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From: Jonathan Rosenberg <jdrosen@cisco.com>
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To: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Provisional response to terminate early dialog
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Robert Sparks wrote:

> How is this bound to 64T1?
> 
> It's an INVITE transaction - it can pend potentially indefinitely 
> (though timer C will likely kick in to muck up the works).

Oops, you are right. My brain is still not up to speed due to the nasty 
cold I suffered through this weekend.


> 
> I can see the kind of cases Christer is considering:
> 
>   1) You parallel fork to several places. All initiate early media. All 
> but one
>        return an error code that doesn't stimulate immediate forwarding 
> in the
>        proxy response context (say, 401just to see the situation, as odd 
> as that
>        would be to mix w/ early media). One leg just stays pending. The 
> initiating
>        UA will hold onto the resources for early media for _all_ of 
> those legs until
>        the pending leg completes.
> 
> 2) If you have a serial forking proxy, and each leg initiates early 
> media, at the
>     end of the fork, the initiating UA is holding early media resources 
> for each
>     previous leg - it has no way to let go before it gets a final response.
> 
> Personally, I see this as just one of the consequences of the
> terrible thing we did when we allowed early media in the first place.
> I am reticent to further complicate the protocol by introducing 
> provisionals
> along the lines Christer is proposing - that step takes us most, if not 
> all, of the way to having
> transactions within transactions. Taken to an extreme, this could solve 
> HERFP
> by tunneling all the final responses on the other side of a proxy back 
> to the UA
> so it might be able to do something about it (say they were 421 Extension
> Required responses: A UA could deal with those immediately by initiating a
> new branch, parallel forking at the UA (yuck)). This could even lead to 
> pressure
> to change the merging rules so that a UA can deal with 401s as part of 
> early dialogs
> by reissuing the request on a new branch (double-yuck).

Well, you may recall that the initial proposed solution to HERFP was to 
use a provisional response instead of a final one for errors that are 
fixable. As such, I would argue that the problem Christer is worried 
about *is* the HERFP problem, and if we want to solve anything, we 
should solve that more generic one rather than this specific 
instantiation of it.

Now, do we want to really solve the HERFP problem? Initially I thought 
we would for sure, but reality and actual usage has yet to show this as 
a pressing problem...

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   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] Abbreviated WGLC
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comments:

1. the title should indicate that this draft has something to do with SIP.

2. Abstract is too short. It will be bounced by rfc-ed in this form. You 
need to provide a little more context here.

Propose:

The REFER method has been defined for the Session Initiation Protocol 
(SIP) for the purposes of asking a user agent to generate a request to a 
specified Uniform Resource Identifier (URI). Sending a REFER request 
creates an implicit subscription that allows the referor to get updates 
on the status of the request generated by the REFER. However, experience 
has shown that in many cases this implicit subscription is not 
desirable. This specification defines a new SIP option tag value that 
causes the recipient of the REFER to suppress the implicit subscription.

3. The introduction seems stranded without the motivation. I would 
rename "Motivation" to "introduction" and copy the only paragraph in the 
introduction to the end of the motivation section.

> REFER-Recipient to the REFER-Issuer.  The REFER-Recipient may choose
>     to cancel the implicit subscription with this NOTIFY.  

by setting the Subscription-State header field value in the NOTIFY to 
"terminated".

> The "norefersub" option tag MUST be used by the REFER-Issuer only
>     when the REFER-Issuer can be certain that the REFER request will not
>     be forked.

I think we need to say how it knows that. Sending the REFER to the GRUU 
is one way. Its really the only reliable way I think....

>  The REFER-Issuer can place the "norefersub" option tag either in the
>     Require header or in the Supported header of the REFER request,
>     subject to application requirements.

note on terminology: the right term here is "header field" and not 
"header". "header" refers to everything from the start line to the 
message body, and thus includes all of the header fields.

> If the REFER-Recipient is willing to grant the "norefersub" behavior
>     for the issued REFER request, it MUST insert a Supported: norefersub
>     header in the 2xx response to the REFER-Issuer.  In this case no
>     dialog is created.
> 

This is not correct usage of Supported/Require. If you place Supported 
in a request, it means that an extension is understood. If the server 
sends a response which requires that the client understand the extension 
in order to process it, it includes a Require in the response. Thus, in 
the case above, the norefersub needs to appear in a Require in the 2xx, 
not a Supported.

That said, I see little value in having the REFER-Recipient make the 
choice about whether to create the implicit subscription. Isnt it the 
REFER-issuer that really understands the usage behind the REFER, and 
thus, whether notifies are needed? I'd propose that it be used only in 
the Require header field in the SUBSCRIBE.

> Preventing Forking of REFER Requests
> 
>     The REFER specification allows for the possibility of forking a REFER
>     request which is sent outside of an existing dialog.  The
>     REFER-Issuer can ensure that REFER doesn't get forked by sending
>     REFER to a REFER-Recipient which has GRUU properties according to
>     definitions of [5].

ahh, here is that text. I'd propsoe moving it up into the previous 
section where you say that you MUST only use this when you know it won't 
be forked.

That said, we don't have a way for a UA to look at a URI and determine 
that it has the GRUU property. How then can this requirement be met?

> Example
> 
>     An example of REFER which suppresses the implicit subscription is
>     shown below:
> 
>     REFER sip:pc-b@tradewind.com SIP/2.0
>     Via: SIP/2.0/TCP issuer.tradewind.com;branch=z9hG4bK-a-1
>     From: <sip:a@tradewind.com>;tag=1a
>     To: <sip:pc-b@tradewind.com>
>     Call-ID: 1@issuer.tradewind.com
>     CSeq: 234234 REFER
>     Max-Forwards: 70
>     Refer-To: <sip:c@tradewind.com;method=INVITE>
>     Require: norefersub
>     Accept-Contact: *;audio;require
>     Contact: sip:a@issuer.tradewind.com
>     Content-Type: message/sipfrag
>     Content-Id: <1239103912039@issuer.tradewind.com>
>     Content-Length: ...

all example messages have to use domains from example.com or 
example.org. You cannot use tradewind.com.

The tag in the From is invalid. It is insufficiently random. See section 
  19.3 of rfc3261. Same with call-id.

The message indicates that the body is a sipfrag, but it is not present 
thus the message is invalid. Why do you want to put a sipfrag in there 
anyway?

The message uses caller prefs headers. I suggest not to do that, it 
complicates the story.

If you use the ellipses (...) to avoid calculating the content-length, 
you must state that explicitly.

> This document defines a new option tag, "norefersub", which specifies
>     that no implicit subscription should be created as a result of
>     accepting the REFER request.  This option tag is only meaningful for
>     the REFER request defined in RFC3515.

this doesn't follow standard templates for iana registrations. I promise 
you will introduce delays like this.

Suggest:

This specification registers a new SIP option tag, as per the
    guidelines in Section 27.1 of RFC 3261.
    Name: norefersub
    Description: This option tag specifies
     that no implicit subscription should be created as a result of
     accepting the REFER request.  This option tag is only meaningful for
     the REFER request defined in RFC3515.


btw, based on this definition even, it only makes sense to place ths in 
Require.

> Security Considerations
> 
>     This extension doesn't introduce new security threads beyond those
>     identified and addressed in the core SIP specifications.

You are going to need to explain why.

> Acknowledgements
> 
>     The SIP community would like to thank Sriram Parameswar for his ideas
>     being originally presented in draft-parameswar-sipping-norefersub-00
>     and incorporated in this document.

usually the acknowledgements are from the authors, not the community. 
its a nit...

> 10.2  Informational References
> 
>     [5]  Rosenberg, J., "Obtaining and Using Globally Routable User Agent
>          (UA) URIs (GRUU) in the  Session Initiation Protocol (SIP)",
>          draft-ietf-sip-gruu-02 (work in progress), July 2004.

careful here...

you specify that a client MUST only send a refer to a URI if it doesnt 
fork, and then say later that this is done with gruu. This would make 
gruu a normative reference I think.

-Jonathan R.

Rohan Mahy wrote:

> Hi Folks,
> 
> I'd like to begin an abbreviated WGLC for:
> 
>     http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-with- 
> norefersub-00.txt
> 
> Since this document is extremely short and has been discussed  
> extensively, WGLC will end on February 1st.
> 
> thanks,
> -rohan
> 
> 
> _______________________________________________
> 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
Director, Service Provider VoIP Architecture   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: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Minor anomaly in RFC 3261 Registration example?
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And in particular, this bug is in bugzilla already:
http://bugs.sipit.net/show_bug.cgi?id=682

-Jonathan R.

Robert Sparks wrote:

> The bugzilla database is available at bugs.sipit.net.
> 
> RjS
> 
> On Jan 14, 2005, at 4:06 PM, Jain, Rajnish wrote:
> 
>> Folks:
>>
>> Not sure if this minor anomaly has been reported before and is already 
>> logged in SIP bugzilla (which seems to be down right now). Apologies 
>> if this is a repetition.
>>
>>
>>
>> Section 10.3 of RFC 3261 states the following:
>>
>>       8. The registrar returns a 200 (OK) response.  The response MUST
>>           contain Contact header field values enumerating all current
>>           bindings.  Each Contact value MUST feature an "expires"
>>           parameter indicating its expiration interval chosen by the
>>           registrar.  The response SHOULD include a Date header field.
>>
>>
>>
>> Section 24.1 of RFC 3261 gives an example of 200 OK for a REGISTER as 
>> the following:
>>
>>         SIP/2.0 200 OK
>>          Via: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=z9hG4bKnashds7
>>          ;received=192.0.2.4
>>         To: Bob <sip:bob@biloxi.com>;tag=2493k59kd
>>         From: Bob <sip:bob@biloxi.com>;tag=456248
>>         Call-ID: 843817637684230@998sdasdh09
>>         CSeq: 1826 REGISTER
>>          Contact: <sip:bob@192.0.2.4>
>>         Expires: 7200
>>          Content-Length: 0
>>
>>
>>
>> The "expires" parameter in the 200 OK's Contact header is missing in 
>> this example whereas it's a MUST in the text.
>>
>> Thanks,
>>  Rajnish
>>
>> Email     mailto:rjain@excelswitching.com
>> SIP       sip:rjain@voip.excelswitching.com
>> PSTN      508-862-3215
>> _______________________________________________
>> 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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   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 Feb  1 02:11:56 2005
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Robert Sparks'" <rjsparks@nostrum.com>,
        Jonathan Rosenberg
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Subject: RE: [Sip] Provisional response to terminate early dialog
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Hi,

In my use-case I actually meant to say SERIAL forking, not parallel. Of course, from a UAC perspective it doesn't matter (it doesn't know how the forking is done).

However, the fact that we can receive early media from multiple UASs in the parallel case I probably something we can do very little about, and if many nodes send early media at the same time it may not help much to have an indication on when a dialog is terminated.

In the serial case, however, there will only be early media from one node at a time. But, when one fork is terminated, the media may "switch" to another dialog, initiated by the next fork. But, again, since the UAC doesn't knows that there will be no more early media from the first fork it may keep the resources reserved for the first fork until it receives a final response from another fork.

Jonathan said in his second reply that we need to find out if this will be a problem in reality.

The reason for my initial question was basically due to the work on supplementary services which is currently ongoing in the ETSI TISPAN and 3GPP bodies. Some services will provide early media/announcments, and one idea is that they will be associated to a specific early dialog. The media coming from the actual UAS will then be associated with another dialog. To describe it easy, from a SIP perspective there would be a proxy doing serial forking, first to the announcement UAS, and then to the actual UAS.

The problem is that a 3GPP UAC may choose not to accept (for a number of technical reasons) multiple early dialogs, so in this case it would probably accept the first "announcement" dialog, but not the UAC-UAS dialog - which is of course bad. However, if the node doing the forking would be able to directly, when the announcement has finished, indicate to the UAC that the announcement dialog is terminated, the UAC could free the resources associated with that and then accept the UAC-UAS dialog.

So, I think that would be a very good feature, and I don't think it would take much to define it. And, using a provisional response would (as I said earlier) also be backward compatible.


Now, about forwarding mutliple FINAL responses - I don't think we should go that way - at least not until we have real requirements for it. It would require much more work - not only to forward the responses themselves, but also to handle the possible actions that may/should/should not be taken by the UAC when receiving multiple final responses. There would probably also be backward compability issues.

As Jonathan said, one thing could be to include the final response code in the new provisional response. The UAC could then, depedning on the response code, its implementation logic etc, choose to CANCEL the WHOLE INVITE, and re-issue a new INVITE (for example including extensions which were required). I do NOT think that we should go the transaction-within-transactions way, and start sending "new INVITEs" for certain early dialogs.

Regards,

Christer Holmberg
Ericsson Finland






> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> Sent: 31. tammikuuta 2005 19:31
> To: Jonathan Rosenberg
> Cc: 'sip@ietf.org'; Christer Holmberg (JO/LMF)
> Subject: Re: [Sip] Provisional response to terminate early dialog
> 
> 
> How is this bound to 64T1?
> 
> It's an INVITE transaction - it can pend potentially indefinitely 
> (though timer C will likely kick in to muck up the works).
> 
> I can see the kind of cases Christer is considering:
> 
>    1) You parallel fork to several places. All initiate early 
> media. All 
> but one
>         return an error code that doesn't stimulate immediate 
> forwarding 
> in the
>         proxy response context (say, 401just to see the situation, as 
> odd as that
>         would be to mix w/ early media). One leg just stays 
> pending. The 
> initiating
>         UA will hold onto the resources for early media for _all_ of 
> those legs until
>         the pending leg completes.
> 
> 2) If you have a serial forking proxy, and each leg initiates early 
> media, at the
>      end of the fork, the initiating UA is holding early 
> media resources 
> for each
>      previous leg - it has no way to let go before it gets a final 
> response.
> 
> Personally, I see this as just one of the consequences of the
> terrible thing we did when we allowed early media in the first place.
> I am reticent to further complicate the protocol by introducing 
> provisionals
> along the lines Christer is proposing - that step takes us 
> most, if not 
> all, of the way to having
> transactions within transactions. Taken to an extreme, this 
> could solve 
> HERFP
> by tunneling all the final responses on the other side of a 
> proxy back 
> to the UA
> so it might be able to do something about it (say they were 421 
> Extension
> Required responses: A UA could deal with those immediately by 
> initiating a
> new branch, parallel forking at the UA (yuck)). This could 
> even lead to 
> pressure
> to change the merging rules so that a UA can deal with 401s 
> as part of 
> early dialogs
> by reissuing the request on a new branch (double-yuck).
> 
> In shorter words, I think this is a problem we should admit we have, 
> but not
> try to do anything about. If we _do_ choose to do something about it, 
> then we
> better carefully bound the scope.
> 
> RjS
> 
> On Jan 30, 2005, at 8:39 AM, Jonathan Rosenberg wrote:
> 
> >
> >
> > Christer Holmberg (JO/LMF) wrote:
> >
> >> Hi,
> >> Today, when a request gets forked, the forking proxy will 
> "collect" 
> >> final responses received from the different dialogs.
> >> The calling UAC, however, does now know when specific 
> early dialogs 
> >> are
> >> terminated, since the forking proxy will not forward the 
> final answer.
> >> That means that the UAC may have to keep resources etc that it 
> >> reserved when the early dialog was created (by a 18x provisional 
> >> response)
> >> available until the proxy forwards a final response.
> >
> > This is a relatively small amount of time - 32 seconds.
> >
> >> Would it somehow be possible to define a "dialog has been 
> terminated" 
> >> response code that the proxy could
> >> send to the UAC directly each time it receives an error 
> response from 
> >> any of the forked dialogs. A provisional response code 
> would probably 
> >> be good, because it would be backward compatible (nodes that don't 
> >> support it simply keeps the early dialog alive until the final 
> >> response comes from the forking proxy, as today).
> >
> > I don't really see what the problem is. Is this 32 seconds 
> or so that 
> > critical? Besides shaving a few seconds off the amount of 
> time that a 
> > UAC holds transaction state, is there any other benefit?
> >
> > -Jonathan R.
> >
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > Director, Service Provider VoIP Architecture   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 Feb  1 03:55:44 2005
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From: Simone Leggio <simone.leggio@cs.helsinki.fi>
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Subject: [Sip] Questions on the SIP identity draft
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Hi all,

I have a couple of questions about the SIP identity draft, i.e.

Enhancements for Authenticated Identity Management
in the Session Initiation Protocol (SIP)

draft-ietf-sip-identity-03

1) In the background section, I quote:
>  Note that the scope of this
>    document is limited to providing this identity assurance for SIP
>    requests; solving this problem for SIP responses is more complicated,
>    and is a subject for future work.

Why can't SIP Identity be applied to responses as well? Why can't an 
Identity header field be added based on the relevant response fields? At 
least, final responses like 200 OK.
For more security, instead of adding only the addr-spec component of the 
To header field, it may be added also the tag part. Am I forgetting some 
relevant security issue?

2) Section 10 says that

> The REGISTER method uses Contact header fields in very unusual ways that
>    complicate its applicability to this mechanism.  Accordingly, the
>    Identity and Identity-Info header MUST NOT appear in REGISTER

Why can't the REGISTER be protected? Usually authentication happens 
because user agent and server share a common secret, i.e. password. What 
if they would share a key? The user agent signs the REGISTER with its 
key, and the server uses it for verifying the identity of the 
authenticating entity. Of course, the user agent has communicated 
previously its key to the server, probably when the user registers for 
the first time in that SIP domain. Anyway, how key communication is 
performed would be out of scope for SIP Identity.

Best regards,

	Simone Leggio

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From sip-bounces@ietf.org  Tue Feb  1 07:11:23 2005
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Hi,

I have the following doubts and need some help.

a) In order to contruct DNS query, I googled-up and found RFCs 1034 and
1035. But it doesn't talk of SIP. Can those guidelines be used for
constructing the DNS for SIP related queries?

b) Where to send the DNS query from UAC?

c) Unlike RFC 3261 which clearly tells how to construct invite, ack, etc...
messages, RFC 3263 specifies no syntax or query format to construct the DNS
query.

Regards,
B.Suganya


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From: Joseph Chooracken John-A19500 <johnc@motorola.com>
To: "'Bhattacharyya, Dipankar'" <DBhattacharyya@empirix.com>, sip@ietf.org
Subject: RE: [Sip] Query on Early Media ( 183 Session in Progress ) 
Date: Thu, 27 Jan 2005 12:32:02 +0530
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Dipankar,
 
   A dialog/Session is not established unless a 2xx response is sent back by
the UAS [Gateway]. Hence no media is sent out before 183 session in
progress.
 
Thanks,
John
 
 

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Bhattacharyya, Dipankar
Sent: Thursday, January 27, 2005 5:27 AM
To: sip@ietf.org
Subject: [Sip] Query on Early Media ( 183 Session in Progress ) 



Hi All,

    This is query regarding Early Media ( 183 Session in Progress ) .

Can an  Implementation ( Gateway ) send Media before sending out the 183
Session in Progress over UDP Transport in response to a INVITE   ? Is there
any RFC/draft on the details of 183 Session in progress and Media handling ?


Thanks in advance .

Best Regards,   

 

Dipankar Bhattacharyya 
Systems Engineer 
Communication Infrastructure Group

p |  972.401.4112 f | 972.596.4909 m | 214.766.1479
email | dbhattacharyya@empirix.com <mailto:dbhattacharyya@empirix.com> 
6100 Ohio Drive , Suite 2011 |   Plano, TX  |  75024 .

E M P I R I X 
 
<file:///C:\Documents%20and%20Settings\rknaub\Application%20Data\Microsoft\S
ignatures\www.empirix.com> www.empirix.com

 

 


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From sip-bounces@ietf.org  Tue Feb  1 14:31:53 2005
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From: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
To: "'sip@ietf.org'" <sip@ietf.org>
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Subject: [Sip] Requests within SIP Dialog and Contact Header
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The Contact header is defined as a Mandatory Header in some of Request messages,
like NOTIFY as a mandatory header. 
 
I am trying to understand if this is required for SIP header like Contact to be mandatory
in SIP requests that are sent in a dialog that is already setup with a INVITE transaction.
If it is required, then what is the intent of making it Mandatory and how it should be used.
 
In my opinion, I think a distinction needs to be made between the case how a request
such as NOTIFY is used. If it is used to setup a dialog, then SIP headers like Contact 
should be mandatory, otherwise they should be not even be present.
 
Nagesh Shekar 
Email: shek@lucent.com 

 


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<DIV><SPAN class=007061519-01022005>The Contact header is defined as a Mandatory 
Header in some of Request messages,</SPAN></DIV>
<DIV><SPAN class=007061519-01022005>like NOTIFY as a mandatory header. 
</SPAN></DIV>
<DIV><SPAN class=007061519-01022005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=007061519-01022005>I am trying to understand if this is 
required for SIP header like Contact to be mandatory</SPAN></DIV>
<DIV><SPAN class=007061519-01022005>in SIP requests that are sent in a dialog 
that is already setup with a INVITE transaction.</SPAN></DIV>
<DIV><SPAN class=007061519-01022005>If it is required, then what is the intent 
of making it Mandatory and how it should be used.</SPAN></DIV>
<DIV><SPAN class=007061519-01022005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=007061519-01022005>In my opinion, I think a distinction needs 
to be made between the case how a request</SPAN></DIV>
<DIV><SPAN class=007061519-01022005>such as NOTIFY is used. If it is used to 
setup a dialog, then SIP headers like Contact </SPAN></DIV>
<DIV><SPAN class=007061519-01022005>should be mandatory, otherwise they should 
be not even be present.</SPAN></DIV>
<DIV><SPAN class=007061519-01022005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=007061519-01022005></SPAN><FONT size=5><FONT 
face="Monotype Corsiva"><FONT color=#0000ff><STRONG><EM>Nagesh Shekar</EM> 
</STRONG></FONT></FONT></FONT></DIV>
<DIV align=left><STRONG><FONT size=2>Email: shek@lucent.com 
</FONT></STRONG></DIV>
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_______________________________________________
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From sip-bounces@ietf.org  Tue Feb  1 16:57:12 2005
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To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Draft defining the Accept-Disposition header field
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Hi Cullen,

inline:

Cullen Jennings wrote:

>>>I'm not really sure what the problem is that this solves.
>>
>>it lets a UA know whether the remote UA supports a particular
>>disposition type. Right now, UAs guess this information based on the
>>support for MIME types. For example, if the remote UA supports
>>application/sdp, my UA assumes that the remote UA supports the 'session'
>>disposition type... but what about 'early-session'?
>> 
> 
> Seems like Supported header is a better way to solve this particular issue

Yes, we can define new option tags for everything we want to 
negotiate... but defining an option tag that just means "I support this 
disposition type" seems like duplicating stuff unnecessary (e.g., 
defining an option tag that means that the UA supports the early-session 
disposition type).

In addition, we have an error response (415) that, per its definition, 
should be returned if the UAS does not support a MIME type or a 
disposition type in the request... we have a means to express which MIME 
types the UAS supports if the problem was an unknown MIME type, but we 
do not have anything to express which disposition types the UAS 
supports, in case the problem had to do with a disposition type. This 
information can come handy for debugging.


> I guess I am wondering if using the MIME dispositions is really the right
> way to basically tell the UA what you want it to do with a given SIP command
> - what does recipient-list mean in the context of email or other users of
> MIME?

Yes, it has already been pointed out that disposition types are context 
dependent. The issue is that, in email, any disposition type belongs to 
the same context: the reception of an email message. In SIP, we have 
different contexts. For example, the reception of a body whose 
disposition type is 'session' in an INVITE request makes sense, but what 
happens if it is received in a MESSAGE?

Gonzalo


_______________________________________________
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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  Tue Feb  1 16:57:15 2005
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Hi Cullen,

inline:

Cullen Jennings wrote:

>>>I'm not really sure what the problem is that this solves.
>>
>>it lets a UA know whether the remote UA supports a particular
>>disposition type. Right now, UAs guess this information based on the
>>support for MIME types. For example, if the remote UA supports
>>application/sdp, my UA assumes that the remote UA supports the 'session'
>>disposition type... but what about 'early-session'?
>> 
> 
> Seems like Supported header is a better way to solve this particular issue

Yes, we can define new option tags for everything we want to 
negotiate... but defining an option tag that just means "I support this 
disposition type" seems like duplicating stuff unnecessary (e.g., 
defining an option tag that means that the UA supports the early-session 
disposition type).

In addition, we have an error response (415) that, per its definition, 
should be returned if the UAS does not support a MIME type or a 
disposition type in the request... we have a means to express which MIME 
types the UAS supports if the problem was an unknown MIME type, but we 
do not have anything to express which disposition types the UAS 
supports, in case the problem had to do with a disposition type. This 
information can come handy for debugging.


> I guess I am wondering if using the MIME dispositions is really the right
> way to basically tell the UA what you want it to do with a given SIP command
> - what does recipient-list mean in the context of email or other users of
> MIME?

Yes, it has already been pointed out that disposition types are context 
dependent. The issue is that, in email, any disposition type belongs to 
the same context: the reception of an email message. In SIP, we have 
different contexts. For example, the reception of a body whose 
disposition type is 'session' in an INVITE request makes sense, but what 
happens if it is received in a MESSAGE?

In any event, as Paul suggested, I believe it is good to analyze all the 
issues related to SIP and MIME stuff and make decisions in one way or 
another. At this point, there are issues that haven't got enough attention.

Gonzalo


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From sip-bounces@ietf.org  Tue Feb  1 17:05:16 2005
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Abbreviated WGLC for draft-ietf-sip-refer-with-norefers
	ub-00.txt
Date: Tue, 1 Feb 2005 15:09:12 -0600
To: "Elwell, John" <john.elwell@siemens.com>
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        Orit Levin <oritl@microsoft.com>,
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On Feb 1, 2005, at 2:50 PM, Elwell, John wrote:

> Robert,
>
> Section 3, 2nd paragraph: "One purpose of requiring the implicit
> subscription and initial NOTIFY is to allow for the situation where the
> REFER request gets forked and the REFER-Issuer needs a way to see the
> multiple dialogs that may be established as a result of the forked 
> REFER."
> Surely the REFER-Issuer will see the multiple dialogs as a result of
> receiving multiple 2xx responses to the REFER request?

Exactly not.

The issuer will only see one of those 2xx's.
Proxies only return one final response  to
non-INVITE transactions (like REFER).

That's why we have the immediate NOTIFY in SIP Events.

>
> Consequently, concerning the following statement in section 4: "The
> "norefersub" option tag MUST be used by the REFER-Issuer only when the
> REFER-Issuer can be certain that the REFER request will not be 
> forked." - is
> it really necessary for MUST strength?
Yes. For the reason above.

>
> Then at the end of section 4: "If the REFER-Recipient is willing to 
> grant
> the "norefersub" behavior for the issued REFER request, it MUST insert 
> a
> Supported: norefersub header in the 2xx response to the REFER-Issuer.  
> In
> this case no dialog is created."
> Surely a dialog is created by the 2xx response to REFER?

No, that's the point of the extension. No subscription. The norefersub 
REFER is a transaction
that stands alone and doesn't start any new usages (so it can't start 
any new dialogs).

>
> John
>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
>> Behalf Of Robert Sparks
>> Sent: 31 January 2005 17:34
>> To: Robert Sparks
>> Cc: SIP WG; Rohan Mahy; Orit Levin; Dean Willis
>> Subject: Re: [Sip] Abbreviated WGLC for
>> draft-ietf-sip-refer-with-norefersub-00.txt
>>
>> The LC the chairs initiated ends today.
>>
>> Please read this draft and comment. It's very short - please do this
>> now.
>>
>> We've talked about this idea a lot - let make sure the talk
>> is captured
>> in the draft.
>>
>> RjS
>>
>> On Jan 26, 2005, at 4:22 PM, Robert Sparks wrote:
>>
>>> Initial LC Comments:
>>>
>>> First paragraph of Section 3 calls out 481ing an initial
>> NOTIFY. The
>>> -dialog-usage-
>>> draft shows that is a bad idea, especially for in-dialog REFERs.
>>> Suggest trimming
>>> the sentence to end after (Expires 0).
>>>
>>> Section 4 fourth paragraph: It would help implementers to call out
>>> explicitly that the
>>> 200 response to the REFER will not contain a Supported or Require
>>> header containing
>>> the norefersub token.
>>>
>>> Section 4 fifth paragraph: This needs more definition. How is it
>>> rejected? 603, 420, or
>>> something else? Implementors will need some guidance here
>> on when to
>>> use which
>>> error response. I think (but am not dead certain at the
>> moment) that
>>> 420 is almost always
>>> what you want to use.
>>>
>>> Section 4 sixth paragraph: Is Supported: in the response
>> really the
>>> right thing? Eventlist
>>> uses Require: in the response to a SUBSCRIBE. The intent of the
>>> normative behavior in
>>> this paragraph is to let a Refer-issuer who only included
>> Supported:
>>> know whether or
>>> not a subscription was actually created. Require: (I'm using this
>>> extension) in the response
>>> fits that better than Supported: (I could use this
>> extension if you
>>> want to).
>>>
>>> Section 4 sixth paragraph: Suggest replacing "In this case,
>> no dialog
>>> is created" with
>>> "In this case, no implicit subscription is created. As a
>> consequence,
>>> no new dialog
>>> is created if this REFER was issued outside any existing dialog."
>>>
>>> Section 5: Probably worth also calling out using
>> Max-Forwards: 0 to
>>> prevent forking
>>> for those (rare now, but hopefully not rare forever) cases where
>>> signaling can get to
>>> the refer target without going through a proxy.
>>>
>>> Section 6: Must replace "tradewind" with "example" throughout.
>>>
>>> Section 8: "threads" should be "threats", It would be good
>> to pull any
>>> discussion
>>> that led to this conclusion together for when the IESG asks if we
>>> _really_ thought
>>> about this. Have we talked about what would happen if a SIP
>>> intermediary (or an
>>> unauthorized MitM) just injected a Require: norefersub into
>> a REFER
>>> request?
>>>
>>> RjS
>>>
>>>
>>> On Jan 24, 2005, at 5:07 PM, Rohan Mahy wrote:
>>>
>>>> Hi Folks,
>>>>
>>>> I'd like to begin an abbreviated WGLC for:
>>>>
>>>> 	http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-with-
>>>> norefersub-00.txt
>>>>
>>>> Since this document is extremely short and has been discussed
>>>> extensively, WGLC will end on February 1st.
>>>>
>>>> thanks,
>>>> -rohan
>>>>
>>>>
>>>> _______________________________________________
>>>> 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  Tue Feb  1 17:08:48 2005
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Subject: Re: [Sip] Abbreviated
	WGLC	for	draft-ietf-sip-refer-with-norefersub-00.txt
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inline.

	Paul

Jonathan Rosenberg wrote:

>> The "norefersub" option tag MUST be used by the REFER-Issuer only
>>     when the REFER-Issuer can be certain that the REFER request will not
>>     be forked.
> 
> I think we need to say how it knows that. Sending the REFER to the GRUU 
> is one way. Its really the only reliable way I think....

Saying MUST is fine, but guaranteeing that it is done correctly is 
another thing. So what bad things happen if this is violated, and the 
request is forked?

I don't think anything terrible happens. The REFER gets forked and 
possibly multiple call attempts happen as a result. The referrer 
subscribes to something that it thinks is the target, and probably is 
one of the targets. So it finds out about that call but not the others.

So some unexpected things happen, nobody's phone explodes. Why not just 
make this SHOULD and count on the users of this to minimize their own pain?

>> If the REFER-Recipient is willing to grant the "norefersub" behavior
>>     for the issued REFER request, it MUST insert a Supported: norefersub
>>     header in the 2xx response to the REFER-Issuer.  In this case no
>>     dialog is created.
> 
> This is not correct usage of Supported/Require. If you place Supported 
> in a request, it means that an extension is understood. If the server 
> sends a response which requires that the client understand the extension 
> in order to process it, it includes a Require in the response. Thus, in 
> the case above, the norefersub needs to appear in a Require in the 2xx, 
> not a Supported.
> 
> That said, I see little value in having the REFER-Recipient make the 
> choice about whether to create the implicit subscription. Isnt it the 
> REFER-issuer that really understands the usage behind the REFER, and 
> thus, whether notifies are needed? I'd propose that it be used only in 
> the Require header field in the SUBSCRIBE.

I figured this was intended to allow the referrer to cope with referees 
that don't support this option. In that case it gets all those implicit 
subscriptions, and has to kill them off. But at least it gets its job done.

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From sip-bounces@ietf.org  Tue Feb  1 17:44:39 2005
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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Abbreviated WGLC for draft-ietf-sip-refer-with-norefers
	ub-00.txt
To: Robert Sparks <rjsparks@nostrum.com>
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Robert,

Section 3, 2nd paragraph: "One purpose of requiring the implicit
subscription and initial NOTIFY is to allow for the situation where the
REFER request gets forked and the REFER-Issuer needs a way to see the
multiple dialogs that may be established as a result of the forked REFER."
Surely the REFER-Issuer will see the multiple dialogs as a result of
receiving multiple 2xx responses to the REFER request?

Consequently, concerning the following statement in section 4: "The
"norefersub" option tag MUST be used by the REFER-Issuer only when the
REFER-Issuer can be certain that the REFER request will not be forked." - is
it really necessary for MUST strength?

Then at the end of section 4: "If the REFER-Recipient is willing to grant
the "norefersub" behavior for the issued REFER request, it MUST insert a
Supported: norefersub header in the 2xx response to the REFER-Issuer.  In
this case no dialog is created."
Surely a dialog is created by the 2xx response to REFER?

