0, 99, 18); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 99, 18); =
">RjS</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: =
rgb(0, 99, 18); "><BR style=3D"color: rgb(0, 99, 18); "></DIV><DIV =
style=3D"MARGIN: 0px; color: rgb(0, 99, 18); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(0, 99, 18); ">On Jul 27, =
2006, at 12:29 PM, Jeroen Van Bemmel wrote:</SPAN></DIV><DIV =
style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: rgb(0, 99, 18); "><BR =
style=3D"color: rgb(0, 99, 18); "></DIV><BLOCKQUOTE type=3D"cite"><DIV =
style=3D"MARGIN: 0px; color: rgb(84, 0, 0); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(84, 0, 0); =
">Robert,</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; =
color: rgb(84, 0, 0); "><BR style=3D"color: rgb(84, 0, 0); "></DIV><DIV =
style=3D"MARGIN: 0px; color: rgb(84, 0, 0); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(84, 0, 0); ">RFC4320 bans =
408 responses to NIT, but I think there are situations</SPAN><SPAN =
class=3D"Apple-converted-space"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(84, 0, 0); ">=A0 </SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(84, 0, 0); ">in which it =
is still useful and needed. In particular, consider</SPAN><SPAN =
class=3D"Apple-converted-space"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(84, 0, 0); ">=A0 </SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(84, 0, 0); ">when the UAC =
is using T1=3D2000 and the proxy uses T1=3D500 (eg because</SPAN><SPAN =
class=3D"Apple-converted-space"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(84, 0, 0); ">=A0 </SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(84, 0, 0); ">the client =
is on a slow wireless link)</SPAN></DIV><DIV style=3D"MIN-HEIGHT: 14px; =
MARGIN: 0px; color: rgb(84, 0, 0); "><BR style=3D"color: rgb(84, 0, 0); =
"></DIV><DIV style=3D"MARGIN: 0px; color: rgb(84, 0, 0); "><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(84, 0, 0); ">The CT at =
the proxy times out after 64*T1=3D32 sec, while the client</SPAN><SPAN =
class=3D"Apple-converted-space"><SPAN class=3D"Apple-style-span" =
style=3D"color: rgb(84, 0, 0); ">=A0 </SPAN></SPAN><SPAN =
class=3D"Apple-style-span" style=3D"color: rgb(84, 0, 0); ">is still =
retransmitting. In this case, 408 does help</SPAN></DIV><DIV =
style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; color: rgb(84, 0, 0); "><BR =
style=3D"color: rgb(84, 0, 0); "></DIV><DIV style=3D"MARGIN: 0px; color: =
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From sip-bounces@ietf.org Thu Jul 27 17:06:24 2006
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
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Trying, trying, ....

It seems we have a very rough consensus that there will be some
normative text that specifies the allowable URI schemes for what can be
in the location header.  There are still some voices in opposition to
it, but I think we've heard all the relevant points we're going to get.
I had made a specific proposal on this:

location-param =3D LAQUOT allowed-URI RAQUOT *( SEMI loc-param )=20
;cid-url from RFC2111, http_URL from RFC2616, SIP/SIPS-URI from RFC3261,
PRES-URI from RFC3859=20
allowed-URI =3D cid-uri / http_URL / https_URL / SIP-URI / SIPS-URI
              / PRES-URI

As we seem to be developing another rough consensus on NOT defining an
HTTP dereference mechanism IN THE LOCATION-CONVEYANCE DOCUMENT, this
would now be:
location-param =3D LAQUOT allowed-URI RAQUOT *( SEMI loc-param )=20
;cid-url from RFC2111, SIP/SIPS-URI from RFC3261, PRES-URI from RFC3859=20
allowed-URI =3D cid-uri / SIP-URI / SIPS-URI / PRES-URI

The text would say extensions to this list is by (?standards-track?) RFC
requiring geopriv review as a "using" protocol, but is not a SIP change.

A hypothetical http dereference RFC would have the following ABNF:
;allowed-URI from <location-conveyance>, http_URL from RFC2616
allowed-URI =3D/ http_URL / https_URL

The alternative is to use "AbsoluteURI", and to say roughly the same
thing about RFCs and geopriv review.  I prefer being explicit.  The =
"=3D/"
construction is very simple and very explicit.=20

Can we accept this "allowed-URI" proposal and move on? Does it have to
be standards track, or is Informational okay?  I would think
informational is okay, provided geopriv reviews.  For example, if some
other organization came along and defined a protocol that worked, and
met the geopriv requirements, an informational saying that the way you
use this with SIP Location is blah, blah, blah, I would think, provided
geopriv agreed it met their concerns, that would be okay (even with the
ABNF extension to allowed-URI in the Informational)=20

Brian

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From sip-bounces@ietf.org Thu Jul 27 17:11:59 2006
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	URI
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If HTTP and other mechanisms are not defined in the draft (with which I 
agree), I see no technical reason not to allow other URI schemes in the 
draft ABNF.

At least from the voices I heard, my proposal of not restricting the URI 
scheme has received at least as much support as countervailing 
proposals, so I don't know now how you're counting rough consensus. Or 
are there supervotes?

Rosen, Brian wrote:
> Trying, trying, ....
> 
> It seems we have a very rough consensus that there will be some
> normative text that specifies the allowable URI schemes for what can be
> in the location header.  There are still some voices in opposition to
> it, but I think we've heard all the relevant points we're going to get.
> I had made a specific proposal on this:
> 
> location-param = LAQUOT allowed-URI RAQUOT *( SEMI loc-param ) 
> ;cid-url from RFC2111, http_URL from RFC2616, SIP/SIPS-URI from RFC3261,
> PRES-URI from RFC3859 
> allowed-URI = cid-uri / http_URL / https_URL / SIP-URI / SIPS-URI
>               / PRES-URI
> 

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I did a reasonably careful review of draft-ietf-sip-gruu-09. I was 
mostly looking for remaining technical issues rather than formatting 
nits. For awhile now I have mostly be looking at diffs as new versions 
come out and reading selected sections at need. It was good to read the 
whole thing again to see how it all hangs together.

I found nothing really major, but some stuff that is more than 
editorial. Some of it is a judgment call about whether something is 
clear or not. I won't feel bad to have a number of these comments rejected.

	Paul

General comment:

SIPS is mentioned in many places in this draft. Given the current 
ongoing issues wrt SIPS, the language here for dealing with it may prove 
to be prescient, or may turn out to just be wrong. Can this be dodged 
somehow? Maybe this whole draft should only be defined for SIP, with 
SIPS left for a separate draft. (I will still comment on some places 
containing SIPS, when there are other points than the above.)

Section 5:

    ...  The "opaque" parameter is added by the owner
    of the domain to correlate the GRUU to its instance ID, and to easily
    recognize that the URI has the GRUU property.

I think this isn't quite right. The "gruu" parameter may be used to 
recognize the GRUU property. How about:

    ...  The "opaque" parameter is added by the owner
    of the domain to correlate the GRUU to its instance ID, and
    potentially to help verify that the GRUU has not been forged.

Figure 1:

Its unfortunate that pictures such as this lead to many questions, often 
irrelevant ones. I know we have discussed nuances of this one before.

- as shown, a contact is bound to 1 AOR, at most one Instance ID and at 
most one GRUU. In practice is is possible to bind the same contact to 
multiple AORs if one doesn't need to distinguish between them or has 
some other means to do so. (And section 7.1.1.1 allows it, since 
restriction to a single AOR is only RECOMMENDED.)  It is also possible 
that the same contact will be associated with more than one Instance ID. 
(Ideally there is only one such association at any one time, but over 
time it may change.}

- the figure doesn't show what is in the text:

    A GRUU is associated, in a many-to-one fashion, with the combination
    of an instance ID and an AOR.  This combination is referred to as an
    instance ID/AOR pair.  For each GRUU, there is one instance ID/AOR
    pair, and for each instance ID/AOR pair, there can be one or more
    GRUU.

(The picture doesn't show anything associated with an AOR/Instance ID pair.)

I think it is too hard to make a picture that expresses these 
relationships accurately, and this should be done with text, without the 
figure.

Section 6:

    o  For each GRUU, both the SIP and Session Initiation Protocol Secure
       (SIPS) versions MUST exist.  The SIPS URI may not always work,
       particularly if the proxy cannot establish a secure connection to
       the client.  However, the SIPS URI always exists.

The use of "always" seems problematic. The lifetime of the gruu itself 
is not "always". It is no longer than the associated AOR. Also, this 
goes into the previous discussion of relationships. Are the SIP and SIPS 
forms *separate* AORs, or the same AOR? I'm struggling here to come up 
with something that threads the needle. How about:

    o  When both SIP and SIPS AORs identify the same resource, and the
       AORs differ only in the scheme name, then when a GRUU is assigned
       to the resource, both SIP and SIPS GRUUs will be assigned. Only
       one will be returned, but both will exist.  The SIPS URI may not
       always work, particularly if the proxy cannot establish a secure
       connection to the client.

Section 7.1.1.1:

    Specifically, the contact in the REGISTER request SHOULD
    NOT contain the gruu Contact header field parameter, and the contact
    URI itself SHOULD NOT contain the "grid" URI parameter defined below.
    Any such parameters are ignored by the registrar, as the UA cannot
    propose a GRUU for usage with the contact.

Nit: the last sentence is a little weird, because the "as" clause only 
applies to gruu, not grid. Perhaps:

    Any such parameters are ignored by the registrar and not stored.
    (The UA cannot propose a GRUU for usage with the contact, and
    the grid parameter is populated from the GRUU, not the registered
    contact.)

BUT, later I instead suggest treating the presence of grid as an error 
rather than ignoring it. That requires different wording here.

Later in the section:

    ... However, it is
    permitted for a UA to register any kind of contact it wishes,
    including a GRUU.  A typical use case for registering a GRUU is if a
    user wanted calls to one AOR to be delivered to a specific UA
    instance associated with another AOR.

I just realized this presents a problem wrt grid processing. If the 
contact is a gruu associated with a different AOR, then it could also 
contain a grid parameter, and if so, that parameter might be important. 
One wouldn't want to lose it, or override it.

Perhaps it should be an *error* to register a contact that contains a 
grid URI parameter. That would prevent any confusion.

Section 7.1.1.2:

    ...  The URI will be a SIP URI if the To header
    field in the REGISTER request contained a SIP URI, else (if the To
    header field in the REGISTER request contained a SIPS URI) it will be
    a SIPS URI.

(more sips stuff)

    ... Normally, the GRUU will
    not change in subsequent 2xx responses to REGISTER.  Indeed, even if
    the UA lets the contact expire, when it re-registers it at any later
    time, the registrar will normally provide the same GRUU for the same
    address-of-record and instance ID.  However, as discussed above, this
    property cannot be completely guaranteed, as network failures may
    make it impossible to provide an identifier that persists for all
    time.  As a result, a UA MUST be prepared to receive a different GRUU
    for the same instance ID/AOR pair in a subsequent registration
    response.

As written, this puts a burden on a UA. Any time it refreshes a 
registration a previously assigned gruu may change. That gruu may be in 
use in a dialog. Then the UA would have to to a target refresh to fix 
it. That is a lot to ask, and has potential race conditions if the prior 
gruu has become invalid.

IMO it would be better to put the burden on the registrar to keep the 
gruu the same for as long as a registration remains active.

I think it might also be good to insert a note into this section:

     NOTE: the 2xx response to REGISTER returns all the contacts
     bound to the AOR. This may include contacts bound by other
     UAs. Any or all of these may include a "gruu" Contact header field
     parameter. The UA is responsible for extracting and saving the
     GRUU assigned to the contact(s) it has registered.

Section 7.1.2:

    A registrar MUST create both the SIP and SIPS versions of the GRUU,
    such that if the GRUU exists, both URI exist.

(more sips stuff)

Section 7.1.2.1:

Nothing is mentioned in this section about handling of a "grid" URI 
parameter in the contact. If this is present it can screw things up in a 
variety of ways. This is sufficiently severe that it probably isn't wise 
to just ignore it if present. Instead, should probably reject the 
request. (403 again?) The problems that can result:

- regular calls to the AOR will look (to the UE) like calls to a gruu
- if the contact was itself a GRUU, and had a particular grid value
   for some purpose, then when calls are made to the gruu assigned as
   part of this registration, the grid value will be messed up.

(Note that forbidding grid values here doesn't fully solve the problem. 
Using a gruu as a contact in another registration has unresolvable 
problems about the namespace for grid values. It can only work right if 
the two share a common namespace for the values.)

Same section:

    ... If the To
    header field in the REGISTER request contains a SIP URI, the SIP
    version of the GRUU is returned.  If the To header field in the
    REGISTER request contains a SIPS URI, the SIPS version of the GRUU is
    returned.

(more sips stuff)

Section 8.1.1:

    When using a GRUU as a remote target, a UA MAY add the "grid" URI

Maybe its just me, but here and elsewhere I am confused by use of the 
term "remote target". In a dialog, I think of the "remote target" as the 
R-URI I use to reference the *other* end of the dialog. I would then 
expect the term "local target" to refer to the URI I expect the other 
end to use in the R-URI of requests addressed to me. But I think you are 
using it differently.

In the above, I am left wondering if others who receive my GRUU will be 
filling in the grid parameter. Of course this is wrong, but I can read 
it that way.

If you agree this is an issue, I guess it will require a careful 
laundering of all the places that use the term "remote target".

Section 8.2.1:

First, I think it would be worthwhile to add a statement at the 
beginning of this section stating that this only applies when there are 
no Route headers. Maybe this is obvious, but maybe not. Stating this 
removes any ambiguity about how a Path header interacts with existing 
Route headers.

The first paragraph could be interpreted as an *alternative* to the 
cases mentioned in the 2nd and 3rd paragraphs. I believe it is not 
intended that way, and is instead just a modifier that applies 
regardless of which of the other cases are followed. I'm not sure that 
this paragraph even needs to be here. It places a dependency from this 
draft on -outbound behavior over and above the definition of instance 
id. The third paragraph already has language that covers the interaction 
case when both drafts are mucking with the same behavior.

Later in the section

    ... If the grid was already present
    in the contact bound to the GRUU, it is overwritten in this process.

As I noted earlier, this seems dangerous to me, so I think it should be 
prevented earlier.

Section 8.2.2:

    The proxy that accesses the location service (called the home proxy
    here) MUST record-route under two circumstances.  Firstly, if the
    home proxy receives a dialog forming request from a UA in its own
    domain (an originating request) whose Contact header field contains a
    GRUU (indicated by the presence of the "gruu" URI parameter), the
    proxy MUST record-route.

This really only applies if the proxy is responsible for the GRUU.

================

That's all!

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At 05:04 PM 7/27/2006 -0400, Rosen, Brian wrote:
>Trying, trying, ....
>
>It seems we have a very rough consensus that there will be some
>normative text that specifies the allowable URI schemes for what can be
>in the location header.  There are still some voices in opposition to
>it, but I think we've heard all the relevant points we're going to get.
>I had made a specific proposal on this:
>
>location-param = LAQUOT allowed-URI RAQUOT *( SEMI loc-param )
>;cid-url from RFC2111, http_URL from RFC2616, SIP/SIPS-URI from RFC3261,
>PRES-URI from RFC3859
>allowed-URI = cid-uri / http_URL / https_URL / SIP-URI / SIPS-URI
>               / PRES-URI
>
>As we seem to be developing another rough consensus on NOT defining an
>HTTP dereference mechanism IN THE LOCATION-CONVEYANCE DOCUMENT, this
>would now be:
>location-param = LAQUOT allowed-URI RAQUOT *( SEMI loc-param )
>;cid-url from RFC2111, SIP/SIPS-URI from RFC3261, PRES-URI from RFC3859
>allowed-URI = cid-uri / SIP-URI / SIPS-URI / PRES-URI

I think the proper cid reference is RFC 2392


>The text would say extensions to this list is by (?standards-track?) RFC
>requiring geopriv review as a "using" protocol, but is not a SIP change.
>
>A hypothetical http dereference RFC would have the following ABNF:
>;allowed-URI from <location-conveyance>, http_URL from RFC2616
>allowed-URI =/ http_URL / https_URL
>
>The alternative is to use "AbsoluteURI", and to say roughly the same
>thing about RFCs and geopriv review.  I prefer being explicit.  The "=/"
>construction is very simple and very explicit.
>
>Can we accept this "allowed-URI" proposal and move on? Does it have to
>be standards track, or is Informational okay?

A vote for Standards Track, as this ensures IETF LC for everyone's chance 
at a proposal., also so some other WG doesn't attempt to slip something by 
anyone (meaning "by Geopriv")

>I would think
>informational is okay, provided geopriv reviews.

But I don't believe we can state who the reviewers are, we're limited to 
the type of RFC (if one is required)

>For example, if some
>other organization came along and defined a protocol that worked, and
>met the geopriv requirements, an informational saying that the way you
>use this with SIP Location is blah, blah, blah, I would think, provided
>geopriv agreed it met their concerns, that would be okay (even with the
>ABNF extension to allowed-URI in the Informational)
>
>Brian
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv

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At 04:20 PM 7/27/2006 -0500, James M. Polk wrote:
>>Can we accept this "allowed-URI" proposal and move on? Does it have to
>>be standards track, or is Informational okay?
>
>A vote for Standards Track, as this ensures IETF LC for everyone's chance 
>at a proposal., also so some other WG doesn't attempt to slip something by 
>anyone (meaning "by Geopriv")

As a WG chair, I should know this off the top of my head, but I'm 
bewildered by some of the comments here (and in another area), therefore 
confused...

I believe that each subsequent schema proposed would be an official 
'update' to this SIP RFC (to be), which is Standards Track, meaning I don't 
believe an INFO RFC can officially 'update' a Standards Track RFC.  This 
would mean updating this RFC (to be) would take a Standards Track RFC, if 
my math is right...


>>I would think
>>informational is okay, provided geopriv reviews.

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From sip-bounces@ietf.org Thu Jul 27 17:47:21 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
	"Rosen, Brian" <Brian.Rosen@neustar.biz>
Subject: RE: [Sip] Re: [Geopriv] How Location Conveyance specifies the
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I don't claim there is a consensus on how the protocols are limited, only
that they are.  I don't think you are arguing that to put a URI in the
Location Header makes it a protocol needing geopriv review ("using protocol"
notwithstanding the wording problems).  The text would have to say that,
wouldn't you agree?

The net effect is the same, you can't use any URI in the header, it has to
be a protocol that has had geopriv review as a using protocol in an RFC.

I stated why I think the allowed-URI construction is better; it makes the
grammar precise and explicit.  AbsoluteURI makes the grammar imprecise and
implicit.

What is wrong with what I proposed, and what makes no ABNF restrictions
(absoluteURI) better?  Using absoluteURI means that the using protocol RFC
wouldn't have to have the one line ABNF extension to allowed-URI.  Is that a
benefit?  Is there any other difference?

Brian

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Thursday, July 27, 2006 5:10 PM
> To: Rosen, Brian
> Cc: IETF SIP List; GEOPRIV
> Subject: [Sip] Re: [Geopriv] How Location Conveyance specifies the
> allowedURI
> 
> If HTTP and other mechanisms are not defined in the draft (with which I
> agree), I see no technical reason not to allow other URI schemes in the
> draft ABNF.
> 
> At least from the voices I heard, my proposal of not restricting the URI
> scheme has received at least as much support as countervailing
> proposals, so I don't know now how you're counting rough consensus. Or
> are there supervotes?
> 
> Rosen, Brian wrote:
> > Trying, trying, ....
> >
> > It seems we have a very rough consensus that there will be some
> > normative text that specifies the allowable URI schemes for what can be
> > in the location header.  There are still some voices in opposition to
> > it, but I think we've heard all the relevant points we're going to get.
> > I had made a specific proposal on this:
> >
> > location-param = LAQUOT allowed-URI RAQUOT *( SEMI loc-param )
> > ;cid-url from RFC2111, http_URL from RFC2616, SIP/SIPS-URI from RFC3261,
> > PRES-URI from RFC3859
> > allowed-URI = cid-uri / http_URL / https_URL / SIP-URI / SIPS-URI
> >               / PRES-URI
> >
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From: "Brian Rosen" <br@brianrosen.net>
To: "'Andrew Newton'" <andy@hxr.us>,
	"'James M. Polk'" <jmpolk@cisco.com>
Date: Thu, 27 Jul 2006 14:57:12 -0400
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Subject: [Sip] RE: [Geopriv] Teasing apart: constraints on dereference URI
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It just seems so unlikely to me that geopriv would sanction any URI
dereference mechanism that didn't have an RFC it could inject its review
into.  It's got to be limited to things that are RFCs.  We can argue about
whether it has to be standards track, we can argue about if the limit is
enforced with ABNF, IANA registration or normative wording, but it's gotta
be an RFC.

Brian

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Thursday, July 27, 2006 2:42 PM
> To: James M. Polk
> Cc: IETF SIP List; GEOPRIV
> Subject: Re: [Geopriv] Teasing apart: constraints on dereference URI
> 
> James M. Polk wrote:
> >> Finally, how many people desire a middle ground?  Perhaps where the
> >> ABNF, XML Schema or other grammar specification allows for any URI
> >> scheme but there is an IANA registry listing the allowed dereference
> >> URI schemes and associated protocols?
> >
> > I think any IANA Registry here will point back to a Standards Track RFC
> > that defined the usage and semantics, which your question is not
> > allowing.  Therefore, for what you have immediately above, I disagree.
> 
> I guess that may have been too much hand waving on my part.  So let me
> ask you this question:  if the grammar were to only specify "URI" and no
> particular kind of URI, yet the kinds of URIs allowed must have to pass
> some sort of review before their use in the Location header is
> considered acceptable, would you find favor in this approach?
> 
> And if so, how would you implement it?
> 
> -andy
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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From sip-bounces@ietf.org Thu Jul 27 17:47:21 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
	"Rosen, Brian" <Brian.Rosen@neustar.biz>
Subject: RE: [Sip] Re: [Geopriv] How Location Conveyance specifies the
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Date: Thu, 27 Jul 2006 17:45:19 -0400
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I don't claim there is a consensus on how the protocols are limited, only
that they are.  I don't think you are arguing that to put a URI in the
Location Header makes it a protocol needing geopriv review ("using protocol"
notwithstanding the wording problems).  The text would have to say that,
wouldn't you agree?

The net effect is the same, you can't use any URI in the header, it has to
be a protocol that has had geopriv review as a using protocol in an RFC.

I stated why I think the allowed-URI construction is better; it makes the
grammar precise and explicit.  AbsoluteURI makes the grammar imprecise and
implicit.

What is wrong with what I proposed, and what makes no ABNF restrictions
(absoluteURI) better?  Using absoluteURI means that the using protocol RFC
wouldn't have to have the one line ABNF extension to allowed-URI.  Is that a
benefit?  Is there any other difference?

Brian

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Thursday, July 27, 2006 5:10 PM
> To: Rosen, Brian
> Cc: IETF SIP List; GEOPRIV
> Subject: [Sip] Re: [Geopriv] How Location Conveyance specifies the
> allowedURI
> 
> If HTTP and other mechanisms are not defined in the draft (with which I
> agree), I see no technical reason not to allow other URI schemes in the
> draft ABNF.
> 
> At least from the voices I heard, my proposal of not restricting the URI
> scheme has received at least as much support as countervailing
> proposals, so I don't know now how you're counting rough consensus. Or
> are there supervotes?
> 
> Rosen, Brian wrote:
> > Trying, trying, ....
> >
> > It seems we have a very rough consensus that there will be some
> > normative text that specifies the allowable URI schemes for what can be
> > in the location header.  There are still some voices in opposition to
> > it, but I think we've heard all the relevant points we're going to get.
> > I had made a specific proposal on this:
> >
> > location-param = LAQUOT allowed-URI RAQUOT *( SEMI loc-param )
> > ;cid-url from RFC2111, http_URL from RFC2616, SIP/SIPS-URI from RFC3261,
> > PRES-URI from RFC3859
> > allowed-URI = cid-uri / http_URL / https_URL / SIP-URI / SIPS-URI
> >               / PRES-URI
> >
> 
> _______________________________________________
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> Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Thu Jul 27 17:47:21 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Andrew Newton'" <andy@hxr.us>,
	"'James M. Polk'" <jmpolk@cisco.com>
Date: Thu, 27 Jul 2006 14:57:12 -0400
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Subject: [Sip] RE: [Geopriv] Teasing apart: constraints on dereference URI
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It just seems so unlikely to me that geopriv would sanction any URI
dereference mechanism that didn't have an RFC it could inject its review
into.  It's got to be limited to things that are RFCs.  We can argue about
whether it has to be standards track, we can argue about if the limit is
enforced with ABNF, IANA registration or normative wording, but it's gotta
be an RFC.