John
 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Robert Sparks
> Sent: 31 January 2005 17:34
> To: Robert Sparks
> Cc: SIP WG; Rohan Mahy; Orit Levin; Dean Willis
> Subject: Re: [Sip] Abbreviated WGLC for 
> draft-ietf-sip-refer-with-norefersub-00.txt
> 
> The LC the chairs initiated ends today.
> 
> Please read this draft and comment. It's very short - please do this  
> now.
> 
> We've talked about this idea a lot - let make sure the talk 
> is captured  
> in the draft.
> 
> RjS
> 
> On Jan 26, 2005, at 4:22 PM, Robert Sparks wrote:
> 
> > Initial LC Comments:
> >
> > First paragraph of Section 3 calls out 481ing an initial 
> NOTIFY. The  
> > -dialog-usage-
> > draft shows that is a bad idea, especially for in-dialog REFERs.  
> > Suggest trimming
> > the sentence to end after (Expires 0).
> >
> > Section 4 fourth paragraph: It would help implementers to call out  
> > explicitly that the
> > 200 response to the REFER will not contain a Supported or Require  
> > header containing
> > the norefersub token.
> >
> > Section 4 fifth paragraph: This needs more definition. How is it  
> > rejected? 603, 420, or
> > something else? Implementors will need some guidance here 
> on when to  
> > use which
> > error response. I think (but am not dead certain at the 
> moment) that  
> > 420 is almost always
> > what you want to use.
> >
> > Section 4 sixth paragraph: Is Supported: in the response 
> really the  
> > right thing? Eventlist
> > uses Require: in the response to a SUBSCRIBE. The intent of the  
> > normative behavior in
> > this paragraph is to let a Refer-issuer who only included 
> Supported:  
> > know whether or
> > not a subscription was actually created. Require: (I'm using this  
> > extension) in the response
> > fits that better than Supported: (I could use this 
> extension if you  
> > want to).
> >
> > Section 4 sixth paragraph: Suggest replacing "In this case, 
> no dialog  
> > is created" with
> > "In this case, no implicit subscription is created. As a 
> consequence,  
> > no new dialog
> > is created if this REFER was issued outside any existing dialog."
> >
> > Section 5: Probably worth also calling out using 
> Max-Forwards: 0 to  
> > prevent forking
> > for those (rare now, but hopefully not rare forever) cases where  
> > signaling can get to
> > the refer target without going through a proxy.
> >
> > Section 6: Must replace "tradewind" with "example" throughout.
> >
> > Section 8: "threads" should be "threats", It would be good 
> to pull any  
> > discussion
> > that led to this conclusion together for when the IESG asks if we  
> > _really_ thought
> > about this. Have we talked about what would happen if a SIP  
> > intermediary (or an
> > unauthorized MitM) just injected a Require: norefersub into 
> a REFER  
> > request?
> >
> > RjS
> >
> >
> > On Jan 24, 2005, at 5:07 PM, Rohan Mahy wrote:
> >
> >> Hi Folks,
> >>
> >> I'd like to begin an abbreviated WGLC for:
> >>
> >> 	http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-with- 
> >> norefersub-00.txt
> >>
> >> Since this document is extremely short and has been discussed  
> >> extensively, WGLC will end on February 1st.
> >>
> >> thanks,
> >> -rohan
> >>
> >>
> >> _______________________________________________
> >> 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  Wed Feb  2 13:29:13 2005
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From: "Brett Tate" <brett@broadsoft.com>
To: "'Shekar, Nagesh Soma \(Shekar\)** CTR **'" <shek@lucent.com>,
        <sip@ietf.org>
Subject: RE: [Sip] Requests within SIP Dialog and Contact Header
Date: Wed, 2 Feb 2005 13:16:39 -0500
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> The Contact header is defined as a Mandatory 
> Header in some of Request messages,
> like NOTIFY as a mandatory header. 
> 
> I am trying to understand if this is required 
> for SIP header like Contact to be mandatory
> in SIP requests that are sent in a dialog that 
> is already setup with a INVITE transaction.
> If it is required, then what is the intent of 
> making it Mandatory and how it should be used.

The Contact was made mandatory in some requests (NOTIFY, SUBSCRIBE, INVITE)
to help with dialog establishment and dialog recovery.

NOTIFY, SUBSCRIBE, INVITE, and UPDATE are target refresh requests.

Thus a Contact change in these requests indicate an attempt update the
dialog.

http://www.sipwg.org/show_bug.cgi?id=699

http://www1.ietf.org/mail-archive/web/sip/current/msg06960.html




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From sip-bounces@ietf.org  Wed Feb  2 15:19:35 2005
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From: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
To: "'Brett Tate'" <brett@broadsoft.com>, sip@ietf.org
Subject: RE: [Sip] Requests within SIP Dialog and Contact Header
Date: Wed, 2 Feb 2005 15:10:19 -0500 
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Brett,

Thanks for the clarification, if the NOTIFY is expected 
to be  a target  refresh request, then it makes sense 
that the Contact Header be mandatory. Unfortunately, 
there are some implementations that do not include Contact 
Header in NOTIFY messages (Ex. Pingtel Instant xpressa).

I am assuming that the bug is still not fixed. How does the 
bug fixing process work with RFCs, will there be a draft that 
will have the bug fixes.

Nagesh Shekar.

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: Wednesday, February 02, 2005 1:17 PM
> To: 'Shekar, Nagesh Soma (Shekar)** CTR **'; sip@ietf.org
> Subject: RE: [Sip] Requests within SIP Dialog and Contact Header
> 
> 
> > The Contact header is defined as a Mandatory 
> > Header in some of Request messages,
> > like NOTIFY as a mandatory header. 
> > 
> > I am trying to understand if this is required 
> > for SIP header like Contact to be mandatory
> > in SIP requests that are sent in a dialog that 
> > is already setup with a INVITE transaction.
> > If it is required, then what is the intent of 
> > making it Mandatory and how it should be used.
> 
> The Contact was made mandatory in some requests (NOTIFY, 
> SUBSCRIBE, INVITE)
> to help with dialog establishment and dialog recovery.
> 
> NOTIFY, SUBSCRIBE, INVITE, and UPDATE are target refresh requests.
> 
> Thus a Contact change in these requests indicate an attempt update the
> dialog.
> 
> http://www.sipwg.org/show_bug.cgi?id=699
> 
> http://www1.ietf.org/mail-archive/web/sip/current/msg06960.html
> 
> 
> 

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From sip-bounces@ietf.org  Wed Feb  2 17:06:33 2005
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A late thought:

Please point out in the norefersub draft that REFER is a
target refresh request even when this extension is used.

RjS


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Subject: [Sipforum-discussion] Re: [Sip] pri and sip mapping issue
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Hi,
I have sent a request for information to the author about the ClearMode
Draft and have not received any responce. Are we sure that this draft is
alive?

Thanks
Ramesh
Tom Taylor wrote:

>Hi.  My responses below marked with [PTT].
>
>FCG Wang JuLin wrote:
>  
>
>> 
>>
>>Hi,list,
>>
>>Is there any standard in  pri and sip mapping for the following three 
>>items:
>>
>>1)user to user infomation
>>    
>>
>
>[PTT]  To carry the information itself, the obvious method is to add a body of
>type application/octet-stream.  The issue comes in how to signal the service --
>especially, how to indicate what that body is for.
>  
>
>>2)bear capacity( 64K data),UDI.
>>    
>>
>
>[PTT] You can signal and use CLEARMODE
>(http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmode-05.txt).  This
>has been approved by the IESG but is stuck in the RFC Editor's queue because it
>has a dependency on draft-ietf-mmusic-sdp-new-23.txt, which still has to pass Last
>Call.
>  
>
>>3) call transfer service indication mapping.
>>    
>>
>
>[PTT] Leaving aside 3xx redirection responses, you may want to look at the
>Request_History header field
>(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
>  
>
>>Thank you for your response  !
>>
>>Best Regards
>>Wang Julin(Jerry Wang)
>>Product Management
>>Alcatel Shanghai  Bell
>>Phone:+86-21-58541240-9531
>>Fax:+86-21-58342941
>>Mobile: 13817861240
>>E-mail:Julin.Wang@alcatel-sbell.com.cn
>>
>>
>> 
>>
>> 
>>
>> 
>>
>>
>>------------------------------------------------------------------------
>>
>>_______________________________________________
>>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] Questions on the SIP identity draft
Date: Wed, 2 Feb 2005 19:56:00 -0500
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Thread-Topic: [Sip] Questions on the SIP identity draft
Thread-Index: AcUIPHE1Fqd8NVPYSSiCP4QG43xsQQBIwIig
From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "Simone Leggio" <simone.leggio@cs.helsinki.fi>, <sip@ietf.org>
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Hello Simone,

>=20
> I have a couple of questions about the SIP identity draft, i.e.
<snip>
>=20
> Why can't SIP Identity be applied to responses as well? Why can't an=20
> Identity header field be added based on the relevant response fields? =
At=20
> least, final responses like 200 OK.

The answer is that it is possible to do so, but this won't be specified =
in the current sip-identity draft.

The scope of the sip-identity mechanism is to prevent certain trivial =
impersonation attacks that arise primarily in requests. It is much =
easier for me to impersonate the source of a request than it is for me =
to impersonate the destination of the request; the former just requires =
me to erroneously populate the From header field, the latter, however, =
requires me to know that a request is in process, intercept it, =
formulate a plausible response, and submit it in such a way that any =
genuine response is squelched or something. The level of sophistication =
required for response impersonation is far greater than that of request =
impersonation.

Accordingly, we felt that it was valuable to do request identity alone. =
Since there is some amount of controversy surrounding possible =
approaches to response identity, we've left this as an item for future =
work in the current identity draft. Expect to see response identity work =
sometime soon.

> For more security, instead of adding only the addr-spec component of =
the=20
> To header field, it may be added also the tag part. Am I forgetting =
some=20
> relevant security issue?
>=20
> 2) Section 10 says that
>=20
> > The REGISTER method uses Contact header fields in very unusual ways =
that
> >    complicate its applicability to this mechanism.  Accordingly, the
> >    Identity and Identity-Info header MUST NOT appear in REGISTER
>=20
> Why can't the REGISTER be protected? Usually authentication happens=20
> because user agent and server share a common secret, i.e. password. =
What=20
> if they would share a key?=20

As you note, REGISTER is currently protected by the Digest =
authentication system in SIP. sip-identity leverages Digest when the =
authentication service role is instantiated by an intermediary - that =
is, the same Digest credentials that we be used in REGISTER are given to =
an authentication service in order to get an Identity signature. Thus it =
would be kind of pointless to use sip-identity to secure REGISTER when =
the authentication service is instantiated by an intermediary.

Since it is possible, and sufficient, for a user agent to authenticate =
itself to a registrar via Digest, I saw no reason to go out of our way =
and recommend a second potential mechanism for authenticating the source =
of a registration.

> The user agent signs the REGISTER with its=20
> key, and the server uses it for verifying the identity of the=20
> authenticating entity.=20

I hear what you're saying (this is a case where the user agent =
instantiates the auth service role), but sip-identity is based on a =
concept of namespace authorization. The identity that you prove is a =
URI, and URI has a host portion - like sip:jon@example.com. To sign an =
Identity header, one must have a certificate (not just a key) whose =
subject corresponds to the host portion of the identity which you want =
to sign (e.g. example.com). When a UA registers, it is associating =
itself with an AoR controlled by a registrar (say, the regisrar of =
example.com, where the user is 'jon'). But if the AoR of a user is =
controlled by a registrar, it is rare that the user will possess the =
certificate for that AoR - which is necessary in order for it to act as =
an auth svc for that AoR.

At a high level, I guess it would be possible to do this, but it would =
have very limited applicability, and it doesn't seem to fill any holes =
in SIP's security story.

Jon Peterson
NeuStar, Inc.

> Of course, the user agent has communicated=20
> previously its key to the server, probably when the user registers for =

> the first time in that SIP domain. Anyway, how key communication is=20
> performed would be out of scope for SIP Identity.
>=20
> Best regards,
>=20
> 	Simone Leggio
>=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  Thu Feb  3 01:20:24 2005
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From: Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
Subject: RE: [Sip] sip and dns
In-reply-to: <CMELJPNMCKECANMIHPDLEEGPCAAA.suganya@dgbmicro.com>
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Hi,
SIP uses DND procedure to resolve SIP URI into
IP address, Port and Transport of the  next hop to contact.

In order to get this support your SIP stack should support the Client DNS
procedures
defined in RFC 3263. Also your stack should have support for sending NAPTR
and SRV query as well as maintaioning the list of all resolved addresses.

If your SIP stack is supporting then your application can use them one by
one in case of Message send failure.
Your SIP stack should be configured with a list of DNS servers and domain
suffix.

DNS query from UAC is sent to DNS server.

Rajeew Kumar Singh



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Suganya
Sent: Wednesday, February 02, 2005 5:34 AM
To: sip@ietf.org
Subject: [Sip] sip and dns


Hi,

I have the following doubts and need some help.

a) In order to contruct DNS query, I googled-up and found RFCs 1034 and
1035. But it doesn't talk of SIP. Can those guidelines be used for
constructing the DNS for SIP related queries?

b) Where to send the DNS query from UAC?

c) Unlike RFC 3261 which clearly tells how to construct invite, ack, etc...
messages, RFC 3263 specifies no syntax or query format to construct the DNS
query.

Regards,
B.Suganya


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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 Feb  3 04:57:08 2005
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Cc: sip@ietf.org, jdrosen@dynamicsoft.com, schulzrinne@cs.columbia.edu
Subject: [Sip] Re: draft-ietf-sip-sctp-05 clarification
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Hi,

thanks for your comments. Answers inline:

> Section 3.2
> 
> Using TCP as transport protocol normally guarenteed
> that the messages are delivered in sequence. However,
> the usage of SCTP nullifies that, if proper optinos
> are not used.

Subsequent sections of the draft specify how to use those options.

> Multihoming in SCTP can be used as load sharing too,
> which needs a mention because this needs to be known
> at the SIP application level so that messages do not
> end up at wrong places.

That is SCTP standard stuff. We do not need to specify anything new so 
that applications can use the multihoming capabilities that SCTP provides.

By the way, I circulated a preliminary version 06 of the draft in the 
list some time ago. Given that I have not received any extra comment on 
this new version (which includes Cullen's comments), I will be 
submitting it shortly.

Thanks for the comments,

Gonzalo


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From sip-bounces@ietf.org  Thu Feb  3 07:52:49 2005
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From: Rayees Khan <rakhanhss@yahoo.com>
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Subject: [Sip] draft-ietf-sip-sctp-05 clarification
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Hi,

I have noted the following point while going through
the mentioned Draft.


Section 3.2

Using TCP as transport protocol normally guarenteed
that the messages are delivered in sequence. However,
the usage of SCTP nullifies that, if proper optinos
are not used.

Multihoming in SCTP can be used as load sharing too,
which needs a mention because this needs to be known
at the SIP application level so that messages do not
end up at wrong places.


regards
Rayees




		
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From: "Nuno Filipe Oliveira Novo" <nnovo@av.it.pt>
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Subject: [Sip] SIP Loose routing
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Dear all,
When studing loose routing in Sip i stumble in a question.
imagine that we have the situation:
Alice . . .atlanta.com . . . . . biloxi.com . . . Bob

Alice is registered in atlanta.com and Bob in biloxi.com. 
Let's assume that atlanta.com is a loose router proxy and 
biloxi.com is a strict router proxy.
When Alice try to contact Bob she send a INVITE to the 
proxy atlanta.com with the request URI filled with Bob sip 
URI.So far so good.

Now the proxy atlanta.com needs to contact with the 
biloxi.com.

The question is does the proxy atlanta.com, using a search 
mechanism based on DNS, knows that the proxy biloxi.com is 
a strict router?
Or proxy atlanta.com assume that bilibox.com is a stric 
router because atlanta.com doesn't have that kind of 
information about bilibo.com?
Is there any way of a proxy knowing if the next proxy is a 
strict or loose router in this situation?

Best Regards:
Nuno Novo.

_______________________________________________
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: [Sipforum-discussion] Re: [Sip] pri and sip mapping issue
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Hi.  My responses below marked with [PTT].

FCG Wang JuLin wrote:
>  
> 
> Hi,list,
> 
> Is there any standard in  pri and sip mapping for the following three 
> items:
> 
> 1)user to user infomation

[PTT]  To carry the information itself, the obvious method is to add a body of
type application/octet-stream.  The issue comes in how to signal the service --
especially, how to indicate what that body is for.
> 
> 2)bear capacity( 64K data),UDI.

[PTT] You can signal and use CLEARMODE
(http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmode-05.txt).  This
has been approved by the IESG but is stuck in the RFC Editor's queue because it
has a dependency on draft-ietf-mmusic-sdp-new-23.txt, which still has to pass Last
Call.
> 
> 3) call transfer service indication mapping.

[PTT] Leaving aside 3xx redirection responses, you may want to look at the
Request_History header field
(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
> 
> Thank you for your response  !
> 
> Best Regards
> Wang Julin(Jerry Wang)
> Product Management
> Alcatel Shanghai  Bell
> Phone:+86-21-58541240-9531
> Fax:+86-21-58342941
> Mobile: 13817861240
> E-mail:Julin.Wang@alcatel-sbell.com.cn
> 
> 
>  
> 
>  
> 
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> 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

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com
------
To unsubscribe from the Sipforum-discussion mailing list visit https://mbox.su.se/mailman/listinfo/sipforum-discussion



_______________________________________________
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: taylor@nortelnetworks.com, justin.millner@ietf.org
Subject: Re: [Sipforum-discussion] Re: [Sip] pri and sip mapping issue
From: Takuya Sawada <tu-sawada@kddi.com>
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Hi,

> > 1)user to user infomation
> 
> [PTT]  To carry the information itself, the obvious method is to add a body of
> type application/octet-stream.  The issue comes in how to signal the service --
> especially, how to indicate what that body is for.

I think one possible solution is to use ITU-T Q.1980.1.
MIME type will be the following;

  Content-Type: application/nss
  Content-Disposition: signal

The problem of this solution is that Q.1980.1 is not so popular, I think.

Am I correct?

Regards,
Takuya

> Hi.  My responses below marked with [PTT].
> 
> FCG Wang JuLin wrote:
> >  
> > 
> > Hi,list,
> > 
> > Is there any standard in  pri and sip mapping for the following three 
> > items:
> > 
> > 1)user to user infomation
> 
> [PTT]  To carry the information itself, the obvious method is to add a body of
> type application/octet-stream.  The issue comes in how to signal the service --
> especially, how to indicate what that body is for.
> > 
> > 2)bear capacity( 64K data),UDI.
> 
> [PTT] You can signal and use CLEARMODE
> (http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmode-05.txt).  This
> has been approved by the IESG but is stuck in the RFC Editor's queue because it
> has a dependency on draft-ietf-mmusic-sdp-new-23.txt, which still has to pass Last
> Call.
> > 
> > 3) call transfer service indication mapping.
> 
> [PTT] Leaving aside 3xx redirection responses, you may want to look at the
> Request_History header field
> (http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
> > 
> > Thank you for your response  !
> > 
> > Best Regards
> > Wang Julin(Jerry Wang)
> > Product Management
> > Alcatel Shanghai  Bell
> > Phone:+86-21-58541240-9531
> > Fax:+86-21-58342941
> > Mobile: 13817861240
> > E-mail:Julin.Wang@alcatel-sbell.com.cn
> > 
> > 
> >  
> > 
> >  
> > 
> >  
> > 
> > 
> > ------------------------------------------------------------------------
> > 
> > _______________________________________________
> > 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
> 
> -- 
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
> ------
> To unsubscribe from the Sipforum-discussion mailing list visit https://mbox.su.se/mailman/listinfo/sipforum-discussion
> 
> 
> 
> _______________________________________________
> 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


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.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


From sip-bounces@ietf.org  Thu Feb  3 11:54:38 2005
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Requests within SIP Dialog and Contact Header
Date: Thu, 3 Feb 2005 10:34:43 -0600
To: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
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REFER is also a target refresh request, and a similar bug has been
posted against 3515 for clarification.

To your last question: You can't change RFCs once they are published.
You can issue new RFCs that update older ones. We queue these 
clarifications
up in our bug database until it is appropriate to issue an updating 
RFC. Whenever
possible, we mention those clarifications when they affect other new 
work.

RjS

On Feb 2, 2005, at 2:10 PM, Shekar, Nagesh Soma (Shekar)** CTR ** wrote:

> Brett,
>
> Thanks for the clarification, if the NOTIFY is expected
> to be  a target  refresh request, then it makes sense
> that the Contact Header be mandatory. Unfortunately,
> there are some implementations that do not include Contact
> Header in NOTIFY messages (Ex. Pingtel Instant xpressa).
>
> I am assuming that the bug is still not fixed. How does the
> bug fixing process work with RFCs, will there be a draft that
> will have the bug fixes.
>
> Nagesh Shekar.
>
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com]
>> Sent: Wednesday, February 02, 2005 1:17 PM
>> To: 'Shekar, Nagesh Soma (Shekar)** CTR **'; sip@ietf.org
>> Subject: RE: [Sip] Requests within SIP Dialog and Contact Header
>>
>>
>>> The Contact header is defined as a Mandatory
>>> Header in some of Request messages,
>>> like NOTIFY as a mandatory header.
>>>
>>> I am trying to understand if this is required
>>> for SIP header like Contact to be mandatory
>>> in SIP requests that are sent in a dialog that
>>> is already setup with a INVITE transaction.
>>> If it is required, then what is the intent of
>>> making it Mandatory and how it should be used.
>>
>> The Contact was made mandatory in some requests (NOTIFY,
>> SUBSCRIBE, INVITE)
>> to help with dialog establishment and dialog recovery.
>>
>> NOTIFY, SUBSCRIBE, INVITE, and UPDATE are target refresh requests.
>>
>> Thus a Contact change in these requests indicate an attempt update the
>> dialog.
>>
>> http://www.sipwg.org/show_bug.cgi?id=699
>>
>> http://www1.ietf.org/mail-archive/web/sip/current/msg06960.html
>>
>>
>>
>
> _______________________________________________
> 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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Use sipping@ietf.org for new developments on the application of sip


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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] SIP Loose routing
Date: Thu, 3 Feb 2005 10:41:45 -0600
To: "Nuno Filipe Oliveira Novo" <nnovo@av.it.pt>
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Inline:

On Feb 2, 2005, at 11:59 AM, Nuno Filipe Oliveira Novo wrote:

> Dear all,
> When studing loose routing in Sip i stumble in a question.
> imagine that we have the situation:
> Alice . . .atlanta.com . . . . . biloxi.com . . . Bob
>
> Alice is registered in atlanta.com and Bob in biloxi.com. Let's assume 
> that atlanta.com is a loose router proxy and biloxi.com is a strict 
> router proxy.
> When Alice try to contact Bob she send a INVITE to the proxy 
> atlanta.com with the request URI filled with Bob sip URI.So far so 
> good.
>
> Now the proxy atlanta.com needs to contact with the biloxi.com.
>
> The question is does the proxy atlanta.com, using a search mechanism 
> based on DNS, knows that the proxy biloxi.com is a strict router?
> Or proxy atlanta.com assume that bilibox.com is a stric router because 
> atlanta.com doesn't have that kind of information about bilibo.com?
> Is there any way of a proxy knowing if the next proxy is a strict or 
> loose router in this situation?

Inside an established dialog (with a route-set built with Record-Route, 
atlanta.com knows whether biloxi.com is a loose
or strict router by looking for ";lr" in the next Route URI.

I think you're asking about the case where Alice is sending an initial 
INVITE to sip:bob@biloxi.com using a preloaded
route that sends her request through atlanta.com. In this case, Alice's 
initial request is going to look like:
INVITE sip:bob@biloxi.com
Route: <sip:atlanta.com;lr>

altanta.com will consume that route header. There is no subsequent 
route header, so it makes no
difference if biloxi.com is a strict or loose router.  The request 
going from atlanta  to biloxi will be
just

INVITE sip:bob@biloxi.com

RjS

> Best Regards:
> Nuno Novo.
>
> _______________________________________________
> 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 sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Thu Feb  3 12:26:15 2005
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To: "'Robert Sparks'" <rjsparks@nostrum.com>,
        "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
Subject: RE: [Sip] Requests within SIP Dialog and Contact Header
Date: Thu, 3 Feb 2005 11:51:59 -0500 
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Is there way to get a consolidated list of pending changes 
for a particular RFC. It appears that the window between
a RFC baseline and an update can be quite long, how
do implementations typically take into consideration the
changes pending for an RFC.

Nagesh Shekar.

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> Sent: Thursday, February 03, 2005 11:35 AM
> To: Shekar, Nagesh Soma (Shekar)** CTR **
> Cc: 'Brett Tate'; sip@ietf.org
> Subject: Re: [Sip] Requests within SIP Dialog and Contact Header
> 
> 
> REFER is also a target refresh request, and a similar bug has been
> posted against 3515 for clarification.
> 
> To your last question: You can't change RFCs once they are published.
> You can issue new RFCs that update older ones. We queue these 
> clarifications
> up in our bug database until it is appropriate to issue an updating 
> RFC. Whenever
> possible, we mention those clarifications when they affect other new 
> work.
> 
> RjS
> 
> On Feb 2, 2005, at 2:10 PM, Shekar, Nagesh Soma (Shekar)** 
> CTR ** wrote:
> 
> > Brett,
> >
> > Thanks for the clarification, if the NOTIFY is expected
> > to be  a target  refresh request, then it makes sense
> > that the Contact Header be mandatory. Unfortunately,
> > there are some implementations that do not include Contact
> > Header in NOTIFY messages (Ex. Pingtel Instant xpressa).
> >
> > I am assuming that the bug is still not fixed. How does the
> > bug fixing process work with RFCs, will there be a draft that
> > will have the bug fixes.
> >
> > Nagesh Shekar.
> >
> >> -----Original Message-----
> >> From: Brett Tate [mailto:brett@broadsoft.com]
> >> Sent: Wednesday, February 02, 2005 1:17 PM
> >> To: 'Shekar, Nagesh Soma (Shekar)** CTR **'; sip@ietf.org
> >> Subject: RE: [Sip] Requests within SIP Dialog and Contact Header
> >>
> >>
> >>> The Contact header is defined as a Mandatory
> >>> Header in some of Request messages,
> >>> like NOTIFY as a mandatory header.
> >>>
> >>> I am trying to understand if this is required
> >>> for SIP header like Contact to be mandatory
> >>> in SIP requests that are sent in a dialog that
> >>> is already setup with a INVITE transaction.
> >>> If it is required, then what is the intent of
> >>> making it Mandatory and how it should be used.
> >>
> >> The Contact was made mandatory in some requests (NOTIFY,
> >> SUBSCRIBE, INVITE)
> >> to help with dialog establishment and dialog recovery.
> >>
> >> NOTIFY, SUBSCRIBE, INVITE, and UPDATE are target refresh requests.
> >>
> >> Thus a Contact change in these requests indicate an 
> attempt update the
> >> dialog.
> >>
> >> http://www.sipwg.org/show_bug.cgi?id=699
> >>
> >> http://www1.ietf.org/mail-archive/web/sip/current/msg06960.html
> >>
> >>
> >>
> >
> > _______________________________________________
> > 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  Thu Feb  3 12:28:21 2005
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Requests within SIP Dialog and Contact Header
Date: Thu, 3 Feb 2005 10:58:15 -0600
To: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
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Right now the best way is to browse the bug database at
bugs.sipit.net (aka www.sipwg.org).

RjS

On Feb 3, 2005, at 10:51 AM, Shekar, Nagesh Soma (Shekar)** CTR ** 
wrote:

> Is there way to get a consolidated list of pending changes
> for a particular RFC. It appears that the window between
> a RFC baseline and an update can be quite long, how
> do implementations typically take into consideration the
> changes pending for an RFC.
>
> Nagesh Shekar.
>
>> -----Original Message-----
>> From: Robert Sparks [mailto:rjsparks@nostrum.com]
>> Sent: Thursday, February 03, 2005 11:35 AM
>> To: Shekar, Nagesh Soma (Shekar)** CTR **
>> Cc: 'Brett Tate'; sip@ietf.org
>> Subject: Re: [Sip] Requests within SIP Dialog and Contact Header
>>
>>
>> REFER is also a target refresh request, and a similar bug has been
>> posted against 3515 for clarification.
>>
>> To your last question: You can't change RFCs once they are published.
>> You can issue new RFCs that update older ones. We queue these
>> clarifications
>> up in our bug database until it is appropriate to issue an updating
>> RFC. Whenever
>> possible, we mention those clarifications when they affect other new
>> work.
>>
>> RjS
>>
>> On Feb 2, 2005, at 2:10 PM, Shekar, Nagesh Soma (Shekar)**
>> CTR ** wrote:
>>
>>> Brett,
>>>
>>> Thanks for the clarification, if the NOTIFY is expected
>>> to be  a target  refresh request, then it makes sense
>>> that the Contact Header be mandatory. Unfortunately,
>>> there are some implementations that do not include Contact
>>> Header in NOTIFY messages (Ex. Pingtel Instant xpressa).
>>>
>>> I am assuming that the bug is still not fixed. How does the
>>> bug fixing process work with RFCs, will there be a draft that
>>> will have the bug fixes.
>>>
>>> Nagesh Shekar.
>>>
>>>> -----Original Message-----
>>>> From: Brett Tate [mailto:brett@broadsoft.com]
>>>> Sent: Wednesday, February 02, 2005 1:17 PM
>>>> To: 'Shekar, Nagesh Soma (Shekar)** CTR **'; sip@ietf.org
>>>> Subject: RE: [Sip] Requests within SIP Dialog and Contact Header
>>>>
>>>>
>>>>> The Contact header is defined as a Mandatory
>>>>> Header in some of Request messages,
>>>>> like NOTIFY as a mandatory header.
>>>>>
>>>>> I am trying to understand if this is required
>>>>> for SIP header like Contact to be mandatory
>>>>> in SIP requests that are sent in a dialog that
>>>>> is already setup with a INVITE transaction.
>>>>> If it is required, then what is the intent of
>>>>> making it Mandatory and how it should be used.
>>>>
>>>> The Contact was made mandatory in some requests (NOTIFY,
>>>> SUBSCRIBE, INVITE)
>>>> to help with dialog establishment and dialog recovery.
>>>>
>>>> NOTIFY, SUBSCRIBE, INVITE, and UPDATE are target refresh requests.
>>>>
>>>> Thus a Contact change in these requests indicate an
>> attempt update the
>>>> dialog.
>>>>
>>>> http://www.sipwg.org/show_bug.cgi?id=699
>>>>
>>>> http://www1.ietf.org/mail-archive/web/sip/current/msg06960.html
>>>>
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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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To: Takuya Sawada <tu-sawada@kddi.com>
Subject: Re: [Sipforum-discussion] Re: [Sip] pri and sip mapping issue
References: <60BBB1231E9EA443BD400F30A47C0A7026A181@asbmail1.sbell.com.cn>	<070601c50973$384963f0$2464a8c0@dsmsystems.local>
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Q.1980.1 is text-encoded international ISUP.  In other words, it is an alternative 
to RFC 3204 binary ISUP encapsulation.  Since the question was about PRI rather 
thyan ISUP, neither Q.1980.1 nor RFC 3204 is a direct answer.

Takuya Sawada wrote:
> Hi,
> 
> 
>>>1)user to user infomation
>>
>>[PTT]  To carry the information itself, the obvious method is to add a body of
>>type application/octet-stream.  The issue comes in how to signal the service --
>>especially, how to indicate what that body is for.
> 
> 
> I think one possible solution is to use ITU-T Q.1980.1.
> MIME type will be the following;
> 
>   Content-Type: application/nss
>   Content-Disposition: signal
> 
> The problem of this solution is that Q.1980.1 is not so popular, I think.
> 
> Am I correct?
> 
> Regards,
> Takuya
> 
[snip]

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortel.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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To: taylor@nortel.com, tu-sawada@kddi.com
Subject: Re: [Sipforum-discussion] Re: [Sip] pri and sip mapping issue
From: Takuya Sawada <tu-sawada@kddi.com>
References: <60BBB1231E9EA443BD400F30A47C0A7026A181@asbmail1.sbell.com.cn>
	<070601c50973$384963f0$2464a8c0@dsmsystems.local>
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Hi,

In my understanding, the main difference between Q.1980.1 and RFC 3204,
other than format, is that the former allows partial message contents
and the latter only allows the whole complete message.

PRI gateway can not know how to construct the whole ISUP message which
is required to use RFC 3204. It can only know end-to-end information
like UUI or HLC/LLC, which is conveyed in ATP parameter in ISUP.
Then I think it may be useful for PRI gateway to set UUI infomation
in Q.1980.1 message body to carry it to the peer or the ISUP gateway,
theoretically.

* Should we move this discussion to sipping, if it continues? 
  or stop it now (as it is not an IETF work)?

Regards,
Takuya

> Q.1980.1 is text-encoded international ISUP.  In other words, it is an alternative 
> to RFC 3204 binary ISUP encapsulation.  Since the question was about PRI rather 
> thyan ISUP, neither Q.1980.1 nor RFC 3204 is a direct answer.
> 
> Takuya Sawada wrote:
> > Hi,
> > 
> > 
> >>>1)user to user infomation
> >>
> >>[PTT]  To carry the information itself, the obvious method is to add a body of
> >>type application/octet-stream.  The issue comes in how to signal the service --
> >>especially, how to indicate what that body is for.
> > 
> > 
> > I think one possible solution is to use ITU-T Q.1980.1.
> > MIME type will be the following;
> > 
> >   Content-Type: application/nss
> >   Content-Disposition: signal
> > 
> > The problem of this solution is that Q.1980.1 is not so popular, I think.
> > 
> > Am I correct?
> > 
> > Regards,
> > Takuya
> > 
> [snip]
> 
> -- 
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortel.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


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.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: [Sipforum-discussion] Re: [Sip] pri and sip mapping issue
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From: "Takuya Sawada" <tu-sawada@kddi.com>
References: <60BBB1231E9EA443BD400F30A47C0A7026A181@asbmail1.sbell.com.cn><070601c50973$384963f0$2464a8c0@dsmsystems.local><200502032236.DGC58161.TBX-UBBVE@kddi.com><4202B2D9.10506@nortel.com>
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Hi,

In my understanding, the main difference between Q.1980.1 and RFC 3204,
other than format, is that the former allows partial message contents
and the latter only allows the whole complete message.

PRI gateway can not know how to construct the whole ISUP message which
is required to use RFC 3204. It can only know end-to-end information
like UUI or HLC/LLC, which is conveyed in ATP parameter in ISUP.
Then I think it may be useful for PRI gateway to set UUI infomation
in Q.1980.1 message body to carry it to the peer or the ISUP gateway,
theoretically.

* Should we move this discussion to sipping, if it continues? 
  or stop it now (as it is not an IETF work)?

Regards,
Takuya

> Q.1980.1 is text-encoded international ISUP.  In other words, it is an alternative 
> to RFC 3204 binary ISUP encapsulation.  Since the question was about PRI rather 
> thyan ISUP, neither Q.1980.1 nor RFC 3204 is a direct answer.
> 
> Takuya Sawada wrote:
> > Hi,
> > 
> > 
> >>>1)user to user infomation
> >>
> >>[PTT]  To carry the information itself, the obvious method is to add a body of
> >>type application/octet-stream.  The issue comes in how to signal the service --
> >>especially, how to indicate what that body is for.
> > 
> > 
> > I think one possible solution is to use ITU-T Q.1980.1.
> > MIME type will be the following;
> > 
> >   Content-Type: application/nss
> >   Content-Disposition: signal
> > 
> > The problem of this solution is that Q.1980.1 is not so popular, I think.
> > 
> > Am I correct?
> > 
> > Regards,
> > Takuya
> > 
> [snip]
> 
> -- 
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortel.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


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com
------
To unsubscribe from the Sipforum-discussion mailing list visit https://mbox.su.se/mailman/listinfo/sipforum-discussion



_______________________________________________
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] processing bottleneck in proxy
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Dear All,

is there any experience/data regarding the processing load that comes 
from  managing transport level connections (e.g., TCP, TLS states) and 
manipulating SIP messages in SIP proxies? Which is the dominant one?

(I posted this question on the sip-implementors list, but there was no 
response)

Best Regards
Peter



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This list is for NEW development of the core SIP Protocol
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Hi,

Can someone tell me what is the status of
draft-johnston-sip-osp-token-06. Currently, it had expired on December
2004.
=20
Thanks,
Tim

=20


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<pre><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Can someone tell me what is =
the status of </span></font>draft-johnston-sip-osp-token-06. Currently, =
it had expired on December 2004.<o:p></o:p></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
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style=3D'font-size:10.0pt'>Thanks,<o:p></o:p></span></font></pre><pre><fo=
nt
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_______________________________________________
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--===============0782899250==--



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Date: Sun, 06 Feb 2005 02:20:52 +0000
From: "Johnston, Alan" <alan.johnston@mci.com>
Subject: RE: [Sip] status of draft-johnston-sip-osp-token-06
To: Tim Wong <twong@asg.bellsouth.net>, sip@ietf.org
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Hi Tim,
=20
It hasn't expired (it is still in the archives).  It is currently under
review by the IESG - don't have a current status.  One day, it will  be
an RFC...
=20
Thanks,
Alan Johnston
sip:alan@sipstation.com

	-----Original Message-----
	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Tim Wong
	Sent: Friday, February 04, 2005 2:40 PM
	To: sip@ietf.org
	Subject: [Sip] status of draft-johnston-sip-osp-token-06
=09
=09

	Hi,

	Can someone tell me what is the status of
draft-johnston-sip-osp-token-06. Currently, it had expired on December
2004.
	=20
	Thanks,
	Tim

	=20


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<DIV><SPAN class=3D442121602-06022005><FONT face=3DArial color=3D#0000ff =
size=3D2>Hi=20
Tim,</FONT></SPAN></DIV>
<DIV><SPAN class=3D442121602-06022005><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D442121602-06022005><FONT face=3DArial color=3D#0000ff =
size=3D2>It=20
hasn't expired (it is still in the archives).&nbsp; It is currently =
under review=20
by the IESG - don't have a current status.&nbsp; One day, it will&nbsp; =
be an=20
RFC...</FONT></SPAN></DIV>
<DIV><SPAN class=3D442121602-06022005><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D442121602-06022005><FONT face=3DArial color=3D#0000ff =

size=3D2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=3D442121602-06022005><FONT face=3DArial color=3D#0000ff =
size=3D2>Alan=20
Johnston</FONT></SPAN></DIV>
<DIV><SPAN class=3D442121602-06022005><FONT face=3DArial color=3D#0000ff =

size=3D2>sip:alan@sipstation.com</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of =
</B>Tim=20
  Wong<BR><B>Sent:</B> Friday, February 04, 2005 2:40 PM<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] status of=20
  draft-johnston-sip-osp-token-06<BR><BR></FONT></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Hi,<o:p></o:p></SPAN></FONT></P><PRE><FONT face=3DArial =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Can someone =
tell me what is the status of =
</SPAN></FONT>draft-johnston-sip-osp-token-06. Currently, it had expired =
on December 2004.<o:p></o:p></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt">Thanks,<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt">Tim<o:p></o:p></SPAN></FONT></PRE>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P></DIV></BLOCKQUOTE></BODY></HTM=
L>
=00=

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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
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--===============1513682410==--



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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		: The Stream Control Transmission Protocol (SCTP) 
			  as a Transport for the Session Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg, et al.
	Filename	: draft-ietf-sip-sctp-06.txt
	Pages		: 11
	Date		: 2005-2-4
	
This document specifies a mechanism for usage of SCTP (the Stream
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
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Subject: [Sip] Refer-To header/RFC3261 ambiguity?
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All,

We've recently hit a problem with REFER interoperability, on receipt
of a REFER request with a Refer-To header like this:

   Refer-To: sip:user@domain.com;user=3Dphone?Replaces=3Dsomething

The issue is whether the ";user=3Dphone" should be parsed as a URI
parameter or a header parameter.  Obviously the intent here is that it
is a URI parameter, but allowing this runs counter to existing SIP
precedent.

The problem appears to be that the specification of the Refer-To
header (in RFC 3515 section 2.1) is missing some key words that other
headers of similar format (namely the To, From, Contact and Reply-To
headers) have in RFC 3261.

RFC3261 section 20 states:

   The Contact, From, and To header fields contain a URI.  If the URI
   contains a comma, question mark or semicolon, the URI MUST be
   enclosed in angle brackets (< and >).  Any URI parameters are
   contained within these brackets.  If the URI is not enclosed in angle
   brackets, any semicolon-delimited parameters are header-parameters,
   not URI parameters.

RFC3261 section 20.31 (Reply-To) states:

   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, question mark, or
   semicolon.  Syntax issues are discussed in Section 7.3.1.

Note that To, Reply-To, From and Contact are the only RFC3261 headers
that are defined as "addr-spec / name-addr ...", similarly to=20
Refer-To.

For the RFC 3261 headers - if we are understanding correctly - these
words exist in order to force the protocol to be clear about the
structural difference between

   Contact: <sip:user@domain.com;transport=3DUDP>

and

   Contact: <sip:user@domain.com>;expires=3D30

It seems that it would be a good idea to impose the same requirement
for the Refer-To header.  Our guess, therefore, is that RFC 3515
should have the same words for the Refer-To header as RFC 3261 has for
the To/From/Contact headers.  Are we right?

Regards,
Paul D Smith
Network Protocols Group=20
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com=20
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com


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charset=3Diso-8859-1">
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<!-- Converted from text/rtf format -->

<P><FONT SIZE=3D2 FACE=3D"Courier New">All,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">We've recently hit a problem with =
REFER interoperability, on receipt</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">of a REFER request with a =
Refer-To header like this:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; Refer-To: =
sip:user@domain.com;user=3Dphone?Replaces=3Dsomething</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The issue is whether the =
&quot;;user=3Dphone&quot; should be parsed as a URI</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">parameter or a header =
parameter.&nbsp; Obviously the intent here is that it</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">is a URI parameter, but allowing =
this runs counter to existing SIP</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">precedent.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The problem appears to be that =
the specification of the Refer-To</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">header (in RFC 3515 section 2.1) =
is missing some key words that other</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">headers of similar format =
(namely the To, From, Contact and Reply-To</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">headers) have in RFC =
3261.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">RFC3261 section 20 states:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; The Contact, From, =
and To header fields contain a URI.&nbsp; If the URI</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; contains a comma, =
question mark or semicolon, the URI MUST be</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; enclosed in angle =
brackets (&lt; and &gt;).&nbsp; Any URI parameters are</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; contained within =
these brackets.&nbsp; If the URI is not enclosed in angle</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; brackets, any =
semicolon-delimited parameters are header-parameters,</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; not URI =
parameters.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">RFC3261 section 20.31 (Reply-To) =
states:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; Even if the =
&quot;display-name&quot; is empty, the &quot;name-addr&quot; form MUST =
be</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; used if the =
&quot;addr-spec&quot; contains a comma, question mark, or</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; semicolon.&nbsp; =
Syntax issues are discussed in Section 7.3.1.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Note that To, Reply-To, From and =
Contact are the only RFC3261 headers</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">that are defined as =
&quot;addr-spec / name-addr ...&quot;, similarly to </FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">Refer-To.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">For the RFC 3261 headers - if we =
are understanding correctly - these</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">words exist in order to force =
the protocol to be clear about the</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">structural difference =
between</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; Contact: =
&lt;sip:user@domain.com;transport=3DUDP&gt;</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">and</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; Contact: =
&lt;sip:user@domain.com&gt;;expires=3D30</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">It seems that it would be a good =
idea to impose the same requirement</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">for the Refer-To header.&nbsp; =
Our guess, therefore, is that RFC 3515</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">should have the same words for =
the Refer-To header as RFC 3261 has for</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">the To/From/Contact =
headers.&nbsp; Are we right?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Regards,</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">Paul D Smith</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">Network Protocols Group </FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">Data Connection Ltd (DCL)</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">Tel: +44 20 8366 1177&nbsp; =
Email: paul.d.smith@dataconnection.com </FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">Fax: +44 20 8363 1039&nbsp; =
Web:&nbsp;&nbsp; <A =
HREF=3D"http://www.dataconnection.com">http://www.dataconnection.com</A><=
/FONT>
</P>

</BODY>
</HTML>
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Subject: Re: [Sip] Refer-To header/RFC3261 ambiguity?
From: Scott Lawrence <slawrence@pingtel.com>
To: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
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On Tue, 2005-02-08 at 09:37 +0000, Paul D.Smith wrote:
> All,
> 
> We've recently hit a problem with REFER interoperability, on receipt 
> of a REFER request with a Refer-To header like this:
> 
>    Refer-To: sip:user@domain.com;user=phone?Replaces=something
> 
> The issue is whether the ";user=phone" should be parsed as a URI 
> parameter or a header parameter.  Obviously the intent here is that
> it 
> is a URI parameter, but allowing this runs counter to existing SIP 
> precedent.
> 
> The problem appears to be that the specification of the Refer-To 
> header (in RFC 3515 section 2.1) is missing some key words that other 
> headers of similar format (namely the To, From, Contact and Reply-To 
> headers) have in RFC 3261.

> It seems that it would be a good idea to impose the same requirement 
> for the Refer-To header.  Our guess, therefore, is that RFC 3515 
> should have the same words for the Refer-To header as RFC 3261 has
> for 
> the To/From/Contact headers.  Are we right?

I think it would be an even better fix to change the 3261 text to
clarify that the rule applies to any use of addr-spec or name-addr.



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Thread-Topic: wording in the dialog package on mid-dialog requests
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: "Rohan Mahy" <rohan@ekabal.com>, "SIP WG" <sip@ietf.org>
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Subject: [Sip] RE: wording in the dialog package on mid-dialog requests
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Rohan,

I think the text is better and now ensures that a NOTIFY response
does not take down the dialog.

However I am still concerned about the whole idea that sending
a BYE ends a dialog.  I believe this is tantamount to making a
BYE pointless.  The reason for saying this is we are removing
ability to either challenge a BYE, or to indicate that a BYE
failed to reach the UAS as a result of a temporary problem.

If I receive either a 407, 401 or 486 response, I MUST be able
to resend the BYE but by your definition, the dialog has
already ended.  This seems simply wrong and, as I've pointed
out before, totally contrary to the text in RFC3261.  Now if
this is the route that the WG wants to take then so be it, but
we then need to start making associated changes to
SIP.

If we are going to say "when you send a BYE, the dialog is ended"
then I propose that we need to clarify SIP behavior as follows.

1. A BYE cannot be challenged.  Umm, can't see people liking that
   but since the dialog ends on sending the first BYE, there is=20
   no dialog so its over!  Perhaps we could indicate that a
   UAS can, should it wish, reject an unauthenticated BYE but it
   should NOT expect to receive a better BYE.

[There is obviously a DoS attack possible if we allow any
 unauthenticated BYE to end a dialog]

2. A UAS must determine for itself when a dialog has ended in case
   a proxy sends a 486 and the UAC never resends a BYE.  Now this
   is clearly an application level requirement already in case,=20
   for example, a lost network connection rules out all chance
   of a BYE ever arriving.  But your change makes is a much more
   common occurrence ?? caused by a short-term overload.

Thoughts?
Paul D Smith
Network Protocols Group=20
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com=20
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com

-----Original Message-----
From: Rohan Mahy [mailto:rohan@ekabal.com]
Sent: 28 January 2005 20:32
To: SIP
Subject: wording in the dialog package on mid-dialog requests


Hi Paul,

To clarify the difference between a 481 to a NOTIFY (intended for a=20
subscribe usage), and a 481 to a request specific to the INVITE usage=20
of a dialog, I propose the following small change to the text in the=20
dialog package.  I believe this addresses your concern with respect to=20
the dialog package specification itself. In general, we need to be more=20
specific about the context of error responses.

thanks,
-rohan

Current Text
> Once in the confirmed state, the call is active. It can transition to=20
> the terminated state if the UAC sends a BYE or receives a BYE=20
> (corresponding to the "local-bye" and "remote-bye" events as=20
> appropriate), if a mid-dialog request generates a 481 or 408 response=20
> (corresponding to the "error" event), or a mid-dialog request=20
> generates no response (corresponding to the "timeout" event).

Proposed Text
> Once in the confirmed state, the call is active. It can transition to=20
> the terminated state if the UAC sends a BYE or receives a BYE=20
> (corresponding to the "local-bye" and "remote-bye" events as=20
> appropriate), if a mid-dialog request corresponding to the INVITE=20
> usage of that dialog (currently PRACK, UPDATE, INFO, and reINVITE=20
> requests) generates a 481 or 408 response (corresponding to the=20
> "error" event), or if any mid-dialog request generates no response=20
> (corresponding to the "timeout" event).




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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: wording in the dialog package on mid-dialog requests
Date: Tue, 8 Feb 2005 13:10:49 -0500
To: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
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On Feb 8, 2005, at 12:31 PM, Paul D.Smith wrote:

> Rohan,
>
> I think the text is better and now ensures that a NOTIFY response
> does not take down the dialog.
>
> However I am still concerned about the whole idea that sending
> a BYE ends a dialog.  I believe this is tantamount to making a
> BYE pointless.
No more pointless than a TCP FIN. It's a polite way of saying "I'm 
throwing my dialog state away so  it probably makes sense for you to 
trash yours too". Perhaps you think it's pointless because you thought 
a BYE was somehow had more semantics?

> The reason for saying this is we are removing
> ability to either challenge a BYE, or to indicate that a BYE
> failed to reach the UAS as a result of a temporary problem.
>
Where does the text say I can't challenge a BYE?  I certainly don't 
want to believe a BYE unless it is properly authenticated because that 
opens a DoS attack. I don't follow your second sentence, since I can't 
respond to a request I never received, so there's not way to indicate 
this from the UAS to the UAC.

> If I receive either a 407, 401 or 486 response, I MUST be able
> to resend the BYE
why?

> but by your definition, the dialog has
> already ended.
and by mine. You can't cause dialog state that has evaporated (which is 
what I think 401, 407 and 486 each imply) to somehow get reconstituted 
such that a BYE will be processed as if the dialog existed. We could 
change the semantics of BYE such that it is idempotent, but I bet 
people would react to that the way they react to similar suggestions 
I've made over the years (I remember the flameage I got for a directory 
system where if you deleted a non-existent directory entry, the system 
just said "OK, deleted". In fact, if you tried to delete an entry you 
didn't have read privileges on, it also said "OK deleted". I'll leave 
it to you to figure out why this has nice security properties.).

>   This seems simply wrong and, as I've pointed
> out before, totally contrary to the text in RFC3261.  Now if
> this is the route that the WG wants to take then so be it, but
> we then need to start making associated changes to
> SIP.
>
> If we are going to say "when you send a BYE, the dialog is ended"
> then I propose that we need to clarify SIP behavior as follows.
>
Since a dialog is a distributed state machine, it is not possible to 
say this, and I don't think that it does say this anywhere.

> 1. A BYE cannot be challenged.  Umm, can't see people liking that
>    but since the dialog ends on sending the first BYE, there is
>    no dialog so its over!  Perhaps we could indicate that a
>    UAS can, should it wish, reject an unauthenticated BYE but it
>    should NOT expect to receive a better BYE.
>
This would be wrong and I don't think anybody is suggesting it.
> [There is obviously a DoS attack possible if we allow any
>  unauthenticated BYE to end a dialog]
>
> 2. A UAS must determine for itself when a dialog has ended in case
>    a proxy sends a 486 and the UAC never resends a BYE.  Now this
>    is clearly an application level requirement already in case,
>    for example, a lost network connection rules out all chance
>    of a BYE ever arriving.  But your change makes is a much more
>    common occurrence ?? caused by a short-term overload.
>
I don't follow this. If you get a 486 from a proxy you can be pretty 
sure no other request is going to get through for that dialog, so 
what's the point of keeping your state around or sending more requests?

> Thoughts?
> Paul D Smith
> Network Protocols Group
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
>
> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]
> Sent: 28 January 2005 20:32
> To: SIP
> Subject: wording in the dialog package on mid-dialog requests
>
>
> Hi Paul,
>
> To clarify the difference between a 481 to a NOTIFY (intended for a
> subscribe usage), and a 481 to a request specific to the INVITE usage
> of a dialog, I propose the following small change to the text in the
> dialog package.  I believe this addresses your concern with respect to
> the dialog package specification itself. In general, we need to be more
> specific about the context of error responses.
>
> thanks,
> -rohan
>
> Current Text
>> Once in the confirmed state, the call is active. It can transition to
>> the terminated state if the UAC sends a BYE or receives a BYE
>> (corresponding to the "local-bye" and "remote-bye" events as
>> appropriate), if a mid-dialog request generates a 481 or 408 response
>> (corresponding to the "error" event), or a mid-dialog request
>> generates no response (corresponding to the "timeout" event).
>
> Proposed Text
>> Once in the confirmed state, the call is active. It can transition to
>> the terminated state if the UAC sends a BYE or receives a BYE
>> (corresponding to the "local-bye" and "remote-bye" events as
>> appropriate), if a mid-dialog request corresponding to the INVITE
>> usage of that dialog (currently PRACK, UPDATE, INFO, and reINVITE
>> requests) generates a 481 or 408 response (corresponding to the
>> "error" event), or if any mid-dialog request generates no response
>> (corresponding to the "timeout" event).
>
>
>
>
> _______________________________________________
> 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
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@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


From sip-bounces@ietf.org  Tue Feb  8 13:46:01 2005
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To: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: wording in the dialog package on mid-dialog requests
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I would be much happier if every reference to "dialog" was replaced with 
"invite-dialog-usage" or something like that. The whole concept that a 
BYE causes a *dialog* to go away, when there could still be some other 
usage of that dialog in progress, is broken.

Othewise I agree with Dave.

	Paul

David R Oran wrote:
> 
> On Feb 8, 2005, at 12:31 PM, Paul D.Smith wrote:
> 
>> Rohan,
>>
>> I think the text is better and now ensures that a NOTIFY response
>> does not take down the dialog.
>>
>> However I am still concerned about the whole idea that sending
>> a BYE ends a dialog.  I believe this is tantamount to making a
>> BYE pointless.
> 
> No more pointless than a TCP FIN. It's a polite way of saying "I'm 
> throwing my dialog state away so  it probably makes sense for you to 
> trash yours too". Perhaps you think it's pointless because you thought a 
> BYE was somehow had more semantics?
> 
>> The reason for saying this is we are removing
>> ability to either challenge a BYE, or to indicate that a BYE
>> failed to reach the UAS as a result of a temporary problem.
>>
> Where does the text say I can't challenge a BYE?  I certainly don't want 
> to believe a BYE unless it is properly authenticated because that opens 
> a DoS attack. I don't follow your second sentence, since I can't respond 
> to a request I never received, so there's not way to indicate this from 
> the UAS to the UAC.
> 
>> If I receive either a 407, 401 or 486 response, I MUST be able
>> to resend the BYE
> 
> why?
> 
>> but by your definition, the dialog has
>> already ended.
> 
> and by mine. You can't cause dialog state that has evaporated (which is 
> what I think 401, 407 and 486 each imply) to somehow get reconstituted 
> such that a BYE will be processed as if the dialog existed. We could 
> change the semantics of BYE such that it is idempotent, but I bet people 
> would react to that the way they react to similar suggestions I've made 
> over the years (I remember the flameage I got for a directory system 
> where if you deleted a non-existent directory entry, the system just 
> said "OK, deleted". In fact, if you tried to delete an entry you didn't 
> have read privileges on, it also said "OK deleted". I'll leave it to you 
> to figure out why this has nice security properties.).
> 
>>   This seems simply wrong and, as I've pointed
>> out before, totally contrary to the text in RFC3261.  Now if
>> this is the route that the WG wants to take then so be it, but
>> we then need to start making associated changes to
>> SIP.
>>
>> If we are going to say "when you send a BYE, the dialog is ended"
>> then I propose that we need to clarify SIP behavior as follows.
>>
> Since a dialog is a distributed state machine, it is not possible to say 
> this, and I don't think that it does say this anywhere.
> 
>> 1. A BYE cannot be challenged.  Umm, can't see people liking that
>>    but since the dialog ends on sending the first BYE, there is
>>    no dialog so its over!  Perhaps we could indicate that a
>>    UAS can, should it wish, reject an unauthenticated BYE but it
>>    should NOT expect to receive a better BYE.
>>
> This would be wrong and I don't think anybody is suggesting it.
> 
>> [There is obviously a DoS attack possible if we allow any
>>  unauthenticated BYE to end a dialog]
>>
>> 2. A UAS must determine for itself when a dialog has ended in case
>>    a proxy sends a 486 and the UAC never resends a BYE.  Now this
>>    is clearly an application level requirement already in case,
>>    for example, a lost network connection rules out all chance
>>    of a BYE ever arriving.  But your change makes is a much more
>>    common occurrence ?? caused by a short-term overload.
>>
> I don't follow this. If you get a 486 from a proxy you can be pretty 
> sure no other request is going to get through for that dialog, so what's 
> the point of keeping your state around or sending more requests?
> 
>> Thoughts?
>> Paul D Smith
>> Network Protocols Group
>> Data Connection Ltd (DCL)
>> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com
>> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
>>
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan@ekabal.com]
>> Sent: 28 January 2005 20:32
>> To: SIP
>> Subject: wording in the dialog package on mid-dialog requests
>>
>>
>> Hi Paul,
>>
>> To clarify the difference between a 481 to a NOTIFY (intended for a
>> subscribe usage), and a 481 to a request specific to the INVITE usage
>> of a dialog, I propose the following small change to the text in the
>> dialog package.  I believe this addresses your concern with respect to
>> the dialog package specification itself. In general, we need to be more
>> specific about the context of error responses.
>>
>> thanks,
>> -rohan
>>
>> Current Text
>>
>>> Once in the confirmed state, the call is active. It can transition to
>>> the terminated state if the UAC sends a BYE or receives a BYE
>>> (corresponding to the "local-bye" and "remote-bye" events as
>>> appropriate), if a mid-dialog request generates a 481 or 408 response
>>> (corresponding to the "error" event), or a mid-dialog request
>>> generates no response (corresponding to the "timeout" event).
>>
>>
>> Proposed Text
>>
>>> Once in the confirmed state, the call is active. It can transition to
>>> the terminated state if the UAC sends a BYE or receives a BYE
>>> (corresponding to the "local-bye" and "remote-bye" events as
>>> appropriate), if a mid-dialog request corresponding to the INVITE
>>> usage of that dialog (currently PRACK, UPDATE, INFO, and reINVITE
>>> requests) generates a 481 or 408 response (corresponding to the
>>> "error" event), or if any mid-dialog request generates no response
>>> (corresponding to the "timeout" event).
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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
>>
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@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
> 

_______________________________________________
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 all,

I am currently confused how DTMF digits can be carried via the
signalling path using SIP. If I am not mistaking, then a SIP INFO
message should be used, but the confusion that I have is how should
the content look like ?

There are some drafts, namely:

 -  /draft-choudhuri-sip-info-digit-00.txt expired October 2000
/ - ///draft-mahy-sipping-signaled-digits-00.txt expired May 2002/

but as you can see, all expired.

Some think RFC2833 should be used withing a SIP message.  But
it seems rather unlogical to use a sip info message to carry digits as
explained in rfc2833 which targets rtp packets ... ?

I have checked the implementation of some MGW manufacturers:
Cisco, Snom and AudioCodes all use draft 
/mahy-sipping-signaled-digits-00.txt/,
but is this the way to go ? Or is there maybe a new draft or standard
that tell us what to use ?

Thanks in advance,

    Van Opstal Jan
    R&D Alcatel Antwerp, Belgium

Note:
1) an example of a SIP INFO message using RFC2833:

INFO sip:192.168.50.7:5060 SIP/2.0
From: <sip:496420033@192.168.50.5;user=phone>;tag=0082-00000002-0021
To: <sip:025552024@192.168.50.5;user=phone>;tag=0082-000000dd-05d1i0
Call-ID: 0082-00000002-00214204b9f1-167e-433ca000@sipua
CSeq: 4 INFO
Max-Forwards: 29
Contact: "sipua" <sip:192.168.50.6:5060>
Content-Type: audio/telephone-event
Content-Length: 4

??,2,100

(the body part is not really correct: should be extracted as
explained in RFC2833)

2) an example of a SIP INFO msg using draft 
/mahy-sipping-signaled-digits-00.txt/:

INFO sip:2143302100@172.17.2.33 SIP/2.0
Via: SIP/2.0/UDP 172.80.2.100:5060
From: <sip:9724401003@172.80.2.100>;tag=43
To: <sip:2143302100@172.17.2.33>;tag=9753.0207
Call-ID: 984072_15401962@172.80.2.100
CSeq: 25634 INFO
Content-Length: 26
Content-Type: application/dtmf-relay

Signal= 7
Duration= 300

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<body bgcolor="#ffffff" text="#000000">
Hi all,<br>
<br>
I am currently confused how DTMF digits can be carried via the <br>
signalling path using SIP. If I am not mistaking, then a SIP INFO <br>
message should be used, but the confusion that I have is how should <br>
the content look like ?<br>
<br>
There are some drafts, namely:<br>
<br>
&nbsp;-&nbsp; <i>draft-choudhuri-sip-info-digit-00.txt expired October 2000<br>
</i>&nbsp;-&nbsp;<i></i><i>draft-mahy-sipping-signaled-digits-00.txt expired May
2002</i><br>
<br>
but as you can see, all expired.<br>
<br>
Some think RFC2833 should be used withing a SIP message.&nbsp;
But <br>
it seems rather unlogical to use a sip info message to carry digits
as <br>
explained in rfc2833 which targets rtp packets ... ?<br>
<br>
I have checked the implementation of some MGW manufacturers:<br>
Cisco, Snom and AudioCodes all use draft <i>mahy-sipping-signaled-digits-00.txt</i>,<br>
but is this the way to go ? Or is there maybe a new draft or standard<br>
that tell us what to use ?<br>
<br>
Thanks in advance,<br>
<br>
&nbsp;&nbsp;&nbsp; Van Opstal Jan<br>
&nbsp;&nbsp;&nbsp; R&amp;D Alcatel Antwerp, Belgium<br>
<br>
Note: <br>
1) an example of a SIP INFO message using RFC2833:<br>
<br>
<tt>INFO sip:192.168.50.7:5060 SIP/2.0<br>
From:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:496420033@192.168.50.5;user=phone">&lt;sip:496420033@192.168.50.5;user=phone&gt;</a>;tag=0082-00000002-0021<br>
To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:025552024@192.168.50.5;user=phone">&lt;sip:025552024@192.168.50.5;user=phone&gt;</a>;tag=0082-000000dd-05d1i0<br>
Call-ID: 0082-00000002-00214204b9f1-167e-433ca000@sipua<br>
CSeq: 4 INFO<br>
Max-Forwards: 29<br>
Contact: "sipua" &lt;sip:192.168.50.6:5060&gt;<br>
Content-Type: audio/telephone-event<br>
Content-Length: 4<br>
<br>
&#143;&#143;,2,100<br>
<br>
(the body part is not really correct: should be extracted as<br>
explained in RFC2833)<br>
</tt><br>
2) an example of a SIP INFO msg using draft <i>mahy-sipping-signaled-digits-00.txt</i>:<br>
<br>
<tt>INFO <a class="moz-txt-link-abbreviated" href="mailto:sip:2143302100@172.17.2.33">sip:2143302100@172.17.2.33</a> SIP/2.0 <br>
Via: SIP/2.0/UDP 172.80.2.100:5060 <br>
From: <a class="moz-txt-link-rfc2396E" href="mailto:sip:9724401003@172.80.2.100">&lt;sip:9724401003@172.80.2.100&gt;</a>;tag=43 <br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:2143302100@172.17.2.33">&lt;sip:2143302100@172.17.2.33&gt;</a>;tag=9753.0207 <br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:984072_15401962@172.80.2.100">984072_15401962@172.80.2.100</a> <br>
CSeq: 25634 INFO <br>
Content-Length: 26 <br>
Content-Type: application/dtmf-relay <br>
<br>
Signal= 7 <br>
Duration= 300 </tt><br>
</body>
</html>

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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
--===============0432726442==--



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Subject: RE: [Sip] SIP INFO & DTMF
Date: Wed, 9 Feb 2005 09:10:38 -0000
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Thread-Topic: [Sip] SIP INFO & DTMF
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From: "Chris Boulton" <cboulton@ubiquity.net>
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Take a look at the application interaction work:-

=20

http://www.ietf.org/internet-drafts/draft-ietf-sipping-app-interaction-fram=
ework-03.txt

=20

in association with the 'Key Pad Markup Language':-

=20

http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt

=20

HTH,

=20

Chris.

=20

=20

=20

-----Original Message-----
From: Jan.Van_Opstal@alcatel.be [mailto:Jan.Van_Opstal@alcatel.be]=20
Sent: 08 February 2005 08:15
To: sip@ietf.org
Subject: [Sip] SIP INFO & DTMF

=20

Hi all,

I am currently confused how DTMF digits can be carried via the=20
signalling path using SIP. If I am not mistaking, then a SIP INFO=20
message should be used, but the confusion that I have is how should=20
the content look like ?

There are some drafts, namely:

 -  draft-choudhuri-sip-info-digit-00.txt expired October 2000
 - draft-mahy-sipping-signaled-digits-00.txt expired May 2002

but as you can see, all expired.

Some think RFC2833 should be used withing a SIP message.  But=20
it seems rather unlogical to use a sip info message to carry digits as=20
explained in rfc2833 which targets rtp packets ... ?

I have checked the implementation of some MGW manufacturers:
Cisco, Snom and AudioCodes all use draft mahy-sipping-signaled-digits-00.tx=
t,
but is this the way to go ? Or is there maybe a new draft or standard
that tell us what to use ?