Brian

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Thursday, July 27, 2006 2:42 PM
> To: James M. Polk
> Cc: IETF SIP List; GEOPRIV
> Subject: Re: [Geopriv] Teasing apart: constraints on dereference URI
> 
> James M. Polk wrote:
> >> Finally, how many people desire a middle ground?  Perhaps where the
> >> ABNF, XML Schema or other grammar specification allows for any URI
> >> scheme but there is an IANA registry listing the allowed dereference
> >> URI schemes and associated protocols?
> >
> > I think any IANA Registry here will point back to a Standards Track RFC
> > that defined the usage and semantics, which your question is not
> > allowing.  Therefore, for what you have immediately above, I disagree.
> 
> I guess that may have been too much hand waving on my part.  So let me
> ask you this question:  if the grammar were to only specify "URI" and no
> particular kind of URI, yet the kinds of URIs allowed must have to pass
> some sort of review before their use in the Location header is
> considered acceptable, would you find favor in this approach?
> 
> And if so, how would you implement it?
> 
> -andy
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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From sip-bounces@ietf.org Thu Jul 27 17:50:39 2006
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Subject: RE: [Sip] Re: [Geopriv] How Location Conveyance specifies the allowed
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I could be wrong, but I think it is possible to define such an extension
as not updating the original document.  Nearly every SIP extension
effectively modifies one or more ABNF lines in 3261, but none of them
are considered updates to 3261.

Brian

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]
> Sent: Thursday, July 27, 2006 5:28 PM
> To: Rosen, Brian; IETF SIP List; GEOPRIV
> Subject: Re: [Sip] Re: [Geopriv] How Location Conveyance specifies the
> allowed URI
>=20
> At 04:20 PM 7/27/2006 -0500, James M. Polk wrote:
> >>Can we accept this "allowed-URI" proposal and move on? Does it have
to
> >>be standards track, or is Informational okay?
> >
> >A vote for Standards Track, as this ensures IETF LC for everyone's
chance
> >at a proposal., also so some other WG doesn't attempt to slip
something
> by
> >anyone (meaning "by Geopriv")
>=20
> As a WG chair, I should know this off the top of my head, but I'm
> bewildered by some of the comments here (and in another area),
therefore
> confused...
>=20
> I believe that each subsequent schema proposed would be an official
> 'update' to this SIP RFC (to be), which is Standards Track, meaning I
> don't
> believe an INFO RFC can officially 'update' a Standards Track RFC.
This
> would mean updating this RFC (to be) would take a Standards Track RFC,
if
> my math is right...
>=20
>=20
> >>I would think
> >>informational is okay, provided geopriv reviews.

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> >;cid-url from RFC2111, SIP/SIPS-URI from RFC3261, PRES-URI from
RFC3859
> >allowed-URI =3D cid-uri / SIP-URI / SIPS-URI / PRES-URI
>=20
> I think the proper cid reference is RFC 2392
Sorry, I'll fix it (if the idea is accepted)

..snip.
> >I would think
> >informational is okay, provided geopriv reviews.
>=20
> But I don't believe we can state who the reviewers are, we're limited
to
> the type of RFC (if one is required)
SIP Change states things like this.  I think we can say it.

BTW, I am not opposed to specifying standards track, I'm just don't
think it's absolutely necessary.


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Subject: [Sip] RE: [Geopriv] How Location Conveyance specifies the allowed
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At 05:51 PM 7/27/2006 -0400, Rosen, Brian wrote:
> > >;cid-url from RFC2111, SIP/SIPS-URI from RFC3261, PRES-URI from
>RFC3859
> > >allowed-URI = cid-uri / SIP-URI / SIPS-URI / PRES-URI
> >
> > I think the proper cid reference is RFC 2392
>Sorry, I'll fix it (if the idea is accepted)

2392 is already in -03, so there needs to be a call to remove 2392 as being 
improper 

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From sip-bounces@ietf.org Thu Jul 27 18:15:47 2006
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Rosen, Brian" <Brian.Rosen@neustar.biz>,
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Subject: Re: [Sip] Re: [Geopriv] How Location Conveyance specifies the
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Brian,

Yes, but there is one big difference with what you are proposing: extensions 
to SIP specialize existing, clearly defined extension points in the RFC3261 
ABNF. For example, "generic-param  =  token [ EQUAL gen-value ]" is often 
extended by giving a specific definition of 'token' and 'gen-value', for a 
specific header parameter

Some headers currently don't allow parameters. That sometimes limits 
possible solutions, since extensions are not allowed to add parameters to 
such headers (since it would break backwards compatibility). In other words, 
modifications to the ABNF are not arbitrary.

SIP elements are required to ignore headers that are not understood (RFC3261 
section 8.2.2). However, early implementations of location conveyance will 
read the Location header. If there is no generic "AbsoluteURI" (the 
equivalent of generic-param) as part of the ABNF for the Location header, 
then additions to this field will give parsing errors for these early 
implementations. A typical implementation will then return "400 Bad 
Request", which does not help debugging. And what if the location is not 
critical to processing the request? then there should not be an error 
response, but the parser cannot make that distinction

Regards,

Jeroen

----- Original Message ----- 
From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "James M. Polk" <jmpolk@cisco.com>; "IETF SIP List" <sip@ietf.org>; 
"GEOPRIV" <geopriv@ietf.org>
Sent: Thursday, July 27, 2006 11:48 PM
Subject: RE: [Sip] Re: [Geopriv] How Location Conveyance specifies the 
allowedURI


I could be wrong, but I think it is possible to define such an extension
as not updating the original document.  Nearly every SIP extension
effectively modifies one or more ABNF lines in 3261, but none of them
are considered updates to 3261.

Brian

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]
> Sent: Thursday, July 27, 2006 5:28 PM
> To: Rosen, Brian; IETF SIP List; GEOPRIV
> Subject: Re: [Sip] Re: [Geopriv] How Location Conveyance specifies the
> allowed URI
>
> At 04:20 PM 7/27/2006 -0500, James M. Polk wrote:
> >>Can we accept this "allowed-URI" proposal and move on? Does it have
to
> >>be standards track, or is Informational okay?
> >
> >A vote for Standards Track, as this ensures IETF LC for everyone's
chance
> >at a proposal., also so some other WG doesn't attempt to slip
something
> by
> >anyone (meaning "by Geopriv")
>
> As a WG chair, I should know this off the top of my head, but I'm
> bewildered by some of the comments here (and in another area),
therefore
> confused...
>
> I believe that each subsequent schema proposed would be an official
> 'update' to this SIP RFC (to be), which is Standards Track, meaning I
> don't
> believe an INFO RFC can officially 'update' a Standards Track RFC.
This
> would mean updating this RFC (to be) would take a Standards Track RFC,
if
> my math is right...
>
>
> >>I would think
> >>informational is okay, provided geopriv reviews.

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From sip-bounces@ietf.org Thu Jul 27 18:30:06 2006
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Jeroen van Bemmel wrote:
> Because CANCEL and ACK can loop like any other request. The concern 
> mostly applies to *stateless* proxies

Ah, ok; I forgot about stateless proxies.  If you want to keep
stateless proxies simple, then just don't worry about loop
detection there.  The loop will be detected in the next stateful
proxy anyway.

> (stateful proxies mostly dont "forward" but "generate" CANCEL and ACK
> to non-2xx), but it matters also to a stateful proxy that receives a
> CANCEL with no matching INVITE ST (RFC3261 says "MUST forward
> statelessly").

Yes, I see where it says that, but my interpretation of the
phrase "MUST forward statelessly" is colored by what it says
next (see end of Section 16.10).  RFC3261 says that CANCELs MUST
be forwarded statelessly because the proxy may have statelessly
proxied the preceeding INVITE.

Assume you are a stateful proxy and never proxy an INVITE
statelessly.  When you get a CANCEL, you know for a fact that
you handle all INVITEs statefully so forwarding the CANCEL
statelessly causes needless network traffic.

Case 1: Consider the case where a stateful proxy receives a
CANCEL after just having proxied a 2xx response.  The 2xx will
put the ST in Terminated state at the proxy, whereupon it may
delete the ST, or keep it in memory for a few seconds longer.
But regardless, the proxy knows that it never sends any
INVITE transactions statelessly, so proxying a CANCEL is
pointless.

Case 2: A proxy receives a CANCEL request out of the blue.
Since it knows that it always sends INVITE statefully, proxying
the CANCEL is pointless.  Better to 481 it right there.

Ultimately, it is an implementation decision; if you ever
proxied a INV statelessly, then you must do the same for a
CANCEL.  Otherwise, statelessly proxying a CANCEL just adds
to network traffic.

Thanks.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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Michael Hammer (mhammer) wrote:
> Then we will need an RFC to point to the index RFCs that point to the
> base RFCs?

Nah, it's not as bad as that.  We are talking about 2 RFCs at
most, really.

> A single RFC pointing document would be best.

If we go the single RFC route, then that means obsoleting that
RFC at least once, or maybe twice.  Now, while you and I are
adept at making our way through how IETF numbers their RFCs
and what happens when one RFC is obsoleted, and so on ... I
think the majority of the people using our specifications may
not have this context.  They grab an RFC -- obsolete or not --
and run with it.

To take this to the absurd, I am aware of products based on
*expired* I-Ds, not to mention obsolete RFCs!

In summary, my preference is for (1), with two RFCs produced
at most.  But, if the WG decides to go with (2), then that
is fine...

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)


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From sip-bounces@ietf.org Thu Jul 27 20:52:41 2006
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From: "Dawson, Martin" <Martin.Dawson@andrew.com>
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Hi Andy,

I'll take a stab at it. Not that HTTP is a valid *using protocol* but,
rather, that the location reference does not actually identify a using
protocol and, in fact, HTTP would only be the transport protocol...

3693 states:
   The "using protocol" is the protocol that uses (reads or modifies)
   the Location Object.  A protocol that just transports the LO as a
   string of bits, without looking at them (like an IP storage protocol
   could do), is not a using protocol, but only a transport protocol.
   Nevertheless, the entity or protocol that caused the transport
   protocol to move the LO is responsible for the appropriate
   distribution, protection, usage, retention, and storage of the LO
   based on the rules that apply to that LO.

A location reference which provided an HTTP URI indicates that the
entity that is going to be used to request the LO is to be communicated
with using HTTP. Because of the rules associated with the PIDF-LO, it
can be inferred that the entity and whatever semantics are associated
with the HTTP communications with that entity will honor the
requirements. That is, HTTP is only the transport protocol. An HTTP GET
has to be interpreted by the receiving entity within the scope of a
using protocol - with its associated set of rules and semantics - which
go beyond HTTP and which do respect the geopriv requirements. Indeed,
this is generally the case - sending an HTTP GET to an arbitrary IP node
can yield any number of results depending what that node actually does.
The client needs to understand the purpose of that server in the scope
of some higher level protocol.

To use another example, a location reference could be an IP address.
There's nothing inherent in IP that honors the geopriv rules, but that
doesn't stop the reference being valid nor say that the protocol used
between a client and that IP address would not honor the rules.

The purpose of the SIP conveyance specification is to say how a LO or
reference can be transferred from one entity to another using SIP. It's
out of scope for a SIP specification to mandate anything about the
transport protocol or to infer anything about using protocol implied by
that transport. As has been previously mentioned, this would be subject
to separate specification and BCP - lest the SIP conveyance
specification become overly rigid and subject to unnecessary maintenance
and churn.

Cheers,
Martin

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Thursday, 27 July 2006 10:45 PM
> To: IETF SIP List; GEOPRIV
> Subject: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>=20
> There have been a number of opinions regarding HTTP as a using
protocol.
>=20
> I'd like to know, how many people believe that HTTP requires no or
> very little specification to meet the requirements in RFC 3693 and
> RFC 3694.  If you believe this, what basis do you have for this
> conclusion?  Or do you believe more specification needs to be done,
> but it is possible for HTTP to be a GEOPRIV using protocol?
>=20
> Finally, how many people believe that HTTP should not be a using
> protocol?
>=20
> -andy
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

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From sip-bounces@ietf.org Thu Jul 27 21:39:04 2006
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	"Brian Rosen" <br@brianrosen.net>
From: Randall Gellens <randy@qualcomm.com>
Subject: [Geopriv] Re: [Sip] teasing apart: http as a GEOPRIV using
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At 9:33 AM -0400 7/27/06, Henning Schulzrinne wrote:

>  For all protocols, the hardest part is specifying how 
> authentication works in conjunction with the privacy rules. This is 
> far from obvious for HTTP, for example. (Is it the Digest user 
> identifier? Can Basic over TLS be used instead? What about 
> crypto-random request URIs? Is HTTP ok in that case or is HTTPS 
> needed - the answer isn't obvious since an observer would only see 
> a location, not a location-identity pair.)
>
>  For SIP, this is largely done, except for the crypto-random request 
> URI case and variations, such as user + password.

A while back there was some brief discussion in geopriv about using 
"pawn ticket" URIs for location information.  That is, URLAUTH from 
RFC 4467, which is a URL that has been signed by the server and can 
be used by the holder to gain access to the information, subject to 
expiration time and/or role restrictions.  Role restrictions limit 
the URL to being used by an entity that is known to the location 
server as performing a specified role.  Additionally, RFC 4467 allows 
all currently-issued URLs to be revoked by changing the key.

I wonder if there is any value in this?
-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
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	Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: Re: [Geopriv] Re: [Sip] teasing apart: http as a GEOPRIV
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At 5:36 PM +0200 7/27/06, Hannes Tschofenig wrote:

>  here is the reason why I think we are not talking about a 'using 
> protocol' here: All we want is to perform dereferencing.
>
>  We don't want something that does all the fancy features of a 
> presence solution.

Perhaps it is a question of who is doing the dereferencing: if the 
location URI is given to the target, that seems to me to be 
provisioning, the same as the DHCP case; if instead the URI is given 
to anyone else, that looks like location disclosure (or at least the 
first half of it).  In the disclosure case, dereferencing the URI is, 
I think, a using protocol.
-- 
Randall Gellens
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Hi Martin,
	Having not seen your note, I already took a stab
it, along much the same lines you suggest.  Clearly, the attached
-00 (not yet in the drafts directories) is the barest of bare bones,
but I thought it might be useful to sketch that out as a contrast
to some of the real applications which might use HTTP as a
using protocol.
	I believe this also differs from RELO in that it is
focused on the transport of PIDF-LO, which is slightly different
from RELO's remit.  But clearly, there are strong connections there.
	And if the phrase "strawman" in the title doesn't
signal that I don't think this is a done piece of work, allow
this too.
			regards,
				Ted Hardie


At 7:50 PM -0500 7/27/06, Dawson, Martin wrote:
>Hi Andy,
>
>I'll take a stab at it. Not that HTTP is a valid *using protocol* but,
>rather, that the location reference does not actually identify a using
>protocol and, in fact, HTTP would only be the transport protocol...
>
>3693 states:
>   The "using protocol" is the protocol that uses (reads or modifies)
>   the Location Object.  A protocol that just transports the LO as a
>   string of bits, without looking at them (like an IP storage protocol
>   could do), is not a using protocol, but only a transport protocol.
>   Nevertheless, the entity or protocol that caused the transport
>   protocol to move the LO is responsible for the appropriate
>   distribution, protection, usage, retention, and storage of the LO
>   based on the rules that apply to that LO.
>
>A location reference which provided an HTTP URI indicates that the
>entity that is going to be used to request the LO is to be communicated
>with using HTTP. Because of the rules associated with the PIDF-LO, it
>can be inferred that the entity and whatever semantics are associated
>with the HTTP communications with that entity will honor the
>requirements. That is, HTTP is only the transport protocol. An HTTP GET
>has to be interpreted by the receiving entity within the scope of a
>using protocol - with its associated set of rules and semantics - which
>go beyond HTTP and which do respect the geopriv requirements. Indeed,
>this is generally the case - sending an HTTP GET to an arbitrary IP node
>can yield any number of results depending what that node actually does.
>The client needs to understand the purpose of that server in the scope
>of some higher level protocol.
>
>To use another example, a location reference could be an IP address.
>There's nothing inherent in IP that honors the geopriv rules, but that
>doesn't stop the reference being valid nor say that the protocol used
>between a client and that IP address would not honor the rules.
>
>The purpose of the SIP conveyance specification is to say how a LO or
>reference can be transferred from one entity to another using SIP. It's
>out of scope for a SIP specification to mandate anything about the
>transport protocol or to infer anything about using protocol implied by
>that transport. As has been previously mentioned, this would be subject
>to separate specification and BCP - lest the SIP conveyance
>specification become overly rigid and subject to unnecessary maintenance
>and churn.
>
>Cheers,
>Martin
>
>> -----Original Message-----
>> From: Andrew Newton [mailto:andy@hxr.us]
>> Sent: Thursday, 27 July 2006 10:45 PM
>> To: IETF SIP List; GEOPRIV
>> Subject: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>>
>> There have been a number of opinions regarding HTTP as a using
>protocol.
>>
>> I'd like to know, how many people believe that HTTP requires no or
>> very little specification to meet the requirements in RFC 3693 and
>> RFC 3694.  If you believe this, what basis do you have for this
>> conclusion?  Or do you believe more specification needs to be done,
>> but it is possible for HTTP to be a GEOPRIV using protocol?
> >
>> Finally, how many people believe that HTTP should not be a using
>> protocol?
>>
>> -andy
>>
>> _______________________________________________
>> Geopriv mailing list
>> Geopriv@ietf.org
>> https://www1.ietf.org/mailman/listinfo/geopriv
>
>------------------------------------------------------------------------------------------------
>This message is for the designated recipient only and may
>contain privileged, proprietary, or otherwise private information. 
>If you have received it in error, please notify the sender
>immediately and delete the original.  Any unauthorized use of
>this email is prohibited.
>------------------------------------------------------------------------------------------------
>[mf2]
>
>_______________________________________________
>Geopriv mailing list
>Geopriv@ietf.org
>https://www1.ietf.org/mailman/listinfo/geopriv

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--============_-1058070887==_============--




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From: "Bridget Grady" <BridgetGrady@mail.ru>
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To: "Drage, Keith \(Keith\)" <drage@lucent.com>
Subject: Re: [Sip] Hitchhikers guide - when is it done?
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I think once a year is reasonable. Hopefully this document itself is not 
controversial and there would be a relatively quick path through WGLC, 
review and IESG approval.

-Jonathan R.

Drage, Keith (Keith) wrote:

> Lets indicate an expectation of once every 12 months. Within that period
> there ought to be at least 10/20 references to either add or change.
> 
> Regards
> 
> Keith 
> 
> 
>>-----Original Message-----
>>From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com] 
>>Sent: 27 July 2006 19:05
>>To: Jonathan Rosenberg
>>Cc: IETF SIP List
>>Subject: RE: [Sip] Hitchhikers guide - when is it done?
>>
>>I am aware about this. On the other hand, the problem with 
>>RFCs is that they tend to be updated at a slow pace. How 
>>often will the Hitchhiker Guide be updated? - every two, 
>>three years? I am looking at a semi-official solution to be 
>>used in-between, maintained at a known well place by a 
>>registrar nominated by the SIP WG or RAI Area. 
>>
>> Dan
>>
>> 
>>
>>
>>>-----Original Message-----
>>>From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
>>>Sent: Wednesday, July 26, 2006 6:33 PM
>>>To: Romascanu, Dan (Dan)
>>>Cc: IETF SIP List
>>>Subject: Re: [Sip] Hitchhikers guide - when is it done?
>>>
>>>The problem with any kind of website reference is that they 
>>
>>are never 
>>
>>>'official'. There are countless web sites enumerating SIP related 
>>>documents. What makes something an official output of the 
>>
>>IETF is that 
>>
>>>its an RFC.
>>>
>>>-Jonathan R.
>>>
>>>Romascanu, Dan (Dan) wrote:
>>>
>>>
>>>>Because of the overhead and clerical delays involved in the 
>>>>publication of RFCs I would suggest a refinement to option
>>>
>>>2. Provide
>>>
>>>>a reference in the RFC to a Web page maintained and
>>>
>>>refreshed by the
>>>
>>>>SIP WG or the RAI Area including the 
>>
>>'to-the-best-of-our-knowledge' 
>>
>>>>updates in-between two versions of the RFC.
>>>>
>>>>Dan
>>>>
>>>>
>>>>
>>>>
>>>> 
>>>> 
>>>>
>>>>
>>>>
>>>>>-----Original Message-----
>>>>>From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
>>>>>Sent: Tuesday, July 25, 2006 8:56 PM
>>>>>To: IETF SIP List
>>>>>Subject: [Sip] Hitchhikers guide - when is it done?
>>>>>
>>>>>During the IETF meeting, I discussed the one current open
>>>
>>>issue in the
>>>
>>>>>Hitchhikers guide. The issue is: when does the document
>>>
>>>finish? When
>>>
>>>>>do we send to IESG to issue an RFC?
>>>>>
>>>>>There were two approaches:
>>>>>
>>>>>1. split the document in two. One part contains only the
>>>
>>>core specs,
>>>
>>>>>and issues relatively soon. The other part has everything 
>>
>>else, and 
>>
>>>>>basically ends up being the last RFC produced by the SIP WG
>>>
>>>before it
>>>
>>>>>shuts its doors (which has always been 1.5 years from now
>>>
>>>for the last
>>>
>>>>>5 years).
>>>>>
>>>>>2. keep it as one document. Issue an RFC at some point, and then 
>>>>>revise it a few times as needed. We need to decide when 
>>
>>to issue it.
>>
>>>>>
>>>>>I'd like to propose option 2. I think people get more 
>>
>>value from it 
>>
>>>>>as a consolidated document. Furthermore, I'd propose that we send 
>>>>>the document to IESG for consideration as informational at the 
>>>>>instant all of the documents listed as 'core' have RFC numbers 
>>>>>issued by RFC-ed.
>>>>>
>>>>>Comments?
>>>>>
>>>>>-Jonathan R.
>>>>>-- 
>>>>>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>>>>>Cisco Fellow                                   Parsippany, NJ 
>>>>>07054-2711
>>>>>Cisco Systems
>>>>>jdrosen@cisco.com                              FAX:   
>>
>>(973) 952-5050
>>
>>>>>http://www.jdrosen.net                         PHONE: 
>>
>>(973) 952-5000
>>
>>>>>http://www.cisco.com
>>>>>
>>>>>_______________________________________________
>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>This list is for NEW development of the core SIP Protocol Use 
>>>>>sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>>>sipping@ietf.org for new developments on the application of sip
>>>>>
>>>>
>>>>
>>>-- 
>>>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>>>Cisco Fellow                                   Parsippany, NJ 
>>>07054-2711
>>>Cisco Systems
>>>jdrosen@cisco.com                              FAX:   (973) 952-5050
>>>http://www.jdrosen.net                         PHONE: (973) 952-5000
>>>http://www.cisco.com
>>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol Use 
>>sip-implementors@cs.columbia.edu for questions on current sip 
>>Use sipping@ietf.org for new developments on the application of sip
>>
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



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Oops - sorry to change my mind on this. I vaguely recall discussing this 
issue previously, but didn't remember that we concluded anything. There 
are some actual arguments in the threads and links I found arguing why 
URIs is better than URI (namely, its clear whether its plural or 
singular), that apply to GRUU too. So, I do think we should go with 
GRUUs and I've already made this edit in my working version of -10.

-Jonathan R.

Paul Kyzivat wrote:

> 
> 
> Jonathan Rosenberg wrote:
> 
>> Dale.Worley@comcast.net wrote:
> 
> 
>>> There seems to be disagreement as to what the plural of GRUU is --
>>> GRUU or GRUUs.  E.g., section 13, page 31 uses both "Furthermore,
>>> GRUUs do not address" and "These GRUU will be passed".  There is a
>>> similar inconsistency regarding the plural of AOR.
>>
>>
>> Well, this topic has come up for the term URI, which is the last U in 
>> GRUU. So it would stand to reason that if the plural of URI is URIs, 
>> then the plural of GRUU is GRUUs. I found some email threads on this 
>> topic, for example:
>>
>> http://lists.w3.org/Archives/Public/www-tag/2002Aug/0124.html
>>
>> and apparently this was an open issue in the preparation of the 
>> revision to 2396:
>>
>> http://www.gbiv.com/protocols/uri/rev-2002/issues.html#023-URI-plural
>>
>> the conclusion appears to be that the plural of URI is URIs. So, I 
>> suggest we follow suit and have the plural of GRUU be GRUUs.
> 
> 
> Hey, that's not fair!
> 
> I was looking for a precedent when cleaning up 
> draft-ietf-sipping-gruu-reg-event-06, and noticed there was 
> inconsistency in draft-ietf-sip-gruu. We discussed this, and I thought 
> we decided that the plural is GRUU. I made my draft consistent with that.
> 
> If you decide this way then I should respin 
> draft-ietf-sipping-gruu-reg-event.
> 
>     Paul
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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_______________________________________________
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From sip-bounces@ietf.org Fri Jul 28 01:26:00 2006
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From: "Dawson, Martin" <Martin.Dawson@andrew.com>
To: "Ted Hardie" <hardie@qualcomm.com>, "Andrew Newton" <andy@hxr.us>,
	"IETF SIP List" <sip@ietf.org>, "GEOPRIV" <geopriv@ietf.org>
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Thread-Topic: [Geopriv] teasing apart: http as a GEOPRIV using protocol
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Subject: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV using protocol
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G'day Ted,

Strawman or not, I think what you've written is a good example of why
the location reference does not need to say anything about whether the
geopriv rules for a PIF-LO using protocol are honored or not. The point
being that once the reference is used to establish a "using protocol
session" it must be the case that these rules are then honored.