Thanks in advance,

    Van Opstal Jan
    R&D Alcatel Antwerp, Belgium

Note:=20
1) an example of a SIP INFO message using RFC2833:

INFO sip:192.168.50.7:5060 SIP/2.0
From: <sip:496420033@192.168.50.5;user=3Dphone> <mailto:sip:496420033@192.1=
68.50.5;user=3Dphone> ;tag=3D0082-00000002-0021
To: <sip:025552024@192.168.50.5;user=3Dphone> <mailto:sip:025552024@192.168=
.50.5;user=3Dphone> ;tag=3D0082-000000dd-05d1i0
Call-ID: 0082-00000002-00214204b9f1-167e-433ca000@sipua
CSeq: 4 INFO
Max-Forwards: 29
Contact: "sipua" <sip:192.168.50.6:5060>
Content-Type: audio/telephone-event
Content-Length: 4

=8F=8F,2,100

(the body part is not really correct: should be extracted as
explained in RFC2833)

2) an example of a SIP INFO msg using draft mahy-sipping-signaled-digits-00=
.txt:

INFO sip:2143302100@172.17.2.33 SIP/2.0=20
Via: SIP/2.0/UDP 172.80.2.100:5060=20
From: <sip:9724401003@172.80.2.100> <mailto:sip:9724401003@172.80.2.100> ;t=
ag=3D43=20
To: <sip:2143302100@172.17.2.33> <mailto:sip:2143302100@172.17.2.33> ;tag=
=3D9753.0207=20
Call-ID: 984072_15401962@172.80.2.100=20
CSeq: 25634 INFO=20
Content-Length: 26=20
Content-Type: application/dtmf-relay=20

Signal=3D 7=20
Duration=3D 300=20



This email and any attached files are confidential and copyright protected.=
  If you are not the addressee, any dissemination, distribution or copying =
of this communication is strictly prohibited.  Unless otherwise expressly a=
greed in writing, nothing stated in this communication shall be legally bin=
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Take a look at the application interac=
tion
work:-</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-app-interact=
ion-framework-03.txt">http://www.ietf.org/internet-drafts/draft-ietf-sippin=
g-app-interaction-framework-03.txt</a></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>in association with the &#8216;Key Pad
Markup Language&#8217;:-</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt"=
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt</a></sp=
an></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>HTH,</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Chris.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 color=3Dbl=
ack
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma;color:wind=
owtext'>-----Original
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> Jan.Van_Opstal@alcatel=
.be
[mailto:Jan.Van_Opstal@alcatel.be] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 08 February 2005 08:15=
<br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] SIP INFO &amp=
; DTMF</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3 color=3Dbl=
ack
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;</span></fo=
nt></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3 color=3Dbl=
ack
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Hi all,<br>
<br>
I am currently confused how DTMF digits can be carried via the <br>
signalling path using SIP. If I am not mistaking, then a SIP INFO <br>
message should be used, but the confusion that I have is how should <br>
the content look like ?<br>
<br>
There are some drafts, namely:<br>
<br>
&nbsp;-&nbsp; <i><span style=3D'font-style:italic'>draft-choudhuri-sip-info=
-digit-00.txt
expired October 2000<br>
</span></i>&nbsp;-&nbsp;<i><span style=3D'font-style:italic'>draft-mahy-sip=
ping-signaled-digits-00.txt
expired May 2002</span></i><br>
<br>
but as you can see, all expired.<br>
<br>
Some think RFC2833 should be used withing a SIP message.&nbsp; But <br>
it seems rather unlogical to use a sip info message to carry digits as <br>
explained in rfc2833 which targets rtp packets ... ?<br>
<br>
I have checked the implementation of some MGW manufacturers:<br>
Cisco, Snom and AudioCodes all use draft <i><span style=3D'font-style:itali=
c'>mahy-sipping-signaled-digits-00.txt</span></i>,<br>
but is this the way to go ? Or is there maybe a new draft or standard<br>
that tell us what to use ?<br>
<br>
Thanks in advance,<br>
<br>
&nbsp;&nbsp;&nbsp; Van Opstal Jan<br>
&nbsp;&nbsp;&nbsp; R&amp;D Alcatel Antwerp, Belgium<br>
<br>
Note: <br>
1) an example of a SIP INFO message using RFC2833:<br>
<br>
</span></font><tt><font size=3D2 face=3D"Courier New"><span style=3D'font-s=
ize:10.0pt'>INFO
sip:192.168.50.7:5060 SIP/2.0</span></font></tt><font size=3D2 face=3D"Cour=
ier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'><br>
<tt><font face=3D"Courier New">From: <a
href=3D"mailto:sip:496420033@192.168.50.5;user=3Dphone">&lt;sip:496420033@1=
92.168.50.5;user=3Dphone&gt;</a>;tag=3D0082-00000002-0021</font></tt><br>
<tt><font face=3D"Courier New">To: <a
href=3D"mailto:sip:025552024@192.168.50.5;user=3Dphone">&lt;sip:025552024@1=
92.168.50.5;user=3Dphone&gt;</a>;tag=3D0082-000000dd-05d1i0</font></tt><br>
<tt><font face=3D"Courier New">Call-ID:
0082-00000002-00214204b9f1-167e-433ca000@sipua</font></tt><br>
<tt><font face=3D"Courier New">CSeq: 4 INFO</font></tt><br>
<tt><font face=3D"Courier New">Max-Forwards: 29</font></tt><br>
<tt><font face=3D"Courier New">Contact: &quot;sipua&quot;
&lt;sip:192.168.50.6:5060&gt;</font></tt><br>
<tt><font face=3D"Courier New">Content-Type: audio/telephone-event</font></=
tt><br>
<tt><font face=3D"Courier New">Content-Length: 4</font></tt><br>
<br>
<tt><font face=3D"Courier New">=8F=8F,2,100</font></tt><br>
<br>
<tt><font face=3D"Courier New">(the body part is not really correct: should=
 be
extracted as</font></tt><br>
<tt><font face=3D"Courier New">explained in RFC2833)</font></tt><br>
</span></font><br>
2) an example of a SIP INFO msg using draft <i><span style=3D'font-style:it=
alic'>mahy-sipping-signaled-digits-00.txt</span></i>:<br>
<br>
<tt><font size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>IN=
FO <a
href=3D"mailto:sip:2143302100@172.17.2.33">sip:2143302100@172.17.2.33</a> S=
IP/2.0
</span></font></tt><font size=3D2 face=3D"Courier New"><span style=3D'font-=
size:10.0pt;
font-family:"Courier New"'><br>
<tt><font face=3D"Courier New">Via: SIP/2.0/UDP 172.80.2.100:5060 </font></=
tt><br>
<tt><font face=3D"Courier New">From: <a href=3D"mailto:sip:9724401003@172.8=
0.2.100">&lt;sip:9724401003@172.80.2.100&gt;</a>;tag=3D43
</font></tt><br>
<tt><font face=3D"Courier New">To: <a href=3D"mailto:sip:2143302100@172.17.=
2.33">&lt;sip:2143302100@172.17.2.33&gt;</a>;tag=3D9753.0207
</font></tt><br>
<tt><font face=3D"Courier New">Call-ID: <a
href=3D"mailto:984072_15401962@172.80.2.100">984072_15401962@172.80.2.100</=
a> </font></tt><br>
<tt><font face=3D"Courier New">CSeq: 25634 INFO </font></tt><br>
<tt><font face=3D"Courier New">Content-Length: 26 </font></tt><br>
<tt><font face=3D"Courier New">Content-Type: application/dtmf-relay </font>=
</tt><br>
<br>
<tt><font face=3D"Courier New">Signal=3D 7 </font></tt><br>
<tt><font face=3D"Courier New">Duration=3D 300 </font></tt></span></font></=
p>

</div>

<br><br>
<FONT style=3D"BACKGROUND-COLOR: #ffffff">
<DIV>
<DIV><FONT face=3DVerdana color=3D#808080 size=3D1>This email and any attac=
hed files are confidential and copyright protected.&nbsp; If you are not th=
e addressee, any dissemination, distribution or copying of this communicati=
on is strictly prohibited.&nbsp; Unless otherwise expressly agreed in writi=
ng, nothing stated in this communication shall be legally binding.</FONT></=
DIV></DIV></FONT>
</body>

</html>

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--===============0446480794==--



From sip-bounces@ietf.org  Wed Feb  9 14:24:09 2005
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Refer-To header/RFC3261 ambiguity?
Date: Wed, 9 Feb 2005 13:00:27 -0600
To: Scott Lawrence <slawrence@pingtel.com>
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I thought there were outstanding bugs to clarify this wherever possible.
I didn't find them on a quick scan, so I added 765.

As we are waiting for the opportunity to put those clarifications in
place, help spread the word:

->ANYWHERE<- in the header that name-addr / addr-spec is used,
if you want URI parameters, they have to appear inside <>s, otherwise
they are header parameters.

RjS

On Feb 8, 2005, at 7:10 AM, Scott Lawrence wrote:

> On Tue, 2005-02-08 at 09:37 +0000, Paul D.Smith wrote:
>> All,
>>
>> We've recently hit a problem with REFER interoperability, on receipt
>> of a REFER request with a Refer-To header like this:
>>
>>    Refer-To: sip:user@domain.com;user=phone?Replaces=something
>>
>> The issue is whether the ";user=phone" should be parsed as a URI
>> parameter or a header parameter.  Obviously the intent here is that
>> it
>> is a URI parameter, but allowing this runs counter to existing SIP
>> precedent.
>>
>> The problem appears to be that the specification of the Refer-To
>> header (in RFC 3515 section 2.1) is missing some key words that other
>> headers of similar format (namely the To, From, Contact and Reply-To
>> headers) have in RFC 3261.
>
>> It seems that it would be a good idea to impose the same requirement
>> for the Refer-To header.  Our guess, therefore, is that RFC 3515
>> should have the same words for the Refer-To header as RFC 3261 has
>> for
>> the To/From/Contact headers.  Are we right?
>
> I think it would be an even better fix to change the 3261 text to
> clarify that the rule applies to any use of addr-spec or name-addr.
>
>
>
> _______________________________________________
> 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] Need DTMF generation code
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Hello all

I am in urgent need of source code generating DTMF tones.  I need to
send these tones inband.  Any help is appreciated.

Thanks and Best Regards
       Anurag.


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Subject: [Sip] Accept header accept-param m-parameter ambiguity question
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Is there an ambiguity between "*(SEMI accept-param)" and "*(SEMI
m-parameter)" on page 227 of 3261. Could someone explain how you could
tell the difference between a accept-param and m-parameter in a
Accept: header for a SIP message?

ABNF cut&pasted from 25.1
Accept         =  "Accept" HCOLON [ accept-range *(COMMA accept-range) ]
accept-range   =  media-range *(SEMI accept-param)
media-range    =  ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH m-subtype )
                  ) *( SEMI m-parameter )
accept-param =  ("q" EQUAL qvalue) / generic-param
m-parameter  =  m-attribute EQUAL m-value
m-attribute     =  token
m-value          =  token / quoted-string
generic-param  =  token [ EQUAL gen-value ]
gen-value      =  token / host / quoted-string

I don't see how the following header could parse properly.
Accept: */*;mparametername=mparametervalue;acceptparamname=acceptparamvalue

thanks,

-jasson

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To: sip-implementors@cs.columbia.edu, SIP WG <sip@ietf.org>
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Date: Fri, 11 Feb 2005 16:03:02 -0600
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Subject: [Sip] SIPIT 16 Registration is open
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Registration for SIPIT 16 is open. The event will take place
in Banff, Alberta, Canada, April 4-8. Jasomi Networks is hosting.

See
     http://sipit16.jasomi.com
and
     http://www.sipit.net
for more details and to register.

RjS


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Date: Sat, 12 Feb 2005 17:58:59 -0800
From: Cullen Jennings <fluffy@cisco.com>
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> This message is in MIME format. Since your mail reader does not understand
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Nagendra and I put together a draft on using DTLS with SIP. Until it shows
up in the archives you can find it at

http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.
html

(there is a .txt version too)


The abstract is:

   This draft specifies how to use Datagram Transport Layer Security
   (DTLS) as a transport for SIP.  DTLS is a new protocol for providing
   TLS security over a datagram protocol.

--B_3191075939_9781529
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<TITLE>draft-jennings-sip-dtls</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:12.0px'><BR>
Nagendra and I put together a draft on using DTLS with SIP. Until it shows =
up in the archives you can find it at<BR>
<BR>
<a href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-si=
p-dtls-00.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jenni=
ngs-sip-dtls-00.html</a><BR>
<BR>
(there is a .txt version too)<BR>
<BR>
<BR>
The abstract is:<BR>
<BR>
&nbsp;&nbsp;&nbsp;This draft specifies how to use Datagram Transport Layer =
Security<BR>
&nbsp;&nbsp;&nbsp;(DTLS) as a transport for SIP. &nbsp;DTLS is a new protoc=
ol for providing<BR>
&nbsp;&nbsp;&nbsp;TLS security over a datagram protocol.</SPAN></FONT>
</BODY>
</HTML>


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_______________________________________________
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From sip-bounces@ietf.org  Sun Feb 13 03:51:16 2005
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From: "Kowsalya  Subramanian , NPD - Chennai" <kowsalyas@hcltech.com>
To: Anurag Kabra <anuragk@aftek.com>, sip@ietf.org
Subject: RE: [Sip] Need DTMF generation code
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http://www.sephail.net/projects/dtmfkit/ - has code for generating DTMF
tones inband. Look at the file dtmf_encode.c in the dtmfkit.tar

-----Original Message-----
From: Anurag Kabra [mailto:anuragk@aftek.com] 
Sent: Friday, February 11, 2005 4:38 PM
To: sip@ietf.org
Subject: [Sip] Need DTMF generation code

Hello all

I am in urgent need of source code generating DTMF tones.  I need to
send these tones inband.  Any help is appreciated.

Thanks and Best Regards
       Anurag.


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This list is for NEW development of the core SIP Protocol
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Piers

sorry I didn't get back to your questions earlier (no excuses) ... answers 
below

At 06:10 PM 1/24/2005 +0000, Piers O'Hanlon wrote:
>Hi Henning,
>
>I've taken a look at the 05 draft and noticed most of the changes you 
>acknowledged below (for draft-04) don't seem to have made it into this draft.
>
>In addition I noticed that there is a hanging reference from section 3.4 
>to section to 4.2.2 (which is not present in draft-05).

thanks, this is now in section 4.1 of version -06 (thanks for catching this 
one, I went right over it when revising this ID)


>Also just to clarify - If a UAC wants to use an OPTIONS request to 
>determine if a proxy supports Resource-priority - it would send a 
>"Require: Resource-priority" and might expect a "Supported: 
>Resource-priority" header and Accept-Resource-priority: namespace.value, 
>... (given sufficient authorisation)?

I believe the normal procedure in an OPTIONs Request is to include 
extensions supported in an Accept header, the response should include the 
Supported header including the extensions listed in the Request that the 
Server supports.  An A-R-P header could be included as well in the 
Response, but I believe that should be a matter of local policy to dictate 
the behavior of - instead of the ID including some form of compliance 
statement here.


>Wouldn't a 420 response be obliged include a Unsupported header - or is 
>this assumed from RFC3261 compliance?

I am assuming 3261 compliance with this, should I specify this anyway?


>In the definition of strict mode

all the modes are gone/deleted/put in a grave....

I hope and pray version -06 is clearer and more acceptable; as I certainly 
changed a lot of the text to accommodate most of the comments

>Thanks,
>
>Piers O'Hanlon
>--------------
>Computer Science
>University College London


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented

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From sip-bounces@ietf.org  Mon Feb 14 11:26:14 2005
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Subject: RE: [Sip] RE: wording in the dialog package on mid-dialog requests
Date: Mon, 14 Feb 2005 16:19:51 -0000
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Thread-Topic: [Sip] RE: wording in the dialog package on mid-dialog requests
Thread-Index: AcUOCYw51ZSGJrVUSuCEdat+xh2h9wBih7UA
From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: "David R Oran" <oran@cisco.com>, "SIP WG" <sip@ietf.org>,
        "Paul Kyzivat" <pkyzivat@cisco.com>, "Rohan Mahy" <rohan@ekabal.com>
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David, Paul, Rohan,

Having read David's e-mail, I believe there is one question that, if we =
can
answer it, holds the key to resolving this discussion.  The question is:

  "What happens if someone challenges a BYE?"

Now if we assume a couple of proxies between UAC and UAS, and everyone
wants authentication, I expected the flow to be as follows.

UAC          P1          P2           UAS

--- RQ ---->
<-- 407 ----
--- RQ ----> --- RQ ---->
<-- 407 ---- <-- 407 ----
--- RQ ----> --- RQ ----> --- RQ ---->
<-- 401 ---- <-- 401 ---- <-- 401 ----
--- RQ ----> --- RQ ----> --- RQ ---->
<-- 200 ---- <-- 200 ---- <-- 200 ----

I base this on RFC3665, section 3.3 which shows a request (actually
an INVITE) traversing multiple proxies.

>From your (Rohan, David, Paul K.) comments, I believe that the flow you =
are
expecting is the following.

UAC          P1          P2           UAS

--- RQ ----> [Dialog has ended on UAC]
<-- 407 ----

I don't believe a UAC is permitted to send requests on a dialog that is =
in
terminated state.  Therefore since the dialog ends on sending the first =
BYE,
I believe that the UAC cannot send any further requests, so cannot take =
any
further action when challenged. =20

Can you please confirm whether my understanding is correct or clarify =
what
aspect of this I have misunderstood?  We may then continue, or close, =
this
discussion based on our mutual understanding of this scenario.

Regards,
Paul D Smith
Network Protocols Group=20
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com=20
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com

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Paul D.Smith wrote:
> David, Paul, Rohan,
> 
> Having read David's e-mail, I believe there is one question that, if we can
> answer it, holds the key to resolving this discussion.  The question is:
> 
>   "What happens if someone challenges a BYE?"
> 
> Now if we assume a couple of proxies between UAC and UAS, and everyone
> wants authentication, I expected the flow to be as follows.
> 
> UAC          P1          P2           UAS
> 
> --- RQ ---->
> <-- 407 ----
> --- RQ ----> --- RQ ---->
> <-- 407 ---- <-- 407 ----
> --- RQ ----> --- RQ ----> --- RQ ---->
> <-- 401 ---- <-- 401 ---- <-- 401 ----
> --- RQ ----> --- RQ ----> --- RQ ---->
> <-- 200 ---- <-- 200 ---- <-- 200 ----
> 
> I base this on RFC3665, section 3.3 which shows a request (actually
> an INVITE) traversing multiple proxies.
> 
>>From your (Rohan, David, Paul K.) comments, I believe that the flow you are
> expecting is the following.
> 
> UAC          P1          P2           UAS
> 
> --- RQ ----> [Dialog has ended on UAC]
> <-- 407 ----

I have been making a slightly different point. It is that it is not the 
*dialog* that has ended upon sending a BYE. It is the dialog *usage* 
(for the invite) that has perhaps ended. (Or perhaps not, depending on 
the outcome of this discussion.)

This problem arises because of an ambiguity in 3261 between 'dialog' and 
'invite usage of dialog'. That ambiguity is compounded in the "dialog" 
event package, which is really an 'invite usage of dialog' event package.

> I don't believe a UAC is permitted to send requests on a dialog that is in
> terminated state. 

Its not permitted to send messages on a *dialog* that has terminated. It 
might however be permitted to send messages *about* a usage that has 
terminated, in order to ensure that the other party knows that that 
usage has terminated. Until the other party knows that, it won't know 
that the dialog is gone.

> Therefore since the dialog ends on sending the first BYE,
> I believe that the UAC cannot send any further requests, so cannot take any
> further action when challenged.  

This just seems overly simplistic to me. It seems like the UAC should be 
able to keep trying to get the BYE through as long as there is no 
evidence that the *dialog* is gone.

However, this isn't always going to be enough, so there will alsways be 
some weird cases. For instance, suppose at the time the call was 
established the UAC had to prompt the user for a password, in order to 
handle a digest challenge. And it has been using the resulting 
credentials since then. But then suppose those credentials expire before 
the BYE is sent. When the BYE is challenged, it may be necessary to 
prompt the user for a new password before responding to the challenge. 
If so, and if the user has hung up and walked away, that just isn't 
going to happen. In that case the BYE just isn't going to make it. Tough!

	Paul

> Can you please confirm whether my understanding is correct or clarify what
> aspect of this I have misunderstood?  We may then continue, or close, this
> discussion based on our mutual understanding of this scenario.
> 
> Regards,
> Paul D Smith
> Network Protocols Group 
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
> 

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From sip-bounces@ietf.org  Mon Feb 14 13:55:52 2005
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From: "Shetty, Sathish" <sathish.shetty@siemens.com>
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Date: Mon, 14 Feb 2005 10:48:42 -0800
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Subject: [Sip] Question on IM calls.
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Hi,

   We are currently working on SIP Instant messaging. Our goal is to design
a client to support only IM calls using MESSAGE method. I'm just wondering
how to format the IM only SDP message in the INVITE/200 OK message. In case
of IM calls most of the parameters in the SDP are NOT required ( such as
codec, RTP/RTCP port etc)  but RFC 2327 mandates all the parameters. I saw
MSRP proposal for this using SEND method. Is there any other proposals in
conjunction with MESSAGE method for IM only calls?

Is there any RFC or drafts other than
http://www.ietf.org/internet-drafts/draft-ietf-simple-message-sessions-09.tx
t

Please advice..

Thanks,
Sathish.


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Shetty, Sathish wrote:
> Hi,
> 
>    We are currently working on SIP Instant messaging. Our goal is to design
> a client to support only IM calls using MESSAGE method. I'm just wondering
> how to format the IM only SDP message in the INVITE/200 OK message.

Are you developoing an independent system, or a system intended to 
interwork with some other system that already works this way? (E.g. 
Microsoft.)

> In case
> of IM calls most of the parameters in the SDP are NOT required ( such as
> codec, RTP/RTCP port etc)  but RFC 2327 mandates all the parameters. I saw
> MSRP proposal for this using SEND method. Is there any other proposals in
> conjunction with MESSAGE method for IM only calls?

The problem you are running up against is that MESSAGE was not designed 
to support IM *calls*. There has been an attempt to draw a distinction 
between "page mode" (using MESSAGE) and "session mode" (e.g. using MSRP.)

There is (I'm pretty sure, though I don't remember exactly where) an 
explicit prohibition against using MESSAGE for session oriented 
conversations. So you should not be using INVITE to establish a session 
in which MESSAGE will be the way of exchanging IM "media".

> Is there any RFC or drafts other than
> http://www.ietf.org/internet-drafts/draft-ietf-simple-message-sessions-09.tx
> t

There were a number of drafts exploring various possibilities before 
settling on MSRP. But they have all long since expired.

Other than that, there is what Microsoft is currently doing, which is 
not sanctioned by the IETF. If you want to do what they do, you either 
need to get details from them, or reverse engineer their product.

	Paul

> Please advice..
> 
> Thanks,
> Sathish.
> 
> 
> _______________________________________________
> 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: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] RE: wording in the dialog package on mid-dialog requests
Date: Mon, 14 Feb 2005 09:33:51 -0800
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Hi Paul,

BYE is unique in that when my UA sends a BYE, it is done with the 
usage. If the peer challenges the BYE, my UA will of course try to 
automatically resend the BYE with the appropriate credentials, but this 
does not impact my view of the state of the usage.  I am done with it.  
Nothing you can do will bring the usage back from the dead.  The 
challenge may increase the amount of time my peer thinks the usage is 
alive when I think it is dead, and that is perfectly fine.

thanks,
-rohan



On Feb 14, 2005, at 8:19, Paul D.Smith wrote:

> David, Paul, Rohan,
>
> Having read David's e-mail, I believe there is one question that, if 
> we can
> answer it, holds the key to resolving this discussion.  The question 
> is:
>
>   "What happens if someone challenges a BYE?"
>
> Now if we assume a couple of proxies between UAC and UAS, and everyone
> wants authentication, I expected the flow to be as follows.
>
> UAC          P1          P2           UAS
>
> --- RQ ---->
> <-- 407 ----
> --- RQ ----> --- RQ ---->
> <-- 407 ---- <-- 407 ----
> --- RQ ----> --- RQ ----> --- RQ ---->
> <-- 401 ---- <-- 401 ---- <-- 401 ----
> --- RQ ----> --- RQ ----> --- RQ ---->
> <-- 200 ---- <-- 200 ---- <-- 200 ----
>
> I base this on RFC3665, section 3.3 which shows a request (actually
> an INVITE) traversing multiple proxies.
>
>> From your (Rohan, David, Paul K.) comments, I believe that the flow 
>> you are
> expecting is the following.
>
> UAC          P1          P2           UAS
>
> --- RQ ----> [Dialog has ended on UAC]
> <-- 407 ----
>
> I don't believe a UAC is permitted to send requests on a dialog that 
> is in
> terminated state.  Therefore since the dialog ends on sending the 
> first BYE,
> I believe that the UAC cannot send any further requests, so cannot 
> take any
> further action when challenged.
>
> Can you please confirm whether my understanding is correct or clarify 
> what
> aspect of this I have misunderstood?  We may then continue, or close, 
> this
> discussion based on our mutual understanding of this scenario.
>
> Regards,
> Paul D Smith
> Network Protocols Group
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com


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Rohan Mahy wrote:
> 
> Hi Paul,
> 
> BYE is unique in that when my UA sends a BYE, it is done with the usage. 
> If the peer challenges the BYE, my UA will of course try to 
> automatically resend the BYE with the appropriate credentials, but this 
> does not impact my view of the state of the usage.  I am done with it.  
> Nothing you can do will bring the usage back from the dead.  The 
> challenge may increase the amount of time my peer thinks the usage is 
> alive when I think it is dead, and that is perfectly fine.

I'm perfectly fine with that as long as as:
- we are talking about the *usage*, not the *dialog*
- its ok to keep trying to get that BYE through

> thanks,
> -rohan
> 
> 
> 
> On Feb 14, 2005, at 8:19, Paul D.Smith wrote:
> 
>> David, Paul, Rohan,
>>
>> Having read David's e-mail, I believe there is one question that, if 
>> we can
>> answer it, holds the key to resolving this discussion.  The question is:
>>
>>   "What happens if someone challenges a BYE?"
>>
>> Now if we assume a couple of proxies between UAC and UAS, and everyone
>> wants authentication, I expected the flow to be as follows.
>>
>> UAC          P1          P2           UAS
>>
>> --- RQ ---->
>> <-- 407 ----
>> --- RQ ----> --- RQ ---->
>> <-- 407 ---- <-- 407 ----
>> --- RQ ----> --- RQ ----> --- RQ ---->
>> <-- 401 ---- <-- 401 ---- <-- 401 ----
>> --- RQ ----> --- RQ ----> --- RQ ---->
>> <-- 200 ---- <-- 200 ---- <-- 200 ----
>>
>> I base this on RFC3665, section 3.3 which shows a request (actually
>> an INVITE) traversing multiple proxies.
>>
>>> From your (Rohan, David, Paul K.) comments, I believe that the flow 
>>> you are
>>
>> expecting is the following.
>>
>> UAC          P1          P2           UAS
>>
>> --- RQ ----> [Dialog has ended on UAC]
>> <-- 407 ----
>>
>> I don't believe a UAC is permitted to send requests on a dialog that 
>> is in
>> terminated state.  Therefore since the dialog ends on sending the 
>> first BYE,
>> I believe that the UAC cannot send any further requests, so cannot 
>> take any
>> further action when challenged.
>>
>> Can you please confirm whether my understanding is correct or clarify 
>> what
>> aspect of this I have misunderstood?  We may then continue, or close, 
>> this
>> discussion based on our mutual understanding of this scenario.
>>
>> Regards,
>> Paul D Smith
>> Network Protocols Group
>> Data Connection Ltd (DCL)
>> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com
>> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
> 
> 

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From sip-bounces@ietf.org  Tue Feb 15 19:31:41 2005
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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Question on IM calls.
Date: Tue, 15 Feb 2005 18:25:32 -0600
To: "Shetty, Sathish" <sathish.shetty@siemens.com>
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On Feb 14, 2005, at 12:48 PM, Shetty, Sathish wrote:

> Hi,
>
>    We are currently working on SIP Instant messaging. Our goal is to  
> design
> a client to support only IM calls using MESSAGE method. I'm just  
> wondering
> how to format the IM only SDP message in the INVITE/200 OK message. In  
> case
> of IM calls most of the parameters in the SDP are NOT required ( such  
> as
> codec, RTP/RTCP port etc)  but RFC 2327 mandates all the parameters. I  
> saw
> MSRP proposal for this using SEND method. Is there any other proposals  
> in
> conjunction with MESSAGE method for IM only calls?
>
> Is there any RFC or drafts other than
> http://www.ietf.org/internet-drafts/draft-ietf-simple-message- 
> sessions-09.tx
> t
>
>

Note that this is probably a question best left to the sip-implementors  
list, but it does raise a protocol usage question that might should  
have been on the SIPPING list, so if we need to debate this at any  
length we should move it there (or off-line).

There's probably something wrong at the religion level with what I'm  
about to propose, so I've just donned my asbestos undergarment . . .

It seems reasonable to me to have a null offer-answer -- that is, no  
SDP in the INVITE, none in the 200 OK, and none in the ACK, since  
you're not establishing a media channel in the context of this dialog.

The obvious problem is "How do you know to do this?"

For example, one might send an INVITE from such a client to another  
client that supports voice as well as null-media sessions. This UAS  
might respond with an SDP offer in the 200 OK, making life complicated.

One answer might be that the set of AoRs used for the application is  
restricted, so you just know it isn't going to happen. This might work  
adequately for a lab exercise, a student project, or something like  
that. But eventually, you're going to want to go live on the Internet  
and talk to strangers (and their user agents).

The caller-preferences/callee-caps mechanisms give us a framework for  
capability negotiations. So we could define a media feature tag  
(perhaps "nomedia") which the caller could include in the INVITE  
(perhaps with the require" modifier), indicating the the caller does  
not wish to engage in a media offer-answer.

Another, (perhaps more theologically correct) approach would be just to  
send the MESSAGE request to the target AoR, and forget about  
establishing a surrounding dialog-context using INVITE. After all, for  
most uses of pager-mode messaging, the dialog is just overhead you  
didn't really need anyhow.

--
Dean


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This list is for NEW development of the core SIP Protocol
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To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] draft-jennings-sip-dtls
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Cullen,

What seems missing to me from this is requirements and problem 
statements. What is DTLS doing for us that we don't get from TLS?

Though SIP can run over many different transport protocols, I think 
experience over time has shown that more choices here is not necessarily 
a good thing, as SIP has a fair bit of complexity as a result of dealing 
with the differences between UDP and TCP. As such, I don't think its a 
good idea to just add more transport protocols to SIP's list of 
supported ones unless there is a compelling problem that it is solving.

Thanks,
Jonathan R.

Cullen Jennings wrote:

> 
> Nagendra and I put together a draft on using DTLS with SIP. Until it 
> shows up in the archives you can find it at
> 
> http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.html
> 
> (there is a .txt version too)
> 
> 
> The abstract is:
> 
>    This draft specifies how to use Datagram Transport Layer Security
>    (DTLS) as a transport for SIP.  DTLS is a new protocol for providing
>    TLS security over a datagram protocol.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> 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
Director, Service Provider VoIP Architecture   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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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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Date: Tue, 15 Feb 2005 20:08:20 -0800
Subject: Re: [Sip] draft-jennings-sip-dtls
From: Cullen Jennings <fluffy@cisco.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>
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Oops - I meant to put that.... There is pretty much one key thing. No one
has build an single edge proxy that can terminate 100k to 1M connections to
UAs using TLS. In theory it is possible, but in practice it seems hard. The
argument is that this will be easier with DTLS. The issue is not the time it
takes to do the crypto - session resumption deals with that nicely - it just
the issues of dealing with half a million TCP connections to one box. Of
course no one has done it with DTLS either :-)

I believe the argument we made for SCTP was that adding an extensions for
SCTP won't increase the complexity of things that don't support SCTP.

I agree the UDP/TCP complexity made SIP more complicated and I agree that
sip and sips made things more complicated. I'm not sure I buy that both TLS
and TCP made things more complicated.


On 2/15/05 7:15 PM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:

> Cullen, 
> 
> What seems missing to me from this is requirements and problem
> statements. What is DTLS doing for us that we don't get from TLS?
> 
> Though SIP can run over many different transport protocols, I think
> experience over time has shown that more choices here is not necessarily
> a good thing, as SIP has a fair bit of complexity as a result of dealing
> with the differences between UDP and TCP. As such, I don't think its a
> good idea to just add more transport protocols to SIP's list of
> supported ones unless there is a compelling problem that it is solving.
> 
> Thanks, 
> Jonathan R. 
> 
> Cullen Jennings wrote:
> 
>> > 
>> > Nagendra and I put together a draft on using DTLS with SIP. Until it
>> > shows up in the archives you can find it at
>> > 
>> > 
>> http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.h
>> tml 
>> > 
>> > (there is a .txt version too)
>> > 
>> > 
>> > The abstract is:
>> > 
>> >    This draft specifies how to use Datagram Transport Layer Security
>> >    (DTLS) as a transport for SIP.  DTLS is a new protocol for providing
>> >    TLS security over a datagram protocol.
>> > 
>> > 
>> > ------------------------------------------------------------------------
>> > 
>> > _______________________________________________
>> > 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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<TITLE>Re: [Sip] draft-jennings-sip-dtls</TITLE>
</HEAD>
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<FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:12.0px'><BR>
Oops - I meant to put that.... There is pretty much one key thing. No one h=
as build an single edge proxy that can terminate 100k to 1M connections to U=
As using TLS. In theory it is possible, but in practice it seems hard. The a=
rgument is that this will be easier with DTLS. The issue is not the time it =
takes to do the crypto - session resumption deals with that nicely - it just=
 the issues of dealing with half a million TCP connections to one box. Of co=
urse no one has done it with DTLS either :-)<BR>
<BR>
I believe the argument we made for SCTP was that adding an extensions for S=
CTP won't increase the complexity of things that don't support SCTP. <BR>
<BR>
I agree the UDP/TCP complexity made SIP more complicated and I agree that s=
ip and sips made things more complicated. I'm not sure I buy that both TLS a=
nd TCP made things more complicated. <BR>
<BR>
<BR>
On 2/15/05 7:15 PM, &quot;Jonathan Rosenberg&quot; &lt;jdrosen@cisco.com&gt=
; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYL=
E=3D'font-size:12.0px'>Cullen, <BR>
<BR>
What seems missing to me from this is requirements and problem <BR>
statements. What is DTLS doing for us that we don't get from TLS? <BR>
<BR>
Though SIP can run over many different transport protocols, I think <BR>
experience over time has shown that more choices here is not necessarily <B=
R>
a good thing, as SIP has a fair bit of complexity as a result of dealing <B=
R>
with the differences between UDP and TCP. As such, I don't think its a <BR>
good idea to just add more transport protocols to SIP's list of <BR>
supported ones unless there is a compelling problem that it is solving. <BR=
>
<BR>
Thanks, <BR>
Jonathan R. <BR>
<BR>
Cullen Jennings wrote: <BR>
<BR>
&gt; <BR>
&gt; Nagendra and I put together a draft on using DTLS with SIP. Until it <=
BR>
&gt; shows up in the archives you can find it at <BR>
&gt; <BR>
&gt; <a href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennin=
gs-sip-dtls-00.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-=
jennings-sip-dtls-00.html</a> <BR>
&gt; <BR>
&gt; (there is a .txt version too) <BR>
&gt; <BR>
&gt; <BR>
&gt; The abstract is: <BR>
&gt; <BR>
&gt; &nbsp;&nbsp;&nbsp;This draft specifies how to use Datagram Transport L=
ayer Security <BR>
&gt; &nbsp;&nbsp;&nbsp;(DTLS) as a transport for SIP. &nbsp;DTLS is a new p=
rotocol for providing <BR>
&gt; &nbsp;&nbsp;&nbsp;TLS security over a datagram protocol. <BR>
&gt; <BR>
&gt; <BR>
&gt; ----------------------------------------------------------------------=
-- <BR>
&gt; <BR>
&gt; _______________________________________________ <BR>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip <BR>
&gt; This list is for NEW development of the core SIP Protocol <BR>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip <BR>
&gt; Use sipping@ietf.org for new developments on the application of sip <B=
R>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:12.0px'><BR>
</SPAN></FONT>
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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
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Date: Wed, 16 Feb 2005 02:03:37 -0500
From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [Sip] Update to draft-ietf-sip-guidelines
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Folks,

I've just submitted an update to draft-ietf-sip-guidelines based on 
comments received from IESG during their review.

Until it appears in the archives, you can pick up a copy at:
http://www.jdrosen.net/papers/draft-ietf-sip-guidelines-09.txt


The following are the main changes to the document:

* extensions MUST NOT only function if SIP and media paths are
coupled;  P-headers are needed in such cases.

* defining compact forms is a MAY if the header is expected to be
heavily used, and only if they are over 6
characters (previously, it was four). A definition is given for
"heavily used". Sigcomp is mentioned as a reason why this is not so
important.

* previously, the draft said that header field names had to be
composed of ascii characters and marks. This is actually wrong. Its
been changed to: Header field names are defined by the
"token" production in RFC 3261 Section 25.1, and thus include the
upper and lowercase letters, the digits 0 through 9, the HYPHEN-MINUS
(-), FULL STOP (.), EXCLAMATION MARK (!), PERCENT SIGN (%), ASTERISK
(*), LOW LINE (_), PLUS SIGN (+), GRAVE ACCENT (`), APOSTROPHE ('),
and TILDE (~)

* clarified the definition of "large bodies" - several times the
ethernet MTU

* added a paragraph discussing interactions between signaling and
media security

* header field extensions that include URI have to be explicit about
which URI schemes can be used

* added registration of URI parameters and header field parameters to
section on IANA considerations for extensions

* the section on writing iana cosniderations metnions whether or not
the extension has to be standards track or not to register a
particular parameter type (header, option tag, event package, etc.)

* RFC 2393 reference updated to RFC 3173

* informational reference added to draft-iab-model. The section on
writing overviews also mentions that you should expand acronyms and
write this for non-experts in sip

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   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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This list is for NEW development of the core SIP Protocol
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Subject: [Sip] SIP Interop Test Plan
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Hi All,

Does anyone know or have a SIP Interoperability test plan (a SIP Proxy
internetworked to a PBX environment)?  


JR





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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Wed Feb 16 15:35:48 2005
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To: "'Cullen Jennings'" <fluffy@cisco.com>,
        "'Jonathan Rosenberg'"
	<jdrosen@cisco.com>
Subject: RE: [Sip] draft-jennings-sip-dtls
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I would agree with Jonathan.
 
Yes, some implementations of SIP/TCP have performance issues compared to
SIP/UDP, but not necessarily all of them. 
 
Also, if we need to address the other problems of UDP transport (like
fragmentation, and others), then it is not clear to me that we are saving
much in the first place by using UDP/DTLS instead of TCP/TLS.
 
I'd like to see real data before we add yet another thing we'll have to
implement...

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Cullen
Jennings
Sent: Tuesday, February 15, 2005 20:08
To: Jonathan Rosenberg
Cc: sip@ietf.org
Subject: Re: [Sip] draft-jennings-sip-dtls



Oops - I meant to put that.... There is pretty much one key thing. No one
has build an single edge proxy that can terminate 100k to 1M connections to
UAs using TLS. In theory it is possible, but in practice it seems hard. The
argument is that this will be easier with DTLS. The issue is not the time it
takes to do the crypto - session resumption deals with that nicely - it just
the issues of dealing with half a million TCP connections to one box. Of
course no one has done it with DTLS either :-)

I believe the argument we made for SCTP was that adding an extensions for
SCTP won't increase the complexity of things that don't support SCTP. 

I agree the UDP/TCP complexity made SIP more complicated and I agree that
sip and sips made things more complicated. I'm not sure I buy that both TLS
and TCP made things more complicated. 


On 2/15/05 7:15 PM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:



Cullen, 

What seems missing to me from this is requirements and problem 
statements. What is DTLS doing for us that we don't get from TLS? 

Though SIP can run over many different transport protocols, I think 
experience over time has shown that more choices here is not necessarily 
a good thing, as SIP has a fair bit of complexity as a result of dealing 
with the differences between UDP and TCP. As such, I don't think its a 
good idea to just add more transport protocols to SIP's list of 
supported ones unless there is a compelling problem that it is solving. 

Thanks, 
Jonathan R. 

Cullen Jennings wrote: 

> 
> Nagendra and I put together a draft on using DTLS with SIP. Until it 
> shows up in the archives you can find it at 
> 
>
http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.
html
<http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00
.html>  
> 
> (there is a .txt version too) 
> 
> 
> The abstract is: 
> 
>    This draft specifies how to use Datagram Transport Layer Security 
>    (DTLS) as a transport for SIP.  DTLS is a new protocol for providing 
>    TLS security over a datagram protocol. 
> 
> 
> ------------------------------------------------------------------------ 
> 
> _______________________________________________ 
> 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><SPAN class=705474319-16022005><FONT face=Arial color=#800000 size=2>I 
would agree with Jonathan.</FONT></SPAN></DIV>
<DIV><SPAN class=705474319-16022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=705474319-16022005><FONT face=Arial color=#800000 size=2>Yes, 
some implementations of SIP/TCP have performance issues compared to SIP/UDP, but 
not necessarily all of them. </FONT></SPAN></DIV>
<DIV><SPAN class=705474319-16022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=705474319-16022005><FONT face=Arial color=#800000 size=2>Also, 
if we need to address the other problems of UDP transport (like fragmentation, 
and others), then it is not clear to me that we are saving much in the first 
place by using UDP/DTLS instead of TCP/TLS.</FONT></SPAN></DIV>
<DIV><SPAN class=705474319-16022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=705474319-16022005><FONT face=Arial color=#800000 size=2>I'd 
like to see real data before we add yet another thing we'll have to 
implement...</FONT></SPAN></DIV>
<BLOCKQUOTE 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT 
  face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> 
  sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Cullen 
  Jennings<BR><B>Sent:</B> Tuesday, February 15, 2005 20:08<BR><B>To:</B> 
  Jonathan Rosenberg<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> Re: [Sip] 
  draft-jennings-sip-dtls<BR><BR></FONT></DIV><FONT 
  face="Verdana, Helvetica, Arial"><SPAN style="FONT-SIZE: 12px"><BR>Oops - I 
  meant to put that.... There is pretty much one key thing. No one has build an 
  single edge proxy that can terminate 100k to 1M connections to UAs using TLS. 
  In theory it is possible, but in practice it seems hard. The argument is that 
  this will be easier with DTLS. The issue is not the time it takes to do the 
  crypto - session resumption deals with that nicely - it just the issues of 
  dealing with half a million TCP connections to one box. Of course no one has 
  done it with DTLS either :-)<BR><BR>I believe the argument we made for SCTP 
  was that adding an extensions for SCTP won't increase the complexity of things 
  that don't support SCTP. <BR><BR>I agree the UDP/TCP complexity made SIP more 
  complicated and I agree that sip and sips made things more complicated. I'm 
  not sure I buy that both TLS and TCP made things more complicated. 
  <BR><BR><BR>On 2/15/05 7:15 PM, "Jonathan Rosenberg" &lt;jdrosen@cisco.com&gt; 
  wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face="Verdana, Helvetica, Arial"><SPAN 
    style="FONT-SIZE: 12px">Cullen, <BR><BR>What seems missing to me from this 
    is requirements and problem <BR>statements. What is DTLS doing for us that 
    we don't get from TLS? <BR><BR>Though SIP can run over many different 
    transport protocols, I think <BR>experience over time has shown that more 
    choices here is not necessarily <BR>a good thing, as SIP has a fair bit of 
    complexity as a result of dealing <BR>with the differences between UDP and 
    TCP. As such, I don't think its a <BR>good idea to just add more transport 
    protocols to SIP's list of <BR>supported ones unless there is a compelling 
    problem that it is solving. <BR><BR>Thanks, <BR>Jonathan R. <BR><BR>Cullen 
    Jennings wrote: <BR><BR>&gt; <BR>&gt; Nagendra and I put together a draft on 
    using DTLS with SIP. Until it <BR>&gt; shows up in the archives you can find 
    it at <BR>&gt; <BR>&gt; <A 
    href="http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.html</A> 
    <BR>&gt; <BR>&gt; (there is a .txt version too) <BR>&gt; <BR>&gt; <BR>&gt; 
    The abstract is: <BR>&gt; <BR>&gt; &nbsp;&nbsp;&nbsp;This draft specifies 
    how to use Datagram Transport Layer Security <BR>&gt; 
    &nbsp;&nbsp;&nbsp;(DTLS) as a transport for SIP. &nbsp;DTLS is a new 
    protocol for providing <BR>&gt; &nbsp;&nbsp;&nbsp;TLS security over a 
    datagram protocol. <BR>&gt; <BR>&gt; <BR>&gt; 
    ------------------------------------------------------------------------ 
    <BR>&gt; <BR>&gt; _______________________________________________ <BR>&gt; 
    Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip <BR>&gt; 
    This list is for NEW development of the core SIP Protocol <BR>&gt; Use 
    sip-implementors@cs.columbia.edu for questions on current sip <BR>&gt; Use 
    sipping@ietf.org for new developments on the application of sip 
    <BR></SPAN></FONT></BLOCKQUOTE><FONT face="Verdana, Helvetica, Arial"><SPAN 
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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
--===============0489298902==--



From sip-bounces@ietf.org  Wed Feb 16 17:16:37 2005
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Subject: RE: [Sip] draft-jennings-sip-dtls
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Thread-Topic: [Sip] draft-jennings-sip-dtls
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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Francois Audet" <audet@nortel.com>, "Cullen Jennings" <fluffy@cisco.com>,
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Today I would recommend SIP operators to avoid TCP (and that includes
TLS) because the processes are simply running out of sockets (think
about an ITSP with one million customers). The UDP/DTLS idea would make
it reasonable simple to have much more connections and on top have
security. And as far as I understood the dtls it finally does solve the
fragmentation pain problem.
=20
That sounds like a great thing to me. I would vote for dtls!
=20
CS


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Francois Audet
	Sent: Wednesday, February 16, 2005 9:47 PM
	To: 'Cullen Jennings'; 'Jonathan Rosenberg'
	Cc: sip@ietf.org
	Subject: RE: [Sip] draft-jennings-sip-dtls
=09
=09
	I would agree with Jonathan.
	=20
	Yes, some implementations of SIP/TCP have performance issues
compared to SIP/UDP, but not necessarily all of them.=20
	=20
	Also, if we need to address the other problems of UDP transport
(like fragmentation, and others), then it is not clear to me that we are
saving much in the first place by using UDP/DTLS instead of TCP/TLS.
	=20
	I'd like to see real data before we add yet another thing we'll
have to implement...

		-----Original Message-----
		From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]
On Behalf Of Cullen Jennings
		Sent: Tuesday, February 15, 2005 20:08
		To: Jonathan Rosenberg
		Cc: sip@ietf.org
		Subject: Re: [Sip] draft-jennings-sip-dtls
	=09
	=09
	=09
		Oops - I meant to put that.... There is pretty much one
key thing. No one has build an single edge proxy that can terminate 100k
to 1M connections to UAs using TLS. In theory it is possible, but in
practice it seems hard. The argument is that this will be easier with
DTLS. The issue is not the time it takes to do the crypto - session
resumption deals with that nicely - it just the issues of dealing with
half a million TCP connections to one box. Of course no one has done it
with DTLS either :-)
	=09
		I believe the argument we made for SCTP was that adding
an extensions for SCTP won't increase the complexity of things that
don't support SCTP.=20
	=09
		I agree the UDP/TCP complexity made SIP more complicated
and I agree that sip and sips made things more complicated. I'm not sure
I buy that both TLS and TCP made things more complicated.=20
	=09
	=09
		On 2/15/05 7:15 PM, "Jonathan Rosenberg"
<jdrosen@cisco.com> wrote:
	=09
	=09

			Cullen,=20
		=09
			What seems missing to me from this is
requirements and problem=20
			statements. What is DTLS doing for us that we
don't get from TLS?=20
		=09
			Though SIP can run over many different transport
protocols, I think=20
			experience over time has shown that more choices
here is not necessarily=20
			a good thing, as SIP has a fair bit of
complexity as a result of dealing=20
			with the differences between UDP and TCP. As
such, I don't think its a=20
			good idea to just add more transport protocols
to SIP's list of=20
			supported ones unless there is a compelling
problem that it is solving.=20
		=09
			Thanks,=20
			Jonathan R.=20
		=09
			Cullen Jennings wrote:=20
		=09
			>=20
			> Nagendra and I put together a draft on using
DTLS with SIP. Until it=20
			> shows up in the archives you can find it at=20
			>=20
			>
http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls
-00.html=20
			>=20
			> (there is a .txt version too)=20
			>=20
			>=20
			> The abstract is:=20
			>=20
			>    This draft specifies how to use Datagram
Transport Layer Security=20
			>    (DTLS) as a transport for SIP.  DTLS is a
new protocol for providing=20
			>    TLS security over a datagram protocol.=20
			>=20
			>=20
			>
------------------------------------------------------------------------

			>=20
			>
_______________________________________________=20
			> Sip mailing list
https://www1.ietf.org/mailman/listinfo/sip=20
			> This list is for NEW development of the core
SIP Protocol=20
			> Use sip-implementors@cs.columbia.edu for
questions on current sip=20
			> Use sipping@ietf.org for new developments on
the application of sip=20
		=09

	=09
	=09


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2604" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>Today I would recommend SIP operators to avoid =
TCP (and=20
that includes TLS) because the processes are simply running out of =
sockets=20
(think about an ITSP with one million customers). The UDP/DTLS idea =
would make=20
it reasonable simple to have much more connections and on top have =
security. And=20
as far as I understood the dtls it finally does solve the fragmentation =
pain=20
problem.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>That sounds like a great thing to me. I would =
vote for=20
dtls!</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Francois=20
  Audet<BR><B>Sent:</B> Wednesday, February 16, 2005 9:47 =
PM<BR><B>To:</B>=20
  'Cullen Jennings'; 'Jonathan Rosenberg'<BR><B>Cc:</B>=20
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
  draft-jennings-sip-dtls<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000 size=3D2>I=20
  would agree with Jonathan.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000 size=3D2>Yes,=20
  some implementations of SIP/TCP have performance issues compared to =
SIP/UDP,=20
  but not necessarily all of them. </FONT></SPAN></DIV>
  <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
  size=3D2>Also, if we need to address the other problems of UDP =
transport (like=20
  fragmentation, and others), then it is not clear to me that we are =
saving much=20
  in the first place by using UDP/DTLS instead of =
TCP/TLS.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000 size=3D2>I'd=20
  like to see real data before we add yet another thing we'll have to=20
  implement...</FONT></SPAN></DIV>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV></DIV>
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
    face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
    sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of=20
    </B>Cullen Jennings<BR><B>Sent:</B> Tuesday, February 15, 2005=20
    20:08<BR><B>To:</B> Jonathan Rosenberg<BR><B>Cc:</B>=20
    sip@ietf.org<BR><B>Subject:</B> Re: [Sip]=20
    draft-jennings-sip-dtls<BR><BR></FONT></DIV><FONT=20
    face=3D"Verdana, Helvetica, Arial"><SPAN style=3D"FONT-SIZE: =
12px"><BR>Oops - I=20
    meant to put that.... There is pretty much one key thing. No one has =
build=20
    an single edge proxy that can terminate 100k to 1M connections to =
UAs using=20
    TLS. In theory it is possible, but in practice it seems hard. The =
argument=20
    is that this will be easier with DTLS. The issue is not the time it =
takes to=20
    do the crypto - session resumption deals with that nicely - it just =
the=20
    issues of dealing with half a million TCP connections to one box. Of =
course=20
    no one has done it with DTLS either :-)<BR><BR>I believe the =
argument we=20
    made for SCTP was that adding an extensions for SCTP won't increase =
the=20
    complexity of things that don't support SCTP. <BR><BR>I agree the =
UDP/TCP=20
    complexity made SIP more complicated and I agree that sip and sips =
made=20
    things more complicated. I'm not sure I buy that both TLS and TCP =
made=20
    things more complicated. <BR><BR><BR>On 2/15/05 7:15 PM, "Jonathan=20
    Rosenberg" &lt;jdrosen@cisco.com&gt; wrote:<BR><BR></SPAN></FONT>
    <BLOCKQUOTE><FONT face=3D"Verdana, Helvetica, Arial"><SPAN=20
      style=3D"FONT-SIZE: 12px">Cullen, <BR><BR>What seems missing to me =
from this=20
      is requirements and problem <BR>statements. What is DTLS doing for =
us that=20
      we don't get from TLS? <BR><BR>Though SIP can run over many =
different=20
      transport protocols, I think <BR>experience over time has shown =
that more=20
      choices here is not necessarily <BR>a good thing, as SIP has a =
fair bit of=20
      complexity as a result of dealing <BR>with the differences between =
UDP and=20
      TCP. As such, I don't think its a <BR>good idea to just add more =
transport=20
      protocols to SIP's list of <BR>supported ones unless there is a =
compelling=20
      problem that it is solving. <BR><BR>Thanks, <BR>Jonathan R. =
<BR><BR>Cullen=20
      Jennings wrote: <BR><BR>&gt; <BR>&gt; Nagendra and I put together =
a draft=20
      on using DTLS with SIP. Until it <BR>&gt; shows up in the archives =
you can=20
      find it at <BR>&gt; <BR>&gt; <A=20
      =
href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-s=
ip-dtls-00.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-j=
ennings-sip-dtls-00.html</A>=20
      <BR>&gt; <BR>&gt; (there is a .txt version too) <BR>&gt; <BR>&gt; =
<BR>&gt;=20
      The abstract is: <BR>&gt; <BR>&gt; &nbsp;&nbsp;&nbsp;This draft =
specifies=20
      how to use Datagram Transport Layer Security <BR>&gt;=20
      &nbsp;&nbsp;&nbsp;(DTLS) as a transport for SIP. &nbsp;DTLS is a =
new=20
      protocol for providing <BR>&gt; &nbsp;&nbsp;&nbsp;TLS security =
over a=20
      datagram protocol. <BR>&gt; <BR>&gt; <BR>&gt;=20
      =
------------------------------------------------------------------------ =

      <BR>&gt; <BR>&gt; _______________________________________________ =
<BR>&gt;=20
      Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip =
<BR>&gt;=20
      This list is for NEW development of the core SIP Protocol <BR>&gt; =
Use=20
      sip-implementors@cs.columbia.edu for questions on current sip =
<BR>&gt; Use=20
      sipping@ietf.org for new developments on the application of sip=20
      <BR></SPAN></FONT></BLOCKQUOTE><FONT face=3D"Verdana, Helvetica, =
Arial"><SPAN=20
    style=3D"FONT-SIZE: =
12px"><BR></BLOCKQUOTE></BLOCKQUOTE></SPAN></FONT></BODY></HTML>

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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
--===============0685188293==--



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Subject: RE: [Sip] RE: wording in the dialog package on mid-dialog requests
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Rohan, Paul K., David,

I believe I now understand what you are intending and I think we can =
close
this thread.  I do have a couple of comments below which will hopefully =
be
helpful in completing this thread.

I think the comments from Rohan and Paul K. (below) have clarified my
confusion.  As Paul K. points out (and I'm not too good about this =
myself!),
we need to be very clear on the use of "dialog" and "dialog-usage".  As =
I
understand it now, draft-ietf-sipping-dialog-package is about =
"dialog-usage"
and _not_ dialogs.  We therefore must clarify this including changing =
the
title of the RFC/draft so that this is explicit.

However this does raise the question of how we define a "dialog-usage". =
The
best document I have found thus far is "Multiple Dialog Usages in the
Session Initiation Protocol", draft-sparks-sipping-dialogusage-00.
Unfortunately this contains the following text.

   A dialog comes into existence with the creation of the first usage,
   and continues to exist until the last usage is terminated (reference
   counting).  Unfortunately, many of the usage management aspects of
   SIP, such as authentication, were originally designed with the
   implicit assumption that there was one usage per dialog.  The
   resulting mechanisms have mixed effects, some influencing the usage,
   and some influencing the entire dialog.

...and...

  401 Unauthorized ,407 Proxy Authentication Required: This request,
      not the subscription or dialog, is being challenged.  The usages
      and dialog are not terminated.

The first paragraph is clearly not consistent with the current proposal
for draft-ietf-sipping-dialog-package since it is allowing a =
dialog-usage
to terminated _but_ the dialog still exists to allow a challenged BYE to
be sent.

Similarly, the second paragraph contravenes the assertion that the
dialog-usage terminates on sending the BYE and has terminated _before_
the 401/407 response is received.

As I'm sure you realise, I'm happy with the definitions given in
draft-sparks-sipping-dialogusage-00 as this was the basis for my
previous comments.

However I appreciate that this is _not_ the current thinking of the WG.
Can I therefore ask that someone takes the task to clarify/change the
definition of a dialog-usage, its lifetime and states?  I believe this
MUST be done in parallel with the current work
on draft-ietf-sipping-dialog-package, otherwise this draft will continue
to be inconsistent with published definitions of dialog-usage.

One suggestion that might be useful is to consider a new state for
dialog-usage known as "terminating".  This is a half-way house between
"active" and "terminated" and has the following properties.

- Any attempt to send a dialog-usage request within that dialog-usage
  fails _except_ for the dialog-usage terminating request [1] (and in
  the same direction as the original dialog-usage terminating request).
- The dialog-usage terminating request may be resent (actually a new
  request in strict SIP terms), for example to take account of
  a challenge being received.
- Once the dialog-usage terminating request has completed (success, 481,
  408, or an application determined retry limit has been reached (for
  example the application may choose to resend a BYE once after a 486
  response), then the dialog-usage enters terminated state.

This would be consistent with existing definitions of the dialog-usage
whilst still allowing the draft-ietf-sipping-dialog-package to indicate
"this dialog-usage" is ending.

Thanks for your patience is discussing this issue.  Hopefully this =
thread
will be a useful reference for the future.

Regards,
Paul D Smith
Network Protocols Group=20
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com=20
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com





Rohan Mahy wrote:
>=20
> Hi Paul,
>=20
> BYE is unique in that when my UA sends a BYE, it is done with the =
usage.=20
> If the peer challenges the BYE, my UA will of course try to=20
> automatically resend the BYE with the appropriate credentials, but =
this=20
> does not impact my view of the state of the usage.  I am done with it. =
=20
> Nothing you can do will bring the usage back from the dead.  The=20
> challenge may increase the amount of time my peer thinks the usage is=20
> alive when I think it is dead, and that is perfectly fine.

I'm perfectly fine with that as long as as:
- we are talking about the *usage*, not the *dialog*
- its ok to keep trying to get that BYE through

> thanks,
> -rohan

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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		: Enhancements for Authenticated Identity Management 
			  in the Session Initiation Protocol (SIP)
	Author(s)	: J. Peterson, C. Jennings
	Filename	: draft-ietf-sip-identity-04.txt
	Pages		: 35
	Date		: 2005-2-16
	
The existing security mechanisms in the Session Initiation Protocol
   are inadequate for cryptographically assuring the identity of the end
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   context.  This document recommends practices and conventions for
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The Session Initiation Protocol (SIP) is a flexible, yet simple tool
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Paul D,

Good summary. I pretty much agree with it. Just a couple of points inline.

	Paul K.

Paul D.Smith wrote:
> Rohan, Paul K., David,
> 
> I believe I now understand what you are intending and I think we can close
> this thread.  I do have a couple of comments below which will hopefully be
> helpful in completing this thread.
> 
> I think the comments from Rohan and Paul K. (below) have clarified my
> confusion.  As Paul K. points out (and I'm not too good about this myself!),
> we need to be very clear on the use of "dialog" and "dialog-usage".  As I
> understand it now, draft-ietf-sipping-dialog-package is about "dialog-usage"
> and _not_ dialogs.  We therefore must clarify this including changing the
> title of the RFC/draft so that this is explicit.

I've been trying to get the *package name* changed for months. (Probably 
years by now.) But I have more or less given up, and would be satisfied 
if there was sufficient text in the draft explaining that what it means 
is somewhat different than what one might conclude from the terms used.

That package isn't about dialogs, and it isn't about dialog-usages in 
general either. It is about 'invite-dialog-usage'.

The same UA may have a bunch of subscriptions, both incoming and 
outgoing, that are dialog usages. But those will never be reflected in 
the content of the 'dialog' package.

> However this does raise the question of how we define a "dialog-usage". The
> best document I have found thus far is "Multiple Dialog Usages in the
> Session Initiation Protocol", draft-sparks-sipping-dialogusage-00.
> Unfortunately this contains the following text.
> 
>    A dialog comes into existence with the creation of the first usage,
>    and continues to exist until the last usage is terminated (reference
>    counting).  Unfortunately, many of the usage management aspects of
>    SIP, such as authentication, were originally designed with the
>    implicit assumption that there was one usage per dialog.  The
>    resulting mechanisms have mixed effects, some influencing the usage,
>    and some influencing the entire dialog.
> 
> ...and...
> 
>   401 Unauthorized ,407 Proxy Authentication Required: This request,
>       not the subscription or dialog, is being challenged.  The usages
>       and dialog are not terminated.
> 
> The first paragraph is clearly not consistent with the current proposal
> for draft-ietf-sipping-dialog-package since it is allowing a dialog-usage
> to terminated _but_ the dialog still exists to allow a challenged BYE to
> be sent.
> 
> Similarly, the second paragraph contravenes the assertion that the
> dialog-usage terminates on sending the BYE and has terminated _before_
> the 401/407 response is received.
> 
> As I'm sure you realise, I'm happy with the definitions given in
> draft-sparks-sipping-dialogusage-00 as this was the basis for my
> previous comments.
> 
> However I appreciate that this is _not_ the current thinking of the WG.
> Can I therefore ask that someone takes the task to clarify/change the
> definition of a dialog-usage, its lifetime and states?  I believe this
> MUST be done in parallel with the current work
> on draft-ietf-sipping-dialog-package, otherwise this draft will continue
> to be inconsistent with published definitions of dialog-usage.
> 
> One suggestion that might be useful is to consider a new state for
> dialog-usage known as "terminating".  This is a half-way house between
> "active" and "terminated" and has the following properties.
> 
> - Any attempt to send a dialog-usage request within that dialog-usage
>   fails _except_ for the dialog-usage terminating request [1] (and in
>   the same direction as the original dialog-usage terminating request).

Probably need to be more lenient here. A BYE could be sent, but be 
challenged. Before a successful retry is completed, a BYE could arrive 
from the other end. I think this should be answered, and terminate the 
dialog usage.

> - The dialog-usage terminating request may be resent (actually a new
>   request in strict SIP terms), for example to take account of
>   a challenge being received.
> - Once the dialog-usage terminating request has completed (success, 481,
>   408, or an application determined retry limit has been reached (for
>   example the application may choose to resend a BYE once after a 486
>   response), then the dialog-usage enters terminated state.
> 
> This would be consistent with existing definitions of the dialog-usage
> whilst still allowing the draft-ietf-sipping-dialog-package to indicate
> "this dialog-usage" is ending.
> 
> Thanks for your patience is discussing this issue.  Hopefully this thread
> will be a useful reference for the future.
> 
> Regards,
> Paul D Smith
> Network Protocols Group 
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
> 
> 
> 
> 
> 
> Rohan Mahy wrote:
> 
>>Hi Paul,
>>
>>BYE is unique in that when my UA sends a BYE, it is done with the usage. 
>>If the peer challenges the BYE, my UA will of course try to 
>>automatically resend the BYE with the appropriate credentials, but this 
>>does not impact my view of the state of the usage.  I am done with it.  
>>Nothing you can do will bring the usage back from the dead.  The 
>>challenge may increase the amount of time my peer thinks the usage is 
>>alive when I think it is dead, and that is perfectly fine.
> 
> 
> I'm perfectly fine with that as long as as:
> - we are talking about the *usage*, not the *dialog*
> - its ok to keep trying to get that BYE through
> 
> 
>>thanks,
>>-rohan
> 
> 

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Registration for SIPIT 16 will close in a couple of weeks.
If you have not already registered, please do so now.

Please note that the special rates at the event hotel are only 
available until March 4th.

See http://www.sipit.net/ for more information.

RjS


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From: "Tolga Asveren" <asveren@ulticom.com>
To: "Christian Stredicke" <Christian.Stredicke@snom.de>,
        "Francois Audet" <audet@nortel.com>,
        "Cullen Jennings" <fluffy@cisco.com>,
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Subject: RE: [Sip] draft-jennings-sip-dtls
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MessageIsn't "running out of sockets" an implementation issue rather than a
protocol one?

And I believe using multiple processes would solve that problem.

  Tolga
  -----Original Message-----
  From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Christian Stredicke
  Sent: Wednesday, February 16, 2005 5:02 PM
  To: Francois Audet; Cullen Jennings; Jonathan Rosenberg
  Cc: sip@ietf.org
  Subject: RE: [Sip] draft-jennings-sip-dtls


  Today I would recommend SIP operators to avoid TCP (and that includes TLS)
because the processes are simply running out of sockets (think about an ITSP
with one million customers). The UDP/DTLS idea would make it reasonable
simple to have much more connections and on top have security. And as far as
I understood the dtls it finally does solve the fragmentation pain problem.

  That sounds like a great thing to me. I would vote for dtls!

  CS



----------------------------------------------------------------------------
    From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Francois Audet
    Sent: Wednesday, February 16, 2005 9:47 PM
    To: 'Cullen Jennings'; 'Jonathan Rosenberg'
    Cc: sip@ietf.org
    Subject: RE: [Sip] draft-jennings-sip-dtls


    I would agree with Jonathan.

    Yes, some implementations of SIP/TCP have performance issues compared to
SIP/UDP, but not necessarily all of them.

    Also, if we need to address the other problems of UDP transport (like
fragmentation, and others), then it is not clear to me that we are saving
much in the first place by using UDP/DTLS instead of TCP/TLS.

    I'd like to see real data before we add yet another thing we'll have to
implement...
      -----Original Message-----
      From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Cullen Jennings
      Sent: Tuesday, February 15, 2005 20:08
      To: Jonathan Rosenberg
      Cc: sip@ietf.org
      Subject: Re: [Sip] draft-jennings-sip-dtls



      Oops - I meant to put that.... There is pretty much one key thing. No
one has build an single edge proxy that can terminate 100k to 1M connections
to UAs using TLS. In theory it is possible, but in practice it seems hard.
The argument is that this will be easier with DTLS. The issue is not the
time it takes to do the crypto - session resumption deals with that nicely -
it just the issues of dealing with half a million TCP connections to one
box. Of course no one has done it with DTLS either :-)

      I believe the argument we made for SCTP was that adding an extensions
for SCTP won't increase the complexity of things that don't support SCTP.

      I agree the UDP/TCP complexity made SIP more complicated and I agree
that sip and sips made things more complicated. I'm not sure I buy that both
TLS and TCP made things more complicated.


      On 2/15/05 7:15 PM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:


        Cullen,

        What seems missing to me from this is requirements and problem
        statements. What is DTLS doing for us that we don't get from TLS?

        Though SIP can run over many different transport protocols, I think
        experience over time has shown that more choices here is not
necessarily
        a good thing, as SIP has a fair bit of complexity as a result of
dealing
        with the differences between UDP and TCP. As such, I don't think its
a
        good idea to just add more transport protocols to SIP's list of
        supported ones unless there is a compelling problem that it is
solving.

        Thanks,
        Jonathan R.