The authentication and security steps that you have identified in your
draft - where https is used first; and it could be any equivalent
function - are an integral part of the semantics of the using protocol
defined in this example. And they are independent of, and not explicit
in, the reference itself.

Thanks for that.

Cheers,
Martin

> -----Original Message-----
> From: Ted Hardie [mailto:hardie@qualcomm.com]
> Sent: Friday, 28 July 2006 11:53 AM
> To: Dawson, Martin; Andrew Newton; IETF SIP List; GEOPRIV
> Subject: RE: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>=20
> Hi Martin,
> 	Having not seen your note, I already took a stab
> it, along much the same lines you suggest.  Clearly, the attached
> -00 (not yet in the drafts directories) is the barest of bare bones,
> but I thought it might be useful to sketch that out as a contrast
> to some of the real applications which might use HTTP as a
> using protocol.
> 	I believe this also differs from RELO in that it is
> focused on the transport of PIDF-LO, which is slightly different
> from RELO's remit.  But clearly, there are strong connections there.
> 	And if the phrase "strawman" in the title doesn't
> signal that I don't think this is a done piece of work, allow
> this too.
> 			regards,
> 				Ted Hardie
>=20
>=20
> At 7:50 PM -0500 7/27/06, Dawson, Martin wrote:
> >Hi Andy,
> >
> >I'll take a stab at it. Not that HTTP is a valid *using protocol*
but,
> >rather, that the location reference does not actually identify a
using
> >protocol and, in fact, HTTP would only be the transport protocol...
> >
> >3693 states:
> >   The "using protocol" is the protocol that uses (reads or modifies)
> >   the Location Object.  A protocol that just transports the LO as a
> >   string of bits, without looking at them (like an IP storage
protocol
> >   could do), is not a using protocol, but only a transport protocol.
> >   Nevertheless, the entity or protocol that caused the transport
> >   protocol to move the LO is responsible for the appropriate
> >   distribution, protection, usage, retention, and storage of the LO
> >   based on the rules that apply to that LO.
> >
> >A location reference which provided an HTTP URI indicates that the
> >entity that is going to be used to request the LO is to be
communicated
> >with using HTTP. Because of the rules associated with the PIDF-LO, it
> >can be inferred that the entity and whatever semantics are associated
> >with the HTTP communications with that entity will honor the
> >requirements. That is, HTTP is only the transport protocol. An HTTP
GET
> >has to be interpreted by the receiving entity within the scope of a
> >using protocol - with its associated set of rules and semantics -
which
> >go beyond HTTP and which do respect the geopriv requirements. Indeed,
> >this is generally the case - sending an HTTP GET to an arbitrary IP
node
> >can yield any number of results depending what that node actually
does.
> >The client needs to understand the purpose of that server in the
scope
> >of some higher level protocol.
> >
> >To use another example, a location reference could be an IP address.
> >There's nothing inherent in IP that honors the geopriv rules, but
that
> >doesn't stop the reference being valid nor say that the protocol used
> >between a client and that IP address would not honor the rules.
> >
> >The purpose of the SIP conveyance specification is to say how a LO or
> >reference can be transferred from one entity to another using SIP.
It's
> >out of scope for a SIP specification to mandate anything about the
> >transport protocol or to infer anything about using protocol implied
by
> >that transport. As has been previously mentioned, this would be
subject
> >to separate specification and BCP - lest the SIP conveyance
> >specification become overly rigid and subject to unnecessary
maintenance
> >and churn.
> >
> >Cheers,
> >Martin
> >
> >> -----Original Message-----
> >> From: Andrew Newton [mailto:andy@hxr.us]
> >> Sent: Thursday, 27 July 2006 10:45 PM
> >> To: IETF SIP List; GEOPRIV
> >> Subject: [Geopriv] teasing apart: http as a GEOPRIV using protocol
> >>
> >> There have been a number of opinions regarding HTTP as a using
> >protocol.
> >>
> >> I'd like to know, how many people believe that HTTP requires no or
> >> very little specification to meet the requirements in RFC 3693 and
> >> RFC 3694.  If you believe this, what basis do you have for this
> >> conclusion?  Or do you believe more specification needs to be done,
> >> but it is possible for HTTP to be a GEOPRIV using protocol?
> > >
> >> Finally, how many people believe that HTTP should not be a using
> >> protocol?
> >>
> >> -andy
> >>
> >> _______________________________________________
> >> Geopriv mailing list
> >> Geopriv@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/geopriv
> >
>
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Subject: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV using protocol
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At 11:14 PM -0500 7/27/06, Thomson, Martin wrote:
>
>Or, are we above the scrabbling for loopholes - after all we[1] wrote the rules, we know the reasons for the rules[2], so why are we playing games with definitions?  This is avoiding RFC 3693, and I'm guilty of that[3] (if I recall, you told me to stop).  Why not simply address the requirements?
>
>The point that Martin (D) was making is related to the fact that conveyance doesn't have to specify how a reference is used.  That doesn't mean that it doesn't have to be specified at all, but SIP can treat this as opaque data (except cid:, of course). 
>
>Cheers,
>Martin

Going through 3693 again was what inspired me to put up a strawman.  I hope
setting it alight gets us some light and not too much heat.

My own reading says that if you want to use HTTP as a transport of PIDF-LO, rather
than as a using protocol, you have to use it in a way that protects the privacy
of the object in flight (encrypting TLS in the strawman, but it could be mandating
s/mime or something else), and you have to be sure that the parties transporting
it are authenticated to each other (to protect against monkey-in-the-middle attacks
as a start).

More importantly, you have to use it in a protocol context where it is handed off
after transport to some other protocol or application which then does the real
work of reading or modifying the object and of obeying the rules.  To me that
means that it can't be used in this way loosely--just implementing this strawman
and calling it a using protocol would be utterly bogus, since once it arrives
at the client, you haven't said what to do with it all or how the geopriv
requirements are met.  But I think it *might* be valid in very specific protocol
contexts where you have said what happens when the *using protocol*
(SIP, in this example) does from there. 

It may be, of course, that I was thinking to hard and need to take a bit of
a lie down to recover.  But I thought about the URI schemes which encode
data in the URI itself, and I thought:  how can those be using protocols?
Obviously, they aren't; they're just ways of carrying the data around for
the benefit of real using protocols.  If that is the case, we may be able
to do something simple to specify how to use a suitably protected transport
to dereference a URI rather than carrying the data inside the using protocol.
But I think that this fundamentally requires that the transport/dereference
occur in a very bounded protocol context inside the real using protocol.
Otherwise, you are absolutely right that it is just an end-run.

To re-iterate:  this a strawman; feel free to rip up the innards and use as
a pillow, flammable materials, or use as you see fit.  I won't be offended if
the response is any of the above.
			regards,
				Ted











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inline.

Paul Kyzivat wrote:

> 
> 
> Jonathan Rosenberg wrote:
> 
>> Cullen Jennings wrote:
> 
> 
>>> Section 6, page 12, 4th para or so ... you are talking about mid- 
>>> dialog requests. Might be nice to mention how a transaction stateful  
>>> proxy knows if something is a middialog request or not.
>>
>>
>> These days I recommend topmost route URI over tags, since its easy to 
>> fool a proxy into thinking something is a mid-dialog request by faking 
>> a tag. Its harder to fake a route URI.
>>
>> Text now reads:
>>
>> <t> Mid-dialog requests will also be sent to GRUUs, as they are
>> included as the remote-target in dialog-forming and target refresh
>> requests and responses. However, in those cases, a proxy SHOULD only
>> apply services that are meaningful for mid-dialog requests, generally
>> speaking. This excludes screening functions, as well as forwarding
>> ones. A proxy can determine that a request is a mid-dialog request
>> based on the Route header field in the request it receives. If the
>> topmost URI matches one that the proxy placed into the Record-Route
>> header field of a dialog-forming request, then the request is a
>> mid-dialog request. </t>
> 
> 
> Is it really harder to fake?

If the objective is to fool a *proxy* into thinking a request is a 
mid-dialog request, then yes.

If a proxy uses the mere presence of the To tag as the indicator of a 
mid-dialog request, then a UA can simply send an initial request with 
any arbitrary tag in the To header field.

If a proxy uses the value of Route header field, then the UA has to 
construct a Route header field value which the proxy will accept. If the 
proxy is halfway intelligent in construction of record-routes, it'll use 
random numbers, hashes, and otherwise proprietary techniques to 
construct the URI. The odds are pretty good that the UA would not be 
able to construct a Route URI that the proxy would recognize as one that 
it would have inserted into a Record-Route.

Now, a request that contains a tag not matching any dialog at the UAS is 
rejected anyway. However, you might be able to work with a cooperating 
UAS to make something useful happen. For example if a proxy is 
generating call start and stop records based on *initial* INVITE and 
BYE, respectively, cooperating UA might be able to get free calls.

> 
> Suppose Alice is permitted to call Bob, but not to subscribe to the reg 
> event package. So Alice first calls Bob, and notes the Record-Route that 
> results. Then in a separate dialog, Alice sends a SUBSCRIBE to the reg 
> event package, including a Route header with a URI of Bob's home proxy 
> extracted from the old R-R.

If the RR is constructed using hashes of the tags this won't work.

> 
> Seems like this strategy also requires some protection against a "replay 
> attack" like this.
> 
> But that isn't really an issue for this draft.

No; but it makes for interesting list discussion ;)

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Fri Jul 28 03:38:05 2006
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On ke, 2006-07-19 at 17:46 -0500, ext Vijay K. Gurbani wrote:
> This remains a problem; I have had a quick chat with Scott
> Lawrence on it and we'll be submitting a draft latter part
> of next week on one way to deal with this.

I'm curious, can that draft really suggest anything other than simply
"MUST NOT use UDP"?

Cheers,
Aki

-- 
Aki Niemi
Nokia Research Center


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Hi,

   I have been earlier proposing to deprecate the SIP over UDP and there
have been arguments about the large SDP etc.

   It will be useful for Page Mode IM, where we will be sending short
IM. Hence we might need DTLS too. Should we still deprecate it..

  Rajnish was arguing that one can cut and paste long text in the body
too. If it is so, then client can use TCP or TLS based on the size.


Thx
Samir


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From sip-bounces@ietf.org Fri Jul 28 05:15:22 2006
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At 05:15 28/07/2006, Jonathan Rosenberg wrote:
> If the proxy is halfway intelligent in construction of record-routes, it'll use random numbers, hashes, and otherwise proprietary 
> techniques to construct the URI. The odds are pretty good that the UA would not be able to construct a Route URI that the proxy 
> would recognize as one that it would have inserted into a Record-Route.

Well, in terms of interesting list discussion: I think this can be actually used to replace the
hostname-based "is it myself" test in processing route header fields. That appears to me more
robust because it removes ambiguities around multiple IP addresses, many names in hosted
environments, having to configure all of that, failover scenarios with backup having 
different IP from primary servers, etc.

>Now, a request that contains a tag not matching any dialog at the UAS is rejected anyway. However, you might be able to work with a cooperating UAS to make something useful happen. For example if a proxy is generating call start and stop records based on *initial* INVITE and BYE, respectively, cooperating UA might be able to get free calls.
>
>>Suppose Alice is permitted to call Bob, but not to subscribe to the reg event package. So Alice first calls Bob, and notes the Record-Route that results. Then in a separate dialog, Alice sends a SUBSCRIBE to the reg event package, including a Route header with a URI of Bob's home proxy extracted from the old R-R.
>
>If the RR is constructed using hashes of the tags this won't work.
>
>>Seems like this strategy also requires some protection against a "replay attack" like this.
>>But that isn't really an issue for this draft.
>
>No; but it makes for interesting list discussion ;)

:)

Replay attack protection is not easy for protocols which rely on notion of piggy-backing
without any more support from client than just mirroring a value from server. We would
have to touch the client to provide for message integrity. 

-jiri


--
Jiri Kuthan            http://iptel.org/~jiri/ 


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Subject: Re: [Sip] draft-niemi-sip-subnot-etags-00.txt
From: Aki Niemi <aki.niemi@nokia.com>
To: "ext Drage, Keith (Keith)" <drage@lucent.com>
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I care. ;)

Anyway, I have updated the draft with the comments received so far. I
think it's starting to shape up, so please take a look:

http://www.ietf.org/internet-drafts/draft-niemi-sip-subnot-etags-01.txt

Cheers,
Aki

On su, 2006-03-26 at 17:24 +0100, ext Drage, Keith (Keith) wrote:
> (As chair)
> 
> http://www.ietf.org/internet-drafts/draft-niemi-sip-subnot-etags-00.txt makes a proposal for:
> 
> "   The Session Initiation Protocol (SIP) events framework enables
>    receiving asynchronous notification of events related to SIP
> systems.
>    This framework defines the procedures for creating, refreshing and
>    terminating subscriptions, as well as fetching and periodic polling
>    of resource state.  These procedures have a serious deficiency in
>    that they do not allow state to persist over a subscription
> refresh,
>    or between two consecutive polls.  Another related but different
>    problem relates to the relative intolerance of the framework to
>    interferences in networking connectivity of subscribers in long-
>    lasting subscriptions.  This document explains the problems in more
>    detail and discusses possible solutions."
> 
> This draft was discussed at the face-to-face meeting at IETF #65 but
> without conclusion.
> 
> We seek views on whether the SIP WG should adopt a draft in this area
> as a charter item, and if so, what scope that charter item should
> have.
> 
> regards
> 
> Keith
> 
> Keith Drage
> drage@lucent.com
> 
> _______________________________________________
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Subject: RE: [Sip] Hitchhikers guide - when is it done?
Date: Fri, 28 Jul 2006 13:47:05 -0400
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Thread-Topic: [Sip] Hitchhikers guide - when is it done?
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Vijay K. Gurbani" <vkg@lucent.com>
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I am anticipating that this needs to be maintained for quite some time
and that each year something new would be needed.  So, I don't believe
the "2 at most" IMHO.

Cheers,
Mike
=20

> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]=20
> Sent: Thursday, July 27, 2006 6:36 PM
> To: Michael Hammer (mhammer)
> Cc: Jonathan Rosenberg (jdrosen); IETF SIP List
> Subject: Re: [Sip] Hitchhikers guide - when is it done?
>=20
> Michael Hammer (mhammer) wrote:
> > Then we will need an RFC to point to the index RFCs that=20
> point to the=20
> > base RFCs?
>=20
> Nah, it's not as bad as that.  We are talking about 2 RFCs at=20
> most, really.
>=20
> > A single RFC pointing document would be best.
>=20
> If we go the single RFC route, then that means obsoleting=20
> that RFC at least once, or maybe twice.  Now, while you and I=20
> are adept at making our way through how IETF numbers their=20
> RFCs and what happens when one RFC is obsoleted, and so on=20
> ... I think the majority of the people using our=20
> specifications may not have this context.  They grab an RFC=20
> -- obsolete or not -- and run with it.
>=20
> To take this to the absurd, I am aware of products based on
> *expired* I-Ds, not to mention obsolete RFCs!
>=20
> In summary, my preference is for (1), with two RFCs produced=20
> at most.  But, if the WG decides to go with (2), then that is fine...
>=20
> Thanks,
>=20
> - vijay
> --
> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> Bell Laboratories, Lucent Technologies, Inc.
> 2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
>=20
>=20

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To: Ted Hardie <hardie@qualcomm.com>,
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	"Andrew Newton" <andy@hxr.us>, "IETF SIP List" <sip@ietf.org>,
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This does look to me to be a "using protocol".

Presumably the HTTP server and the dereferencer have a relationship 
such that HTTP DIGEST can be used.

Perhaps the DIGEST identity can be used when evaluating the policy rules.
-- 
Randall Gellens
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From sip-bounces@ietf.org Fri Jul 28 18:03:06 2006
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Date: Sat, 29 Jul 2006 00:01:26 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV using
	protocol
To: Ted Hardie <hardie@qualcomm.com>,
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Hi Ted,

At what scale do you expect this to work? If the location server is require=
d to have a trust relation with everyone I might want to share my location =
with, it won't work for the Internet.=20

For that, I think something like trait based authorization is needed: a URL=
 that encodes my consent for you to retrieve my location, for a limited per=
iod

Regards,
Jeroen

-Oorspronkelijk bericht -----
Van: "Ted Hardie" <hardie@qualcomm.com>
Aan: "Dawson, Martin" <Martin.Dawson@andrew.com>; "Andrew Newton" <andy@hxr=
us>; "IETF SIP List" <sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
Verzonden: 28-7-06 03:53
Onderwerp: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV using proto=
col

Hi Martin,
	Having not seen your note, I already took a stab
it, along much the same lines you suggest.  Clearly, the attached
-00 (not yet in the drafts directories) is the barest of bare bones,
but I thought it might be useful to sketch that out as a contrast
to some of the real applications which might use HTTP as a
using protocol.
	I believe this also differs from RELO in that it is
focused on the transport of PIDF-LO, which is slightly different
from RELO's remit.  But clearly, there are strong connections there.
	And if the phrase "strawman" in the title doesn't
signal that I don't think this is a done piece of work, allow
this too.
			regards,
				Ted Hardie


At 7:50 PM -0500 7/27/06, Dawson, Martin wrote:
>Hi Andy,
>
>I'll take a stab at it. Not that HTTP is a valid *using protocol* but,
>rather, that the location reference does not actually identify a using
>protocol and, in fact, HTTP would only be the transport protocol...
>
>3693 states:
>   The "using protocol" is the protocol that uses (reads or modifies)
>   the Location Object.  A protocol that just transports the LO as a
>   string of bits, without looking at them (like an IP storage protocol
>   could do), is not a using protocol, but only a transport protocol.
>   Nevertheless, the entity or protocol that caused the transport
>   protocol to move the LO is responsible for the appropriate
>   distribution, protection, usage, retention, and storage of the LO
>   based on the rules that apply to that LO.
>
>A location reference which provided an HTTP URI indicates that the
>entity that is going to be used to request the LO is to be communicated
>with using HTTP. Because of the rules associated with the PIDF-LO, it
>can be inferred that the entity and whatever semantics are associated
>with the HTTP communications with that entity will honor the
>requirements. That is, HTTP is only the transport protocol. An HTTP GET
>has to be interpreted by the receiving entity within the scope of a
>using protocol - with its associated set of rules and semantics - which
>go beyond HTTP and which do respect the geopriv requirements. Indeed,
>this is generally the case - sending an HTTP GET to an arbitrary IP node
>can yield any number of results depending what that node actually does.
>The client needs to understand the purpose of that server in the scope
>of some higher level protocol.
>
>To use another example, a location reference could be an IP address.
>There's nothing inherent in IP that honors the geopriv rules, but that
>doesn't stop the reference being valid nor say that the protocol used
>between a client and that IP address would not honor the rules.
>
>The purpose of the SIP conveyance specification is to say how a LO or
>reference can be transferred from one entity to another using SIP. It's
>out of scope for a SIP specification to mandate anything about the
>transport protocol or to infer anything about using protocol implied by
>that transport. As has been previously mentioned, this would be subject
>to separate specification and BCP - lest the SIP conveyance
>specification become overly rigid and subject to unnecessary maintenance
>and churn.
>
>Cheers,
>Martin
>
>> -----Original Message-----
>> From: Andrew Newton [mailto:andy@hxr.us]
>> Sent: Thursday, 27 July 2006 10:45 PM
>> To: IETF SIP List; GEOPRIV
>> Subject: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>>
>> There have been a number of opinions regarding HTTP as a using
>protocol.
>>
>> I'd like to know, how many people believe that HTTP requires no or
>> very little specification to meet the requirements in RFC 3693 and
>> RFC 3694.  If you believe this, what basis do you have for this
>> conclusion?  Or do you believe more specification needs to be done,
>> but it is possible for HTTP to be a GEOPRIV using protocol?
> >
>> Finally, how many people believe that HTTP should not be a using
>> protocol?
>>
>> -andy
>>
>> _______________________________________________
>> Geopriv mailing list
>> Geopriv@ietf.org
>> https://www1.ietf.org/mailman/listinfo/geopriv
>
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>If you have received it in error, please notify the sender
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>
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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Jeroen Van Bemmel" <jbemmel@zonnet.nl>,
	"Ted Hardie" <hardie@qualcomm.com>,
	"Dawson, Martin" <Martin.Dawson@andrew.com>, "Andrew Newton" <andy@hxr.us>,
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Date: Sat, 29 Jul 2006 05:40:21 -0500
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprotocol
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Isn't the limited concent component covered by common policy in the form of=
 a rule validity period?
=20

________________________________

From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
Sent: Fri 28/07/2006 5:01 PM
To: Ted Hardie; Dawson, Martin; Andrew Newton; IETF SIP List; GEOPRIV
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprot=
ocol



Hi Ted,

At what scale do you expect this to work? If the location server is require=
d to have a trust relation with everyone I might want to share my location =
with, it won't work for the Internet.

For that, I think something like trait based authorization is needed: a URL=
 that encodes my consent for you to retrieve my location, for a limited per=
iod

Regards,
Jeroen

-Oorspronkelijk bericht -----
Van: "Ted Hardie" <hardie@qualcomm.com>
Aan: "Dawson, Martin" <Martin.Dawson@andrew.com>; "Andrew Newton" <andy@hxr=
us>; "IETF SIP List" <sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
Verzonden: 28-7-06 03:53
Onderwerp: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV using proto=
col

Hi Martin,
        Having not seen your note, I already took a stab
it, along much the same lines you suggest.  Clearly, the attached
-00 (not yet in the drafts directories) is the barest of bare bones,
but I thought it might be useful to sketch that out as a contrast
to some of the real applications which might use HTTP as a
using protocol.
        I believe this also differs from RELO in that it is
focused on the transport of PIDF-LO, which is slightly different
from RELO's remit.  But clearly, there are strong connections there.
        And if the phrase "strawman" in the title doesn't
signal that I don't think this is a done piece of work, allow
this too.
                        regards,
                                Ted Hardie


At 7:50 PM -0500 7/27/06, Dawson, Martin wrote:
>Hi Andy,
>
>I'll take a stab at it. Not that HTTP is a valid *using protocol* but,
>rather, that the location reference does not actually identify a using
>protocol and, in fact, HTTP would only be the transport protocol...
>
>3693 states:
>   The "using protocol" is the protocol that uses (reads or modifies)
>   the Location Object.  A protocol that just transports the LO as a
>   string of bits, without looking at them (like an IP storage protocol
>   could do), is not a using protocol, but only a transport protocol.
>   Nevertheless, the entity or protocol that caused the transport
>   protocol to move the LO is responsible for the appropriate
>   distribution, protection, usage, retention, and storage of the LO
>   based on the rules that apply to that LO.
>
>A location reference which provided an HTTP URI indicates that the
>entity that is going to be used to request the LO is to be communicated
>with using HTTP. Because of the rules associated with the PIDF-LO, it
>can be inferred that the entity and whatever semantics are associated
>with the HTTP communications with that entity will honor the
>requirements. That is, HTTP is only the transport protocol. An HTTP GET
>has to be interpreted by the receiving entity within the scope of a
>using protocol - with its associated set of rules and semantics - which
>go beyond HTTP and which do respect the geopriv requirements. Indeed,
>this is generally the case - sending an HTTP GET to an arbitrary IP node
>can yield any number of results depending what that node actually does.
>The client needs to understand the purpose of that server in the scope
>of some higher level protocol.
>
>To use another example, a location reference could be an IP address.
>There's nothing inherent in IP that honors the geopriv rules, but that
>doesn't stop the reference being valid nor say that the protocol used
>between a client and that IP address would not honor the rules.
>
>The purpose of the SIP conveyance specification is to say how a LO or
>reference can be transferred from one entity to another using SIP. It's
>out of scope for a SIP specification to mandate anything about the
>transport protocol or to infer anything about using protocol implied by
>that transport. As has been previously mentioned, this would be subject
>to separate specification and BCP - lest the SIP conveyance
>specification become overly rigid and subject to unnecessary maintenance
>and churn.
>
>Cheers,
>Martin
>
>> -----Original Message-----
>> From: Andrew Newton [mailto:andy@hxr.us]
>> Sent: Thursday, 27 July 2006 10:45 PM
>> To: IETF SIP List; GEOPRIV
>> Subject: [Geopriv] teasing apart: http as a GEOPRIV using protocol
>>
>> There have been a number of opinions regarding HTTP as a using
>protocol.
>>
>> I'd like to know, how many people believe that HTTP requires no or
>> very little specification to meet the requirements in RFC 3693 and
>> RFC 3694.  If you believe this, what basis do you have for this
>> conclusion?  Or do you believe more specification needs to be done,
>> but it is possible for HTTP to be a GEOPRIV using protocol?
> >
>> Finally, how many people believe that HTTP should not be a using
>> protocol?
>>
>> -andy
>>
>> _______________________________________________
>> Geopriv mailing list
>> Geopriv@ietf.org
>> https://www1.ietf.org/mailman/listinfo/geopriv
>
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>[mf2]
>
>_______________________________________________
>Geopriv mailing list
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>https://www1.ietf.org/mailman/listinfo/geopriv


_______________________________________________
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Geopriv@ietf.org
https://www1.ietf.org/mailman/listinfo/geopriv



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If you have received it in error, please notify the sender
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[mf2]

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Sat Jul 29 10:23:28 2006
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Date: Sat, 29 Jul 2006 16:22:05 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprotocol
To: "Winterbottom, James" <James.Winterbottom@andrew.com>,
	Ted Hardie <hardie@qualcomm.com>, "Dawson,
	Martin" <Martin.Dawson@andrew.com>, 
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Could be, but that would mean the UA is forced to "publish" a location obje=
ct to a server each time it wants to convey it

By putting the expiration time in the URL, it can reuse the same object eac=
h time, offering a different "view" on it to each conveyee. When the locati=
on is constant or one of a set of common places (for example, "at home"), t=
his can be  more convenient

The UA could even encrypt its actual location in the URL, and have the serv=
er generate a PIDF-LO with that data. Then you have a snapshot by reference=
, without being concerned about state on the server, but with additional ac=
cess control possibilities

Regards,

Jeroen

-----Oorspronkelijk bericht -----
Van: "Winterbottom, James" <James.Winterbottom@andrew.com>
Aan: "Jeroen Van Bemmel" <jbemmel@zonnet.nl>; "Ted Hardie" <hardie@qualcomm=
com>; "Dawson, Martin" <Martin.Dawson@andrew.com>; "Andrew Newton" <andy@h=
xr.us>; "IETF SIP List" <sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
Verzonden: 29-7-06 12:40
Onderwerp: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingpr=
otocol

Isn't the limited concent component covered by common policy in the form of=
 a rule validity period?
=20

________________________________

From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
Sent: Fri 28/07/2006 5:01 PM
To: Ted Hardie; Dawson, Martin; Andrew Newton; IETF SIP List; GEOPRIV
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprot=
ocol



Hi Ted,

At what scale do you expect this to work? If the location server is require=
d to have a trust relation with everyone I might want to share my location =
with, it won't work for the Internet.