        Cullen Jennings wrote:

        >
        > Nagendra and I put together a draft on using DTLS with SIP. Until
it
        > shows up in the archives you can find it at
        >
        >
http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.
html
        >
        > (there is a .txt version too)
        >
        >
        > The abstract is:
        >
        >    This draft specifies how to use Datagram Transport Layer
Security
        >    (DTLS) as a transport for SIP.  DTLS is a new protocol for
providing
        >    TLS security over a datagram protocol.
        >
        >






  > ------------------------------------------------------------------------
        >
        > _______________________________________________
        > 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><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1479" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D814205418-17022005>Isn't=20
"running out of sockets" an implementation issue rather than a protocol =
one?=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D814205418-17022005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D814205418-17022005>And I=20
believe using multiple processes would solve that =
problem.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D814205418-17022005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D814205418-17022005>&nbsp;&nbsp;Tolga</SPAN></FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> =
sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christian=20
  Stredicke<BR><B>Sent:</B> Wednesday, February 16, 2005 5:02 =
PM<BR><B>To:</B>=20
  Francois Audet; Cullen Jennings; Jonathan Rosenberg<BR><B>Cc:</B>=20
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
  draft-jennings-sip-dtls<BR><BR></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2>Today I would recommend SIP operators to =
avoid TCP (and=20
  that includes TLS) because the processes are simply running out of =
sockets=20
  (think about an ITSP with one million customers). The UDP/DTLS idea =
would make=20
  it reasonable simple to have much more connections and on top have =
security.=20
  And as far as I understood the dtls it finally does solve the =
fragmentation=20
  pain problem.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2>That sounds like a great thing to me. I would =
vote for=20
  dtls!</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Francois=20
    Audet<BR><B>Sent:</B> Wednesday, February 16, 2005 9:47 =
PM<BR><B>To:</B>=20
    'Cullen Jennings'; 'Jonathan Rosenberg'<BR><B>Cc:</B>=20
    sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
    draft-jennings-sip-dtls<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000 size=3D2>I=20
    would agree with Jonathan.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
    size=3D2>Yes, some implementations of SIP/TCP have performance =
issues compared=20
    to SIP/UDP, but not necessarily all of them. </FONT></SPAN></DIV>
    <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
    size=3D2>Also, if we need to address the other problems of UDP =
transport (like=20
    fragmentation, and others), then it is not clear to me that we are =
saving=20
    much in the first place by using UDP/DTLS instead of=20
    TCP/TLS.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
    size=3D2>I'd like to see real data before we add yet another thing =
we'll have=20
    to implement...</FONT></SPAN></DIV>
    <BLOCKQUOTE=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 =
2px solid; MARGIN-RIGHT: 0px">
      <DIV></DIV>
      <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
      face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
      sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of =

      </B>Cullen Jennings<BR><B>Sent:</B> Tuesday, February 15, 2005=20
      20:08<BR><B>To:</B> Jonathan Rosenberg<BR><B>Cc:</B>=20
      sip@ietf.org<BR><B>Subject:</B> Re: [Sip]=20
      draft-jennings-sip-dtls<BR><BR></FONT></DIV><FONT=20
      face=3D"Verdana, Helvetica, Arial"><SPAN style=3D"FONT-SIZE: =
12px"><BR>Oops -=20
      I meant to put that.... There is pretty much one key thing. No one =
has=20
      build an single edge proxy that can terminate 100k to 1M =
connections to=20
      UAs using TLS. In theory it is possible, but in practice it seems =
hard.=20
      The argument is that this will be easier with DTLS. The issue is =
not the=20
      time it takes to do the crypto - session resumption deals with =
that nicely=20
      - it just the issues of dealing with half a million TCP =
connections to one=20
      box. Of course no one has done it with DTLS either :-)<BR><BR>I =
believe=20
      the argument we made for SCTP was that adding an extensions for =
SCTP won't=20
      increase the complexity of things that don't support SCTP. =
<BR><BR>I agree=20
      the UDP/TCP complexity made SIP more complicated and I agree that =
sip and=20
      sips made things more complicated. I'm not sure I buy that both =
TLS and=20
      TCP made things more complicated. <BR><BR><BR>On 2/15/05 7:15 PM,=20
      "Jonathan Rosenberg" &lt;jdrosen@cisco.com&gt;=20
wrote:<BR><BR></SPAN></FONT>
      <BLOCKQUOTE><FONT face=3D"Verdana, Helvetica, Arial"><SPAN=20
        style=3D"FONT-SIZE: 12px">Cullen, <BR><BR>What seems missing to =
me from=20
        this is requirements and problem <BR>statements. What is DTLS =
doing for=20
        us that we don't get from TLS? <BR><BR>Though SIP can run over =
many=20
        different transport protocols, I think <BR>experience over time =
has=20
        shown that more choices here is not necessarily <BR>a good =
thing, as SIP=20
        has a fair bit of complexity as a result of dealing <BR>with the =

        differences between UDP and TCP. As such, I don't think its a =
<BR>good=20
        idea to just add more transport protocols to SIP's list of =
<BR>supported=20
        ones unless there is a compelling problem that it is solving.=20
        <BR><BR>Thanks, <BR>Jonathan R. <BR><BR>Cullen Jennings wrote:=20
        <BR><BR>&gt; <BR>&gt; Nagendra and I put together a draft on =
using DTLS=20
        with SIP. Until it <BR>&gt; shows up in the archives you can =
find it at=20
        <BR>&gt; <BR>&gt; <A=20
        =
href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-s=
ip-dtls-00.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-j=
ennings-sip-dtls-00.html</A>=20
        <BR>&gt; <BR>&gt; (there is a .txt version too) <BR>&gt; =
<BR>&gt;=20
        <BR>&gt; The abstract is: <BR>&gt; <BR>&gt; =
&nbsp;&nbsp;&nbsp;This draft=20
        specifies how to use Datagram Transport Layer Security <BR>&gt;=20
        &nbsp;&nbsp;&nbsp;(DTLS) as a transport for SIP. &nbsp;DTLS is a =
new=20
        protocol for providing <BR>&gt; &nbsp;&nbsp;&nbsp;TLS security =
over a=20
        datagram protocol. <BR>&gt; <BR>&gt; <BR>&gt;=20
        =
------------------------------------------------------------------------ =

        <BR>&gt; <BR>&gt; =
_______________________________________________=20
        <BR>&gt; Sip mailing list=20
        &nbsp;https://www1.ietf.org/mailman/listinfo/sip <BR>&gt; This =
list is=20
        for NEW development of the core SIP Protocol <BR>&gt; Use=20
        sip-implementors@cs.columbia.edu for questions on current sip =
<BR>&gt;=20
        Use sipping@ietf.org for new developments on the application of =
sip=20
        <BR></SPAN></FONT></BLOCKQUOTE><FONT=20
      face=3D"Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: =
12px"><BR></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></SPAN></FONT></BODY></H=
TML>

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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
--===============1868686227==--




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Subject: RE: [Sip] draft-jennings-sip-dtls
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From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Tolga Asveren" <asveren@ulticom.com>, "Francois Audet" <audet@nortel.com>,
        "Cullen Jennings" <fluffy@cisco.com>,
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16 bit set an upper limit of 64 K connections. My feeling is that TCP
was not designed for this purpose.
=20
Sorry for being so pragmatic. Maybe someone can show me an
implementation that can handle 100 K connections on one IP address.
=20
CS


________________________________

	From: Tolga Asveren [mailto:asveren@ulticom.com]=20
	Sent: Thursday, February 17, 2005 8:16 PM
	To: Christian Stredicke; Francois Audet; Cullen Jennings;
Jonathan Rosenberg
	Cc: sip@ietf.org
	Subject: RE: [Sip] draft-jennings-sip-dtls
=09
=09
	Isn't "running out of sockets" an implementation issue rather
than a protocol one?=20
	=20
	And I believe using multiple processes would solve that problem.
	=20
	  Tolga

		-----Original Message-----
		From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org]On Behalf Of Christian Stredicke
		Sent: Wednesday, February 16, 2005 5:02 PM
		To: Francois Audet; Cullen Jennings; Jonathan Rosenberg
		Cc: sip@ietf.org
		Subject: RE: [Sip] draft-jennings-sip-dtls
	=09
	=09
		Today I would recommend SIP operators to avoid TCP (and
that includes TLS) because the processes are simply running out of
sockets (think about an ITSP with one million customers). The UDP/DTLS
idea would make it reasonable simple to have much more connections and
on top have security. And as far as I understood the dtls it finally
does solve the fragmentation pain problem.
		=20
		That sounds like a great thing to me. I would vote for
dtls!
		=20
		CS


________________________________

			From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On Behalf Of Francois Audet
			Sent: Wednesday, February 16, 2005 9:47 PM
			To: 'Cullen Jennings'; 'Jonathan Rosenberg'
			Cc: sip@ietf.org
			Subject: RE: [Sip] draft-jennings-sip-dtls
		=09
		=09
			I would agree with Jonathan.
			=20
			Yes, some implementations of SIP/TCP have
performance issues compared to SIP/UDP, but not necessarily all of them.

			=20
			Also, if we need to address the other problems
of UDP transport (like fragmentation, and others), then it is not clear
to me that we are saving much in the first place by using UDP/DTLS
instead of TCP/TLS.
			=20
			I'd like to see real data before we add yet
another thing we'll have to implement...

				-----Original Message-----
				From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On Behalf Of Cullen Jennings
				Sent: Tuesday, February 15, 2005 20:08
				To: Jonathan Rosenberg
				Cc: sip@ietf.org
				Subject: Re: [Sip]
draft-jennings-sip-dtls
			=09
			=09
			=09
				Oops - I meant to put that.... There is
pretty much one key thing. No one has build an single edge proxy that
can terminate 100k to 1M connections to UAs using TLS. In theory it is
possible, but in practice it seems hard. The argument is that this will
be easier with DTLS. The issue is not the time it takes to do the crypto
- session resumption deals with that nicely - it just the issues of
dealing with half a million TCP connections to one box. Of course no one
has done it with DTLS either :-)
			=09
				I believe the argument we made for SCTP
was that adding an extensions for SCTP won't increase the complexity of
things that don't support SCTP.=20
			=09
				I agree the UDP/TCP complexity made SIP
more complicated and I agree that sip and sips made things more
complicated. I'm not sure I buy that both TLS and TCP made things more
complicated.=20
			=09
			=09
				On 2/15/05 7:15 PM, "Jonathan Rosenberg"
<jdrosen@cisco.com> wrote:
			=09
			=09

				Cullen,=20
			=09
				What seems missing to me from this is
requirements and problem=20
				statements. What is DTLS doing for us
that we don't get from TLS?=20
			=09
				Though SIP can run over many different
transport protocols, I think=20
				experience over time has shown that more
choices here is not necessarily=20
				a good thing, as SIP has a fair bit of
complexity as a result of dealing=20
				with the differences between UDP and
TCP. As such, I don't think its a=20
				good idea to just add more transport
protocols to SIP's list of=20
				supported ones unless there is a
compelling problem that it is solving.=20
			=09
				Thanks,=20
				Jonathan R.=20
			=09
				Cullen Jennings wrote:=20
			=09
				>=20
				> Nagendra and I put together a draft on
using DTLS with SIP. Until it=20
				> shows up in the archives you can find
it at=20
				>=20
				>
http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls
-00.html=20
				>=20
				> (there is a .txt version too)=20
				>=20
				>=20
				> The abstract is:=20
				>=20
				>    This draft specifies how to use
Datagram Transport Layer Security=20
				>    (DTLS) as a transport for SIP.
DTLS is a new protocol for providing=20
				>    TLS security over a datagram
protocol.=20
				>=20
				>=20
				>
------------------------------------------------------------------------

				>=20
				>
_______________________________________________=20
				> Sip mailing list
https://www1.ietf.org/mailman/listinfo/sip=20
				> This list is for NEW development of
the core SIP Protocol=20
				> Use sip-implementors@cs.columbia.edu
for questions on current sip=20
				> Use sipping@ietf.org for new
developments on the application of sip=20
			=09

			=09
			=09


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2604" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>16 bit set an upper limit of 64 K connections. =
My feeling=20
is that TCP was not designed for this purpose.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>Sorry for being so pragmatic. Maybe someone can =
show me an=20
implementation that can handle 100 K connections on one IP=20
address.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Tolga Asveren=20
  [mailto:asveren@ulticom.com] <BR><B>Sent:</B> Thursday, February 17, =
2005 8:16=20
  PM<BR><B>To:</B> Christian Stredicke; Francois Audet; Cullen Jennings; =

  Jonathan Rosenberg<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: =
[Sip]=20
  draft-jennings-sip-dtls<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D814205418-17022005>Isn't "running out of sockets" an =
implementation=20
  issue rather than a protocol one? </SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D814205418-17022005></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D814205418-17022005>And=20
  I believe using multiple processes would solve that=20
  problem.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D814205418-17022005></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D814205418-17022005>&nbsp;&nbsp;Tolga</SPAN></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> =
sip-bounces@ietf.org=20
    [mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christian=20
    Stredicke<BR><B>Sent:</B> Wednesday, February 16, 2005 5:02 =
PM<BR><B>To:</B>=20
    Francois Audet; Cullen Jennings; Jonathan Rosenberg<BR><B>Cc:</B>=20
    sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
    draft-jennings-sip-dtls<BR><BR></FONT></DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
    color=3D#0000ff size=3D2>Today I would recommend SIP operators to =
avoid TCP (and=20
    that includes TLS) because the processes are simply running out of =
sockets=20
    (think about an ITSP with one million customers). The UDP/DTLS idea =
would=20
    make it reasonable simple to have much more connections and on top =
have=20
    security. And as far as I understood the dtls it finally does solve =
the=20
    fragmentation pain problem.</FONT></SPAN></DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
    color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
    color=3D#0000ff size=3D2>That sounds like a great thing to me. I =
would vote for=20
    dtls!</FONT></SPAN></DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
    color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
    color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr =
align=3Dleft>
      <HR tabIndex=3D-1>
      <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
      [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Francois=20
      Audet<BR><B>Sent:</B> Wednesday, February 16, 2005 9:47 =
PM<BR><B>To:</B>=20
      'Cullen Jennings'; 'Jonathan Rosenberg'<BR><B>Cc:</B>=20
      sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
      draft-jennings-sip-dtls<BR></FONT><BR></DIV>
      <DIV></DIV>
      <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
      size=3D2>I would agree with Jonathan.</FONT></SPAN></DIV>
      <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
      size=3D2></FONT></SPAN>&nbsp;</DIV>
      <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
      size=3D2>Yes, some implementations of SIP/TCP have performance =
issues=20
      compared to SIP/UDP, but not necessarily all of them. =
</FONT></SPAN></DIV>
      <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
      size=3D2></FONT></SPAN>&nbsp;</DIV>
      <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
      size=3D2>Also, if we need to address the other problems of UDP =
transport=20
      (like fragmentation, and others), then it is not clear to me that =
we are=20
      saving much in the first place by using UDP/DTLS instead of=20
      TCP/TLS.</FONT></SPAN></DIV>
      <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
      size=3D2></FONT></SPAN>&nbsp;</DIV>
      <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
      size=3D2>I'd like to see real data before we add yet another thing =
we'll=20
      have to implement...</FONT></SPAN></DIV>
      <BLOCKQUOTE=20
      style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 =
2px solid; MARGIN-RIGHT: 0px">
        <DIV></DIV>
        <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
        face=3DTahoma size=3D2>-----Original =
Message-----<BR><B>From:</B>=20
        sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf =
Of=20
        </B>Cullen Jennings<BR><B>Sent:</B> Tuesday, February 15, 2005=20
        20:08<BR><B>To:</B> Jonathan Rosenberg<BR><B>Cc:</B>=20
        sip@ietf.org<BR><B>Subject:</B> Re: [Sip]=20
        draft-jennings-sip-dtls<BR><BR></FONT></DIV><FONT=20
        face=3D"Verdana, Helvetica, Arial"><SPAN style=3D"FONT-SIZE: =
12px"><BR>Oops=20
        - I meant to put that.... There is pretty much one key thing. No =
one has=20
        build an single edge proxy that can terminate 100k to 1M =
connections to=20
        UAs using TLS. In theory it is possible, but in practice it =
seems hard.=20
        The argument is that this will be easier with DTLS. The issue is =
not the=20
        time it takes to do the crypto - session resumption deals with =
that=20
        nicely - it just the issues of dealing with half a million TCP=20
        connections to one box. Of course no one has done it with DTLS =
either=20
        :-)<BR><BR>I believe the argument we made for SCTP was that =
adding an=20
        extensions for SCTP won't increase the complexity of things that =
don't=20
        support SCTP. <BR><BR>I agree the UDP/TCP complexity made SIP =
more=20
        complicated and I agree that sip and sips made things more =
complicated.=20
        I'm not sure I buy that both TLS and TCP made things more =
complicated.=20
        <BR><BR><BR>On 2/15/05 7:15 PM, "Jonathan Rosenberg"=20
        &lt;jdrosen@cisco.com&gt; wrote:<BR><BR></SPAN></FONT>
        <BLOCKQUOTE><FONT face=3D"Verdana, Helvetica, Arial"><SPAN=20
          style=3D"FONT-SIZE: 12px">Cullen, <BR><BR>What seems missing =
to me from=20
          this is requirements and problem <BR>statements. What is DTLS =
doing=20
          for us that we don't get from TLS? <BR><BR>Though SIP can run =
over=20
          many different transport protocols, I think <BR>experience =
over time=20
          has shown that more choices here is not necessarily <BR>a good =
thing,=20
          as SIP has a fair bit of complexity as a result of dealing =
<BR>with=20
          the differences between UDP and TCP. As such, I don't think =
its a=20
          <BR>good idea to just add more transport protocols to SIP's =
list of=20
          <BR>supported ones unless there is a compelling problem that =
it is=20
          solving. <BR><BR>Thanks, <BR>Jonathan R. <BR><BR>Cullen =
Jennings=20
          wrote: <BR><BR>&gt; <BR>&gt; Nagendra and I put together a =
draft on=20
          using DTLS with SIP. Until it <BR>&gt; shows up in the =
archives you=20
          can find it at <BR>&gt; <BR>&gt; <A=20
          =
href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-s=
ip-dtls-00.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-j=
ennings-sip-dtls-00.html</A>=20
          <BR>&gt; <BR>&gt; (there is a .txt version too) <BR>&gt; =
<BR>&gt;=20
          <BR>&gt; The abstract is: <BR>&gt; <BR>&gt; =
&nbsp;&nbsp;&nbsp;This=20
          draft specifies how to use Datagram Transport Layer Security =
<BR>&gt;=20
          &nbsp;&nbsp;&nbsp;(DTLS) as a transport for SIP. &nbsp;DTLS is =
a new=20
          protocol for providing <BR>&gt; &nbsp;&nbsp;&nbsp;TLS security =
over a=20
          datagram protocol. <BR>&gt; <BR>&gt; <BR>&gt;=20
          =
------------------------------------------------------------------------ =

          <BR>&gt; <BR>&gt; =
_______________________________________________=20
          <BR>&gt; Sip mailing list=20
          &nbsp;https://www1.ietf.org/mailman/listinfo/sip <BR>&gt; This =
list is=20
          for NEW development of the core SIP Protocol <BR>&gt; Use=20
          sip-implementors@cs.columbia.edu for questions on current sip =
<BR>&gt;=20
          Use sipping@ietf.org for new developments on the application =
of sip=20
          <BR></SPAN></FONT></BLOCKQUOTE><FONT=20
        face=3D"Verdana, Helvetica, Arial"><SPAN=20
      style=3D"FONT-SIZE: =
12px"><BR></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></SPAN></FO=
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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
--===============0967825894==--



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From: "Vijay K. Gurbani" <vkg@lucent.com>
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To: Tolga Asveren <asveren@ulticom.com>
Subject: Re: [Sip] draft-jennings-sip-dtls
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Tolga Asveren wrote:
> Isn't "running out of sockets" an implementation issue rather than a 
> protocol one?

It's actually a little bit of both.  See
http://www.ietf.org/internet-drafts/draft-gurbani-sipping-connection-guidelines-01.txt

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] draft-jennings-sip-dtls
Date: Thu, 17 Feb 2005 18:40:57 -0600
To: "Christian Stredicke" <Christian.Stredicke@snom.de>
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        Francois Audet <audet@nortel.com>, Tolga Asveren <asveren@ulticom.com>
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On Feb 17, 2005, at 2:00 PM, Christian Stredicke wrote:

> 16 bit set an upper limit of 64 K connections. My feeling is that TCP=20=

> was not designed for this purpose.
> =A0
> Sorry for being so pragmatic. Maybe someone can show me an=20
> implementation that can handle 100 K connections on one IP address.
> =A0

Actually, we fairly thoroughly flogged this topic a few years ago.

The net-net was that the 64k limit is a kernel limit, induced by the=20
use of 16-bit indices in file descriptor tables and other places.

Some kernels (or embedded IP limitations that are not kernel specific)=20=

don't seem to have this limitation. I'm told by some of the bootserver=20=

engineers I talk with occasionally that they can get several hundred=20
thousand TCP connections on Solaris 9, at a cost of about 4kB per=20
connection. Of course, this eats a boatload of RAM.

But DTLS does have some nice properties, and its ability to scale=20
larger on a larger set of OS platforms is quite possibly one of them.

--
Dean=


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From: "Tolga Asveren" <asveren@ulticom.com>
To: "Christian Stredicke" <Christian.Stredicke@snom.de>,
        "Francois Audet" <audet@nortel.com>,
        "Cullen Jennings" <fluffy@cisco.com>,
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Subject: RE: [Sip] draft-jennings-sip-dtls
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Message<Actually this becomes too implementation dependent, probably need to
switch to implementors list, but just because it is about the
existence/non-existence of certain limitation which might be an argument in
favor of a new proposal, I continue here>

Why is there a limitation because of 16 bit (I assume you are referring to
local/remote port field length in TCP header)? When the server side calls
"accept", a new socket will be assigned but no extra port is consumed.

And yes, there are ways to support the case you mentioned -with one IP
Address/port pair-.

Don't get me wrong, obviously the protocol needs to be implementable, but I
personally do not see a problem related with that issue, neither in TCP
protocol nor how to achieve the desired functionality in mainstream OS.

        Tolga
  -----Original Message-----
  From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Christian Stredicke
  Sent: Thursday, February 17, 2005 3:01 PM
  To: Tolga Asveren; Francois Audet; Cullen Jennings; Jonathan Rosenberg
  Cc: sip@ietf.org
  Subject: RE: [Sip] draft-jennings-sip-dtls


  16 bit set an upper limit of 64 K connections. My feeling is that TCP was
not designed for this purpose.

  Sorry for being so pragmatic. Maybe someone can show me an implementation
that can handle 100 K connections on one IP address.

  CS



----------------------------------------------------------------------------
    From: Tolga Asveren [mailto:asveren@ulticom.com]
    Sent: Thursday, February 17, 2005 8:16 PM
    To: Christian Stredicke; Francois Audet; Cullen Jennings; Jonathan
Rosenberg
    Cc: sip@ietf.org
    Subject: RE: [Sip] draft-jennings-sip-dtls


    Isn't "running out of sockets" an implementation issue rather than a
protocol one?

    And I believe using multiple processes would solve that problem.

      Tolga
      -----Original Message-----
      From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Christian Stredicke
      Sent: Wednesday, February 16, 2005 5:02 PM
      To: Francois Audet; Cullen Jennings; Jonathan Rosenberg
      Cc: sip@ietf.org
      Subject: RE: [Sip] draft-jennings-sip-dtls


      Today I would recommend SIP operators to avoid TCP (and that includes
TLS) because the processes are simply running out of sockets (think about an
ITSP with one million customers). The UDP/DTLS idea would make it reasonable
simple to have much more connections and on top have security. And as far as
I understood the dtls it finally does solve the fragmentation pain problem.

      That sounds like a great thing to me. I would vote for dtls!

      CS



------------------------------------------------------------------------
        From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf
Of Francois Audet
        Sent: Wednesday, February 16, 2005 9:47 PM
        To: 'Cullen Jennings'; 'Jonathan Rosenberg'
        Cc: sip@ietf.org
        Subject: RE: [Sip] draft-jennings-sip-dtls


        I would agree with Jonathan.

        Yes, some implementations of SIP/TCP have performance issues
compared to SIP/UDP, but not necessarily all of them.

        Also, if we need to address the other problems of UDP transport
(like fragmentation, and others), then it is not clear to me that we are
saving much in the first place by using UDP/DTLS instead of TCP/TLS.

        I'd like to see real data before we add yet another thing we'll have
to implement...
          -----Original Message-----
          From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf
Of Cullen Jennings
          Sent: Tuesday, February 15, 2005 20:08
          To: Jonathan Rosenberg
          Cc: sip@ietf.org
          Subject: Re: [Sip] draft-jennings-sip-dtls



          Oops - I meant to put that.... There is pretty much one key thing.
No one has build an single edge proxy that can terminate 100k to 1M
connections to UAs using TLS. In theory it is possible, but in practice it
seems hard. The argument is that this will be easier with DTLS. The issue is
not the time it takes to do the crypto - session resumption deals with that
nicely - it just the issues of dealing with half a million TCP connections
to one box. Of course no one has done it with DTLS either :-)

          I believe the argument we made for SCTP was that adding an
extensions for SCTP won't increase the complexity of things that don't
support SCTP.

          I agree the UDP/TCP complexity made SIP more complicated and I
agree that sip and sips made things more complicated. I'm not sure I buy
that both TLS and TCP made things more complicated.


          On 2/15/05 7:15 PM, "Jonathan Rosenberg" <jdrosen@cisco.com>
wrote:


            Cullen,

            What seems missing to me from this is requirements and problem
            statements. What is DTLS doing for us that we don't get from
TLS?

            Though SIP can run over many different transport protocols, I
think
            experience over time has shown that more choices here is not
necessarily
            a good thing, as SIP has a fair bit of complexity as a result of
dealing
            with the differences between UDP and TCP. As such, I don't think
its a
            good idea to just add more transport protocols to SIP's list of
            supported ones unless there is a compelling problem that it is
solving.

            Thanks,
            Jonathan R.

            Cullen Jennings wrote:

            >
            > Nagendra and I put together a draft on using DTLS with SIP.
Until it
            > shows up in the archives you can find it at
            >
            >
http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-dtls-00.
html
            >
            > (there is a .txt version too)
            >
            >
            > The abstract is:
            >
            >    This draft specifies how to use Datagram Transport Layer
Security
            >    (DTLS) as a transport for SIP.  DTLS is a new protocol for
providing
            >    TLS security over a datagram protocol.
            >
            >










  > ------------------------------------------------------------------------
            >
            > _______________________________________________
            > 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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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1479" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =

size=3D2>&lt;Actually this becomes too implementation dependent, =
probably need=20
to&nbsp;switch to implementors list, but just because it is about the=20
existence/non-existence of certain limitation which might be an argument =
in=20
favor of a new proposal, I continue here&gt;</FONT></SPAN></DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =
size=3D2>Why is=20
there a limitation because of 16 bit (I assume you are referring to =
local/remote=20
port field length in TCP header)? When the server side calls "accept", a =
new=20
socket will be assigned but no extra port is =
consumed.</FONT></SPAN></DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =
size=3D2>And=20
yes, there are ways to support the case you mentioned -with one IP =
Address/port=20
pair-.</FONT></SPAN></DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =
size=3D2>Don't=20
get me wrong, obviously&nbsp;the protocol&nbsp;needs to be =
implementable, but I=20
personally do not see a problem related with that issue, neither in TCP =
protocol=20
nor how to achieve the desired functionality in mainstream=20
OS.</FONT></SPAN></DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D735290822-17022005><FONT face=3DArial color=3D#0000ff =

size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Tolga</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> =
sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christian=20
  Stredicke<BR><B>Sent:</B> Thursday, February 17, 2005 3:01 =
PM<BR><B>To:</B>=20
  Tolga Asveren; Francois Audet; Cullen Jennings; Jonathan=20
  Rosenberg<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
  draft-jennings-sip-dtls<BR><BR></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2>16 bit set an upper limit of 64 K =
connections. My feeling=20
  is that TCP was not designed for this purpose.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2>Sorry for being so pragmatic. Maybe someone =
can show me=20
  an implementation that can handle 100 K connections on one IP=20
  address.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D470075819-17022005><FONT =
face=3DVerdana=20
  color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> Tolga Asveren=20
    [mailto:asveren@ulticom.com] <BR><B>Sent:</B> Thursday, February 17, =
2005=20
    8:16 PM<BR><B>To:</B> Christian Stredicke; Francois Audet; Cullen =
Jennings;=20
    Jonathan Rosenberg<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: =
[Sip]=20
    draft-jennings-sip-dtls<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
    class=3D814205418-17022005>Isn't "running out of sockets" an =
implementation=20
    issue rather than a protocol one? </SPAN></FONT></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
    class=3D814205418-17022005></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
    class=3D814205418-17022005>And I believe using multiple processes =
would solve=20
    that problem.</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
    class=3D814205418-17022005></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
    class=3D814205418-17022005>&nbsp;&nbsp;Tolga</SPAN></FONT></DIV>
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
      size=3D2>-----Original Message-----<BR><B>From:</B> =
sip-bounces@ietf.org=20
      [mailto:sip-bounces@ietf.org]<B>On Behalf Of </B>Christian=20
      Stredicke<BR><B>Sent:</B> Wednesday, February 16, 2005 5:02=20
      PM<BR><B>To:</B> Francois Audet; Cullen Jennings; Jonathan=20
      Rosenberg<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
      draft-jennings-sip-dtls<BR><BR></FONT></DIV>
      <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
      color=3D#0000ff size=3D2>Today I would recommend SIP operators to =
avoid TCP=20
      (and that includes TLS) because the processes are simply running =
out of=20
      sockets (think about an ITSP with one million customers). The =
UDP/DTLS=20
      idea would make it reasonable simple to have much more connections =
and on=20
      top have security. And as far as I understood the dtls it finally =
does=20
      solve the fragmentation pain problem.</FONT></SPAN></DIV>
      <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
      color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
      <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
      color=3D#0000ff size=3D2>That sounds like a great thing to me. I =
would vote=20
      for dtls!</FONT></SPAN></DIV>
      <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
      color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
      <DIV dir=3Dltr align=3Dleft><SPAN class=3D829285521-16022005><FONT =
face=3DVerdana=20
      color=3D#0000ff size=3D2>CS</FONT></SPAN></DIV><BR>
      <BLOCKQUOTE dir=3Dltr=20
      style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
        <DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr =
align=3Dleft>
        <HR tabIndex=3D-1>
        <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
        [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Francois=20
        Audet<BR><B>Sent:</B> Wednesday, February 16, 2005 9:47 =
PM<BR><B>To:</B>=20
        'Cullen Jennings'; 'Jonathan Rosenberg'<BR><B>Cc:</B>=20
        sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
        draft-jennings-sip-dtls<BR></FONT><BR></DIV>
        <DIV></DIV>
        <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
        size=3D2>I would agree with Jonathan.</FONT></SPAN></DIV>
        <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
        size=3D2></FONT></SPAN>&nbsp;</DIV>
        <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
        size=3D2>Yes, some implementations of SIP/TCP have performance =
issues=20
        compared to SIP/UDP, but not necessarily all of them.=20
        </FONT></SPAN></DIV>
        <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
        size=3D2></FONT></SPAN>&nbsp;</DIV>
        <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
        size=3D2>Also, if we need to address the other problems of UDP =
transport=20
        (like fragmentation, and others), then it is not clear to me =
that we are=20
        saving much in the first place by using UDP/DTLS instead of=20
        TCP/TLS.</FONT></SPAN></DIV>
        <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
        size=3D2></FONT></SPAN>&nbsp;</DIV>
        <DIV><SPAN class=3D705474319-16022005><FONT face=3DArial =
color=3D#800000=20
        size=3D2>I'd like to see real data before we add yet another =
thing we'll=20
        have to implement...</FONT></SPAN></DIV>
        <BLOCKQUOTE=20
        style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: =
#800000 2px solid; MARGIN-RIGHT: 0px">
          <DIV></DIV>
          <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
          face=3DTahoma size=3D2>-----Original =
Message-----<BR><B>From:</B>=20
          sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On =
Behalf Of=20
          </B>Cullen Jennings<BR><B>Sent:</B> Tuesday, February 15, 2005 =

          20:08<BR><B>To:</B> Jonathan Rosenberg<BR><B>Cc:</B>=20
          sip@ietf.org<BR><B>Subject:</B> Re: [Sip]=20
          draft-jennings-sip-dtls<BR><BR></FONT></DIV><FONT=20
          face=3D"Verdana, Helvetica, Arial"><SPAN=20
          style=3D"FONT-SIZE: 12px"><BR>Oops - I meant to put that.... =
There is=20
          pretty much one key thing. No one has build an single edge =
proxy that=20
          can terminate 100k to 1M connections to UAs using TLS. In =
theory it is=20
          possible, but in practice it seems hard. The argument is that =
this=20
          will be easier with DTLS. The issue is not the time it takes =
to do the=20
          crypto - session resumption deals with that nicely - it just =
the=20
          issues of dealing with half a million TCP connections to one =
box. Of=20
          course no one has done it with DTLS either :-)<BR><BR>I =
believe the=20
          argument we made for SCTP was that adding an extensions for =
SCTP won't=20
          increase the complexity of things that don't support SCTP. =
<BR><BR>I=20
          agree the UDP/TCP complexity made SIP more complicated and I =
agree=20
          that sip and sips made things more complicated. I'm not sure I =
buy=20
          that both TLS and TCP made things more complicated. =
<BR><BR><BR>On=20
          2/15/05 7:15 PM, "Jonathan Rosenberg" =
&lt;jdrosen@cisco.com&gt;=20
          wrote:<BR><BR></SPAN></FONT>
          <BLOCKQUOTE><FONT face=3D"Verdana, Helvetica, Arial"><SPAN=20
            style=3D"FONT-SIZE: 12px">Cullen, <BR><BR>What seems missing =
to me=20
            from this is requirements and problem <BR>statements. What =
is DTLS=20
            doing for us that we don't get from TLS? <BR><BR>Though SIP =
can run=20
            over many different transport protocols, I think =
<BR>experience over=20
            time has shown that more choices here is not necessarily =
<BR>a good=20
            thing, as SIP has a fair bit of complexity as a result of =
dealing=20
            <BR>with the differences between UDP and TCP. As such, I =
don't think=20
            its a <BR>good idea to just add more transport protocols to =
SIP's=20
            list of <BR>supported ones unless there is a compelling =
problem that=20
            it is solving. <BR><BR>Thanks, <BR>Jonathan R. =
<BR><BR>Cullen=20
            Jennings wrote: <BR><BR>&gt; <BR>&gt; Nagendra and I put =
together a=20
            draft on using DTLS with SIP. Until it <BR>&gt; shows up in =
the=20
            archives you can find it at <BR>&gt; <BR>&gt; <A=20
            =
href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-s=
ip-dtls-00.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-j=
ennings-sip-dtls-00.html</A>=20
            <BR>&gt; <BR>&gt; (there is a .txt version too) <BR>&gt; =
<BR>&gt;=20
            <BR>&gt; The abstract is: <BR>&gt; <BR>&gt; =
&nbsp;&nbsp;&nbsp;This=20
            draft specifies how to use Datagram Transport Layer Security =

            <BR>&gt; &nbsp;&nbsp;&nbsp;(DTLS) as a transport for SIP. =
&nbsp;DTLS=20
            is a new protocol for providing <BR>&gt; =
&nbsp;&nbsp;&nbsp;TLS=20
            security over a datagram protocol. <BR>&gt; <BR>&gt; =
<BR>&gt;=20
            =
------------------------------------------------------------------------ =

            <BR>&gt; <BR>&gt; =
_______________________________________________=20
            <BR>&gt; Sip mailing list=20
            &nbsp;https://www1.ietf.org/mailman/listinfo/sip <BR>&gt; =
This list=20
            is for NEW development of the core SIP Protocol <BR>&gt; Use =

            sip-implementors@cs.columbia.edu for questions on current =
sip=20
            <BR>&gt; Use sipping@ietf.org for new developments on the=20
            application of sip <BR></SPAN></FONT></BLOCKQUOTE><FONT=20
          face=3D"Verdana, Helvetica, Arial"><SPAN=20
        style=3D"FONT-SIZE: =
12px"><BR></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOT=
E></SPAN></FONT></BODY></HTML>

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From sip-bounces@ietf.org  Fri Feb 18 09:12:32 2005
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From: "Tolga Asveren" <asveren@ulticom.com>
To: "Vijay K. Gurbani" <vkg@lucent.com>
Subject: RE: [Sip] draft-jennings-sip-dtls
Date: Fri, 18 Feb 2005 08:40:57 -0500
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Hi Vijay,

Could you point to specific section in the draft -or explain the protocol
limitation briefly-?

   Thanks,
   Tolga

> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
> Sent: Thursday, February 17, 2005 5:14 PM
> To: Tolga Asveren
> Cc: sip@ietf.org
> Subject: Re: [Sip] draft-jennings-sip-dtls
>
>
> Tolga Asveren wrote:
> > Isn't "running out of sockets" an implementation issue rather than a
> > protocol one?
>
> It's actually a little bit of both.  See
> http://www.ietf.org/internet-drafts/draft-gurbani-sipping-connecti
on-guidelines-01.txt

- vijay
--
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From: "Vijay K. Gurbani" <vkg@lucent.com>
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Tolga Asveren wrote:
> Hi Vijay,
> 
> Could you point to specific section in the draft -or explain the protocol
> limitation briefly-?

Section 1 and 2 of the I-D discuss the limitations...

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" />

 

Hi all,

 

I am currently looking at using SigComp with EPIC compression for SIP
messages. After reading relevant RFCs and Internet drafts, a few things are
not clear to me.

 

First, I understand that EPIC relies on a profile to compress and decompress
data. The EPIC reference implementation from Roke Manor Research Ltd
provides a profile for RTP. However, I can't find a profile for SIP.

Is there one out there?

 

I think I can write one but this would likely be a problem for
interoperability since the decompressor MUST use the same profile. One
solution would be to download the profile to the other endpoint but this
does not look too good to me, partly because I have a concern about the
profile size. Furthermore, I don't see how I could instruct the other
endpoint to use a downloaded profile (file name and location ?).

Any other option?

 

Also, in the SigComp User Guide (draft-ietf-rohc-sigcomp-user-guide-00.txt),
section 4.6 is talking about EPIC and SIP but never mentions the need or
existence of a profile.

Without a profile, how can EPIC be used ?

 

The mentioned section also shows compression results for three algorithms,
including EPIC. All algorithms employed a static dictionary and shared
compression.

Exactly how the EPIC results were obtained is not clear to me, in particular
how EPIC was used ?

Only the Hierarchical Huffman algorithm ?

EPIC used in conjunction with DEFLATE ?

Did all three algorithms used the static dictionary described in RFC3485 ?

 

Any help would be really appreciated, Thanks.

 


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style="mso-ansi-language: EN-US"><FONT size=3><FONT 
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<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
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face="Times New Roman"></FONT></FONT></SPAN>&nbsp;</P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Hi 
all,<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">I am 
currently looking at using SigComp with EPIC compression for SIP messages. After 
reading relevant RFCs and Internet drafts, a few things are not clear to 
me.<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">First, I understand that EPIC relies on a profile to 
compress and decompress data. The EPIC reference implementation from Roke Manor 
Research Ltd provides a profile for RTP. However, I can't find a profile for 
SIP.<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Is 
there one out there?<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">I 
think I can write one but this would likely be a problem for interoperability 
since the decompressor MUST use the same profile. One solution would be to 
download the profile to the other endpoint but this does not look too good to 
me, partly because I have a concern about the profile size. Furthermore, I don't 
see how I could instruct the other endpoint to use a downloaded profile (file 
name and location ?).<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Any 
other option?<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Also, 
in the SigComp User Guide (draft-ietf-rohc-sigcomp-user-guide-00.txt), section 
4.6 is talking about EPIC and SIP but never mentions the need or existence of a 
profile.<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">Without a profile, how can EPIC be used 
?<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">The 
mentioned section also shows compression results for three algorithms, including 
EPIC. All algorithms employed a static dictionary and shared 
compression.<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">Exactly how the EPIC results were obtained is not clear 
to me, in particular how EPIC was used ?<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Only 
the Hierarchical Huffman algorithm ?<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">EPIC 
used in conjunction with DEFLATE ?<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Did 
all three algorithms used the static dictionary described in RFC3485 
?<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT 
face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Any 
help would be really appreciated, Thanks.</FONT></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
style="mso-ansi-language: EN-US"><FONT face=Arial 
size=2></FONT></SPAN></FONT>&nbsp;</P></DIV></BODY></HTML>

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From sip-bounces@ietf.org  Sun Feb 20 22:47:18 2005
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From: "Tai-Hsing Yu" <fhubert@ms37.hinet.net>
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Subject: [Sip] Questions of handling large request message
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Hello,
        I have some questions about handling SIP msgs that are larger =
than 1300 bytes. RFC 3261, chapter 18 as listed, mentions that if =
request message is larger than 1300 bytes, it MUST be sent using TCP =
because of the meesage size constraints. But what should the proxy do =
when user registers to Proxy/Registrar with =
=E2=80=9Ctransport=3Dudp=E2=80=9D or without =E2=80=9Ctransport=E2=80=9D =
parameter??

Any acknowledgement will be deeply appreciated.
Sincerely,
Fish


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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		: Suppression of Session Initiation Protocol REFER Method Implicit Subscription
	Author(s)	: O. Levin
	Filename	: draft-ietf-sip-refer-with-norefersub-01.txt
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	Date		: 2005-2-21
	
This specification defines a way to suppress an implicit subscription
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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'Lepine, Jean-Pierre'" <jeanpierre.lepine@smisrd.com>, rohc@ietf.org,
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Subject: [Sip] RE: [rohc] SigComp with EPIC compression for SIP messages
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The EPIC work was an experimental work that was brought to the ROHC WG as a header compression proposal. The work was never adopted by the WG and was terminated about 2 years ago, but some of the ideas about Formal Notation (FN) have been used in the ROHC TCP work. 
 
The SigComp user guide did (I am not sure about the new version just released) refer to EPIC, which is probably a mistake as algorithms are not specified in that document, and there is no reference available for an EPIC algorithm. 
 
BR
/Lars-Erik
 

-----Original Message-----
From: rohc-bounces@ietf.org [mailto:rohc-bounces@ietf.org]On Behalf Of Lepine, Jean-Pierre
Sent: den 18 februari 2005 19:18
To: rohc@ietf.org; sip@ietf.org; sipping@ietf.org
Subject: [rohc] SigComp with EPIC compression for SIP messages




 

Hi all,

 

I am currently looking at using SigComp with EPIC compression for SIP messages. After reading relevant RFCs and Internet drafts, a few things are not clear to me.

 

First, I understand that EPIC relies on a profile to compress and decompress data. The EPIC reference implementation from Roke Manor Research Ltd provides a profile for RTP. However, I can't find a profile for SIP.

Is there one out there?

 

I think I can write one but this would likely be a problem for interoperability since the decompressor MUST use the same profile. One solution would be to download the profile to the other endpoint but this does not look too good to me, partly because I have a concern about the profile size. Furthermore, I don't see how I could instruct the other endpoint to use a downloaded profile (file name and location ?).

Any other option?

 

Also, in the SigComp User Guide (draft-ietf-rohc-sigcomp-user-guide-00.txt), section 4.6 is talking about EPIC and SIP but never mentions the need or existence of a profile.

Without a profile, how can EPIC be used ?

 

The mentioned section also shows compression results for three algorithms, including EPIC. All algorithms employed a static dictionary and shared compression.

Exactly how the EPIC results were obtained is not clear to me, in particular how EPIC was used ?

Only the Hierarchical Huffman algorithm ?

EPIC used in conjunction with DEFLATE ?

Did all three algorithms used the static dictionary described in RFC3485 ?

 

Any help would be really appreciated, Thanks.

 


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<DIV><SPAN class=565274307-21022005><FONT face=Arial color=#0000ff size=2>The 
EPIC work was an experimental work that was brought to the ROHC WG as a header 
compression proposal. The work was never adopted by the WG and was terminated 
about 2 years ago, but some of the ideas&nbsp;about Formal Notation (FN) have 
been&nbsp;used in the ROHC TCP work. </FONT></SPAN></DIV>
<DIV><SPAN class=565274307-21022005><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=565274307-21022005><FONT face=Arial color=#0000ff size=2>The 
SigComp user guide did (I am not sure about the new version just released) refer 
to EPIC, which is probably&nbsp;a mistake as algorithms are not specified in 
that document, and there is no reference available&nbsp;for an EPIC algorithm. 
</FONT></SPAN></DIV>
<DIV><SPAN class=565274307-21022005><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=565274307-21022005><FONT face=Arial color=#0000ff 
size=2>BR</FONT></SPAN></DIV>
<DIV><SPAN class=565274307-21022005><FONT face=Arial color=#0000ff 
size=2>/Lars-Erik</FONT></SPAN></DIV>
<DIV><SPAN class=565274307-21022005></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> rohc-bounces@ietf.org 
  [mailto:rohc-bounces@ietf.org]<B>On Behalf Of </B>Lepine, 
  Jean-Pierre<BR><B>Sent:</B> den 18 februari 2005 19:18<BR><B>To:</B> 
  rohc@ietf.org; sip@ietf.org; sipping@ietf.org<BR><B>Subject:</B> [rohc] 
  SigComp with EPIC compression for SIP messages<BR><BR></FONT></DIV>
  <DIV><FONT face=Arial size=2>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman"><o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman"></FONT></FONT></SPAN>&nbsp;</P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Hi 
  all,<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">I 
  am currently looking at using SigComp with EPIC compression for SIP messages. 
  After reading relevant RFCs and Internet drafts, a few things are not clear to 
  me.<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">First, I understand that EPIC relies on a profile to 
  compress and decompress data. The EPIC reference implementation from Roke 
  Manor Research Ltd provides a profile for RTP. However, I can't find a profile 
  for SIP.<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Is 
  there one out there?<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">I 
  think I can write one but this would likely be a problem for interoperability 
  since the decompressor MUST use the same profile. One solution would be to 
  download the profile to the other endpoint but this does not look too good to 
  me, partly because I have a concern about the profile size. Furthermore, I 
  don't see how I could instruct the other endpoint to use a downloaded profile 
  (file name and location ?).<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Any 
  other option?<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">Also, in the SigComp User Guide 
  (draft-ietf-rohc-sigcomp-user-guide-00.txt), section 4.6 is talking about EPIC 
  and SIP but never mentions the need or existence of a 
  profile.<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">Without a profile, how can EPIC be used 
  ?<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">The 
  mentioned section also shows compression results for three algorithms, 
  including EPIC. All algorithms employed a static dictionary and shared 
  compression.<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">Exactly how the EPIC results were obtained is not clear 
  to me, in particular how EPIC was used ?<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">Only the Hierarchical Huffman algorithm 
  ?<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">EPIC used in conjunction with DEFLATE 
  ?<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Did 
  all three algorithms used the static dictionary described in RFC3485 
  ?<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT 
  face="Times New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT size=3><FONT face="Times New Roman">Any 
  help would be really appreciated, Thanks.</FONT></FONT></SPAN></P>
  <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US 
  style="mso-ansi-language: EN-US"><FONT face=Arial 
  size=2></FONT></SPAN></FONT>&nbsp;</P></DIV></BLOCKQUOTE></BODY></HTML>

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    Hi Guys

I am a newbee in this area of protocol testing, I am developing a SIP=
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Looking forward to get some help, thanking in anticipation

Regards,
Akshay





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This is an announcement of RADEXT WG last call on "RADIUS Extension
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The document will be available on the IETF archive here:
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interoperation between RADIUS and Diameter, we encourage participants
to read both documents and comment on them.

The Diameter SIP Application document is available here:
http://www.ietf.org/internet-drafts/draft-ietf-aaa-diameter-sip-app-06.txt

_______________________________________________
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 Feb 23 20:42:31 2005
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> This message is in MIME format. Since your mail reader does not understand
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Kumiko and I submitted a new version of "Example call flows using SIP
security mechanisms". It is in the drafts directory and also html at

http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-sec-flow
s-02.html




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<FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:12.0px'><BR>
Kumiko and I submitted a new version of &quot;Example call flows using SIP =
security mechanisms&quot;. It is in the drafts directory and also html at <B=
R>
<BR>
<a href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-si=
p-sec-flows-02.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-=
jennings-sip-sec-flows-02.html</a><BR>
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_______________________________________________
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--===============0201989602==--



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--NextPart

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		: Communications Resource Priority for the Session Initiation Protocol (SIP)
	Author(s)	: H. Schulzrinne, J. Polk
	Filename	: draft-ietf-sip-resource-priority-06.txt
	Pages		: 33
	Date		: 2005-2-22
	
This document defines two new SIP header fields for communicating 
   resource priority, namely "Resource-Priority" and "Accept-Resource-
   Priority".  The "Resource-Priority" header field can influence the 
   behavior of SIP user agents, such as telephone gateways and IP 
   telephones, and Session Initiation Protocol (SIP) proxies.  It does 
   not directly influence the forwarding behavior of IP routers.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-resource-priority-06.txt

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_______________________________________________
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From sip-bounces@ietf.org  Wed Feb 23 21:55:01 2005
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Date: Wed, 23 Feb 2005 12:10:56 -0600
From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] SIP Agenda Requests, IETF 62
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If you have a topic that you think needs to be discussed at IETF 62 in 
the SIP WG meeting, please send me an agenda request.


Please use the Camarillo format if you can:


     <tr>
       <td><a href="mailto:Gonzalo.Camarillo@ericsson.com">
Gonzalo Camarillo</a></td>
       <td>Exploders</td>
       <td><center>5</center></td>
       <td>
	<a href=
"http://www.ietf.org/internet-drafts/draft-camarillo-sipping-exploders-02.txt">
	draft-camarillo-sipping-exploders-02.txt
	</a>
       </td>
     </tr>


Please keep in mind that WG meeting time is for resolving open isssues 
of the WG, not trying to pitch an idea that you didn't get traction for 
on the mailing list. So the chairs WILL be exercising editorial control 
on the agenda to see to it that WG priorities are met first. If we have 
extra time, we MAY entertain additional relevant topics.

I'll be assembling a requests list page and draft agenda on the 
www.softarmor.com site as time permits.

Thanks,

--
Dean Willis
SIP Co-Chair


_______________________________________________
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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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From: Pekka Pessi <Pekka.Pessi@nokia.com>
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Cc: sip@ietf.org, "'Lepine, Jean-Pierre'" <jeanpierre.lepine@smisrd.com>,
        rohc@ietf.org, sipping@ietf.org
Subject: [Sip] Re: [rohc] SigComp with EPIC compression for SIP messages
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"Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com> writes:
>The SigComp user guide did (I am not sure about the new version just
>released) refer to EPIC, which is probably a mistake as algorithms are not
>specified in that document, and there is no reference available for an EPIC
>algorithm. 

As you can see from the UDVM assembly code (the text also mentions
it) the compression algorithm referred as EPIC is just DEFLATE
with minor modifications. The UDVM assembly code in section 4.6
(and RFC 1951) is all the specification you need. 

A message compressed with DEFLATE consists of literals (bytes
0..255, end-of-block) and pairs (length, distance), where length
is 3..258 and distance in 1..32768. "EPIC" modifies DEFLATE to use
INPUT-HUFFMAN-coding for literals and (length, distance) pairs
insted of the normal static huffman codes, as specified in RFC
1951.

So, unlike DEFLATE bytecode "EPIC" does not need tables and the
literal/length symbols and distance values can be directly decoded
with a single INPUT-HUFFMAN instruction each. Another improvement
(?) is with the literal/length, it has two different codings,
depending whether the last symbol was a literal or length+distance
pair.

--Pekka

>-----Original Message-----
>From: rohc-bounces@ietf.org [mailto:rohc-bounces@ietf.org]On Behalf Of Lepine, Jean-Pierre
>Sent: den 18 februari 2005 19:18
>To: rohc@ietf.org; sip@ietf.org; sipping@ietf.org
>Subject: [rohc] SigComp with EPIC compression for SIP messages

>Hi all,

>I am currently looking at using SigComp with EPIC compression for
>SIP messages. After reading relevant RFCs and Internet drafts, a
>few things are not clear to me.

>First, I understand that EPIC relies on a profile to compress and
>decompress data. The EPIC reference implementation from Roke
>Manor Research Ltd provides a profile for RTP. However, I can't
>find a profile for SIP.

>Is there one out there?

>I think I can write one but this would likely be a problem for
>interoperability since the decompressor MUST use the same
>profile. One solution would be to download the profile to the
>other endpoint but this does not look too good to me, partly
>because I have a concern about the profile size. Furthermore, I
>don't see how I could instruct the other endpoint to use a
>downloaded profile (file name and location ?).

>Any other option?

>Also, in the SigComp User Guide
>(draft-ietf-rohc-sigcomp-user-guide-00.txt), section 4.6 is
>talking about EPIC and SIP but never mentions the need or
>existence of a profile.

>Without a profile, how can EPIC be used ?

>The mentioned section also shows compression results for three
>algorithms, including EPIC. All algorithms employed a static
>dictionary and shared compression.

>Exactly how the EPIC results were obtained is not clear to me, in
>particular how EPIC was used ?

>Only the Hierarchical Huffman algorithm ?

>EPIC used in conjunction with DEFLATE ?

>Did all three algorithms used the static dictionary described in RFC3485 ?



>Any help would be really appreciated, Thanks.

>_______________________________________________
>Rohc mailing list
>Rohc@ietf.org
>https://www1.ietf.org/mailman/listinfo/rohc

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--NextPart

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-03.txt
	Pages		: 33
	Date		: 2005-2-23
	
Several applications of the Session Initiation Protocol (SIP) require
a user agent (UA) to construct and distribute a URI which can be used
by anyone on the Internet to route a call to that specific UA
instance. A URI which 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-03.txt

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From sip-bounces@ietf.org  Thu Feb 24 00:10:44 2005
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Date: Tue, 22 Feb 2005 16:45:59 -0800
From: Cullen Jennings <fluffy@cisco.com>
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Subject: [Sip] draft-jennings-sip-hashcash-01
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

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I updated this draft on "SIP Computational Puzzles" so it has enough detail
to implement it. (Previous version did not). It is in the drafts directory
and also in HTML form at:

http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcash
-01.html

Cullen


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</HEAD>
<BODY>
<FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:12.0px'><BR>
I updated this draft on &quot;SIP Computational Puzzles&quot; so it has eno=
ugh detail to implement it. (Previous version did not). It is in the drafts =
directory and also in HTML form at:<BR>
<BR>
<a href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-si=
p-hashcash-01.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-j=
ennings-sip-hashcash-01.html</a><BR>
<BR>
Cullen<BR>
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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
--===============1796959537==--



From sip-bounces@ietf.org  Thu Feb 24 00:57:00 2005
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From: "Peter Hedman (LD/EAB)" <peter.hedman@ericsson.com>
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Subject: [SIP] SDP in OPTIONS
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Hi,

3GPP are discussing capability exchange using SIP OPTIONS. Specifically, to be able to optimise an exchange by including UAC capabilities within the OPTIONS request, for information to those UAS that understand it. The current understanding is that a UAS following RFC 3261 would simply ignore the SDP in the OPTIONS request and normal operations take place. 

Also, even if the UAS would read the SDP in an OPTIONS request, the SDP in the response would also be a capability document that will not in any way follow the offer/answer model.

It would be great to get the groups opinion on the above assumptions.

BR,
Peter

Ericsson AB
Phone: +46 46 231760
Mobile: +46 703602149
Mail: peter.hedman@ericsson.com


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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 Feb 24 01:01:55 2005
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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] TESTING Requirements
Date: Wed, 23 Feb 2005 11:20:20 -0600
To: "Akshay Bangalorekar" <akshayb@mahindrabt.com>
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On Feb 20, 2005, at 11:53 PM, Akshay Bangalorekar wrote:

> =A0=A0=A0 Hi Guys
> =A0
> I am a newbee in this area of protocol testing, I am developing a SIP=20=

> application. So wanted to know the requirements in detail for testing=20=

> the SIP application.
> =A0
> Looking forward to get some help, thanking in anticipation
>

I recommend prayer, fortitude, vigilance, and a great toleration for=20
ambiguity. And followups to the sip-implementors list:

mailto:sip-implementors@cs.columbia.edu

and I also believe that SIP Forum is working on some testing guidelines:

http://www.sipforum.org/

--
Dean


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From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [Sip] updated GRUU spec
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Folks,

I posted an updated version of the GRUU spec, which just popped out:

http://www.ietf.org/internet-drafts/draft-ietf-sip-gruu-03.txt

In the end, I did NOT change the fundamental mechanism per our 
discussions on the list; its still based on getting the gruu in the 
register response. There were lots of other text changes to improve 
clarity and make it simpler. The biggest changes were that the server 
will now overwrite the registration from a particular instance when a 
new one arrives. I've also punted on the issue we've been debating the 
last two meetings, on whether multiple contacts from a single instance 
can be registered.

Here are the detailed changes:



* instance parameter has to be a URN

* equality rules for instance ID are now equal to the function rules
for that URN scheme, else lexical URN rules if the scheme is
unknown. Mentioned that this is unliekly to be a problem, and that
clients should just use the same instance ID in all registrations

* rewrote introduction to do a much better job at motivating why you
should care about this draft

* added back the GRUU property of long-livedness, but comment that
this property cannot be guaranteed

* clarified that gruus are resolved using normal rfc3263 procedures

* substantial rewrite of the normative behaviors in order to simplify
the explanation, including an updated simpler UML model.

* previously, behavior was not specified when a contact expired. That
is now specified - its binding to both the AOR and GRUU are removed

* previously, if a Contact with an instance ID arrived and there
already was a contact with that instance ID registered, an error
response is returned. Now, the new contact replaces the previous
one. The 425 response code is therefore no longer needed, and has been
removed.

* SIP/SIPS handling is now specified - both a SIP and SIPS version of
a GRUU have to exist. The scheme used for the GRUU in the register
response is equal to that of the To in the REGISTER request. Little
else is said about sips processing; it should work equally to a sips
AOR (which is admittedly underspecified in RFC3261, but its not GRUU's
problem to fix that)

* clarified proxy processing by showing exactly how the processes of
section 16 in rfc3261 are affected - its just a different abstract
locatoin service

* note that Path (rfc3327) processing applies to GRUU as well

* sustantial text added to discuss the interactions with
record-routing and how to avoid the spiral that can ensue if you are
not careful

* clarify that if the contact in a register had a grid for some reason
(not a good idea), it would be overwritten with the grid value in a
gruu that is translated to that contact

* during discussions in the last few meetings, it was proposed that
multiple contacts of the same instance be allowed to register for a
gruu. This is useful only for supporting high availability in
clients. However, the requirements do not include such a
capability. As such, I went with the simpler approach as currently
documented, with a single contact for each instance. The high
availability work will require us to rethink the registration data
model more seriously, and should be relegated to a different spec that
can modify gruu processing if needed. draft-jennings-sipping-outbound
seems the right forum for that

* updated example based on new rules for conflict resolutions

* removed IANA registration for response code

* the conflict resoltuion algorithm provides a neat way to remove any
contact with a specific instance ID; the equivalent of * but targeted
to a specific instance ID, and this is documented

* tried to clean up terminology, since "contact" and its variants are
used all over the place with different meaning


Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   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 Feb 24 01:59:55 2005
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From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [Sip] a middle ground approach for knowing AOR of a gruu
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We had a long discussion in another thread about whether or not we had 
an additional requirement to allow the recipient of a GRUU to be able to 
determine the AOR associated with that gruu. This seemed to be an issue 
for systems that generate call logs or bills, since you'll have these 
entries that are meaningless.

Though I didnt make any changes to gruu in order to meet this 
requirement, there may be a simple approach that we can provide that 
fits with the current gruu mechanism.

The spec is clear that there are no constraints on how a domain 
constructs a gruu. The appendix includes a few sample algorithms (though 
I'll note one of them is broken...). As such, there is nothing wrong 
with a domain constructing a gruu by including the instance ID and AOR 
in the clear in the user part of the URI:

sip:jdrosen.instance.00sd8ggdgdasd===asd9@cisco.com

for example. Thus, we could list such a technique as a viable approach, 
and recommend that a particular means of encoding the URI be used in 
order to allow other entities to extract the aor.

Thus, we would have a way for the recipient of a domain to get a hint 
about the possible AOR (no guarantee though), without changing the 
mechanism at all.

Would that represent a reasonable compromise?

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   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 Feb 24 02:29:00 2005
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From: "kkuei" <kkuei@above.net.tw>
To: Jonathan Rosenberg <jdrosen@cisco.com>, sip@ietf.org
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Subject: [Sip] rfc3264 question
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I'm reading rfc3264 (offer/answer model with SDP). May I ask you a question 
about that? 

in 5.1, it says: 
  For sendrecv RTP 
  streams, the payload type numbers indicate the value of the payload 
  type field the offerer expects to receive, and would prefer to send. 
  However, for sendonly and sendrecv streams, the answer might indicate 
  different payload type numbers for the same codecs, in which case, 
  the offerer MUST send with the payload type numbers from the answer. 

in 6.1, it says: 
  In the case of RTP, if a particular codec was referenced with a 
  specific payload type number in the offer, that same payload type 
  number SHOULD be used for that codec in the answer.  Even if the same 
  payload type number is used, the answer MUST contain rtpmap 
  attributes to define the payload type mappings for dynamic payload 
  types, and SHOULD contain mappings for static payload types.  The 
  media formats in the "m=" line MUST be listed in order of preference, 
  with the first format listed being preferred.  In this case, 
  preferred means that the offerer SHOULD use the format with the 
  highest preference from the answer. 

My question is, the payload type number in answer may be different than in 
offer. Here is the case: 

1. Offer 
PCM-WB: 98 
ISAC: 101 
iLBC: 102 

2. Answer 
iLBC: 98 

This means both parties agreed to use iLBC. 
My question is , what payload type number should be use in constructing RTP 
pkts (in both direction). 

In sec5.1, it says offerer MUST respect the answer. It MUST put "98" in 
payload type number in RTP (offerer->answerer). 

But what about in answerer ?? should it respect the offer? which payload type 
number (102 or 98) it should put in RTP (answerer->offerer)?? 

I tried to read the rfc3264 for several times.. but still not quite clear. Do 
u have any idea about that? 

-- 
Kevin Kuei 


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From sip-bounces@ietf.org  Thu Feb 24 03:04:34 2005
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From: Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
Subject: RE: [Sip] rfc3264 question
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Hi,
 Here is my understanding on your offer-answer scenerio
Let us suppose two UA having said capabilities

UA-A
	          PT
PCM-WB:        98
ISAC:             101
iLBC:              102

UA-B		PT
	iLBC:    98

Note : All PT falls under dynamic payload range

Two possible cases for offer-answer

Case1: If offer is from UA-A to UA-B
	A---------------Offer1-----------------> B
	A <------------Answer1--------------B
Offer1:
PCM-WB: 98
ISAC: 101
iLBC: 102

Then Answer1 will be
iLBC: 102

Here UA-B will change the dynamic paylod type for iLBC to 102 from 98 and
use this in RTP packets.
Both UA-A and UA-B will use PT 102 in RTP Packets

Case2: If the offer is from UA-B to UA-A

 	A<-------------------------Offer2------------------B
	A-------------------------Answer2--------------->B

Offer2:
iLBC:    98

Then answer2 will be
iLBC:    98

Here UA-A will change the dynamic payload type for iLBC to 98 from 102 and
will use this PT 98 in RTP packets.
Both UA will use PT 98 in RTP Packets.

Hope it will help you.

Regards
Rajeew Kumar Singh

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of kkuei
Sent: Thursday, February 24, 2005 12:56 PM
To: Jonathan Rosenberg; sip@ietf.org
Subject: [Sip] rfc3264 question


I'm reading rfc3264 (offer/answer model with SDP). May I ask you a question
about that?

in 5.1, it says:
  For sendrecv RTP
  streams, the payload type numbers indicate the value of the payload
  type field the offerer expects to receive, and would prefer to send.
  However, for sendonly and sendrecv streams, the answer might indicate
  different payload type numbers for the same codecs, in which case,
  the offerer MUST send with the payload type numbers from the answer.

in 6.1, it says:
  In the case of RTP, if a particular codec was referenced with a
  specific payload type number in the offer, that same payload type
  number SHOULD be used for that codec in the answer.  Even if the same
  payload type number is used, the answer MUST contain rtpmap
  attributes to define the payload type mappings for dynamic payload
  types, and SHOULD contain mappings for static payload types.  The
  media formats in the "m=" line MUST be listed in order of preference,
  with the first format listed being preferred.  In this case,
  preferred means that the offerer SHOULD use the format with the
  highest preference from the answer.

My question is, the payload type number in answer may be different than in
offer. Here is the case:

1. Offer
PCM-WB: 98
ISAC: 101
iLBC: 102

2. Answer
iLBC: 98

This means both parties agreed to use iLBC.
My question is , what payload type number should be use in constructing RTP
pkts (in both direction).

In sec5.1, it says offerer MUST respect the answer. It MUST put "98" in
payload type number in RTP (offerer->answerer).

But what about in answerer ?? should it respect the offer? which payload
type
number (102 or 98) it should put in RTP (answerer->offerer)??

I tried to read the rfc3264 for several times.. but still not quite clear.
Do
u have any idea about that?

--
Kevin Kuei


_______________________________________________
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From sip-bounces@ietf.org  Thu Feb 24 06:19:36 2005
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Subject: [Sip] Wireless/WiFi Convergence - Paris - France
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The Wireless/WiFi Convergence conference will stand in Paris on November 29
to December 2, 2005.
A call for papers has been launched, effective until April 15, 2005. The
proposals submitted will be examined by a scientific committee. 

 

Get more details at:

 
<http://www.upperside.fr/wirelessconvergence2005/wwconvergence2005intro.htm>
http://www.upperside.fr/wirelessconvergence2005/wwconvergence2005intro.htm

 


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>The <strong><b><font face=3DArial><span
style=3D'font-family:Arial'>Wireless/WiFi =
Convergence</span></font></b></strong>
conference will stand in <st1:place w:st=3D"on"><st1:City =
w:st=3D"on"><span
  class=3Dstyle24>Paris</span></st1:City></st1:place><span =
class=3Dstyle24> on
November 29 to December 2, 2005.</span><br>
A call for papers has been launched, effective <u>until April 15, =
2005</u>. The
proposals submitted will be examined by a scientific committee. =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>Get more details =
at:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
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href=3D"http://www.upperside.fr/wirelessconvergence2005/wwconvergence2005=
intro.htm"
title=3D"http://www.upperside.fr/wirelessconvergence2005/wwconvergence200=
5intro.htm"><span
lang=3DEN-GB>http://www.upperside.fr/wirelessconvergence2005/wwconvergenc=
e2005intro.htm</span></a></span></font><span
class=3Dkeynote><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p></o:p></span></font></span></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
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_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip
--===============0788614701==--




From sip-bounces@ietf.org  Thu Feb 24 08:23:43 2005
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From: "Scott Godin" <slgodin@icescape.com>
To: "'Peter Hedman \(LD/EAB\)'" <peter.hedman@ericsson.com>, <sip@ietf.org>
Subject: RE: [SIP] SDP in OPTIONS
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My understanding is that if an SDP if provided in the OPTIONS request the
SDP in the 200 response should be the same as it would be if the UA was
responding to an INVITE request - of course the SDP exchange should not be
acted on - it is only used for capability discovery.

Scott

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Peter
Hedman (LD/EAB)
Sent: Wednesday, February 23, 2005 6:51 AM
To: sip@ietf.org
Subject: [SIP] SDP in OPTIONS

Hi,

3GPP are discussing capability exchange using SIP OPTIONS. Specifically, to
be able to optimise an exchange by including UAC capabilities within the
OPTIONS request, for information to those UAS that understand it. The
current understanding is that a UAS following RFC 3261 would simply ignore
the SDP in the OPTIONS request and normal operations take place. 

Also, even if the UAS would read the SDP in an OPTIONS request, the SDP in
the response would also be a capability document that will not in any way
follow the offer/answer model.

It would be great to get the groups opinion on the above assumptions.

BR,
Peter

Ericsson AB
Phone: +46 46 231760
Mobile: +46 703602149
Mail: peter.hedman@ericsson.com


_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Thu Feb 24 08:47:30 2005
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [Sip] a middle ground approach for knowing AOR of a gruu
References: <421D6367.8050703@cisco.com>
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I find this sub-optimal from a general design point of view. I thought 
we have had a general preference for being explicit about semantics, 
rather than overloading other elements or mechanisms. (See discussions 
on INFO and other similar topics.) I think many of the proposed labeling 
mechanisms would work and supporting at least one, even if somewhat 
arbitrarily selected, will get us to do the right thing in the majority 
of cases.

This is particularly important since this is not just a local matter, 
i.e., other parties not familiar with the conventions of the sending 
side will need to be able to robustly extract this information.

Jonathan Rosenberg wrote:
> We had a long discussion in another thread about whether or not we had 
> an additional requirement to allow the recipient of a GRUU to be able to 
> determine the AOR associated with that gruu. This seemed to be an issue 
> for systems that generate call logs or bills, since you'll have these 
> entries that are meaningless.
> 
> Though I didnt make any changes to gruu in order to meet this 
> requirement, there may be a simple approach that we can provide that 
> fits with the current gruu mechanism.
> 
> The spec is clear that there are no constraints on how a domain 
> constructs a gruu. The appendix includes a few sample algorithms (though 
> I'll note one of them is broken...). As such, there is nothing wrong 
> with a domain constructing a gruu by including the instance ID and AOR 
> in the clear in the user part of the URI:
> 
> sip:jdrosen.instance.00sd8ggdgdasd===asd9@cisco.com
> 
> for example. Thus, we could list such a technique as a viable approach, 
> and recommend that a particular means of encoding the URI be used in 
> order to allow other entities to extract the aor.
> 
> Thus, we would have a way for the recipient of a domain to get a hint 
> about the possible AOR (no guarantee though), without changing the 
> mechanism at all.
> 
> Would that represent a reasonable compromise?
> 
> Thanks,
> Jonathan R.


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From sip-bounces@ietf.org  Thu Feb 24 10:59:08 2005
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From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [SIP] SDP in OPTIONS
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At 08:20 AM 2/24/2005 -0500, Scott Godin wrote:
>My understanding is that if an SDP if provided in the OPTIONS request the
>SDP in the 200 response should be the same as it would be if the UA was
>responding to an INVITE request

I thought the SDP in the 200 OK could/would include all the session 
parameters the UAS was willing to reveal (including all the codecs the UAS 
was willing to communicate with).

>  - of course the SDP exchange should not be
>acted on - it is only used for capability discovery.
>
>Scott


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented

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From: Scott Godin <slgodin@icescape.com>
To: "'James M. Polk'" <jmpolk@cisco.com>, Scott Godin <slgodin@icescape.com>,
        "'Peter Hedman (LD/EAB)'" <peter.hedman@ericsson.com>, sip@ietf.org
Subject: RE: [SIP] SDP in OPTIONS
Date: Thu, 24 Feb 2005 12:12:53 -0500
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After re-reading Section 11 of RFC3261 - there doesn't really seem to be any
definitive answer to this.  The following quotes are what I based my
assumptions on:

"The response code chosen MUST be the same that would have been chosen had
the request
been an INVITE. That is, a 200 (OK) would be returned if the UAS is ready to
accept a call, a 486 (Busy
Here) would be returned if the UAS is busy, etc. This allows an OPTIONS
request to be used to determine
the basic state of a UAS, which can be an indication of whether the UAS will
accept an INVITE request."

Technically this quote refers to the response code - but it would seem to
make sense to apply to the other parts of the options response too.

"A message body MAY be sent, the type of which is determined by the Accept
header field in the OPTIONS
request (application/sdp is the default if the Accept header field is not
present). If the types include one that
can describe media capabilities, the UAS SHOULD include a body in the
response for that purpose. Details
on the construction of such a body in the case of application/sdp are
described in [12]."

This quote references RFC3264 (SDP Offer Answer) as a mechanism to form the
SDP response.

Scott

-----Original Message-----
From: James M. Polk [mailto:jmpolk@cisco.com] 
Sent: Thursday, February 24, 2005 10:56 AM
To: Scott Godin; 'Peter Hedman (LD/EAB)'; sip@ietf.org
Subject: RE: [SIP] SDP in OPTIONS

At 08:20 AM 2/24/2005 -0500, Scott Godin wrote:
>My understanding is that if an SDP if provided in the OPTIONS request the
>SDP in the 200 response should be the same as it would be if the UA was
>responding to an INVITE request

I thought the SDP in the 200 OK could/would include all the session 
parameters the UAS was willing to reveal (including all the codecs the UAS 
was willing to communicate with).