For that, I think something like trait based authorization is needed: a URL=
 that encodes my consent for you to retrieve my location, for a limited per=
iod

Regards,
Jeroen


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Sat Jul 29 11:25:41 2006
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Subject: RE: [Sip] SIP Security mechanism?
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Actually, I don't think TLS would be required and I don't think you need
to trust the proxies anymore than you do now.  Diffie-Hellman (if I
understand it correctly, which is not a certainty) essentially causes
the endpoints to dynamically generate public keys that have
corresponding private keys.  The receiver sends the public key to the
sender to be used for encryption and uses the private key for
decryption.  This mechanism would allow for the public keys to be passed
in the clear in the initial exchange and that means it could be used
with any type of transport.

The proposal has weaknesses (i.e. difficulties with authentication, MitM
attacks), but it does allow for simple key exchange that doesn't require
ANY infrastructure which should facilitate encryption of bodies if
proxies are present and entire messages if the UAs are B2B.  The
proposal is focused mainly on the difficulties associated with key and
certificate management that have been observed operationally.  By
eliminating that part of the process, perhaps encryption would become
more pervasive, which would be a good thing IMHO.

Thanks,
FM



On Wed, 2006-07-26 at 08:47 +0200, Fries, Steffen wrote:
> Hi Frank,
> 
> the method you are describing at least requires the usage of TLS to
> protect the key exchange between Alice and Bob.  Moreover, you do need
> to trust the proxies in between, as they have access to the keys
> (assumed you send the key from Alice to Bob in the SDP). You may combine
> the SIP_CERTS approach with draft-ono-sipping-smime-keyreuse-01.txt to
> have one S/MIME encryption based on certificates and use the key-reuse
> for the rest of the session. But as I wrote last time, the draft
> regarding key-usage has expired  and I'm not sure if it will be updated.
> 
> Ciao
> 	Steffen
> 
> > -----Original Message-----
> > From: Frank W. Miller [mailto:fwmiller@cornfed.com] 
> > Sent: Saturday, July 22, 2006 3:58 AM
> > To: Cullen Jennings
> > Cc: sip@ietf.org
> > Subject: Re: [Sip] SIP Security mechanism?
> > 
> > 
> > Thanks for the pointer.  I have scanned this draft over.
> > 
> > The goal of your draft is similar to what I'm targeting but 
> > slightly different.  Please correct me if I am 
> > misunderstanding something.  The draft (which is now 
> > draft-ietf-sip-certs-01 I believe) has the goal is to provide 
> > a way to make the distribution of certificates more automated 
> > by providing specific infrastructure that is based on simple 
> > user actions.  The draft specifies publication and an event 
> > package that allows for notification to interested User 
> > Agents of certificates.
> > 
> > The great benefit to this approach is its preservation of 
> > user authentication.
> > 
> > My thought was much simpler.  Let me drop in a quote from the 
> > ZRTP draft, draft-zimmermann-avt-zrtp-01.txt:
> > 
> > ZRTP is key agreement protocol which performs Diffie-Hellman key
> >    exchange during call setup in-band in the Real-time Transport
> >    Protocol (RTP) [1] media stream which has been established 
> > using some
> >    other signaling protocol such as Session Initiation Protocol (SIP)
> >    [11].  This generates a shared secret which is then used 
> > to generate
> >    keys and salt for a Secure RTP (SRTP) [2] session.
> > 
> > Although it uses
> >    a public key algorithm, it does not rely on a public key
> >    infrastructure (PKI).
> > 
> > Were we to put in place some sort of simple transaction that 
> > did this key exchange before any other signaling, we could 
> > use the keys to do encryption of the message bodies (similar 
> > to what S/MIME is doing) without having to deal with PKI at 
> > all.  Since the main problem with the use of S/MIME to date 
> > has been the certificate management, this approach is also 
> > targeting the problem that your draft is.
> > 
> > However, there are a couple of differences.  First, your 
> > draft approach preserves full S/MIME certificate based 
> > authentication, this approach would likely not do that, it 
> > would be to allow for encryption only.  The ZRTP draft 
> > proposes prompting the user with a short authentication 
> > string for the media but I don't know if that would be useful 
> > for the signaling.  Second, this approach would be much 
> > simpler, likely requiring only a couple of messages between 
> > the UA endpoints.  It would not require the centralized 
> > server elements that would be necessary for the 
> > PUBLISH/SUBSCRIBE/NOTIFY components of your draft.
> > 
> > Consider the messages used to do key exchange in ZRTP (again 
> > from the ZRTP draft):
> > 
> >  Alice                                      Bob
> >      |                                         |
> >      | Alice and Bob establish a media session.|
> >      |                                         |
> >      |                   RTP                   |
> >      |<=======================================>|
> >      |                                         |
> >      | Hello (ver,cid,hash,cipher,pkt,sas,Alice's ZID) F1
> >      |---------------------------------------->|
> >      |                             HelloACK F2 |
> >      |<----------------------------------------|
> >      | Hello (ver,cid,hash,cipher,pkt,sas,Bob's ZID) F3
> >      |<----------------------------------------|
> >      | HelloACK F4                             |
> >      |---------------------------------------->|
> >      |                                         |
> >      |        Bob acts as the initiator        |
> >      |                                         |
> >      |   Commit (Bob's ZID,hash,cipher,pkt,hvi) F5
> >      |<----------------------------------------|
> >      | DHPart1 (pvr,rs1IDr,rs2IDr,sigsIDr,srtpsIDr,other_secretIDr) F6
> >      |---------------------------------------->|
> >      | DHPart2 (pvi,rs1IDi,rs2IDi,sigsIDi,ssrtpIDi,other_secretIDi) F7
> >      |<----------------------------------------|
> >      |                                         |
> >      | Alice and Bob generate SRTP session key.|
> >      |                                         |
> >      |               SRTP begins               |
> >      |<=======================================>|
> >      |                                         |
> >      | Confirm1 (plaintext,sasflag,hmac) F8    |
> >      |---------------------------------------->|
> >      |    Confirm2 (plaintext,sasflag,hmac) F9 |
> >      |<----------------------------------------|
> >      | Confirm2AK F10                          |
> >      |---------------------------------------->|
> > 
> > 
> > They are generating keys for each of media streams here, i.e. 
> > an inband message exchange is happening in each direction.  
> > Something like this could probably be simplified to just two 
> > messages for the signaling, something like:
> > 
> >     Bob                             Alice
> >      |           HELLO                |
> >      |------------------------------->|
> >      |           200 OK               |
> >      |<-------------------------------|
> >      |                                |
> > 
> > This example uses a new method I made up called HELLO but I 
> > suppose this could be folded into the INVITE transaction too, 
> > using a mechanism similar to the 407.  The key that Alice is 
> > supposed to use to encrypt the bodies sent to Bob would be in 
> > the HELLO and the key that Bob is supposed to use encrypt the 
> > bodies sent to Alice would be in the 200 OK.
> > 
> > Is this clearer?  Does it make sense?  Its simpler but the 
> > authentication may not be there.  That works for the media 
> > because you can possibly recognize someone's voice.  In the 
> > signaling the authentication thing may be a bigger issue.
> > 
> > FM
> > 
> > 
> > On Fri, 2006-07-21 at 00:17 -0700, Cullen Jennings wrote:
> > > you might want to have a read of 
> > draft-ietf-sipping-certs-04 if your 
> > > goal is to make sure that users can find the keys of other 
> > users for 
> > > use with S/MIME
> > > 
> > > On Jul 19, 2006, at 12:17 PM, fwmiller@cornfed.com wrote:
> > > 
> > > >
> > > >
> > > > I've been reading about the ZRTP lately.  I read 
> > Henning's comment 
> > > > at the IETF meeting about how S/MIME relies on the 
> > receiver's public 
> > > > key to be available to allow for encryption/decryption to work.
> > > >
> > > > Why couldnt we put in place a mechanism that did a key 
> > exchange for 
> > > > SIP signaling similar to whats done in ZRTP (i.e. modified Diffie-
> > > > Hellman) prior to any signaling traffic, then use the 
> > exchanged keys 
> > > > as the basis for S/MIME encryption of the bodies?  Or we 
> > could even 
> > > > get really carried away and use it to do encryption of the entire 
> > > > SIP message if the signaling did not traverse proxies?  
> > This could 
> > > > be useful for encrypting UDP signaling, again if no proxy were 
> > > > involved.
> > > >
> > > > Thoughts?
> > > > FM
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use 
> > > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > > sipping@ietf.org for new developments on the application of sip
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use 
> > sip-implementors@cs.columbia.edu for questions on current sip 
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> > 
> 
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Hi Randy,

Maybe there is such a relationship. I do, however, suspect that in the 
typical case there isn't since this places the deployment bar quite 
high: Every location recipient (e.g., PSAP, SIP user agent, Location 
Server, etc.) needs to have a relationship (or even a shared secret) 
with the access network provider (L3 provider) since this is the place 
where the HTTP server needs to be located.

The identity used for digest authentication can indeed be used in 
combination with the common policy rules. You just need to define how 
the digest identity and the digest realm part are matched with the 
content of the 'id' and the 'domain' part to the common policy rules.

With the Pawn Ticket URLs the solution would, however, be simpler (as 
simple as with URLs with a randomly create and unguessable component): 
If you know the URI then you are allowed to retrieve the PIDF-LO.

(This assumes that I understood the Pawn Ticket approach properly.)

Ciao
Hannes

Randall Gellens wrote:
> This does look to me to be a "using protocol".
> 
> Presumably the HTTP server and the dereferencer have a relationship such 
> that HTTP DIGEST can be used.
> 
> Perhaps the DIGEST identity can be used when evaluating the policy rules.


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Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Thu, 27 Jul 2006 15:02:00 -0500
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From: "Stark, Barbara" <Barbara.Stark@BellSouth.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
	"Brian Rosen" <br@brianrosen.net>
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This gets my vote.
Barbara

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Wednesday, July 26, 2006 11:25 PM
> To: Brian Rosen
> Cc: IETF SIP List; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
> location-conveyance
>=20
>=20
> My suggestion would be
>=20
> - to focus the draft on just defining the Location header syntax, =20
> without ruling out any particular mechanism or scheme; this would =20
> make the draft really short (which I would find helpful).
>=20
> - and then add one section to the draft that prototypically defines =20
> what a particular scheme that wants to appear in the Location header =20
> needs to define, using the non-controversial cid mechanism as the =20
> example. Other, separate, drafts can then define other modes, =20
> following the same template. Documents such as the phone-bcp=20
> can then =20
> pick some suitable must-implement subset and make emergency-call-=20
> specific recommendations, rather than sprinkling these=20
> throughout the =20
> conveyance document and repeating some in the phone-bcp. Most of =20
> these are, in any event, operational recommendations, not for =20
> ensuring protocol-level interoperability.
>=20
> That way, the conveyance draft can be extremely short (even shorter =20
> if the requirements are omitted...) and can get done, and we avoid =20
> engineering operational requirements on the fly, commingling =20
> emergency and non-emergency cases, as seems to be happening now.
>=20
> I think protocol documents should define what's needed for =20
> interoperability; BCPs can make more specific, situation-specific =20
> recommendations. Commingling the two seems to be a recipe for =20
> deadlock or arbitrary, non-technically-grounded decisions that are =20
> more lawyer-like than engineering-like ("you did not mention =20
> technique X in some meeting N years ago and thus you can't bring it =20
> up now!")
>=20
> Henning
>=20
> On Jul 26, 2006, at 11:08 PM, Brian Rosen wrote:
>=20
> > At this point, I think we need some chair guidance and by that I =20
> > think it's
> > a combo of geopriv and sip chairs.
> >
> > We seem to have a couple of significant open questions.  I think =20
> > most of the
> > other stuff (like the "allow sending to router") will converge.
> >
> > There seems to be some opposition to allowing sip/sips/pres: in a =20
> > Location
> > Header.  The opposition seems to be that despite 4119, somehow, SIP
> > (presence subscription) is not a "using protocol".  I "think" a =20
> > ruling from
> > the bench on that would work.
> >
> > There is also opposition to defining the allowed schemes as a =20
> > specific list,
> > with a defined (easy) extension mechanism.
> >
> > Related to that, there are some who want location-conveyance to =20
> > define an
> > HTTP based location dereference mechanism, and on the=20
> opposite end, =20
> > we have
> > people saying we haven't even agreed to support any form of
> > location-by-reference.
> >
> > The authors would like to know what to do now.
> >
> > Brian
> >
> >> -----Original Message-----
> >> From: Andrew Newton [mailto:andy@hxr.us]
> >> Sent: Wednesday, July 26, 2006 9:58 PM
> >> To: Winterbottom, James
> >> Cc: sip@ietf.org; geopriv@ietf.org; Abbott, Nadine B; Henning =20
> >> Schulzrinne
> >> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> >> conveyance
> >>
> >>
> >> On Jul 26, 2006, at 9:43 PM, Winterbottom, James wrote:
> >>
> >>> Andy,
> >>>
> >>> RFC-4119 says "However, the usage of this location object=20
> format is
> >>> not
> >>>    limited to presence-using protocols-- any protocol that can
> >>> carry XML
> >>>    or MIME types can carry PIDF."
> >>>
> >>> It seems to me that restricting the allowed schema in what is =20
> >>> supposed
> >>> to be a location conveyance specification is in opposition to this
> >>> statement.
> >>
> >> Did somebody say that HTTP was a presence-using protocol?
> >>
> >> -andy
> >>
> >> ps.  you can find my location with the following reference URIs:
> >>
> >> gopher://fatchance.hxr.us/you-still-have-a-gopher-client
> >> news:alt.locations.people.secretly-being-watched
> >> telnet://screen-scrapers.are.us/
> >> dns:cached-publicly-around-the-world.com?type=3DHOST
> >>
> >>
> >>
> >> _______________________________________________
> >> Geopriv mailing list
> >> Geopriv@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/geopriv
> >
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20

*****

The information transmitted is intended only for the person or entity to =
which it is addressed and may contain confidential, proprietary, and/or =
privileged material. Any review, retransmission, dissemination or other =
use of, or taking of any action in reliance upon this information by =
persons or entities other than the intended recipient is prohibited. If =
you received this in error, please contact the sender and delete the =
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To: Hannes Tschofenig <Hannes.Tschofenig@gmx.net>
From: Randall Gellens <randy@qualcomm.com>
Subject: Re: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV
	using 	protocol
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At 7:31 PM +0200 7/29/06, Hannes Tschofenig wrote:

>  Hi Randy,
>
>  Maybe there is such a relationship. I do, however, suspect that in 
> the typical case there isn't since this places the deployment bar 
> quite high: Every location recipient (e.g., PSAP, SIP user agent, 
> Location Server, etc.) needs to have a relationship (or even a 
> shared secret) with the access network provider (L3 provider) since 
> this is the place where the HTTP server needs to be located.

I agree.  That's the question I was raising: my reading of the draft 
suggests that, since it relies on HTTP DIGEST, there must be a 
relationship in order for there to be credentials.

>
>  The identity used for digest authentication can indeed be used in 
> combination with the common policy rules. You just need to define 
> how the digest identity and the digest realm part are matched with 
> the content of the 'id' and the 'domain' part to the common policy 
> rules.
>
>  With the Pawn Ticket URLs the solution would, however, be simpler 
> (as simple as with URLs with a randomly create and unguessable 
> component): If you know the URI then you are allowed to retrieve 
> the PIDF-LO.
>
>  (This assumes that I understood the Pawn Ticket approach properly.)
>
>  Ciao
>  Hannes
>
>  Randall Gellens wrote:
>>  This does look to me to be a "using protocol".
>>
>>  Presumably the HTTP server and the dereferencer have a 
>> relationship such that HTTP DIGEST can be used.
>>
>>  Perhaps the DIGEST identity can be used when evaluating the policy rules.
>
>
>  _______________________________________________
>  Geopriv mailing list
>  Geopriv@ietf.org
>  https://www1.ietf.org/mailman/listinfo/geopriv


-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly-selected tag: ---------------
Language is a virus from outer space.  --William S. Burroughs

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From sip-bounces@ietf.org Sat Jul 29 15:30:52 2006
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	protocol
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It seems I misunderstood your comment.
In this case I agree with your observation.

Randall Gellens wrote:
> At 7:31 PM +0200 7/29/06, Hannes Tschofenig wrote:
> 
>>  Hi Randy,
>>
>>  Maybe there is such a relationship. I do, however, suspect that in 
>> the typical case there isn't since this places the deployment bar 
>> quite high: Every location recipient (e.g., PSAP, SIP user agent, 
>> Location Server, etc.) needs to have a relationship (or even a shared 
>> secret) with the access network provider (L3 provider) since this is 
>> the place where the HTTP server needs to be located.
> 
> 
> I agree.  That's the question I was raising: my reading of the draft 
> suggests that, since it relies on HTTP DIGEST, there must be a 
> relationship in order for there to be credentials.
> 
>>
>>  The identity used for digest authentication can indeed be used in 
>> combination with the common policy rules. You just need to define how 
>> the digest identity and the digest realm part are matched with the 
>> content of the 'id' and the 'domain' part to the common policy rules.
>>
>>  With the Pawn Ticket URLs the solution would, however, be simpler (as 
>> simple as with URLs with a randomly create and unguessable component): 
>> If you know the URI then you are allowed to retrieve the PIDF-LO.
>>
>>  (This assumes that I understood the Pawn Ticket approach properly.)
>>
>>  Ciao
>>  Hannes
>>
>>  Randall Gellens wrote:
>>
>>>  This does look to me to be a "using protocol".
>>>
>>>  Presumably the HTTP server and the dereferencer have a relationship 
>>> such that HTTP DIGEST can be used.
>>>
>>>  Perhaps the DIGEST identity can be used when evaluating the policy 
>>> rules.
>>
>>
>>
>>  _______________________________________________
>>  Geopriv mailing list
>>  Geopriv@ietf.org
>>  https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 
> 


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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Jeroen Van Bemmel" <jbemmel@zonnet.nl>,
	"Ted Hardie" <hardie@qualcomm.com>,
	"Dawson, Martin" <Martin.Dawson@andrew.com>, "Andrew Newton" <andy@hxr.us>,
	"IETF SIP List" <sip@ietf.org>, "GEOPRIV" <geopriv@ietf.org>
Date: Sat, 29 Jul 2006 17:30:22 -0500
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprotocol
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Jeroen,
=20
I think you are coming from a slightly different premise to me. I am not of=
 the opinion that the Target always knows where it is, sure it can ask, but=
 there are circumstances where A) it is not capable of asking, B) doesn't w=
ant to ask it is happy for certain other entities just to know when they as=
k. I don't believe that in either of these cases that the UA has to publish=
 a PIDF-LO at all.
=20
It simply tells a local server (the thing that the URI points to) not to re=
lease the location information to anyone but but X, and only if X asks for =
it within Y. The local server is able to determine location, and it is my v=
iew that the server can generate a PIDF-LO using an unlinked pseudonym as t=
he entity identifier, RFC-4119 gives us defaults for all other parts we nee=
d. In this way I have the same URI for everbody, but I can restrict how lon=
g it is valid for anyone person.
=20
Cheers
James

________________________________

From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
Sent: Sat 29/07/2006 9:22 AM
To: Winterbottom, James; Ted Hardie; Dawson, Martin; Andrew Newton; IETF SI=
P List; GEOPRIV
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprot=
ocol



Could be, but that would mean the UA is forced to "publish" a location obje=
ct to a server each time it wants to convey it

By putting the expiration time in the URL, it can reuse the same object eac=
h time, offering a different "view" on it to each conveyee. When the locati=
on is constant or one of a set of common places (for example, "at home"), t=
his can be  more convenient

The UA could even encrypt its actual location in the URL, and have the serv=
er generate a PIDF-LO with that data. Then you have a snapshot by reference=
, without being concerned about state on the server, but with additional ac=
cess control possibilities

Regards,

Jeroen

-----Oorspronkelijk bericht -----
Van: "Winterbottom, James" <James.Winterbottom@andrew.com>
Aan: "Jeroen Van Bemmel" <jbemmel@zonnet.nl>; "Ted Hardie" <hardie@qualcomm=
com>; "Dawson, Martin" <Martin.Dawson@andrew.com>; "Andrew Newton" <andy@hx=
r.us>; "IETF SIP List" <sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
Verzonden: 29-7-06 12:40
Onderwerp: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingpr=
otocol

Isn't the limited concent component covered by common policy in the form of=
 a rule validity period?


________________________________

From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
Sent: Fri 28/07/2006 5:01 PM
To: Ted Hardie; Dawson, Martin; Andrew Newton; IETF SIP List; GEOPRIV
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprot=
ocol



Hi Ted,

At what scale do you expect this to work? If the location server is require=
d to have a trust relation with everyone I might want to share my location =
with, it won't work for the Internet.

For that, I think something like trait based authorization is needed: a URL=
 that encodes my consent for you to retrieve my location, for a limited per=
iod

Regards,
Jeroen




---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
contain privileged, proprietary, or otherwise private information. =20
If you have received it in error, please notify the sender
immediately and delete the original.  Any unauthorized use of
this email is prohibited.
---------------------------------------------------------------------------=
---------------------
[mf2]

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From sip-bounces@ietf.org Sat Jul 29 19:03:05 2006
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I will be out of the office starting  29/07/2006 and will not return until
21/08/2006.

I will respond to your message when I return.



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From sip-bounces@ietf.org Sun Jul 30 04:37:53 2006
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Date: Sun, 30 Jul 2006 10:36:28 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
To: "Stark, Barbara" <Barbara.Stark@BellSouth.com>,
	Henning Schulzrinne <hgs@cs.columbia.edu>, Brian Rosen <br@brianrosen.net>
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Yes,mine too, with one additional remark: the location based routing use ca=
se should be removed

It is a special case of a more general one: route to some party in area X. =
This need not have anything to do with the caller's location, in fact for p=
rivacy reasons it is better not.

LBR is better handled using caller prefs/callee capabilities (eg define +ge=
o.zip, not using PIDF at all for simplicity)

Given the above, proxy removal of location and data URIs are less of an iss=
ue

Regards,
Jeroen

-----Oorspronkelijk bericht -----
Van: "Stark, Barbara" <Barbara.Stark@BellSouth.com>
Aan: "Henning Schulzrinne" <hgs@cs.columbia.edu>; "Brian Rosen" <br@brianro=
sen.net>
CC: "IETF SIP List" <sip@ietf.org>; geopriv@ietf.org
Verzonden: 27-7-06 22:02
Onderwerp: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveya=
nce

This gets my vote.
Barbara

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Wednesday, July 26, 2006 11:25 PM
> To: Brian Rosen
> Cc: IETF SIP List; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
> location-conveyance
>=20
>=20
> My suggestion would be
>=20
> - to focus the draft on just defining the Location header syntax, =20
> without ruling out any particular mechanism or scheme; this would =20
> make the draft really short (which I would find helpful).
>=20
> - and then add one section to the draft that prototypically defines =20
> what a particular scheme that wants to appear in the Location header =20
> needs to define, using the non-controversial cid mechanism as the =20
> example. Other, separate, drafts can then define other modes, =20
> following the same template. Documents such as the phone-bcp=20
> can then =20
> pick some suitable must-implement subset and make emergency-call-=20
> specific recommendations, rather than sprinkling these=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org Sun Jul 30 04:37:53 2006
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Date: Sun, 30 Jul 2006 10:36:28 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
To: "Stark, Barbara" <Barbara.Stark@BellSouth.com>,
	Henning Schulzrinne <hgs@cs.columbia.edu>, Brian Rosen <br@brianrosen.net>
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Yes,mine too, with one additional remark: the location based routing use ca=
se should be removed

It is a special case of a more general one: route to some party in area X. =
This need not have anything to do with the caller's location, in fact for p=
rivacy reasons it is better not.

LBR is better handled using caller prefs/callee capabilities (eg define +ge=
o.zip, not using PIDF at all for simplicity)

Given the above, proxy removal of location and data URIs are less of an iss=
ue

Regards,
Jeroen

-----Oorspronkelijk bericht -----
Van: "Stark, Barbara" <Barbara.Stark@BellSouth.com>
Aan: "Henning Schulzrinne" <hgs@cs.columbia.edu>; "Brian Rosen" <br@brianro=
sen.net>
CC: "IETF SIP List" <sip@ietf.org>; geopriv@ietf.org
Verzonden: 27-7-06 22:02
Onderwerp: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveya=
nce

This gets my vote.
Barbara

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Wednesday, July 26, 2006 11:25 PM
> To: Brian Rosen
> Cc: IETF SIP List; geopriv@ietf.org
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to
> location-conveyance
>=20
>=20
> My suggestion would be
>=20
> - to focus the draft on just defining the Location header syntax, =20
> without ruling out any particular mechanism or scheme; this would =20
> make the draft really short (which I would find helpful).
>=20
> - and then add one section to the draft that prototypically defines =20
> what a particular scheme that wants to appear in the Location header =20
> needs to define, using the non-controversial cid mechanism as the =20
> example. Other, separate, drafts can then define other modes, =20
> following the same template. Documents such as the phone-bcp=20
> can then =20
> pick some suitable must-implement subset and make emergency-call-=20
> specific recommendations, rather than sprinkling these=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

From sip-bounces@ietf.org Sun Jul 30 04:37:53 2006
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Date: Sun, 30 Jul 2006 10:36:41 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprotocol
To: "Winterbottom, James" <James.Winterbottom@andrew.com>,
	Ted Hardie <hardie@qualcomm.com>, "Dawson,
	Martin" <Martin.Dawson@andrew.com>, 
	Andrew Newton <andy@hxr.us>, IETF SIP List <sip@ietf.org>,
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James,

=20
>It simply tells a local server (the thing that the URI points to) not to r=
elease the location information to anyone but but X, and only if X asks for=
 it within Y. =20

(JvB) I was only talking about the benefits of encoding authorization and e=
xpiry in the URL, and the same argument applies here: As you describe it, t=
he UA will need (because of the 'within Y' part) to contact the server each=
 time it conveys its location. Encoding this in the URL can avoid that. It =
can be used regardless whether the UA knows its location or not

Regards,

Jeroen=


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Date: Sun, 30 Jul 2006 10:36:41 +0200
From: Jeroen Van Bemmel <jbemmel@zonnet.nl>
Subject: RE: [Sip] RE: [Geopriv] teasing apart: http as a GEOPRIV usingprotocol
To: "Winterbottom, James" <James.Winterbottom@andrew.com>,
	Ted Hardie <hardie@qualcomm.com>, "Dawson,
	Martin" <Martin.Dawson@andrew.com>, 
	Andrew Newton <andy@hxr.us>, IETF SIP List <sip@ietf.org>,
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James,

=20
>It simply tells a local server (the thing that the URI points to) not to r=
elease the location information to anyone but but X, and only if X asks for=
 it within Y. =20

(JvB) I was only talking about the benefits of encoding authorization and e=
xpiry in the URL, and the same argument applies here: As you describe it, t=
he UA will need (because of the 'within Y' part) to contact the server each=
 time it conveys its location. Encoding this in the URL can avoid that. It =
can be used regardless whether the UA knows its location or not

Regards,

Jeroen=


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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
Date: Sun, 30 Jul 2006 09:01:24 -0400
To: Jeroen Van Bemmel <jbemmel@zonnet.nl>
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> LBR is better handled using caller prefs/callee capabilities (eg  
> define +geo.zip, not using PIDF at all for simplicity)
>

This is completely new suggestion; emergency calling is nothing but a  
special case of location-based routing. I agree that it is useful to  
separate location information for routing from that shown to end  
systems, as they often have different resolution needs. (To route to  
the right branch of your pizza delivery franchiser, you may only need  
a zip code, but for delivery, you need the full street address. The  
practical problem is that such boundaries don't always agree with zip  
codes or any other higher-level geographic unit, such as state,  
county or town.)

Henning

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From sip-bounces@ietf.org Sun Jul 30 10:53:14 2006
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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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Hi Henning,

Yes, it was intended as a new angle. And you are right that 'zip' is 
probably not enough to cover all cases, my intention in providing the 'geo' 
feature tree as example was to suggest that GEOPRIV might define other 
feature parameters in this space ("country","state" etc are some obvious 
candidates, and "geo.lat" and "geo.lon" perhaps?)

To me, the "emergency calling" use case should indeed be viewed as a 
scenario in which "location conveyance" and "location based routing" are 
combined.

Regards,

Jeroen

----- Original Message ----- 
From: "Henning Schulzrinne" <hgs@cs.columbia.edu>
To: "Jeroen Van Bemmel" <jbemmel@zonnet.nl>
Cc: "IETF SIP List" <sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
Sent: Sunday, July 30, 2006 3:01 PM
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance


>> LBR is better handled using caller prefs/callee capabilities (eg  define 
>> +geo.zip, not using PIDF at all for simplicity)
>>
>
> This is completely new suggestion; emergency calling is nothing but a 
> special case of location-based routing. I agree that it is useful to 
> separate location information for routing from that shown to end  systems, 
> as they often have different resolution needs. (To route to  the right 
> branch of your pizza delivery franchiser, you may only need  a zip code, 
> but for delivery, you need the full street address. The  practical problem 
> is that such boundaries don't always agree with zip  codes or any other 
> higher-level geographic unit, such as state,  county or town.)
>
> Henning 


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From sip-bounces@ietf.org Sun Jul 30 13:47:25 2006
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Hi Jeroen,

in past discussions we concluded that we don't want to have a special 
encoding for location-based routing.

I agree with Henning that the level of granularity might, in some cases, 
be different depending on the consumer of the PIDF-LO. This is, however, 
more a location-based scenario rather than an emergency calling 
scenario. The problem is that the end host does not necessarily know the 
service boundary ahead of time and hence you don't know the required 
granularity of the location object. If you omit too many elements from 
the PIDF-LO then LoST might just return incorrect information.

Ciao
Hannes

Jeroen van Bemmel wrote:
> Hi Henning,
> 
> Yes, it was intended as a new angle. And you are right that 'zip' is 
> probably not enough to cover all cases, my intention in providing the 
> 'geo' feature tree as example was to suggest that GEOPRIV might define 
> other feature parameters in this space ("country","state" etc are some 
> obvious candidates, and "geo.lat" and "geo.lon" perhaps?)
> 
> To me, the "emergency calling" use case should indeed be viewed as a 
> scenario in which "location conveyance" and "location based routing" are 
> combined.
> 
> Regards,
> 
> Jeroen
> 
> ----- Original Message ----- From: "Henning Schulzrinne" 
> <hgs@cs.columbia.edu>
> To: "Jeroen Van Bemmel" <jbemmel@zonnet.nl>
> Cc: "IETF SIP List" <sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
> Sent: Sunday, July 30, 2006 3:01 PM
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to 
> location-conveyance
> 
  >
>>> LBR is better handled using caller prefs/callee capabilities (eg  
>>> define +geo.zip, not using PIDF at all for simplicity)
>>>
>>
>> This is completely new suggestion; emergency calling is nothing but a 
>> special case of location-based routing. I agree that it is useful to 
>> separate location information for routing from that shown to end  
>> systems, as they often have different resolution needs. (To route to  
>> the right branch of your pizza delivery franchiser, you may only need  
>> a zip code, but for delivery, you need the full street address. The  
>> practical problem is that such boundaries don't always agree with zip  
>> codes or any other higher-level geographic unit, such as state,  
>> county or town.)
>>
>> Henning 
> 
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
> 
> 


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From sip-bounces@ietf.org Sun Jul 30 14:31:17 2006
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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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Hi Hannes,

see inline

Regards,

jeroen

----- Original Message ----- 
From: "Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
Cc: "Henning Schulzrinne" <hgs@cs.columbia.edu>; "IETF SIP List" 
<sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
Sent: Sunday, July 30, 2006 7:45 PM
Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance


> Hi Jeroen,
>
> in past discussions we concluded that we don't want to have a special 
> encoding for location-based routing.
>

I was not aware of any such conclusion. IMHO PIDF-LO is not the answer to 
all location-based problems, but before we delve into that discussion 
again - I was argueing that "location conveyance" and "location based 
routing" are separate cases. Do you agree with that?

A simple illustrative example: suppose I will be going on holiday to Paris 
next week. I want to make a hotel reservation, and call some well-known 
chain of hotels. My location will be somewhere in Holland (at the moment of 
calling), and I want my call to go to their Paris office. How would this 
work with only 1 PIDF-LO?

> I agree with Henning that the level of granularity might, in some cases, 
> be different depending on the consumer of the PIDF-LO. This is, however, 
> more a location-based scenario rather than an emergency calling scenario. 
> The problem is that the end host does not necessarily know the service 
> boundary ahead of time and hence you don't know the required granularity 
> of the location object. If you omit too many elements from the PIDF-LO 
> then LoST might just return incorrect information.

OK, so perhaps the required granularity for matching would have to be 
specified as part of the callee-capabilities. Then an error would occur (eg 
the proxy would return '485 Ambiguous', with candidate contacts) when no 
locations match the given criteria. For specific cases (like Emergency) I 
imagine the required granularity would be defined, such that the UA would 
know in advance.

I understand the desire to limit the number of different formats and 
encodings being used, but caller prefs / callee capabilities is more than a 
format - it is a specific, existing SIP mechanism that can address several 
useful scenarios. It has many benefits - for one, in this way LBR could be 
introduced for existing proxies (that support caller prefs).

Regards,

jeroen



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Hi Jeroen,

Jeroen van Bemmel wrote:
> Hi Hannes,
> 
> see inline
> 
> Regards,
> 
> jeroen
> 
> ----- Original Message ----- From: "Hannes Tschofenig" 
> <Hannes.Tschofenig@gmx.net>
> To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
> Cc: "Henning Schulzrinne" <hgs@cs.columbia.edu>; "IETF SIP List" 
> <sip@ietf.org>; "GEOPRIV" <geopriv@ietf.org>
> Sent: Sunday, July 30, 2006 7:45 PM
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to 
> location-conveyance
> 
> 
>> Hi Jeroen,
>>
>> in past discussions we concluded that we don't want to have a special 
>> encoding for location-based routing.
>>
> 
> I was not aware of any such conclusion. IMHO PIDF-LO is not the answer 
> to all location-based problems, but before we delve into that discussion 
> again - I was argueing that "location conveyance" and "location based 
> routing" are separate cases. Do you agree with that?

I agree with you in various ways. The use case is different, the 
security properties are different and the description in the draft needs 
to describe both cases.

> 
> A simple illustrative example: suppose I will be going on holiday to 
> Paris next week. I want to make a hotel reservation, and call some 
> well-known chain of hotels. My location will be somewhere in Holland (at 
> the moment of calling), and I want my call to go to their Paris office. 
> How would this work with only 1 PIDF-LO?

I guess the use cases most GEOPRIV/ECRIT folks have in mind is a bit 
different.
 From a protocol point of view there is, however, no difference.

> 
>> I agree with Henning that the level of granularity might, in some 
>> cases, be different depending on the consumer of the PIDF-LO. This is, 
>> however, more a location-based scenario rather than an emergency 
>> calling scenario. The problem is that the end host does not 
>> necessarily know the service boundary ahead of time and hence you 
>> don't know the required granularity of the location object. If you 
>> omit too many elements from the PIDF-LO then LoST might just return 
>> incorrect information.
> 
> 
> OK, so perhaps the required granularity for matching would have to be 
> specified as part of the callee-capabilities. Then an error would occur 
> (eg the proxy would return '485 Ambiguous', with candidate contacts) 
> when no locations match the given criteria. For specific cases (like 
> Emergency) I imagine the required granularity would be defined, such 
> that the UA would know in advance.


I have no problem with two PIDF-LOs in a single message: One for routing 
and the other one for end host consumption. Would this solve your 
concerns already?

For emergency services we know that you have to provide location 
information at a fine grain granularity. In recent discussions we argued 
about the need to include composite location with civic information at a 
floor level.

> 
> I understand the desire to limit the number of different formats and 
> encodings being used, but caller prefs / callee capabilities is more 
> than a format - it is a specific, existing SIP mechanism that can 
> address several useful scenarios. It has many benefits - for one, in 
> this way LBR could be introduced for existing proxies (that support 
> caller prefs).

I understood your previous comment as a request to specify a different 
location information format (not PIDF-LO) for location-based routing.

Ciao
Hannes

> 
> Regards,
> 
> jeroen
> 
> 
> 


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From sip-bounces@ietf.org Sun Jul 30 15:54:07 2006
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Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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> I understood your previous comment as a request to specify a different 
> location information format (not PIDF-LO) for location-based routing.

That depends, I would be happy with a mapping of PIDF-LO onto callee 
capabilities / caller prefs. Would that be considered a different format?

Jeroen 


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From sip-bounces@ietf.org Mon Jul 31 09:18:05 2006
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	"'Hannes Tschofenig'" <Hannes.Tschofenig@gmx.net>
Subject: RE: [Sip] RE: [Geopriv] Consensus on changes to location-conveyance
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We have spent a couple of YEARS working on this draft, it has always
considers location based routing one of its uses, and at this point, I think
this idea is not interesting.

Does anyone else besides Jeroen support a different mechanism to convey
location for routing than one for conveying location to the endpoint?

I remind you that many (but not all) location based routing use cases DO
involve conveyance of the location to an endpoint.  The usual case we use
for location based routing besides the emergency case is the pizza delivery
scenario.  You use location to route to the correct store, and the store
gets your location to deliver the pizza.  You don't ALWAYS want to do this,
of course, but it is very common.

At this point, we need fewer options (not only in the draft, in the SIP
world), not more.

Brian

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Sunday, July 30, 2006 3:52 PM
> To: Hannes Tschofenig
> Cc: IETF SIP List; GEOPRIV; Henning Schulzrinne
> Subject: Re: [Sip] RE: [Geopriv] Consensus on changes to location-
> conveyance
> 
> > I understood your previous comment as a request to specify a different
> > location information format (not PIDF-LO) for location-based routing.
> 
> That depends, I would be happy with a mapping of PIDF-LO onto callee
> capabilities / caller prefs. Would that be considered a different format?
> 
> Jeroen
> 
> 
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv


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From sip-bounces@ietf.org Mon Jul 31 09:32:55 2006
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Brian Rosen wrote:
> At this point, we need fewer options (not only in the draft, in the SIP
> world), not more.

I agree.  Not only do I not see a need for this, but such a suggestion 
this late in the game without any implementation, specification, or 
formalized use case is not helpful.

-andy


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On Jul 5, 2006, at 9:33 AM, Hadriel Kaplan wrote:

>
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Sunday, July 02, 2006 1:02 PM
>>
>> On Jun 30, 2006, at 12:57 PM, Hadriel Kaplan wrote:
>>
>>> I agree with you, but it is used for detecting suspicious From
>>> headers even
>>> for non-anonymous calls.  Obviously if the proxy inserting the PAI
>>> _knows_ a
>>>> From is spoofed, it could/should reject it.  But if the proxy does
>>>> _not_
>>> know whether the From is spoofed or not, it just doesn't insert the
>>> PAI.
>>> For example, ingress proxy receives a request from another domain -
>>> if it
>>> trusts the domain it passes the PAI (or inserts one).  If it does
>>> not trust
>>> the domain, it removes PAI, but doesn't want to reject the request
>>> (else it
>>> wouldn't accept requests from such untrusted domains to begin with).
>>
>> I see your point, I can see how a proxy that is deciding not to
>> remove a PAI might not know a From was spoofed. I don't see how a
>> proxy that was going to *insert* a PAI would know what to put in the
>> PAI and not be able to know if a PAI was spoofed.
>
> I'm not sure I follow - maybe you meant the word "From" instead of  
> "PAI" in
> the "if a PAI was spoofed" bit?
> I meant in the context of implicitly trusted connections.  For  
> example: an
> operator receiving a request from a directly-attached business  
> customer
> proxy, where the operator's ingress proxy may insert a PAI of
> bob@company.com based on the From, even though company.com's egress  
> proxy
> did not insert the PAI itself.  Obviously for an anonymous request the
> ingress proxy would not know enough to put that URI in, and it's  
> certainly
> the case that it can't be 100% sure that the request really came  
> from bob to
> begin with (e.g., the company proxy could be hijacked, etc.).  But  
> that
> could be the case with Identity in use as well.
>
> -hadriel

Oops - yes I meant From.



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Dear SIP Forum member --

Five of eight current SIP Forum Board of Director (BoD) seats are  
expiring shortly.  Elections for new Board members will be held at  
our annual meeting in September this year.  (Watch for the formal  
meeting announcement that will be sent in mid-August.)

Nominations are now open for new Directors.  Any SIP Forum member  
(participating/free or full/paid) can be nominated to serve a two- 
year term.  NOMINATIONS MUST BE RECEIVED WITHIN TWO WEEKS, on or  
before AUGUST 14, 2006.

The requirements to be elected, the duties of Board members, and the  
process to make a nomination, are all described on the SIP Forum  
website here:  http://www.sipforum.org/content/view/272/226/ .  Note:  
You may nominate yourself.

If you are passionate about the advancement of SIP, and please take a  
moment to nominate a passionate person who wants to help.

Thank you.
Cheers
-jb

----------
Jay Batson
Acting Managing Director, Chairman of the Board
batsonjay@sipforum.org
+1-978-824-0111


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From sip-bounces@ietf.org Mon Jul 31 11:50:48 2006
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I have lost track of where we stand on whether the only mentioned URI
schemes in location-conveyance are sip/sips/pres.  I "think" we have
agreed that an HTTP/HTTPS dereference is in a separate document, but I'm
not sure.

This is NOT the discussion of what the ABNF of the location header looks
like (although it would NOT have http/https in it for sure if we agree
that an HTTP dereference protocol is a separate draft).

If you think location-conveyance DOES have HTTP dereference text in it,
please say so, and please explain why you would not be satisfied if
there were a separate draft on the subject.

Brian

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From sip-bounces@ietf.org Mon Jul 31 14:04:09 2006
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Subject: Re: [Sip] Updated acr-code draft
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On Jun 21, 2006, at 1:11 AM, Jeroen van Bemmel wrote:

> Jonathan,
>
> (Only) one comment: What about privacy service behavior? In  
> particular I am
> thinking about a privacy service that anonymized the request, and  
> receives a
> 433.
>
> If the privacy service is acting as a B2BUA, it should probably not  
> retry
> the request "non-anonymously" (this could constitute a downgrade  
> attack).
> This is also something worth mentioning in the security  
> considerations: the
> retrying of the request SHOULD be based on an explicit decision by  
> the user
> making the request, it SHOULD NOT be done automatically

Any response on this before we move to last call?

--
Dean


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Thanks for the detailed review, Paul. Responses below.

Paul Kyzivat wrote:

> I did a reasonably careful review of draft-ietf-sip-gruu-09. I was 
> mostly looking for remaining technical issues rather than formatting 
> nits. For awhile now I have mostly be looking at diffs as new versions 
> come out and reading selected sections at need. It was good to read the 
> whole thing again to see how it all hangs together.
> 
> I found nothing really major, but some stuff that is more than 
> editorial. Some of it is a judgment call about whether something is 
> clear or not. I won't feel bad to have a number of these comments rejected.
> 
>     Paul
> 
> General comment:
> 
> SIPS is mentioned in many places in this draft. Given the current 
> ongoing issues wrt SIPS, the language here for dealing with it may prove 
> to be prescient, or may turn out to just be wrong. Can this be dodged 
> somehow? Maybe this whole draft should only be defined for SIP, with 
> SIPS left for a separate draft. (I will still comment on some places 
> containing SIPS, when there are other points than the above.)

Well, it used to say nothing about sips and I think you complained it 
needed to a long time back ;)

I'm not awful inclined to hold the draft up for us to figure all of that 
out, and I'm also not inclined to say nothing. My preference is to 
address it as best we can based on the current mechanism, and the 
in-progress sips update will need to tackle gruu.

> 
> Section 5:
> 
>    ...  The "opaque" parameter is added by the owner
>    of the domain to correlate the GRUU to its instance ID, and to easily
>    recognize that the URI has the GRUU property.
> 
> I think this isn't quite right. The "gruu" parameter may be used to 
> recognize the GRUU property. How about:
> 
>    ...  The "opaque" parameter is added by the owner
>    of the domain to correlate the GRUU to its instance ID, and
>    potentially to help verify that the GRUU has not been forged.

Dale flagged this one too.  I'll use your text.


> 
> Figure 1:
> 
> Its unfortunate that pictures such as this lead to many questions, often 
> irrelevant ones. I know we have discussed nuances of this one before.
> 
> - as shown, a contact is bound to 1 AOR, at most one Instance ID and at 
> most one GRUU. In practice is is possible to bind the same contact to 
> multiple AORs if one doesn't need to distinguish between them or has 
> some other means to do so. (And section 7.1.1.1 allows it, since 
> restriction to a single AOR is only RECOMMENDED.) 

Fair point. The cardinality in this direction isn't particularly useful. 
Rather than changing to 'n' and then needing to explain, I just removed 
the cardinality entirely. The main thing to show here is that there are 
0..n contacts bound to the AOR and GRUU.


> It is also possible 
> that the same contact will be associated with more than one Instance ID. 
> (Ideally there is only one such association at any one time, but over 
> time it may change.}

One at a time is what is meant to be captured here. Changing this to 'n' 
I think will be very confusing.


> 
> - the figure doesn't show what is in the text:
> 
>    A GRUU is associated, in a many-to-one fashion, with the combination
>    of an instance ID and an AOR.  This combination is referred to as an
>    instance ID/AOR pair.  For each GRUU, there is one instance ID/AOR
>    pair, and for each instance ID/AOR pair, there can be one or more
>    GRUU.
> 
> (The picture doesn't show anything associated with an AOR/Instance ID 
> pair.)

Right. In earlier versions of GRUU I used to have an object called 
instance ID/AOR pair. It cluttered the picture way too much and made it 
even harder to understand.

> 
> I think it is too hard to make a picture that expresses these 
> relationships accurately, and this should be done with text, without the 
> figure.

It is in the text. Are you suggesting removing the figure entirely?

> 
> Section 6:
> 
>    o  For each GRUU, both the SIP and Session Initiation Protocol Secure
>       (SIPS) versions MUST exist.  The SIPS URI may not always work,
>       particularly if the proxy cannot establish a secure connection to
>       the client.  However, the SIPS URI always exists.
> 
> The use of "always" seems problematic. The lifetime of the gruu itself 
> is not "always". It is no longer than the associated AOR. Also, this 
> goes into the previous discussion of relationships. Are the SIP and SIPS 
> forms *separate* AORs, or the same AOR? I'm struggling here to come up 
> with something that threads the needle. How about:
> 
>    o  When both SIP and SIPS AORs identify the same resource, and the
>       AORs differ only in the scheme name, then when a GRUU is assigned
>       to the resource, both SIP and SIPS GRUUs will be assigned. Only
>       one will be returned, but both will exist.  The SIPS URI may not
>       always work, particularly if the proxy cannot establish a secure
>       connection to the client.

I think the first bit of this is not needed. It makes it sound like its 
OK for a SIP and SIPS AOR to not identify the same resource. I think 
we've concluded all along (and in the sipsec adhoc in Montreal) that 
they always do.

How about:

	    <t>When a GRUU is assigned to an instance ID/AOR pair,
	    both SIP and SIPS GRUUs will be assigned. Only one will be
	    returned, but both will exist.  The SIPS URI may not
	    always work, particularly if the proxy cannot establish a
	    secure connection to the client.
	    </t>

> 
> Section 7.1.1.1:
> 
>    Specifically, the contact in the REGISTER request SHOULD
>    NOT contain the gruu Contact header field parameter, and the contact
>    URI itself SHOULD NOT contain the "grid" URI parameter defined below.
>    Any such parameters are ignored by the registrar, as the UA cannot
>    propose a GRUU for usage with the contact.
> 
> Nit: the last sentence is a little weird, because the "as" clause only 
> applies to gruu, not grid. Perhaps:
> 
>    Any such parameters are ignored by the registrar and not stored.
>    (The UA cannot propose a GRUU for usage with the contact, and
>    the grid parameter is populated from the GRUU, not the registered
>    contact.)

I just removed the 'as' bit entirely.

> 
> BUT, later I instead suggest treating the presence of grid as an error 
> rather than ignoring it. That requires different wording here.

No, I think its OK. See below.

> 
> Later in the section:
> 
>    ... However, it is
>    permitted for a UA to register any kind of contact it wishes,
>    including a GRUU.  A typical use case for registering a GRUU is if a
>    user wanted calls to one AOR to be delivered to a specific UA
>    instance associated with another AOR.
> 
> I just realized this presents a problem wrt grid processing. If the 
> contact is a gruu associated with a different AOR, then it could also 
> contain a grid parameter, and if so, that parameter might be important. 
> One wouldn't want to lose it, or override it.

Right.

> 
> Perhaps it should be an *error* to register a contact that contains a 
> grid URI parameter. That would prevent any confusion.

I think we need to prevent it. The current spec is wrong in this regard; 
it appears to allow it, but it would cause overriding which is wrong.

So, here is my proposal. I think there are two cases here. One, where 
the contact is a 'plain old contact' and is not bound to a gruu at all. 
In such a case, the contact would not contain the instance ID and can 
contain gruu and grid params if the contact happens to be a gruu. No 
gruu gets bound to this contact. If the contact is a specific instance, 
then it contains the instance ID, in which case it cannot contain a gruu 
or grid. With this change, btw, you can register a GRUU to its own AOR, 
and it works as expected. I changed the text to reflect this. It now reads:

			<t>Besides the procedures discussed above, the
		REGISTER request is constructed as it is in the case
		where this extension was not understood. Specifically,
		if a contact contains an instance ID parameter, then
		it SHOULD NOT contain the gruu Contact header field
		parameter, and the contact URI itself SHOULD NOT
		contain the "grid" or "gruu" URI parameters defined below. Any
		such parameters are ignored by the registrar and not
		stored.  Typically, the contact URI is not itself a
		GRUU and will not contain
		these URI parameters. There are use cases where a
		contact can contain a GRUU, however. A typical
		use case is when a user wanted calls to one AOR to be
		delivered to a specific UA instance associated with
		another AOR. If a client does include a contact which
		contains a GRUU, it MUST NOT contain the
		instance ID parameter. This will ensure that the
		contact URI does not get bound to any GRUU, avoiding
		an infinite loop which might otherwise occur.
		</t>


I made updates in a few other places too; most notably where it talks 
about overriding the grid in the contact. That can never happen with the 
change here.

> 
> Section 7.1.1.2:
> 
>    ...  The URI will be a SIP URI if the To header
>    field in the REGISTER request contained a SIP URI, else (if the To
>    header field in the REGISTER request contained a SIPS URI) it will be
>    a SIPS URI.
> 
> (more sips stuff)

Per above I propose to keep this.

> 
>    ... Normally, the GRUU will
>    not change in subsequent 2xx responses to REGISTER.  Indeed, even if
>    the UA lets the contact expire, when it re-registers it at any later
>    time, the registrar will normally provide the same GRUU for the same
>    address-of-record and instance ID.  However, as discussed above, this
>    property cannot be completely guaranteed, as network failures may
>    make it impossible to provide an identifier that persists for all
>    time.  As a result, a UA MUST be prepared to receive a different GRUU
>    for the same instance ID/AOR pair in a subsequent registration
>    response.
> 
> As written, this puts a burden on a UA. Any time it refreshes a 
> registration a previously assigned gruu may change. That gruu may be in 
> use in a dialog. Then the UA would have to to a target refresh to fix 
> it. That is a lot to ask, and has potential race conditions if the prior 
> gruu has become invalid.
> 
> IMO it would be better to put the burden on the registrar to keep the 
> gruu the same for as long as a registration remains active.

Well, registrations can last almost indefinitely (at least until there 
is a power failure). Thus, I'm not sure this changes much practically 
speaking.

I also think that the principle of robustness favors keeping the current 
text. By having endpoints prepare for this, the system works in various 
degrees of server reliability.

Actually, this got me thinking. I was about to write, "note also that a 
dialog can exist without a registration, so that you could have a dialog 
in place when a registration expires". This may be true *prior* to GRUU, 
but with GRUU, it is actually false! Because you are using a gruu as the 
remote contact, mid-dialog requests require the home proxy to have a 
valid registration in order to handle the request. Thus, gruu 
effectively requires registrations for sending or receiving a dialog 
forming request. This is similar to the 3gpp constraint, which many 
folks from IETF fought long and hard against. Funny that, between this 
and sip-outbound, we've basically arrived at the same place anyway.

It is worth pointing that out. I've added the following text in the 
front of 8.1.1.:

	      <t>
		A UA first obtains a GRUU using the procedures of
		<xref target="sec-reg"/>, or by other means outside the scope of this
		specification. If a UA is using registrations to
		obtain GRUUs, it MUST have an active registration
		prior to using a GRUU. It cannot simply reuse a
		previously cached GRUU, even if the URI itself doesn't
		ever change. This is because mid-dialog requests, which
		will be targeted at a GRUU, cannot be processed
		correctly at the home proxy unless a registration is
		active.
	      </t>





> 
> I think it might also be good to insert a note into this section:
> 
>     NOTE: the 2xx response to REGISTER returns all the contacts
>     bound to the AOR. This may include contacts bound by other
>     UAs. Any or all of these may include a "gruu" Contact header field
>     parameter. The UA is responsible for extracting and saving the
>     GRUU assigned to the contact(s) it has registered.

Yes, something like this needs to be said. I rather it not be a 'NOTE' 
since that tends to confuse folks about whether it is informational or 
normative behavior. In fact, you are pointing out an important normative 
thing. So, I put some normative text up front:

		<t>The 2xx response to REGISTER returns all the
		contacts bound to the AOR. This may include contacts
		bound by other UAs. The UA MUST extract the contacts
		it has just registered, and perform the processing on
		those contacts as described here. </t>

		<t> If the response is a 2xx, each Contact header
		field that contained the "+sip.instance" Contact


> 
> Section 7.1.2:
> 
>    A registrar MUST create both the SIP and SIPS versions of the GRUU,
>    such that if the GRUU exists, both URI exist.
> 
> (more sips stuff)
> 
> Section 7.1.2.1:
> 
> Nothing is mentioned in this section about handling of a "grid" URI 
> parameter in the contact. If this is present it can screw things up in a 
> variety of ways. This is sufficiently severe that it probably isn't wise 
> to just ignore it if present. Instead, should probably reject the 
> request. (403 again?) The problems that can result:
> 
> - regular calls to the AOR will look (to the UE) like calls to a gruu
> - if the contact was itself a GRUU, and had a particular grid value
>   for some purpose, then when calls are made to the gruu assigned as
>   part of this registration, the grid value will be messed up.
> 
> (Note that forbidding grid values here doesn't fully solve the problem. 
> Using a gruu as a contact in another registration has unresolvable 
> problems about the namespace for grid values. It can only work right if 
> the two share a common namespace for the values.)

You are right that the spec doesn't say what should happen. Per my above 
suggestion, I've specifide now that if there is an instance ID, you 
ignore and don't store the gruu and grid param.


> 
> Same section:
> 
>    ... If the To
>    header field in the REGISTER request contains a SIP URI, the SIP
>    version of the GRUU is returned.  If the To header field in the
>    REGISTER request contains a SIPS URI, the SIPS version of the GRUU is
>    returned.
> 
> (more sips stuff)
> 
> Section 8.1.1:
> 
>    When using a GRUU as a remote target, a UA MAY add the "grid" URI
> 
> Maybe its just me, but here and elsewhere I am confused by use of the 
> term "remote target". In a dialog, I think of the "remote target" as the 
> R-URI I use to reference the *other* end of the dialog. I would then 
> expect the term "local target" to refer to the URI I expect the other 
> end to use in the R-URI of requests addressed to me. But I think you are 
> using it differently.

I'm using it as defined in the intro (and in rfc3261):

	<t hangText="remote target:">The term "remote target" refers
	to a URI that a user agent uses to identify itself for receipt
	of both mid-dialog and out-of-dialog requests. A remote target
	is established by placing a URI in the Contact header field of
	a dialog-forming request or response and updated by target
	refresh requests.  </t>


> 
> In the above, I am left wondering if others who receive my GRUU will be 
> filling in the grid parameter. Of course this is wrong, but I can read 
> it that way.
> 
> If you agree this is an issue, I guess it will require a careful 
> laundering of all the places that use the term "remote target".

In this particular paragraph I see how it can be confusing. I reworded to:

	      <t> When using a GRUU as a remote target (that is,
	      when populating the Contact header field of an INVITE or
	      200 OK response), the UA placing the GRUU into the
	      request or response MAY add the "grid" URI
	      parameter to the GRUU, and if one is added, MUST include a value for
	      it. This parameter MAY take any value permitted by the grammar 
defined



> 
> Section 8.2.1:
> 
> First, I think it would be worthwhile to add a statement at the 
> beginning of this section stating that this only applies when there are 
> no Route headers. Maybe this is obvious, but maybe not. Stating this 
> removes any ambiguity about how a Path header interacts with existing 
> Route headers.

This ties back into the much bigger issue about what to do with 
pre-loaded route headers in dialog forming requests. For now I think it 
suffices to say that this section is in fact not addressing what happens 
if the request also contains route headers. But if it does, its hard to 
say what the right hting is. In some cases there may be legitimate 
reasons for it that are not attacks.

For now I will merely clarify as you suggest. The specific words I used 
were:

	      <t> The processing in this section applies to
	      requests that are received outside of a dialog. These
	      requests are identified by their lack of a Route header
	      field when received by the proxy. </t>

This doesn't say its wrong to receive a dialog-forming request without a 
route; it simply says that the rules here don't apply to that case.

> 
> The first paragraph could be interpreted as an *alternative* to the 
> cases mentioned in the 2nd and 3rd paragraphs. I believe it is not 
> intended that way, and is instead just a modifier that applies 
> regardless of which of the other cases are followed. I'm not sure that 
> this paragraph even needs to be here. It places a dependency from this 
> draft on -outbound behavior over and above the definition of instance 
> id. The third paragraph already has language that covers the interaction 
> case when both drafts are mucking with the same behavior.

Agree that the first paragraph lends nothing. I deleted it.

> 
> Later in the section
> 
>    ... If the grid was already present
>    in the contact bound to the GRUU, it is overwritten in this process.
> 
> As I noted earlier, this seems dangerous to me, so I think it should be 
> prevented earlier.

Yes, this was removed.

> 
> Section 8.2.2:
> 
>    The proxy that accesses the location service (called the home proxy
>    here) MUST record-route under two circumstances.  Firstly, if the
>    home proxy receives a dialog forming request from a UA in its own
>    domain (an originating request) whose Contact header field contains a
>    GRUU (indicated by the presence of the "gruu" URI parameter), the
>    proxy MUST record-route.
> 
> This really only applies if the proxy is responsible for the GRUU.

The words already apply only in cases where the sending UA is in its own 
domain. So, the only case when this is true, but it wouldn't be 
responsible for the GRUU, is if the UA used a different GRUU. For 
example, I'm a customer of abc.com and I happen to get a gruu from 
xyz.com. I then send an INVITE with a contact of xyz.com. The rules in 
the UA processing section require that this request visit abc.com prior 
to being sent. Its still possible that xyz.com home proxy is on the path 
too. In such an odd case, I've clarified that the proxy checks the 
domain of the gruu too.

Thanks for your comments,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Mon Jul 31 16:19:46 2006
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I think it can retry automatically if it was at the request of the user 
to do so. So, I agree with 'explicitly' but that doesn't rule out 
'automatically'. I'll clarify in the security considerations section.

Thanks,
Jonathan R.

Jeroen van Bemmel wrote:

> Jonathan,
> 
> (Only) one comment: What about privacy service behavior? In particular I am
> thinking about a privacy service that anonymized the request, and 
> receives a
> 433.
> 
> If the privacy service is acting as a B2BUA, it should probably not retry
> the request "non-anonymously" (this could constitute a downgrade attack).
> This is also something worth mentioning in the security considerations: the
> retrying of the request SHOULD be based on an explicit decision by the user
> making the request, it SHOULD NOT be done automatically
> 
> Regards,
> 
> Jeroen
> 
> ----- Original Message ----- From: "Jonathan Rosenberg" <jdrosen@cisco.com>
> To: "IETF SIP List" <sip@ietf.org>
> Sent: Tuesday, June 20, 2006 4:42 AM
> Subject: [Sip] Updated acr-code draft
> 
> 
>> I've just submitted a minor update to the ACR code draft. Until it 
>> appears, you can pick it up at:
>>
>> http://www.jdrosen.net/papers/draft-ietf-sip-acr-code-01.txt
>>
>> The changes were to add the required RFC 2119 boilerplate and fix some 
>> spell check errors. The document is complete AFAIK and has had no 
>> discussion since the last revision. I believe it ready for WGLC.
>>
>> -Jonathan R.
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>> Cisco Fellow                                   Parsippany, NJ 07054-2711
>> Cisco Systems
>> jdrosen@cisco.com                              FAX:   (973) 952-5050
>> http://www.jdrosen.net                         PHONE: (973) 952-5000
>> http://www.cisco.com
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip 
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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From sip-bounces@ietf.org Mon Jul 31 16:32:42 2006
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I've submitted a minor update to the ACR code draft. Until it appears, 
you can pick it up at:

http://www.jdrosen.net/papers/draft-ietf-sip-acr-code-02.txt

the changes are:

* replaced P-Asserted-ID with P-Asserted-Identity

* added to security considerations section, consideration
for anonymizers in the network. Namely, don't retry
request on receipt of 433 unless explicitly configured
to do so by user


Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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From sip-bounces@ietf.org Mon Jul 31 16:39:51 2006
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Just a quick comment:

   request without requesting anonymity.  It SHOULD only do so if it
    obtains confirmation from the user that this is desirable.  Such
    confirmation could be obtained through the user interface, or by
    accessing user defined policy.  The UAC SHOULD NOT retry the request
    if user continues to request anonymity.

No guidance is given when the UAC might just retry the request without 
anonymity and without asking. Is "because it feels like it" a good 
enough reason :-)? Can a vendor just decide that all its products don't 
really feel that strongly about anonymity or is this something that only 
users (or administrators) should be able to configure?

I think the right answer is that the UAC MUST NOT do this automatically 
unless the user or administrator has configured an automatic 
non-anonymous retry.


Jonathan Rosenberg wrote:
> I've submitted a minor update to the ACR code draft. Until it appears, 
> you can pick it up at:
> 
> http://www.jdrosen.net/papers/draft-ietf-sip-acr-code-02.txt
> 
> the changes are:
> 
> * replaced P-Asserted-ID with P-Asserted-Identity
> 
> * added to security considerations section, consideration
> for anonymizers in the network. Namely, don't retry
> request on receipt of 433 unless explicitly configured
> to do so by user
> 
> 
> Thanks,
> Jonathan R.

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From: Jonathan Rosenberg <jdrosen@cisco.com>
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I've just submitted an update to the GRUU spec based on comments on the 
list. Until it appears, you can pick it up at:

* noted that, in the outbound direction, if the home proxy is not on
the original request path, another proxy in that domain needs to put in 
a record-route
that points to it. Provide links to some mechanism which allows a home
proxy to get on the outbound request path

* clarified that, if an outbound proxy is used, the request has to
   visit the domain from which the gruu was obtained before being
   forwarded to the party in the r-uri. There are two ways documented
   to know this:

    a. using a service route obtained from the same proxy that sent you
    a gruu

    b. using an outbound proxy; configuration tells you what domain its
    from


* noted that an outbound proxy is not needed; rules above only apply
when one is used

* clarified the interaction of caller prefs and callee caps with gruu

* clarified how a proxy determines that a request is a mid-dialog
request

* clarified that GRUU basically implies you can't have a sip dialog
   without registration

* clarified that a UA, in processing a REGISTER response, is going to
   have to search for and extract its own contacts in case other UA
   registered against the same AOR

* changed rules on inclusion of grid/gruu URI param in the contact. A
   registered contact cannot have a grid or gruu URI
   param if its also got an instance ID. If there is no instance ID its
   ok. Correspondingly, removed check on making sure a contact wasn't a
   gruu for the AOR in the To field. This is not necessary now, and in
   fact is permitted and works fine.

* clarified that proxy processing of 'initial' requests applies when
   there are no Route headers. Consequently, treatment of
   dialog-forming requests which happen to have route headers is not
   specified.

* removed rfc2119 strength text from requirements section



Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Mon Jul 31 17:59:13 2006
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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
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	dereferencemechanism defined
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Brian,
=20
Your second paragraph re-opens the sore I am afraid.
A large number of people have agreed with Henning's statements and these ar=
e clear:
1) Do not constrain the ABNF
2) Provide minimal description of the reference function.
=20
If you insist on saying then that the ABNF will be restricted if there is n=
ot matching text, then I think it unlikely that you will reach a consensus.
=20
Cheers
James

________________________________

From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
Sent: Mon 31/07/2006 8:38 AM
To: IETF SIP List
Cc: GEOPRIV
Subject: [Geopriv] Does Location-Conveyance have an HTTP dereferencemechani=
sm defined



I have lost track of where we stand on whether the only mentioned URI
schemes in location-conveyance are sip/sips/pres.  I "think" we have
agreed that an HTTP/HTTPS dereference is in a separate document, but I'm
not sure.

This is NOT the discussion of what the ABNF of the location header looks
like (although it would NOT have http/https in it for sure if we agree
that an HTTP dereference protocol is a separate draft).

If you think location-conveyance DOES have HTTP dereference text in it,
please say so, and please explain why you would not be satisfied if
there were a separate draft on the subject.

Brian

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Geopriv mailing list
Geopriv@ietf.org
https://www1.ietf.org/mailman/listinfo/geopriv



---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
contain privileged, proprietary, or otherwise private information. =20
If you have received it in error, please notify the sender
immediately and delete the original.  Any unauthorized use of
this email is prohibited.
---------------------------------------------------------------------------=
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[mf2]

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From sip-bounces@ietf.org Mon Jul 31 18:07:31 2006
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
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	dereferencemechanism defined
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Please don't change the subject, I understand the controversy on
constraining the ABNF.  I tried (apparently without success) to separate
those issues. =20

I'm asking if anyone objects to NOT mentioning HTTP in any way with
respect to dereferencing URIs carried by the Location header.  That is
the only question I asked. =20

I'm not trying to evade the constrained ABNF issue, and I'm not implying
anything tricky.  I simply want to know if it's okay if we leave
anything that deals specifically with HTTP and the Location header URI
to another document.

Brian

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Monday, July 31, 2006 5:55 PM
> To: Rosen, Brian; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> dereferencemechanism defined
>=20
> Brian,
>=20
> Your second paragraph re-opens the sore I am afraid.
> A large number of people have agreed with Henning's statements and
these
> are clear:
> 1) Do not constrain the ABNF
> 2) Provide minimal description of the reference function.
>=20
> If you insist on saying then that the ABNF will be restricted if there
is
> not matching text, then I think it unlikely that you will reach a
> consensus.
>=20
> Cheers
> James
>=20
> ________________________________
>=20
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Mon 31/07/2006 8:38 AM
> To: IETF SIP List
> Cc: GEOPRIV
> Subject: [Geopriv] Does Location-Conveyance have an HTTP
> dereferencemechanism defined
>=20
>=20
>=20
> I have lost track of where we stand on whether the only mentioned URI
> schemes in location-conveyance are sip/sips/pres.  I "think" we have
> agreed that an HTTP/HTTPS dereference is in a separate document, but
I'm
> not sure.
>=20
> This is NOT the discussion of what the ABNF of the location header
looks
> like (although it would NOT have http/https in it for sure if we agree
> that an HTTP dereference protocol is a separate draft).
>=20
> If you think location-conveyance DOES have HTTP dereference text in
it,
> please say so, and please explain why you would not be satisfied if
> there were a separate draft on the subject.
>=20
> Brian
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>=20
>=20
>
------------------------------------------------------------------------
--
> ----------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
>
------------------------------------------------------------------------
--
> ----------------------
> [mf2]

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From sip-bounces@ietf.org Mon Jul 31 18:23:14 2006
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Not true
=20
"> This is NOT the discussion of what the ABNF of the location header
looks
> like (although it would NOT have http/https in it for sure if we agree
> that an HTTP dereference protocol is a separate draft)."
=20
This is a discussion on ABNF and what the document contains, and to that ex=
tent they are not separable.
The consensus has been, do not restrict the ABNF, keep the discussion in th=
e document about references limited. I am failing to understand why you can=
't accept that.
=20
=20

________________________________

From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
Sent: Mon 31/07/2006 5:05 PM
To: Winterbottom, James; IETF SIP List
Cc: GEOPRIV
Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP dereferencemec=
hanism defined



Please don't change the subject, I understand the controversy on
constraining the ABNF.  I tried (apparently without success) to separate
those issues.=20

I'm asking if anyone objects to NOT mentioning HTTP in any way with
respect to dereferencing URIs carried by the Location header.  That is
the only question I asked.=20

I'm not trying to evade the constrained ABNF issue, and I'm not implying
anything tricky.  I simply want to know if it's okay if we leave
anything that deals specifically with HTTP and the Location header URI
to another document.

Brian

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Monday, July 31, 2006 5:55 PM
> To: Rosen, Brian; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> dereferencemechanism defined
>
> Brian,
>
> Your second paragraph re-opens the sore I am afraid.
> A large number of people have agreed with Henning's statements and
these
> are clear:
> 1) Do not constrain the ABNF
> 2) Provide minimal description of the reference function.
>
> If you insist on saying then that the ABNF will be restricted if there
is
> not matching text, then I think it unlikely that you will reach a
> consensus.
>
> Cheers
> James
>
> ________________________________
>
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Mon 31/07/2006 8:38 AM
> To: IETF SIP List
> Cc: GEOPRIV
> Subject: [Geopriv] Does Location-Conveyance have an HTTP
> dereferencemechanism defined
>
>
>
> I have lost track of where we stand on whether the only mentioned URI
> schemes in location-conveyance are sip/sips/pres.  I "think" we have
> agreed that an HTTP/HTTPS dereference is in a separate document, but
I'm
> not sure.
>
> This is NOT the discussion of what the ABNF of the location header
looks
> like (although it would NOT have http/https in it for sure if we agree
> that an HTTP dereference protocol is a separate draft).
>
> If you think location-conveyance DOES have HTTP dereference text in
it,
> please say so, and please explain why you would not be satisfied if
> there were a separate draft on the subject.
>
> Brian
>
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv
>
>
>
>
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> ----------------------
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> contain privileged, proprietary, or otherwise private information.
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> immediately and delete the original.  Any unauthorized use of
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---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
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If you have received it in error, please notify the sender
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From sip-bounces@ietf.org Mon Jul 31 18:46:49 2006
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "Winterbottom, James" <James.Winterbottom@andrew.com>,
	"IETF SIP List" <sip@ietf.org>
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Subject: [Sip] RE: [Geopriv] URI restrictions in location conveyance,
	was Does Location-Conveyance have an HTTP
	dereferencemechanism defined
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Can't let go can you?  I changed the subject line to start a different
thread.  I'd still like the answer to the original thread.

There are two separable issues with respect to the ABNF.  The syntactic
and the semantic.

The syntactic issues could be dealt with by saying:

location-param =3D LAQUOT loc-URI RAQUOT *( SEMI loc-param )=20
;cid-url from RFC2392, SIP/SIPS-URI from RFC3261, PRES-URI from RFC3859=20
;Absolute URI from RFC3261
loc-URI =3D cid-uri / SIP-URI / SIPS-URI / PRES-URI / AbsoluteURI

This answers Henning and Jeroen's objection to strictly limiting the
ABNF.
The alternative is to just use AbsoluteURI.

I guess the question is, what does the text say? I can make the ABNF
match what the text says.

Does the text say: any URI can appear in the location header, entities
must use some undefined mechanism to maintain interoperability.

Does it say: location dereferencing is a "using protocol" as defined in
geopriv requirements.  Any URI found in the location header MUST meet
the requirements of a using protocol.

Does it say: This document describes the use of SIP and SIPS as a
dereferencing protocol (and the use of a pres: uri if the resolution of
the pres: scheme in the referenced domain is to a presence service using
the sip/sips protocol).  All entities using the Location header MUST
support SIP dereferencing. =20

Does it say (if the paragraph immediately above is there): Other
protocols used for dereferencing MUST have a (?standards track?) RFC
defining how to dereference.

Brian

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Monday, July 31, 2006 6:19 PM
> To: Rosen, Brian; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> dereferencemechanism defined
>=20
> Not true
>=20
> "> This is NOT the discussion of what the ABNF of the location header
> looks
> > like (although it would NOT have http/https in it for sure if we
agree
> > that an HTTP dereference protocol is a separate draft)."
>=20
> This is a discussion on ABNF and what the document contains, and to
that
> extent they are not separable.
> The consensus has been, do not restrict the ABNF, keep the discussion
in
> the document about references limited. I am failing to understand why
you
> can't accept that.
>=20
>=20
>=20
> ________________________________
>=20
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Mon 31/07/2006 5:05 PM
> To: Winterbottom, James; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> dereferencemechanism defined
>=20
>=20
>=20
> Please don't change the subject, I understand the controversy on
> constraining the ABNF.  I tried (apparently without success) to
separate
> those issues.
>=20
> I'm asking if anyone objects to NOT mentioning HTTP in any way with
> respect to dereferencing URIs carried by the Location header.  That is
> the only question I asked.
>=20
> I'm not trying to evade the constrained ABNF issue, and I'm not
implying
> anything tricky.  I simply want to know if it's okay if we leave
> anything that deals specifically with HTTP and the Location header URI
> to another document.
>=20
> Brian
>=20
> > -----Original Message-----
> > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > Sent: Monday, July 31, 2006 5:55 PM
> > To: Rosen, Brian; IETF SIP List
> > Cc: GEOPRIV
> > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > dereferencemechanism defined
> >
> > Brian,
> >
> > Your second paragraph re-opens the sore I am afraid.
> > A large number of people have agreed with Henning's statements and
> these
> > are clear:
> > 1) Do not constrain the ABNF
> > 2) Provide minimal description of the reference function.
> >
> > If you insist on saying then that the ABNF will be restricted if
there
> is
> > not matching text, then I think it unlikely that you will reach a
> > consensus.
> >
> > Cheers
> > James
> >
> > ________________________________
> >
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Mon 31/07/2006 8:38 AM
> > To: IETF SIP List
> > Cc: GEOPRIV
> > Subject: [Geopriv] Does Location-Conveyance have an HTTP
> > dereferencemechanism defined
> >
> >
> >
> > I have lost track of where we stand on whether the only mentioned
URI
> > schemes in location-conveyance are sip/sips/pres.  I "think" we have
> > agreed that an HTTP/HTTPS dereference is in a separate document, but
> I'm
> > not sure.
> >
> > This is NOT the discussion of what the ABNF of the location header
> looks
> > like (although it would NOT have http/https in it for sure if we
agree
> > that an HTTP dereference protocol is a separate draft).
> >
> > If you think location-conveyance DOES have HTTP dereference text in
> it,
> > please say so, and please explain why you would not be satisfied if
> > there were a separate draft on the subject.
> >
> > Brian
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
> >
> >
> >
> >
>
------------------------------------------------------------------------
> --
> > ----------------------
> > This message is for the designated recipient only and may
> > contain privileged, proprietary, or otherwise private information.
> > If you have received it in error, please notify the sender
> > immediately and delete the original.  Any unauthorized use of
> > this email is prohibited.
> >
>
------------------------------------------------------------------------
> --
> > ----------------------
> > [mf2]
>=20
>=20
>=20
>
------------------------------------------------------------------------
--
> ----------------------
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> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
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>
------------------------------------------------------------------------
--
> ----------------------
> [mf2]

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From sip-bounces@ietf.org Mon Jul 31 18:47:20 2006
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Subject: RE: [Sip] Re: Review of draft-ietf-sip-gruu-09
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I'm keeping my comments to sips-related issues.=20

See below.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Monday, July 31, 2006 1:15 PM
> To: Paul Kyzivat
> Cc: SIP IETF
> Subject: [Sip] Re: Review of draft-ietf-sip-gruu-09
>=20
> > SIPS is mentioned in many places in this draft. Given the current=20
> > ongoing issues wrt SIPS, the language here for dealing=20
> with it may=20
> > prove to be prescient, or may turn out to just be wrong.=20
> Can this be=20
> > dodged somehow? Maybe this whole draft should only be=20
> defined for SIP,=20
> > with SIPS left for a separate draft. (I will still=20
> comment on some=20
> > places containing SIPS, when there are other points than=20
> the above.)
>=20
> Well, it used to say nothing about sips and I think you=20
> complained it needed to a long time back ;)
>=20
> I'm not awful inclined to hold the draft up for us to=20
> figure all of that out, and I'm also not inclined to say=20
> nothing. My preference is to address it as best we can=20
> based on the current mechanism, and the in-progress sips=20
> update will need to tackle gruu.

Yeah, it's a fine line we are walking here... See below.

> > Section 6:
> >=20
> >    o  For each GRUU, both the SIP and Session Initiation=20
> Protocol Secure
> >       (SIPS) versions MUST exist.  The SIPS URI may not=20
> always work,
> >       particularly if the proxy cannot establish a secure=20
> connection to
> >       the client.  However, the SIPS URI always exists.
> >=20
> > The use of "always" seems problematic. The lifetime of=20
> the gruu itself=20
> > is not "always". It is no longer than the associated AOR.=20
> Also, this=20
> > goes into the previous discussion of relationships. Are=20
> the SIP and SIPS=20
> > forms *separate* AORs, or the same AOR? I'm struggling=20
> here to come up=20
> > with something that threads the needle. How about:
> >=20
> >    o  When both SIP and SIPS AORs identify the same=20
> resource, and the
> >       AORs differ only in the scheme name, then when a=20
> GRUU is assigned
> >       to the resource, both SIP and SIPS GRUUs will be=20
> assigned. Only
> >       one will be returned, but both will exist.  The=20
> SIPS URI may not
> >       always work, particularly if the proxy cannot=20
> establish a secure
> >       connection to the client.
>=20
> I think the first bit of this is not needed. It makes it=20
> sound like its=20
> OK for a SIP and SIPS AOR to not identify the same=20
> resource. I think=20
> we've concluded all along (and in the sipsec adhoc in=20
> Montreal) that=20
> they always do.

I wouldn't say it was "all along", but it is definitively
what we concluded out of the Montr=E9al sipsec BOF.

> How about:
>=20
> 	    <t>When a GRUU is assigned to an instance ID/AOR pair,
> 	    both SIP and SIPS GRUUs will be assigned. Only one will be
> 	    returned, but both will exist.  The SIPS URI may not
> 	    always work, particularly if the proxy cannot establish a
> 	    secure connection to the client.
> 	    </t>

That seems OK with me.

I guess one could argued that both are "reserved", as opposed to =
"assigned".=20
You reserve both sip and sips, and only return one. Maybe I would say =
that
they may not be "usable" instead "working".

Something like:
 	    <t>When a GRUU is assigned to an instance ID/AOR pair,
 	    both SIP and SIPS GRUUs will be reserved. Only one will be
 	    returned, but both will exist.  The SIPS URI may not
 	    always be usable, particularly if the proxy cannot establish a
 	    secure connection to the client.
 	    </t>

But I'm fine either way.

Key is that there is one entry in the table for both shemes, as opposed =
to 2 entries.

> > Section 7.1.1.2:
> >=20
> >    ...  The URI will be a SIP URI if the To header
> >    field in the REGISTER request contained a SIP URI,=20
> else (if the To
> >    header field in the REGISTER request contained a SIPS=20
> URI) it will be
> >    a SIPS URI.
> >=20
> > (more sips stuff)
>=20
> Per above I propose to keep this.

Like Paul, I am a little worried about the following statement:

      The URI will be a SIP URI if the To header
      field in the REGISTER request contained a SIP URI, else (if the To
      header field in the REGISTER request contained a SIPS URI) it will =
be
      a SIPS URI.=20

First, it is not written as a SHOULD or MUST (but as a "will"). And more =
importantly,
this is where the parallel text concerning contacts and sips in RRFC =
3261/10.2.1 is kind
of fuzzy. It says:

      If the address-of-record in the To header field of a REGISTER
      request is a SIPS URI, then any Contact header field values in the
      request SHOULD also be SIPS URIs.  Clients should only register
      non-SIPS URIs under a SIPS address-of-record when the security of
      the resource represented by the contact address is guaranteed by
      other means.  This may be applicable to URIs that invoke protocols
      other than SIP, or SIP devices secured by protocols other than
      TLS.

I think we have 3 choices:
(1) We clean up the language so that it matches the spirit of the same =
text in 3261
    (i.e., SHOULD with the fuzzy language).
(2) We have a statement that the same rule as for Contacts applies
(3) We remore the offending statement.

Personally, I believe options (2) and (3) are much better. I prefer (3).

> > Section 7.1.2:
> >=20
> >    A registrar MUST create both the SIP and SIPS versions=20
> of the GRUU,
> >    such that if the GRUU exists, both URI exist.
> >=20
> > (more sips stuff)
> >=20
> > Section 7.1.2.1:
> >    ... If the To
> >    header field in the REGISTER request contains a SIP=20
> URI, the SIP
> >    version of the GRUU is returned.  If the To header field in the
> >    REGISTER request contains a SIPS URI, the SIPS version=20
> of the GRUU is
> >    returned.
> >=20
> > (more sips stuff)

Same comment as above apply. I'm not sure we need this statement here.=20
If we do decide to keep it, it is a MUST or a SHOULD? It shouldn't be =
this
passive form "is".


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Just using absoluteuri seems like the complete simplification...

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Tuesday, 1 August 2006 8:44 AM
> To: Winterbottom, James; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] URI restrictions in location conveyance,was
Does
> Location-Conveyance have an HTTP dereferencemechanism defined
>=20
> Can't let go can you?  I changed the subject line to start a different
> thread.  I'd still like the answer to the original thread.
>=20
> There are two separable issues with respect to the ABNF.  The
syntactic
> and the semantic.
>=20
> The syntactic issues could be dealt with by saying:
>=20
> location-param =3D LAQUOT loc-URI RAQUOT *( SEMI loc-param )
> ;cid-url from RFC2392, SIP/SIPS-URI from RFC3261, PRES-URI from
RFC3859
> ;Absolute URI from RFC3261
> loc-URI =3D cid-uri / SIP-URI / SIPS-URI / PRES-URI / AbsoluteURI
>=20
> This answers Henning and Jeroen's objection to strictly limiting the
> ABNF.
> The alternative is to just use AbsoluteURI.
>=20
> I guess the question is, what does the text say? I can make the ABNF
> match what the text says.
>=20
> Does the text say: any URI can appear in the location header, entities
> must use some undefined mechanism to maintain interoperability.
>=20
> Does it say: location dereferencing is a "using protocol" as defined
in
> geopriv requirements.  Any URI found in the location header MUST meet
> the requirements of a using protocol.
>=20
> Does it say: This document describes the use of SIP and SIPS as a
> dereferencing protocol (and the use of a pres: uri if the resolution
of
> the pres: scheme in the referenced domain is to a presence service
using
> the sip/sips protocol).  All entities using the Location header MUST
> support SIP dereferencing.
>=20
> Does it say (if the paragraph immediately above is there): Other
> protocols used for dereferencing MUST have a (?standards track?) RFC
> defining how to dereference.
>=20
> Brian
>=20
> > -----Original Message-----
> > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > Sent: Monday, July 31, 2006 6:19 PM
> > To: Rosen, Brian; IETF SIP List
> > Cc: GEOPRIV
> > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > dereferencemechanism defined
> >
> > Not true
> >
> > "> This is NOT the discussion of what the ABNF of the location
header
> > looks
> > > like (although it would NOT have http/https in it for sure if we
> agree
> > > that an HTTP dereference protocol is a separate draft)."
> >
> > This is a discussion on ABNF and what the document contains, and to
> that
> > extent they are not separable.
> > The consensus has been, do not restrict the ABNF, keep the
discussion
> in
> > the document about references limited. I am failing to understand
why
> you
> > can't accept that.
> >
> >
> >
> > ________________________________
> >
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Mon 31/07/2006 5:05 PM
> > To: Winterbottom, James; IETF SIP List
> > Cc: GEOPRIV
> > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > dereferencemechanism defined
> >
> >
> >
> > Please don't change the subject, I understand the controversy on
> > constraining the ABNF.  I tried (apparently without success) to
> separate
> > those issues.
> >
> > I'm asking if anyone objects to NOT mentioning HTTP in any way with
> > respect to dereferencing URIs carried by the Location header.  That
is
> > the only question I asked.
> >
> > I'm not trying to evade the constrained ABNF issue, and I'm not
> implying
> > anything tricky.  I simply want to know if it's okay if we leave
> > anything that deals specifically with HTTP and the Location header
URI
> > to another document.
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > Sent: Monday, July 31, 2006 5:55 PM
> > > To: Rosen, Brian; IETF SIP List
> > > Cc: GEOPRIV
> > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > dereferencemechanism defined
> > >
> > > Brian,
> > >
> > > Your second paragraph re-opens the sore I am afraid.
> > > A large number of people have agreed with Henning's statements and
> > these
> > > are clear:
> > > 1) Do not constrain the ABNF
> > > 2) Provide minimal description of the reference function.
> > >
> > > If you insist on saying then that the ABNF will be restricted if
> there
> > is
> > > not matching text, then I think it unlikely that you will reach a
> > > consensus.
> > >
> > > Cheers
> > > James
> > >
> > > ________________________________
> > >
> > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > Sent: Mon 31/07/2006 8:38 AM
> > > To: IETF SIP List
> > > Cc: GEOPRIV
> > > Subject: [Geopriv] Does Location-Conveyance have an HTTP
> > > dereferencemechanism defined
> > >
> > >
> > >
> > > I have lost track of where we stand on whether the only mentioned
> URI
> > > schemes in location-conveyance are sip/sips/pres.  I "think" we
have
> > > agreed that an HTTP/HTTPS dereference is in a separate document,
but
> > I'm
> > > not sure.
> > >
> > > This is NOT the discussion of what the ABNF of the location header
> > looks
> > > like (although it would NOT have http/https in it for sure if we
> agree
> > > that an HTTP dereference protocol is a separate draft).
> > >
> > > If you think location-conveyance DOES have HTTP dereference text
in
> > it,
> > > please say so, and please explain why you would not be satisfied
if
> > > there were a separate draft on the subject.
> > >
> > > Brian
> > >
> > > _______________________________________________
> > > Geopriv mailing list
> > > Geopriv@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/geopriv
> > >
> > >
> > >
> > >
> >
>
------------------------------------------------------------------------
> > --
> > > ----------------------
> > > This message is for the designated recipient only and may
> > > contain privileged, proprietary, or otherwise private information.
> > > If you have received it in error, please notify the sender
> > > immediately and delete the original.  Any unauthorized use of
> > > this email is prohibited.
> > >
> >
>
------------------------------------------------------------------------
> > --
> > > ----------------------
> > > [mf2]
> >
> >
> >
> >
>
------------------------------------------------------------------------
> --
> > ----------------------
> > This message is for the designated recipient only and may
> > contain privileged, proprietary, or otherwise private information.
> > If you have received it in error, please notify the sender
> > immediately and delete the original.  Any unauthorized use of
> > this email is prohibited.
> >
>
------------------------------------------------------------------------
> --
> > ----------------------
> > [mf2]
>=20
> _______________________________________________
> Geopriv mailing list
> Geopriv@ietf.org
> https://www1.ietf.org/mailman/listinfo/geopriv

---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
contain privileged, proprietary, or otherwise private information. =20
If you have received it in error, please notify the sender
immediately and delete the original.  Any unauthorized use of
this email is prohibited.
---------------------------------------------------------------------------=
---------------------
[mf2]

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Mon Jul 31 19:04:10 2006
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Thread-Topic: [Geopriv] URI restrictions in location conveyance,
	was Does Location-Conveyance have an HTTP dereferencemechanism
	defined
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "Dawson, Martin" <Martin.Dawson@andrew.com>,
	"Winterbottom, James" <James.Winterbottom@andrew.com>,
	"IETF SIP List" <sip@ietf.org>
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Cc: GEOPRIV <geopriv@ietf.org>
Subject: [Sip] RE: [Geopriv] URI restrictions in location conveyance,
	was Does Location-Conveyance have an HTTP
	dereferencemechanism defined
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Sure, and so does: "Any URI can be used".  That isn't necessarily a good
thing.

If the text is like the "this document describes how you use
sip/sips/pres" then the ABNF I suggested is more appropriate.

Any ABNF can be significantly reduced if the only thing you are trying
to convey with it is strict syntactic compliance.  No significant
document with large hunks of ABNF in the IETF does that.  We almost
always try to make the ABNF show as much of the semantic meaning as is
reasonable.  If you follow the SIP threads, there are several messages a
month whose answer starts:
"you can't look at the ABNF alone, you have to read the text".  To me,
this is a document failure, unless the ABNF was has to be so ugly to be
completely correct it fails to communicate.  Alas, that happens.  But if
it is possible to simply communicate the semantics in the ABNF, then it
is appropriate to do so.

You can use the "token" construction a whole lot more than we do if all
you want is syntax checking.

Lots of examples in 3261:
Priority        =3D  "Priority" HCOLON priority-value
priority-value  =3D  "emergency" / "urgent" / "normal"
                   / "non-urgent" / other-priority
other-priority  =3D  token


Contact        =3D  ("Contact" / "m" ) HCOLON
                  ( STAR / (contact-param *(COMMA contact-param)))
contact-param  =3D  (name-addr / addr-spec) *(SEMI contact-params)
name-addr      =3D  [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec      =3D  SIP-URI / SIPS-URI / absoluteURI

Let's settle on the words first.  The ABNF will follow.

Brian

> -----Original Message-----
> From: Dawson, Martin [mailto:Martin.Dawson@andrew.com]
> Sent: Monday, July 31, 2006 6:48 PM
> To: Rosen, Brian; Winterbottom, James; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] URI restrictions in location conveyance,was
Does
> Location-Conveyance have an HTTP dereferencemechanism defined
>=20
> Just using absoluteuri seems like the complete simplification...
>=20
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Tuesday, 1 August 2006 8:44 AM
> > To: Winterbottom, James; IETF SIP List
> > Cc: GEOPRIV
> > Subject: RE: [Geopriv] URI restrictions in location conveyance,was
> Does
> > Location-Conveyance have an HTTP dereferencemechanism defined
> >
> > Can't let go can you?  I changed the subject line to start a
different
> > thread.  I'd still like the answer to the original thread.
> >
> > There are two separable issues with respect to the ABNF.  The
> syntactic
> > and the semantic.
> >
> > The syntactic issues could be dealt with by saying:
> >
> > location-param =3D LAQUOT loc-URI RAQUOT *( SEMI loc-param )
> > ;cid-url from RFC2392, SIP/SIPS-URI from RFC3261, PRES-URI from
> RFC3859
> > ;Absolute URI from RFC3261
> > loc-URI =3D cid-uri / SIP-URI / SIPS-URI / PRES-URI / AbsoluteURI
> >
> > This answers Henning and Jeroen's objection to strictly limiting the
> > ABNF.
> > The alternative is to just use AbsoluteURI.
> >
> > I guess the question is, what does the text say? I can make the ABNF
> > match what the text says.
> >
> > Does the text say: any URI can appear in the location header,
entities
> > must use some undefined mechanism to maintain interoperability.
> >
> > Does it say: location dereferencing is a "using protocol" as defined
> in
> > geopriv requirements.  Any URI found in the location header MUST
meet
> > the requirements of a using protocol.
> >
> > Does it say: This document describes the use of SIP and SIPS as a
> > dereferencing protocol (and the use of a pres: uri if the resolution
> of
> > the pres: scheme in the referenced domain is to a presence service
> using
> > the sip/sips protocol).  All entities using the Location header MUST
> > support SIP dereferencing.
> >
> > Does it say (if the paragraph immediately above is there): Other
> > protocols used for dereferencing MUST have a (?standards track?) RFC
> > defining how to dereference.
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > Sent: Monday, July 31, 2006 6:19 PM
> > > To: Rosen, Brian; IETF SIP List
> > > Cc: GEOPRIV
> > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > dereferencemechanism defined
> > >
> > > Not true
> > >
> > > "> This is NOT the discussion of what the ABNF of the location
> header
> > > looks
> > > > like (although it would NOT have http/https in it for sure if we
> > agree
> > > > that an HTTP dereference protocol is a separate draft)."
> > >
> > > This is a discussion on ABNF and what the document contains, and
to
> > that
> > > extent they are not separable.
> > > The consensus has been, do not restrict the ABNF, keep the
> discussion
> > in
> > > the document about references limited. I am failing to understand
> why
> > you
> > > can't accept that.
> > >
> > >
> > >
> > > ________________________________
> > >
> > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > Sent: Mon 31/07/2006 5:05 PM
> > > To: Winterbottom, James; IETF SIP List
> > > Cc: GEOPRIV
> > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > dereferencemechanism defined
> > >
> > >
> > >
> > > Please don't change the subject, I understand the controversy on
> > > constraining the ABNF.  I tried (apparently without success) to
> > separate
> > > those issues.
> > >
> > > I'm asking if anyone objects to NOT mentioning HTTP in any way
with
> > > respect to dereferencing URIs carried by the Location header.
That
> is
> > > the only question I asked.
> > >
> > > I'm not trying to evade the constrained ABNF issue, and I'm not
> > implying
> > > anything tricky.  I simply want to know if it's okay if we leave
> > > anything that deals specifically with HTTP and the Location header
> URI
> > > to another document.
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > > Sent: Monday, July 31, 2006 5:55 PM
> > > > To: Rosen, Brian; IETF SIP List
> > > > Cc: GEOPRIV
> > > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > > dereferencemechanism defined
> > > >
> > > > Brian,
> > > >
> > > > Your second paragraph re-opens the sore I am afraid.
> > > > A large number of people have agreed with Henning's statements
and
> > > these
> > > > are clear:
> > > > 1) Do not constrain the ABNF
> > > > 2) Provide minimal description of the reference function.
> > > >
> > > > If you insist on saying then that the ABNF will be restricted if
> > there
> > > is
> > > > not matching text, then I think it unlikely that you will reach
a
> > > > consensus.
> > > >
> > > > Cheers
> > > > James
> > > >
> > > > ________________________________
> > > >
> > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > Sent: Mon 31/07/2006 8:38 AM
> > > > To: IETF SIP List
> > > > Cc: GEOPRIV
> > > > Subject: [Geopriv] Does Location-Conveyance have an HTTP
> > > > dereferencemechanism defined
> > > >
> > > >
> > > >
> > > > I have lost track of where we stand on whether the only
mentioned
> > URI
> > > > schemes in location-conveyance are sip/sips/pres.  I "think" we
> have
> > > > agreed that an HTTP/HTTPS dereference is in a separate document,
> but
> > > I'm
> > > > not sure.
> > > >
> > > > This is NOT the discussion of what the ABNF of the location
header
> > > looks
> > > > like (although it would NOT have http/https in it for sure if we
> > agree
> > > > that an HTTP dereference protocol is a separate draft).
> > > >
> > > > If you think location-conveyance DOES have HTTP dereference text
> in
> > > it,
> > > > please say so, and please explain why you would not be satisfied
> if
> > > > there were a separate draft on the subject.
> > > >
> > > > Brian
> > > >
> > > > _______________________________________________
> > > > Geopriv mailing list
> > > > Geopriv@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > >
> > > >
> > > >
> > > >
> > >
> >
>
------------------------------------------------------------------------
> > > --
> > > > ----------------------
> > > > This message is for the designated recipient only and may
> > > > contain privileged, proprietary, or otherwise private
information.
> > > > If you have received it in error, please notify the sender
> > > > immediately and delete the original.  Any unauthorized use of
> > > > this email is prohibited.
> > > >
> > >
> >
>
------------------------------------------------------------------------
> > > --
> > > > ----------------------
> > > > [mf2]
> > >
> > >
> > >
> > >
> >
>
------------------------------------------------------------------------
> > --
> > > ----------------------
> > > This message is for the designated recipient only and may
> > > contain privileged, proprietary, or otherwise private information.
> > > If you have received it in error, please notify the sender
> > > immediately and delete the original.  Any unauthorized use of
> > > this email is prohibited.
> > >
> >
>
------------------------------------------------------------------------
> > --
> > > ----------------------
> > > [mf2]
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>=20
>
------------------------------------------------------------------------
--
> ----------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
>
------------------------------------------------------------------------
--
> ----------------------
> [mf2]

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
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	was Does Location-Conveyance have an HTTP
	dereferencemechanism defined
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Brian,
=20
So the two things are linked then?
=20
It is my belief that the consensus is in favour of what Henning suggested, =
and that is what I would like.
Perhaps you would like to include precisely that position in your next ques=
t for consensus, rather than trying to re-word it and losing the intent.
=20
Cheers
James

________________________________

From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
Sent: Mon 31/07/2006 6:01 PM
To: Dawson, Martin; Winterbottom, James; IETF SIP List
Cc: GEOPRIV
Subject: RE: [Geopriv] URI restrictions in location conveyance,was Does Loc=
ation-Conveyance have an HTTP dereferencemechanism defined



Sure, and so does: "Any URI can be used".  That isn't necessarily a good
thing.

If the text is like the "this document describes how you use
sip/sips/pres" then the ABNF I suggested is more appropriate.

Any ABNF can be significantly reduced if the only thing you are trying
to convey with it is strict syntactic compliance.  No significant
document with large hunks of ABNF in the IETF does that.  We almost
always try to make the ABNF show as much of the semantic meaning as is
reasonable.  If you follow the SIP threads, there are several messages a
month whose answer starts:
"you can't look at the ABNF alone, you have to read the text".  To me,
this is a document failure, unless the ABNF was has to be so ugly to be
completely correct it fails to communicate.  Alas, that happens.  But if
it is possible to simply communicate the semantics in the ABNF, then it
is appropriate to do so.

You can use the "token" construction a whole lot more than we do if all
you want is syntax checking.

Lots of examples in 3261:
Priority        =3D  "Priority" HCOLON priority-value
priority-value  =3D  "emergency" / "urgent" / "normal"
                   / "non-urgent" / other-priority
other-priority  =3D  token


Contact        =3D  ("Contact" / "m" ) HCOLON
                  ( STAR / (contact-param *(COMMA contact-param)))
contact-param  =3D  (name-addr / addr-spec) *(SEMI contact-params)
name-addr      =3D  [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec      =3D  SIP-URI / SIPS-URI / absoluteURI

Let's settle on the words first.  The ABNF will follow.

Brian

> -----Original Message-----
> From: Dawson, Martin [mailto:Martin.Dawson@andrew.com]
> Sent: Monday, July 31, 2006 6:48 PM
> To: Rosen, Brian; Winterbottom, James; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] URI restrictions in location conveyance,was
Does
> Location-Conveyance have an HTTP dereferencemechanism defined
>
> Just using absoluteuri seems like the complete simplification...
>
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > Sent: Tuesday, 1 August 2006 8:44 AM
> > To: Winterbottom, James; IETF SIP List
> > Cc: GEOPRIV
> > Subject: RE: [Geopriv] URI restrictions in location conveyance,was
> Does
> > Location-Conveyance have an HTTP dereferencemechanism defined
> >
> > Can't let go can you?  I changed the subject line to start a
different
> > thread.  I'd still like the answer to the original thread.
> >
> > There are two separable issues with respect to the ABNF.  The
> syntactic
> > and the semantic.
> >
> > The syntactic issues could be dealt with by saying:
> >
> > location-param =3D LAQUOT loc-URI RAQUOT *( SEMI loc-param )
> > ;cid-url from RFC2392, SIP/SIPS-URI from RFC3261, PRES-URI from
> RFC3859
> > ;Absolute URI from RFC3261
> > loc-URI =3D cid-uri / SIP-URI / SIPS-URI / PRES-URI / AbsoluteURI
> >
> > This answers Henning and Jeroen's objection to strictly limiting the
> > ABNF.
> > The alternative is to just use AbsoluteURI.
> >
> > I guess the question is, what does the text say? I can make the ABNF
> > match what the text says.
> >
> > Does the text say: any URI can appear in the location header,
entities
> > must use some undefined mechanism to maintain interoperability.
> >
> > Does it say: location dereferencing is a "using protocol" as defined
> in
> > geopriv requirements.  Any URI found in the location header MUST
meet
> > the requirements of a using protocol.
> >
> > Does it say: This document describes the use of SIP and SIPS as a
> > dereferencing protocol (and the use of a pres: uri if the resolution
> of
> > the pres: scheme in the referenced domain is to a presence service
> using
> > the sip/sips protocol).  All entities using the Location header MUST
> > support SIP dereferencing.
> >
> > Does it say (if the paragraph immediately above is there): Other
> > protocols used for dereferencing MUST have a (?standards track?) RFC
> > defining how to dereference.
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > Sent: Monday, July 31, 2006 6:19 PM
> > > To: Rosen, Brian; IETF SIP List
> > > Cc: GEOPRIV
> > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > dereferencemechanism defined
> > >
> > > Not true
> > >
> > > "> This is NOT the discussion of what the ABNF of the location
> header
> > > looks
> > > > like (although it would NOT have http/https in it for sure if we
> > agree
> > > > that an HTTP dereference protocol is a separate draft)."
> > >
> > > This is a discussion on ABNF and what the document contains, and
to
> > that
> > > extent they are not separable.
> > > The consensus has been, do not restrict the ABNF, keep the
> discussion
> > in
> > > the document about references limited. I am failing to understand
> why
> > you
> > > can't accept that.
> > >
> > >
> > >
> > > ________________________________
> > >
> > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > Sent: Mon 31/07/2006 5:05 PM
> > > To: Winterbottom, James; IETF SIP List
> > > Cc: GEOPRIV
> > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > dereferencemechanism defined
> > >
> > >
> > >
> > > Please don't change the subject, I understand the controversy on
> > > constraining the ABNF.  I tried (apparently without success) to
> > separate
> > > those issues.
> > >
> > > I'm asking if anyone objects to NOT mentioning HTTP in any way
with
> > > respect to dereferencing URIs carried by the Location header.
That
> is
> > > the only question I asked.
> > >
> > > I'm not trying to evade the constrained ABNF issue, and I'm not
> > implying
> > > anything tricky.  I simply want to know if it's okay if we leave
> > > anything that deals specifically with HTTP and the Location header
> URI
> > > to another document.
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > > Sent: Monday, July 31, 2006 5:55 PM
> > > > To: Rosen, Brian; IETF SIP List
> > > > Cc: GEOPRIV
> > > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > > dereferencemechanism defined
> > > >
> > > > Brian,
> > > >
> > > > Your second paragraph re-opens the sore I am afraid.
> > > > A large number of people have agreed with Henning's statements
and
> > > these
> > > > are clear:
> > > > 1) Do not constrain the ABNF
> > > > 2) Provide minimal description of the reference function.
> > > >
> > > > If you insist on saying then that the ABNF will be restricted if
> > there
> > > is
> > > > not matching text, then I think it unlikely that you will reach
a
> > > > consensus.
> > > >
> > > > Cheers
> > > > James
> > > >
> > > > ________________________________
> > > >
> > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > Sent: Mon 31/07/2006 8:38 AM
> > > > To: IETF SIP List
> > > > Cc: GEOPRIV
> > > > Subject: [Geopriv] Does Location-Conveyance have an HTTP
> > > > dereferencemechanism defined
> > > >
> > > >
> > > >
> > > > I have lost track of where we stand on whether the only
mentioned
> > URI
> > > > schemes in location-conveyance are sip/sips/pres.  I "think" we
> have
> > > > agreed that an HTTP/HTTPS dereference is in a separate document,
> but
> > > I'm
> > > > not sure.
> > > >
> > > > This is NOT the discussion of what the ABNF of the location
header
> > > looks
> > > > like (although it would NOT have http/https in it for sure if we
> > agree
> > > > that an HTTP dereference protocol is a separate draft).
> > > >
> > > > If you think location-conveyance DOES have HTTP dereference text
> in
> > > it,
> > > > please say so, and please explain why you would not be satisfied
> if
> > > > there were a separate draft on the subject.
> > > >
> > > > Brian
> > > >
> > > > _______________________________________________
> > > > Geopriv mailing list
> > > > Geopriv@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > >
> > > >
> > > >
> > > >
> > >
> >
>
------------------------------------------------------------------------
> > > --
> > > > ----------------------
> > > > This message is for the designated recipient only and may
> > > > contain privileged, proprietary, or otherwise private
information.
> > > > If you have received it in error, please notify the sender
> > > > immediately and delete the original.  Any unauthorized use of
> > > > this email is prohibited.
> > > >
> > >
> >
>
------------------------------------------------------------------------
> > > --
> > > > ----------------------
> > > > [mf2]
> > >
> > >
> > >
> > >
> >
>
------------------------------------------------------------------------
> > --
> > > ----------------------
> > > This message is for the designated recipient only and may
> > > contain privileged, proprietary, or otherwise private information.
> > > If you have received it in error, please notify the sender
> > > immediately and delete the original.  Any unauthorized use of
> > > this email is prohibited.
> > >
> >
>
------------------------------------------------------------------------
> > --
> > > ----------------------
> > > [mf2]
> >
> > _______________________________________________
> > Geopriv mailing list
> > Geopriv@ietf.org
> > https://www1.ietf.org/mailman/listinfo/geopriv
>
>
------------------------------------------------------------------------
--
> ----------------------
> This message is for the designated recipient only and may
> contain privileged, proprietary, or otherwise private information.
> If you have received it in error, please notify the sender
> immediately and delete the original.  Any unauthorized use of
> this email is prohibited.
>
------------------------------------------------------------------------
--
> ----------------------
> [mf2]



---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
contain privileged, proprietary, or otherwise private information. =20
If you have received it in error, please notify the sender
immediately and delete the original.  Any unauthorized use of
this email is prohibited.
---------------------------------------------------------------------------=
---------------------
[mf2]

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From sip-bounces@ietf.org Mon Jul 31 19:26:26 2006
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Subject: [Sip] RE: [Geopriv] URI restrictions in location conveyance,
	was Does Location-Conveyance have an HTTP
	dereferencemechanism defined
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Errors-To: sip-bounces@ietf.org

No James, the two threads are not linked.  The first asks if there is any
reference to HTTP use in dereferencing.  The other discusses what the text,
and the resulting ABNF says about dereferencing in general.

Note in the message below, the token "http" does not appear, yet the text is
readable.

IF we were to include an http based dereferencing protocol in location
conveyance AND we were to have a more semantically correct ABNF, THEN you
would see HTTP in the list of protocols.  However, you can answer the
questions independently without any problem.

Henning said we should not syntactically limit the ABNF.  The "more
specific" ABNF directly addresses that concern, and Jeroen's concern about
diagnostics by including AbsoluteURI as an alternative.  I'm bowing to list
consensus on strictly limiting the ABNF.  

Let's get back to the question, which in this thread is "what does the text
say about what can appear in the Location URI?"

Brian

> -----Original Message-----
> From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> Sent: Monday, July 31, 2006 7:09 PM
> To: Rosen, Brian; Dawson, Martin; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] URI restrictions in location conveyance,was Does
> Location-Conveyance have an HTTP dereferencemechanism defined
> 
> Brian,
> 
> So the two things are linked then?
> 
> It is my belief that the consensus is in favour of what Henning suggested,
> and that is what I would like.
> Perhaps you would like to include precisely that position in your next
> quest for consensus, rather than trying to re-word it and losing the
> intent.
> 
> Cheers
> James
> 
> ________________________________
> 
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: Mon 31/07/2006 6:01 PM
> To: Dawson, Martin; Winterbottom, James; IETF SIP List
> Cc: GEOPRIV
> Subject: RE: [Geopriv] URI restrictions in location conveyance,was Does
> Location-Conveyance have an HTTP dereferencemechanism defined
> 
> 
> 
> Sure, and so does: "Any URI can be used".  That isn't necessarily a good
> thing.
> 
> If the text is like the "this document describes how you use
> sip/sips/pres" then the ABNF I suggested is more appropriate.
> 
> Any ABNF can be significantly reduced if the only thing you are trying
> to convey with it is strict syntactic compliance.  No significant
> document with large hunks of ABNF in the IETF does that.  We almost
> always try to make the ABNF show as much of the semantic meaning as is
> reasonable.  If you follow the SIP threads, there are several messages a
> month whose answer starts:
> "you can't look at the ABNF alone, you have to read the text".  To me,
> this is a document failure, unless the ABNF was has to be so ugly to be
> completely correct it fails to communicate.  Alas, that happens.  But if
> it is possible to simply communicate the semantics in the ABNF, then it
> is appropriate to do so.
> 
> You can use the "token" construction a whole lot more than we do if all
> you want is syntax checking.
> 
> Lots of examples in 3261:
> Priority        =  "Priority" HCOLON priority-value
> priority-value  =  "emergency" / "urgent" / "normal"
>                    / "non-urgent" / other-priority
> other-priority  =  token
> 
> 
> Contact        =  ("Contact" / "m" ) HCOLON
>                   ( STAR / (contact-param *(COMMA contact-param)))
> contact-param  =  (name-addr / addr-spec) *(SEMI contact-params)
> name-addr      =  [ display-name ] LAQUOT addr-spec RAQUOT
> addr-spec      =  SIP-URI / SIPS-URI / absoluteURI
> 
> Let's settle on the words first.  The ABNF will follow.
> 
> Brian
> 
> > -----Original Message-----
> > From: Dawson, Martin [mailto:Martin.Dawson@andrew.com]
> > Sent: Monday, July 31, 2006 6:48 PM
> > To: Rosen, Brian; Winterbottom, James; IETF SIP List
> > Cc: GEOPRIV
> > Subject: RE: [Geopriv] URI restrictions in location conveyance,was
> Does
> > Location-Conveyance have an HTTP dereferencemechanism defined
> >
> > Just using absoluteuri seems like the complete simplification...
> >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > Sent: Tuesday, 1 August 2006 8:44 AM
> > > To: Winterbottom, James; IETF SIP List
> > > Cc: GEOPRIV
> > > Subject: RE: [Geopriv] URI restrictions in location conveyance,was
> > Does
> > > Location-Conveyance have an HTTP dereferencemechanism defined
> > >
> > > Can't let go can you?  I changed the subject line to start a
> different
> > > thread.  I'd still like the answer to the original thread.
> > >
> > > There are two separable issues with respect to the ABNF.  The
> > syntactic
> > > and the semantic.
> > >
> > > The syntactic issues could be dealt with by saying:
> > >
> > > location-param = LAQUOT loc-URI RAQUOT *( SEMI loc-param )
> > > ;cid-url from RFC2392, SIP/SIPS-URI from RFC3261, PRES-URI from
> > RFC3859
> > > ;Absolute URI from RFC3261
> > > loc-URI = cid-uri / SIP-URI / SIPS-URI / PRES-URI / AbsoluteURI
> > >
> > > This answers Henning and Jeroen's objection to strictly limiting the
> > > ABNF.
> > > The alternative is to just use AbsoluteURI.
> > >
> > > I guess the question is, what does the text say? I can make the ABNF
> > > match what the text says.
> > >
> > > Does the text say: any URI can appear in the location header,
> entities
> > > must use some undefined mechanism to maintain interoperability.
> > >
> > > Does it say: location dereferencing is a "using protocol" as defined
> > in
> > > geopriv requirements.  Any URI found in the location header MUST
> meet
> > > the requirements of a using protocol.
> > >
> > > Does it say: This document describes the use of SIP and SIPS as a
> > > dereferencing protocol (and the use of a pres: uri if the resolution
> > of
> > > the pres: scheme in the referenced domain is to a presence service
> > using
> > > the sip/sips protocol).  All entities using the Location header MUST
> > > support SIP dereferencing.
> > >
> > > Does it say (if the paragraph immediately above is there): Other
> > > protocols used for dereferencing MUST have a (?standards track?) RFC
> > > defining how to dereference.
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > > Sent: Monday, July 31, 2006 6:19 PM
> > > > To: Rosen, Brian; IETF SIP List
> > > > Cc: GEOPRIV
> > > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > > dereferencemechanism defined
> > > >
> > > > Not true
> > > >
> > > > "> This is NOT the discussion of what the ABNF of the location
> > header
> > > > looks
> > > > > like (although it would NOT have http/https in it for sure if we
> > > agree
> > > > > that an HTTP dereference protocol is a separate draft)."
> > > >
> > > > This is a discussion on ABNF and what the document contains, and
> to
> > > that
> > > > extent they are not separable.
> > > > The consensus has been, do not restrict the ABNF, keep the
> > discussion
> > > in
> > > > the document about references limited. I am failing to understand
> > why
> > > you
> > > > can't accept that.
> > > >
> > > >
> > > >
> > > > ________________________________
> > > >
> > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > Sent: Mon 31/07/2006 5:05 PM
> > > > To: Winterbottom, James; IETF SIP List
> > > > Cc: GEOPRIV
> > > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > > dereferencemechanism defined
> > > >
> > > >
> > > >
> > > > Please don't change the subject, I understand the controversy on
> > > > constraining the ABNF.  I tried (apparently without success) to
> > > separate
> > > > those issues.
> > > >
> > > > I'm asking if anyone objects to NOT mentioning HTTP in any way
> with
> > > > respect to dereferencing URIs carried by the Location header.
> That
> > is
> > > > the only question I asked.
> > > >
> > > > I'm not trying to evade the constrained ABNF issue, and I'm not
> > > implying
> > > > anything tricky.  I simply want to know if it's okay if we leave
> > > > anything that deals specifically with HTTP and the Location header
> > URI
> > > > to another document.
> > > >
> > > > Brian
> > > >
> > > > > -----Original Message-----
> > > > > From: Winterbottom, James [mailto:James.Winterbottom@andrew.com]
> > > > > Sent: Monday, July 31, 2006 5:55 PM
> > > > > To: Rosen, Brian; IETF SIP List
> > > > > Cc: GEOPRIV
> > > > > Subject: RE: [Geopriv] Does Location-Conveyance have an HTTP
> > > > > dereferencemechanism defined
> > > > >
> > > > > Brian,
> > > > >
> > > > > Your second paragraph re-opens the sore I am afraid.
> > > > > A large number of people have agreed with Henning's statements
> and
> > > > these
> > > > > are clear:
> > > > > 1) Do not constrain the ABNF
> > > > > 2) Provide minimal description of the reference function.
> > > > >
> > > > > If you insist on saying then that the ABNF will be restricted if
> > > there
> > > > is
> > > > > not matching text, then I think it unlikely that you will reach
> a
> > > > > consensus.
> > > > >
> > > > > Cheers
> > > > > James
> > > > >
> > > > > ________________________________
> > > > >
> > > > > From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> > > > > Sent: Mon 31/07/2006 8:38 AM
> > > > > To: IETF SIP List
> > > > > Cc: GEOPRIV
> > > > > Subject: [Geopriv] Does Location-Conveyance have an HTTP
> > > > > dereferencemechanism defined
> > > > >
> > > > >
> > > > >
> > > > > I have lost track of where we stand on whether the only
> mentioned
> > > URI
> > > > > schemes in location-conveyance are sip/sips/pres.  I "think" we
> > have
> > > > > agreed that an HTTP/HTTPS dereference is in a separate document,
> > but
> > > > I'm
> > > > > not sure.
> > > > >
> > > > > This is NOT the discussion of what the ABNF of the location
> header
> > > > looks
> > > > > like (although it would NOT have http/https in it for sure if we
> > > agree
> > > > > that an HTTP dereference protocol is a separate draft).
> > > > >
> > > > > If you think location-conveyance DOES have HTTP dereference text
> > in
> > > > it,
> > > > > please say so, and please explain why you would not be satisfied
> > if
> > > > > there were a separate draft on the subject.
> > > > >
> > > > > Brian
> > > > >
> > > > > _______________________________________________
> > > > > Geopriv mailing list
> > > > > Geopriv@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/geopriv
> > > > >
> > > > >
> > > > >
> > > > >
> > > >
> > >
> >
> ------------------------------------------------------------------------
> > > > --
> > > > > ----------------------
> > > > > This message is for the designated recipient only and may
> > > > > contain privileged, proprietary, or otherwise private
> information.
> > > > > If you have received it in error, please notify the sender
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> This message is for the designated recipient only and may
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> ----------------------
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> 
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> https://www1.ietf.org/mailman/listinfo/geopriv


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From sip-bounces@ietf.org Mon Jul 31 21:05:26 2006
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From: Andrew Newton <andy@hxr.us>
Subject: Re: [Sip] RE: [Geopriv] Does Location-Conveyance have an HTTP
	dereferencemechanism defined
Date: Mon, 31 Jul 2006 21:03:22 -0400
To: "Winterbottom, James" <James.Winterbottom@andrew.com>
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On Jul 31, 2006, at 6:19 PM, Winterbottom, James wrote:

> Not true
>
> "> This is NOT the discussion of what the ABNF of the location header
> looks
>> like (although it would NOT have http/https in it for sure if we  
>> agree
>> that an HTTP dereference protocol is a separate draft)."
>
> This is a discussion on ABNF and what the document contains, and to  
> that extent they are not separable.
> The consensus has been, do not restrict the ABNF, keep the  
> discussion in the document about references limited. I am failing  
> to understand why you can't accept that.

Brian is right.  They are not the same issue, atleast they do not  
have to be.

 From where I sit, I think the consensus is:
   - do not constrain the ABNF
   - HTTP goes in another document

Keep in mind, my hat is labeled "GEOPRIV" and this is a SIP document.

-andy

_______________________________________________
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From: "Winterbottom, James" <James.Winterbottom@andrew.com>
To: "Andrew Newton" <andy@hxr.us>
Date: Mon, 31 Jul 2006 20:11:16 -0500
Subject: RE: [Sip] RE: [Geopriv] Does Location-Conveyance have an HTTP
	dereferencemechanism defined
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Hi Andy,
=20
That is all I have asked for, do not constrain the ABNF. If "how I use HTTP=
 as a reference" goes in a different document that is fine providing the AB=
NF is not constrained in the conveyance specification. This is not how I re=
ad Brian's statement below:
=20
"This is NOT the discussion of what the ABNF of the location header looks
like (although it would NOT have http/https in it for sure if we agree
that an HTTP dereference protocol is a separate draft)."
=20
Which to me clearly says the ABNF will be restrict to exclude HTTP/HTTPS if=
 there is agreement that the conveyance draft doesn't have specific text on=
 how to deference an HTTP reference.
=20
Cheers
James

________________________________

From: Andrew Newton [mailto:andy@hxr.us]
Sent: Mon 31/07/2006 8:03 PM
To: Winterbottom, James
Cc: Rosen, Brian; IETF SIP List; GEOPRIV
Subject: Re: [Sip] RE: [Geopriv] Does Location-Conveyance have an HTTP dere=
ferencemechanism defined




On Jul 31, 2006, at 6:19 PM, Winterbottom, James wrote:

> Not true
>
> "> This is NOT the discussion of what the ABNF of the location header
> looks
>> like (although it would NOT have http/https in it for sure if we=20
>> agree
>> that an HTTP dereference protocol is a separate draft)."
>
> This is a discussion on ABNF and what the document contains, and to=20
> that extent they are not separable.
> The consensus has been, do not restrict the ABNF, keep the=20
> discussion in the document about references limited. I am failing=20
> to understand why you can't accept that.

Brian is right.  They are not the same issue, atleast they do not=20
have to be.

 From where I sit, I think the consensus is:
   - do not constrain the ABNF
   - HTTP goes in another document

Keep in mind, my hat is labeled "GEOPRIV" and this is a SIP document.

-andy



---------------------------------------------------------------------------=
---------------------
This message is for the designated recipient only and may
contain privileged, proprietary, or otherwise private information. =20
If you have received it in error, please notify the sender
immediately and delete the original.  Any unauthorized use of
this email is prohibited.
---------------------------------------------------------------------------=
---------------------
[mf2]

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: Re: [Sip] RE: [Geopriv] Does Location-Conveyance have an HTTP
	dereferencemechanism defined
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On Jul 31, 2006, at 9:11 PM, Winterbottom, James wrote:

> Hi Andy,
>
> That is all I have asked for, do not constrain the ABNF. If "how I  
> use HTTP as a reference" goes in a different document that is fine  
> providing the ABNF is not constrained in the conveyance  
> specification. This is not how I read Brian's statement below:
>
> "This is NOT the discussion of what the ABNF of the location header  
> looks
> like (although it would NOT have http/https in it for sure if we agree
> that an HTTP dereference protocol is a separate draft)."
>
> Which to me clearly says the ABNF will be restrict to exclude HTTP/ 
> HTTPS if there is agreement that the conveyance draft doesn't have  
> specific text on how to deference an HTTP reference.
>
> Cheers
> James

Please keep in mind that this does not stop the draft from using MUST/ 
SHOULD language from stipulating "the URI MUST be a GEOPRIV standards  
track using protocol."

-andy

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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