>  - of course the SDP exchange should not be
>acted on - it is only used for capability discovery.
>
>Scott


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented

_______________________________________________
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: "'Cullen Jennings'" <fluffy@cisco.com>, "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] draft-jennings-sip-hashcash-01
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We may need to explain some of the limitations of this. It seems to me that
this mechanism would not be terribly appropriate for applications that
aggregate a large number of clients onto a single platform. For example,
this may be too computationally intensive for a large PSTN Gateway, or even
a large Proxy.
 
Any ideas of how a UAS would make the decision to send the 419 without
burdening the UAC? Should there be a supported/require header for this
functionality, so that a UAC could "opt out"?

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Cullen
Jennings
Sent: Tuesday, February 22, 2005 16:46
To: sip@ietf.org
Subject: [Sip] draft-jennings-sip-hashcash-01



I updated this draft on "SIP Computational Puzzles" so it has enough detail
to implement it. (Previous version did not). It is in the drafts directory
and also in HTML form at:

http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcash
-01.html
<http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcas
h-01.html> 

Cullen



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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charset=3Dus-ascii">
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<META content=3D"MSHTML 6.00.2800.1491" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D309211819-24022005><FONT face=3DArial =
color=3D#800000 size=3D2>We may=20
need to explain some of the limitations of this. It seems to me that =
this=20
mechanism would not be terribly appropriate for applications that =
aggregate a=20
large number of clients onto a single platform. For example, this may =
be too=20
computationally intensive for a large PSTN Gateway, or even a large=20
Proxy.</FONT></SPAN></DIV>
<DIV><SPAN class=3D309211819-24022005><FONT face=3DArial =
color=3D#800000=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D309211819-24022005><FONT face=3DArial =
color=3D#800000 size=3D2>Any=20
ideas of how a UAS would make the decision to send the 419 without =
burdening the=20
UAC? Should there be a supported/require header for this functionality, =
so that=20
a UAC could "opt out"?</FONT></SPAN></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B>On Behalf Of =
</B>Cullen=20
  Jennings<BR><B>Sent:</B> Tuesday, February 22, 2005 =
16:46<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip]=20
  draft-jennings-sip-hashcash-01<BR><BR></FONT></DIV><FONT=20
  face=3D"Verdana, Helvetica, Arial"><SPAN style=3D"FONT-SIZE: =
12px"><BR>I updated=20
  this draft on "SIP Computational Puzzles" so it has enough detail to =
implement=20
  it. (Previous version did not). It is in the drafts directory and =
also in HTML=20
  form at:<BR><BR><A=20
  =
href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-=
sip-hashcash-01.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/d=
raft-jennings-sip-hashcash-01.html</A><BR><BR>Cullen<BR></BLOCKQUOTE></S=
PAN></FONT></BODY></HTML>

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_______________________________________________
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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
--===============0862487815==--



From sip-bounces@ietf.org  Thu Feb 24 15:20:48 2005
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To: sip@ietf.org
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Subject: [Sip] midcall signalling
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hello,
is there 100% agreement / concensus on how midcall signalling data is to be=
 carried in sip ?
when 2 SIP UA's are alrady in conversation, then if one user presses some k=
eys on his keypad, are they transmitted as DTMF on RTP, or how exactly is i=
t done ?
cheers,
bzie.
--=20
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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] draft-jennings-sip-hashcash-01
Date: Thu, 24 Feb 2005 15:53:09 -0500
To: "Francois Audet" <audet@nortel.com>
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On Feb 24, 2005, at 2:30 PM, Francois Audet wrote:

> We may need to explain some of the limitations of this. It seems to me =
=20
> that this mechanism would not be terribly appropriate for applications =
=20
> that aggregate a large number of clients onto a single platform. For =20=

> example, this may be too computationally intensive for a large PSTN =20=

> Gateway, or even a large Proxy.
> =A0
That's the whole point! It's to prevent too many calls from one place. =20=

It's always under the calling UACs prerogative to give up with "what a =20=

bozo who wants me to blow smoke out of my Pentium for 2000ms. Guess he =20=

doesn't want to get the call, huh"?

I suspect a decently managed UA would only force you not the puzzle =20
palace if you weren't on the buddy list, or some other whitelist.

> Any ideas of how a UAS would make the decision to send the 419 without =
=20
> burdening the UAC? Should there be a supported/require header for this =
=20
> functionality, so that a UAC could "opt out"?
If the UAC can opt out the whole exercise is useless.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =20=

> Cullen Jennings
> Sent: Tuesday, February 22, 2005 16:46
> To: sip@ietf.org
> Subject: [Sip] draft-jennings-sip-hashcash-01
>
>
> I updated this draft on "SIP Computational Puzzles" so it has enough =20=

> detail to implement it. (Previous version did not). It is in the =20
> drafts directory and also in HTML form at:
>
> http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-=20=

> hashcash-01.html
>
> Cullen
> _______________________________________________
> 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
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com

_______________________________________________
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From: Fatih "Eyüp" NAR <fenar@yahoo.com>
Subject: Re: [Sip] midcall signalling
To: Ban Zie <banzie@mail.com>, sip@ietf.org
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SIP DMTF signalling is done with INFO messages or with
RTP messages, the media protocol. This is defined in
http://www.ietf.org/rfc/rfc2833.txt - IETF RFC 2833 


--- Ban Zie <banzie@mail.com> wrote:

> hello,
> is there 100% agreement / concensus on how midcall
> signalling data is to be carried in sip ?
> when 2 SIP UA's are alrady in conversation, then if
> one user presses some keys on his keypad, are they
> transmitted as DTMF on RTP, or how exactly is it
> done ?
> cheers,
> bzie.
> -- 
>
___________________________________________________________
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> 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 Feb 24 18:25:14 2005
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Subject: RE: [Sip] a middle ground approach for knowing AOR of a gruu
Date: Thu, 24 Feb 2005 15:17:55 -0800
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Thread-Topic: [Sip] a middle ground approach for knowing AOR of a gruu
thread-index: AcUad8Uz10AAS/WLQFaid/FQGYwMbQATlwbg
From: "Orit Levin" <oritl@microsoft.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
        "Jonathan Rosenberg" <jdrosen@cisco.com>
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What would be wrong with having a GRUU always formatted as the user AOR
with an additional URL parameter carrying the value "making it a GRUU"
and understood by the user domain/registrar only?

Instead of:
sip:jdrosen.instance.00sd8ggdgdasd=3D=3D=3Dasd9@cisco.com

Make it:
<sip:jdrosen@cisco.com; instance=3D"00sd8ggdgdasd=3D=3D=3Dasd9">

Orit.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Henning Schulzrinne
> Sent: Thursday, February 24, 2005 5:44 AM
> To: Jonathan Rosenberg
> Cc: sip@ietf.org
> Subject: Re: [Sip] a middle ground approach for knowing AOR of a gruu
>=20
> I find this sub-optimal from a general design point of view.=20
> I thought we have had a general preference for being explicit=20
> about semantics, rather than overloading other elements or=20
> mechanisms. (See discussions on INFO and other similar=20
> topics.) I think many of the proposed labeling mechanisms=20
> would work and supporting at least one, even if somewhat=20
> arbitrarily selected, will get us to do the right thing in=20
> the majority of cases.
>=20
> This is particularly important since this is not just a local=20
> matter, i.e., other parties not familiar with the conventions=20
> of the sending side will need to be able to robustly extract=20
> this information.
>=20
> Jonathan Rosenberg wrote:
> > We had a long discussion in another thread about whether or=20
> not we had=20
> > an additional requirement to allow the recipient of a GRUU=20
> to be able=20
> > to determine the AOR associated with that gruu. This seemed=20
> to be an=20
> > issue for systems that generate call logs or bills, since=20
> you'll have=20
> > these entries that are meaningless.
> >=20
> > Though I didnt make any changes to gruu in order to meet this=20
> > requirement, there may be a simple approach that we can=20
> provide that=20
> > fits with the current gruu mechanism.
> >=20
> > The spec is clear that there are no constraints on how a domain=20
> > constructs a gruu. The appendix includes a few sample algorithms=20
> > (though I'll note one of them is broken...). As such, there=20
> is nothing=20
> > wrong with a domain constructing a gruu by including the=20
> instance ID=20
> > and AOR in the clear in the user part of the URI:
> >=20
> > sip:jdrosen.instance.00sd8ggdgdasd=3D=3D=3Dasd9@cisco.com
> >=20
> > for example. Thus, we could list such a technique as a viable=20
> > approach, and recommend that a particular means of encoding=20
> the URI be=20
> > used in order to allow other entities to extract the aor.
> >=20
> > Thus, we would have a way for the recipient of a domain to=20
> get a hint=20
> > about the possible AOR (no guarantee though), without changing the=20
> > mechanism at all.
> >=20
> > Would that represent a reasonable compromise?
> >=20
> > Thanks,
> > Jonathan R.
>=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: "Francois Audet" <audet@nortel.com>
To: "'David R Oran'" <oran@cisco.com>
Subject: RE: [Sip] draft-jennings-sip-hashcash-01
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Hi Dave,

I understand this. I guess I wasn't clear.

It is in the OTHER direction that I see a problem. I mean when the UAC
is, for example, a big gateway.

In other words, if a bunch of SIP "phones" UAS all decide to send 419's =
to
the UAC in the Gateway.


> -----Original Message-----
> From: David R Oran [mailto:oran@cisco.com]=20
> Sent: Thursday, February 24, 2005 12:53
> To: Audet, Francois [SC100:9T51:EXCH]
> Cc: 'Cullen Jennings'; 'sip@ietf.org'
> Subject: Re: [Sip] draft-jennings-sip-hashcash-01
>=20
>=20
>=20
> On Feb 24, 2005, at 2:30 PM, Francois Audet wrote:
>=20
> > We may need to explain some of the limitations of this. It=20
> seems to me
> > that this mechanism would not be terribly appropriate for=20
> applications =20
> > that aggregate a large number of clients onto a single=20
> platform. For =20
> > example, this may be too computationally intensive for a=20
> large PSTN =20
> > Gateway, or even a large Proxy.
> > =A0
> That's the whole point! It's to prevent too many calls from=20
> one place. =20
> It's always under the calling UACs prerogative to give up=20
> with "what a =20
> bozo who wants me to blow smoke out of my Pentium for 2000ms.=20
> Guess he =20
> doesn't want to get the call, huh"?
>=20
> I suspect a decently managed UA would only force you not the puzzle =20
> palace if you weren't on the buddy list, or some other whitelist.
>=20
> > Any ideas of how a UAS would make the decision to send the=20
> 419 without
> > burdening the UAC? Should there be a supported/require=20
> header for this =20
> > functionality, so that a UAC could "opt out"?
> If the UAC can opt out the whole exercise is useless.
>=20
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of
> > Cullen Jennings
> > Sent: Tuesday, February 22, 2005 16:46
> > To: sip@ietf.org
> > Subject: [Sip] draft-jennings-sip-hashcash-01
> >
> >
> > I updated this draft on "SIP Computational Puzzles" so it=20
> has enough =20
> > detail to implement it. (Previous version did not). It is in the =20
> > drafts directory and also in HTML form at:
> >
> >=20
> http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-=20
> > hashcash-01.html
> >
> > Cullen
> > _______________________________________________
> > 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
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
>=20
>=20

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<TITLE>RE: [Sip] draft-jennings-sip-hashcash-01</TITLE>
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<P><FONT SIZE=3D2>Hi Dave,</FONT>
</P>

<P><FONT SIZE=3D2>I understand this. I guess I wasn't clear.</FONT>
</P>

<P><FONT SIZE=3D2>It is in the OTHER direction that I see a problem. I =
mean when the UAC</FONT>
<BR><FONT SIZE=3D2>is, for example, a big gateway.</FONT>
</P>

<P><FONT SIZE=3D2>In other words, if a bunch of SIP &quot;phones&quot; =
UAS all decide to send 419's to</FONT>
<BR><FONT SIZE=3D2>the UAC in the Gateway.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: David R Oran [<A =
HREF=3D"mailto:oran@cisco.com">mailto:oran@cisco.com</A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Thursday, February 24, 2005 12:53</FONT>
<BR><FONT SIZE=3D2>&gt; To: Audet, Francois [SC100:9T51:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Cullen Jennings'; 'sip@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sip] =
draft-jennings-sip-hashcash-01</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; On Feb 24, 2005, at 2:30 PM, Francois Audet =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; We may need to explain some of the =
limitations of this. It </FONT>
<BR><FONT SIZE=3D2>&gt; seems to me</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that this mechanism would not be terribly =
appropriate for </FONT>
<BR><FONT SIZE=3D2>&gt; applications&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that aggregate a large number of clients =
onto a single </FONT>
<BR><FONT SIZE=3D2>&gt; platform. For&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; example, this may be too computationally =
intensive for a </FONT>
<BR><FONT SIZE=3D2>&gt; large PSTN&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Gateway, or even a large Proxy.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =A0</FONT>
<BR><FONT SIZE=3D2>&gt; That's the whole point! It's to prevent too =
many calls from </FONT>
<BR><FONT SIZE=3D2>&gt; one place.&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; It's always under the calling UACs prerogative =
to give up </FONT>
<BR><FONT SIZE=3D2>&gt; with &quot;what a&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; bozo who wants me to blow smoke out of my =
Pentium for 2000ms. </FONT>
<BR><FONT SIZE=3D2>&gt; Guess he&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; doesn't want to get the call, huh&quot;?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I suspect a decently managed UA would only =
force you not the puzzle&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; palace if you weren't on the buddy list, or =
some other whitelist.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Any ideas of how a UAS would make the =
decision to send the </FONT>
<BR><FONT SIZE=3D2>&gt; 419 without</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; burdening the UAC? Should there be a =
supported/require </FONT>
<BR><FONT SIZE=3D2>&gt; header for this&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; functionality, so that a UAC could =
&quot;opt out&quot;?</FONT>
<BR><FONT SIZE=3D2>&gt; If the UAC can opt out the whole exercise is =
useless.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On </FONT>
<BR><FONT SIZE=3D2>&gt; Behalf Of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cullen Jennings</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sent: Tuesday, February 22, 2005 =
16:46</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Subject: [Sip] =
draft-jennings-sip-hashcash-01</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I updated this draft on &quot;SIP =
Computational Puzzles&quot; so it </FONT>
<BR><FONT SIZE=3D2>&gt; has enough&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; detail to implement it. (Previous version =
did not). It is in the&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; drafts directory and also in HTML form =
at:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-=
sip-" =
TARGET=3D"_blank">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft=
-jennings-sip-</A> </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; hashcash-01.html</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cullen</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; David R. Oran</FONT>
<BR><FONT SIZE=3D2>&gt; Cisco Fellow</FONT>
<BR><FONT SIZE=3D2>&gt; Cisco Systems</FONT>
<BR><FONT SIZE=3D2>&gt; 7 Ladyslipper Lane</FONT>
<BR><FONT SIZE=3D2>&gt; Acton, MA 01720 USA</FONT>
<BR><FONT SIZE=3D2>&gt; Tel: +1 978 264 2048</FONT>
<BR><FONT SIZE=3D2>&gt; Email: oran@cisco.com</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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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
--===============1234877586==--



From sip-bounces@ietf.org  Fri Feb 25 00:06:44 2005
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From: "Ban Zie" <banzie@mail.com>
To: "Louis Alexander" <loualexa@cmguc.com>,
        "'Fatih Ey|p NAR'" <fenar@yahoo.com>, sip@ietf.org
Date: Fri, 25 Feb 2005 00:03:24 -0500
Subject: RE: [Sip] midcall signalling
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Thank you for your responses Fatih & Louis, they are very helpful.

Is there a "common minimum" set for SIP phones on which mechanism(s) to use=
 for midcall signalling ?
i.e. the recipient of such signalling (which in my case would be a minimali=
stic IVR / B2BUA), should
be prepared to receive such signalling in all these forms, or is it suffici=
ent to pick one -- say
the inband DTMF ? I do not have access to a SIP hard-phone (yet! but should=
 be getting one soon), so
do the sip phones s.a. CISCO's, PingTel's, SNOM's have some configuration t=
o select which kind of
midcall signalling they'd send ?

cheers,
zie.

----- Original Message -----
From: "Louis Alexander" <loualexa@cmguc.com>
To: "'Fatih Ey|p NAR'" <fenar@yahoo.com>, "'Ban Zie'" <banzie@mail.com>, si=
p@ietf.org
Subject: RE: [Sip] midcall signalling
Date: Thu, 24 Feb 2005 17:00:32 -0500

>=20
> Of course, plain old inband dtmf (the DTMF tones encoded as audio packets,
> NOT 2833 tones) is still legal as well :)
>=20
> --Louis
>=20
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Fat=
ih
> Ey|p NAR
> Sent: Thursday, February 24, 2005 4:45 PM
> To: Ban Zie; sip@ietf.org
> Subject: Re: [Sip] midcall signalling
>=20
> SIP DMTF signalling is done with INFO messages or with
> RTP messages, the media protocol. This is defined in
> http://www.ietf.org/rfc/rfc2833.txt - IETF RFC 2833
>=20
>=20
> --- Ban Zie <banzie@mail.com> wrote:
>=20
> > hello,
> > is there 100% agreement / concensus on how midcall
> > signalling data is to be carried in sip ?
> > when 2 SIP UA's are alrady in conversation, then if
> > one user presses some keys on his keypad, are they
> > transmitted as DTMF on RTP, or how exactly is it
> > done ?
> > cheers,
> > bzie.
> > --
> ___________________________________________________________
> > Sign-up for Ads Free at Mail.com
> > http://promo.mail.com/adsfreejump.htm
> >
> >
> > _______________________________________________
> > 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
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.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
> ______________________________________________________________________
> This email has been scanned by the MessageLabs Email Security System.
> For more information please visit http://www.messagelabs.com/email
> ______________________________________________________________________

--=20
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Fri Feb 25 09:06:51 2005
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Date: Fri, 25 Feb 2005 11:01:59 -0300
From: Mariano Stokle <mstokle@gmail.com>
To: Ban Zie <banzie@mail.com>
Subject: Re: [Sip] midcall signalling
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You should also check
http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt


On Fri, 25 Feb 2005 00:03:24 -0500, Ban Zie <banzie@mail.com> wrote:
> Thank you for your responses Fatih & Louis, they are very helpful.
> 
> Is there a "common minimum" set for SIP phones on which mechanism(s) to use for midcall signalling ?
> i.e. the recipient of such signalling (which in my case would be a minimalistic IVR / B2BUA), should
> be prepared to receive such signalling in all these forms, or is it sufficient to pick one -- say
> the inband DTMF ? I do not have access to a SIP hard-phone (yet! but should be getting one soon), so
> do the sip phones s.a. CISCO's, PingTel's, SNOM's have some configuration to select which kind of
> midcall signalling they'd send ?
> 
> cheers,
> zie.
> 
> ----- Original Message -----
> From: "Louis Alexander" <loualexa@cmguc.com>
> To: "'Fatih Ey|p NAR'" <fenar@yahoo.com>, "'Ban Zie'" <banzie@mail.com>, sip@ietf.org
> Subject: RE: [Sip] midcall signalling
> Date: Thu, 24 Feb 2005 17:00:32 -0500
> 
> >
> > Of course, plain old inband dtmf (the DTMF tones encoded as audio packets,
> > NOT 2833 tones) is still legal as well :)
> >
> > --Louis
> >
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Fatih
> > Ey|p NAR
> > Sent: Thursday, February 24, 2005 4:45 PM
> > To: Ban Zie; sip@ietf.org
> > Subject: Re: [Sip] midcall signalling
> >
> > SIP DMTF signalling is done with INFO messages or with
> > RTP messages, the media protocol. This is defined in
> > http://www.ietf.org/rfc/rfc2833.txt - IETF RFC 2833
> >
> >
> > --- Ban Zie <banzie@mail.com> wrote:
> >
> > > hello,
> > > is there 100% agreement / concensus on how midcall
> > > signalling data is to be carried in sip ?
> > > when 2 SIP UA's are alrady in conversation, then if
> > > one user presses some keys on his keypad, are they
> > > transmitted as DTMF on RTP, or how exactly is it
> > > done ?
> > > cheers,
> > > bzie.
> > > --
> > ___________________________________________________________
> > > Sign-up for Ads Free at Mail.com
> > > http://promo.mail.com/adsfreejump.htm
> > >
> > >
> > > _______________________________________________
> > > 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
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.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
> >
> > ______________________________________________________________________
> > This email has been scanned by the MessageLabs Email Security System.
> > For more information please visit http://www.messagelabs.com/email
> > ______________________________________________________________________
> 
> --
> ___________________________________________________________
> Sign-up for Ads Free at Mail.com
> http://promo.mail.com/adsfreejump.htm
> 
> _______________________________________________
> 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
> 


-- 
Mariano Stokle
SIP Rules

_______________________________________________
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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: "Ban Zie" <banzie@mail.com>
To: "Mariano Stokle" <mstokle@gmail.com>
Date: Fri, 25 Feb 2005 09:25:26 -0500
Subject: Re: [Sip] midcall signalling
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Thank you, all.

After researching the features of various commercial & free sip phones (har=
d & soft), I've reached the conclusion that rfc2833 is the most common appr=
oach, and almost every phone (actually i didn't find an exception so far) s=
eems to implement it.

cheers,
z.
----- Original Message -----
From: "Mariano Stokle" <mstokle@gmail.com>
To: "Ban Zie" <banzie@mail.com>
Subject: Re: [Sip] midcall signalling
Date: Fri, 25 Feb 2005 11:01:59 -0300

>=20
> You should also check
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt
>=20
>=20
> On Fri, 25 Feb 2005 00:03:24 -0500, Ban Zie <banzie@mail.com> wrote:
> > Thank you for your responses Fatih & Louis, they are very helpful.
> >
> > Is there a "common minimum" set for SIP phones on which=20
> > mechanism(s) to use for midcall signalling ?
> > i.e. the recipient of such signalling (which in my case would be=20
> > a minimalistic IVR / B2BUA), should
> > be prepared to receive such signalling in all these forms, or is=20
> > it sufficient to pick one -- say
> > the inband DTMF ? I do not have access to a SIP hard-phone (yet!=20
> > but should be getting one soon), so
> > do the sip phones s.a. CISCO's, PingTel's, SNOM's have some=20
> > configuration to select which kind of
> > midcall signalling they'd send ?
> >
> > cheers,
> > zie.
> >
> > ----- Original Message -----
> > From: "Louis Alexander" <loualexa@cmguc.com>
> > To: "'Fatih Ey|p NAR'" <fenar@yahoo.com>, "'Ban Zie'"=20
> > <banzie@mail.com>, sip@ietf.org
> > Subject: RE: [Sip] midcall signalling
> > Date: Thu, 24 Feb 2005 17:00:32 -0500
> >
> > >
> > > Of course, plain old inband dtmf (the DTMF tones encoded as audio pac=
kets,
> > > NOT 2833 tones) is still legal as well :)
> > >
> > > --Louis
> > >
> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of=
 Fatih
> > > Ey|p NAR
> > > Sent: Thursday, February 24, 2005 4:45 PM
> > > To: Ban Zie; sip@ietf.org
> > > Subject: Re: [Sip] midcall signalling
> > >
> > > SIP DMTF signalling is done with INFO messages or with
> > > RTP messages, the media protocol. This is defined in
> > > http://www.ietf.org/rfc/rfc2833.txt - IETF RFC 2833
> > >
> > >
> > > --- Ban Zie <banzie@mail.com> wrote:
> > >
> > > > hello,
> > > > is there 100% agreement / concensus on how midcall
> > > > signalling data is to be carried in sip ?
> > > > when 2 SIP UA's are alrady in conversation, then if
> > > > one user presses some keys on his keypad, are they
> > > > transmitted as DTMF on RTP, or how exactly is it
> > > > done ?
> > > > cheers,
> > > > bzie.
> > > > --
> > > ___________________________________________________________
> > > > Sign-up for Ads Free at Mail.com
> > > > http://promo.mail.com/adsfreejump.htm
> > > >
> > > >
> > > > _______________________________________________
> > > > 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
> > > >
> > >
> > >
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam protection around
> > > http://mail.yahoo.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
> > >
> > > ______________________________________________________________________
> > > This email has been scanned by the MessageLabs Email Security System.
> > > For more information please visit http://www.messagelabs.com/email
> > > ______________________________________________________________________
> >
> > --
> > ___________________________________________________________
> > Sign-up for Ads Free at Mail.com
> > http://promo.mail.com/adsfreejump.htm
> >
> > _______________________________________________
> > 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
> --
> Mariano Stokle
> SIP Rules

--=20
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From sip-bounces@ietf.org  Fri Feb 25 09:41:07 2005
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From: "Ban Zie" <banzie@mail.com>
To: "Mariano Stokle" <mstokle@gmail.com>
Date: Fri, 25 Feb 2005 09:39:46 -0500
Subject: Re: [Sip] midcall signalling
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Hello Mariano:

> You should also check
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt

I quickly skimmed thru this I-D, and it appears to me that this is more lik=
ely to be associated as a Event delivery medium between and IVR / G/w / Tra=
nscoder and a SIP AppServer. In my scenario, i.e. UA <----> UA, this doesn'=
t seem reasonable thing to do. Here the second-UA is like a Telephonic cred=
it-card payment gateway.

-z
--=20
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Date: Fri, 25 Feb 2005 08:03:28 -0800
Subject: Re: [Sip] draft-jennings-sip-hashcash-01
From: Cullen Jennings <fluffy@cisco.com>
To: Francois Audet <audet@nortel.com>, "sip@ietf.org" <sip@ietf.org>
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The idea would be to have a very skewed amount of computation to help rate
limit attacks. Whatever amount of computation is "reasonable" for a valid
client to do, an attacker is going to be able to apply a lot more
computation to the problem. However, this still limits the number of
messages the attacker can do and thus raises the cost. I discuss this some
what in the SPAM draft.

This could all happen in different ways. Imagine a case where my enterprise
proxy compute the puzzles on behalf of my phone when I call another domain.
Say in a large enterprise, it has 10,000 people and during a busy hour makes
10 calls per second that are to an address external to the domain. Of these
only 2 are not whitelists. If the enterprise proxy was willing to spend say
3GHz worth of CPU at computing challenges for these 2cps, well that is a
pretty hefty amount of hash that could be required.

Another ways is perhaps all the UAC are willing to do about the amount of
hashing that 1 second of G.729 compression would take. It's hard to imagine
that most UAs (even large GWs) could not do this. This would limit the rate
of attack of a single computer to in the 100s of messages per second. This
is a lot less hashing that the example above but contrast this to what is
possible today. Today I can send in the order of 100,000 message per second.
Increasing the cost of an attack by 1000 is good. It slows it down and give
you time to trace the source and cut it off before it is everywhere. It
makes is less worth while to advertise products that a very very small
percentage of the receivers would buy.

As discussed in the SPAM draft, you would always want to use white lists
first then if the person was not on the white list fallback to something
else like this. 

Cullen


On 2/24/05 11:30 AM, "Francois Audet" <audet@nortel.com> wrote:

> We may need to explain some of the limitations of this. It seems to me that
> this mechanism would not be terribly appropriate for applications that
> aggregate a large number of clients onto a single platform. For example, this
> may be too computationally intensive for a large PSTN Gateway, or even a large
> Proxy.
>  
> Any ideas of how a UAS would make the decision to send the 419 without
> burdening the UAC? Should there be a supported/require header for this
> functionality, so that a UAC could "opt out"?
>>  
>>  
>> -----Original Message-----
>> From:  sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Cullen
>> Jennings
>> Sent: Tuesday, February 22, 2005 16:46
>> To:  sip@ietf.org
>> Subject: [Sip]  draft-jennings-sip-hashcash-01
>> 
>> 
>> I updated  this draft on "SIP Computational Puzzles" so it has enough detail
>> to implement  it. (Previous version did not). It is in the drafts directory
>> and also in HTML  form at:
>> 
>> http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcash-
>> 01.html
>> 
>> Cullen
> 
> 
> _______________________________________________
> 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



--B_3192163409_5968357
Content-type: text/html;
	charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [Sip] draft-jennings-sip-hashcash-01</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:12.0px'><BR>
The idea would be to have a very skewed amount of computation to help rate =
limit attacks. Whatever amount of computation is &quot;reasonable&quot; for =
a valid client to do, an attacker is going to be able to apply a lot more co=
mputation to the problem. However, this still limits the number of messages =
the attacker can do and thus raises the cost. I discuss this some what in th=
e SPAM draft. <BR>
<BR>
This could all happen in different ways. Imagine a case where my enterprise=
 proxy compute the puzzles on behalf of my phone when I call another domain.=
 Say in a large enterprise, it has 10,000 people and during a busy hour make=
s 10 calls per second that are to an address external to the domain. Of thes=
e only 2 are not whitelists. If the enterprise proxy was willing to spend sa=
y 3GHz worth of CPU at computing challenges for these 2cps, well that is a p=
retty hefty amount of hash that could be required. <BR>
<BR>
Another ways is perhaps all the UAC are willing to do about the amount of h=
ashing that 1 second of G.729 compression would take. It's hard to imagine t=
hat most UAs (even large GWs) could not do this. This would limit the rate o=
f attack of a single computer to in the 100s of messages per second. This is=
 a lot less hashing that the example above but contrast this to what is poss=
ible today. Today I can send in the order of 100,000 message per second. Inc=
reasing the cost of an attack by 1000 is good. It slows it down and give you=
 time to trace the source and cut it off before it is everywhere. It makes i=
s less worth while to advertise products that a very very small percentage o=
f the receivers would buy. <BR>
<BR>
As discussed in the SPAM draft, you would always want to use white lists fi=
rst then if the person was not on the white list fallback to something else =
like this. <BR>
<BR>
Cullen<BR>
<BR>
<BR>
On 2/24/05 11:30 AM, &quot;Francois Audet&quot; &lt;audet@nortel.com&gt; wr=
ote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:12.0px'><FONT COLOR=3D"#8000=
00"><FONT FACE=3D"Arial">We may need to explain some of the limitations of thi=
s. It seems to me that this mechanism would not be terribly appropriate for =
applications that aggregate a large number of clients onto a single platform=
. For example, this may be too computationally intensive for a large PSTN Ga=
teway, or even a large Proxy.<BR>
</FONT></FONT><FONT FACE=3D"Verdana, Helvetica, Arial"> <BR>
</FONT><FONT COLOR=3D"#800000"><FONT FACE=3D"Arial">Any ideas of how a UAS woul=
d make the decision to send the 419 without burdening the UAC? Should there =
be a supported/require header for this functionality, so that a UAC could &q=
uot;opt out&quot;?<BR>
</FONT></FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:12.0px'><FONT FACE=3D=
"Verdana, Helvetica, Arial"> <BR>
&nbsp;<BR>
</FONT><FONT FACE=3D"Tahoma">-----Original Message-----<BR>
<B>From:</B> &nbsp;sip-bounces@ietf.org [<a href=3D"mailto:sip-bounces@ietf.o=
rg]">mailto:sip-bounces@ietf.org]</a> <B>On Behalf Of </B>Cullen &nbsp;Jenni=
ngs<BR>
<B>Sent:</B> Tuesday, February 22, 2005 16:46<BR>
<B>To:</B> &nbsp;sip@ietf.org<BR>
<B>Subject:</B> [Sip] &nbsp;draft-jennings-sip-hashcash-01<BR>
<BR>
</FONT><FONT FACE=3D"Verdana, Helvetica, Arial"><BR>
I updated &nbsp;this draft on &quot;SIP Computational Puzzles&quot; so it h=
as enough detail to implement &nbsp;it. (Previous version did not). It is in=
 the drafts directory and also in HTML &nbsp;form at:<BR>
<BR>
<a href=3D"http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-si=
p-hashcash-01.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-j=
ennings-sip-hashcash-01.html</a><BR>
<BR>
Cullen<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:12.0px'><FONT FACE=3D"Verda=
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From sip-bounces@ietf.org  Fri Feb 25 12:56:21 2005
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Subject: [Sip] New Resource Priority Draft (-06)
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All

Posted version -06 of the Resource Priority Header ID

http://www.ietf.org/internet-drafts/draft-ietf-sip-resource-priority-06.txt

The ID doesn't have a "changes" section, so here are the changes since the 
-05 version:

- removed all the modes, and nearly all the text from all the modes (yes!)

- introduce and describe what we mean generically by a preferential 
treatment that is priority-queue based (section 4.4.2) for GETS-type 
services (instead of only explaining a preemption policy)

- added section specifically how a UAC behaves in preemption-based 
administrative domains (4.7.1)

- added section specifically how a UAC behaves in priority-queue based 
administrative domains (4.7.2)

- added section specifically how a UAS behaves in priority-queue based 
administrative domains (4.8.2)

- detailing how a congested queue should function within a proxy of both 
behavior types (4.9)

- deleted the old section on Proxies and preemption policy

- Expanded the sections on the individual behaviors of each of the 5 
namespaces created with this document (8.1.1 - 8.1.5)

- kept, but rewrote section 7 (Multiple Namespaces in a Message) to explain 
that the relative priority level order within a namespace must be 
maintained when interleaving one namespace with another in determining 
message treatment within a SIP element.

- Added a section on Future Namespace Creation (8.2)

- rewrote IANA Registrations of Namespaces and Priority-Values, with a new 
(simpler) table (11.4)

- added IANA Registration section on how to register the relative priority 
order of values within a namespace definition (11.5)

- made [I-D.ietf-sipping-reason-header-for-preemption] a normative 
reference (13.1)

- all this while streamlining (cutting and rewriting) the document

comments are welcome



cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented

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From: Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] draft-jennings-sip-hashcash-01
Date: Fri, 25 Feb 2005 17:31:17 -0800
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On Feb 25, 2005, at 8:03 AM, Cullen Jennings wrote:

>
>  The idea would be to have a very skewed amount of computation to help =
=20
> rate limit attacks. Whatever amount of computation is "reasonable" for =
=20
> a valid client to do, an attacker is going to be able to apply a lot =20=

> more computation to the problem. However, this still limits the number =
=20
> of messages the attacker can do and thus raises the cost. I discuss =20=

> this some what in the SPAM draft.
>
>  This could all happen in different ways. Imagine a case where my =20
> enterprise proxy compute the puzzles on behalf of my phone when I call =
=20
> another domain. Say in a large enterprise, it has 10,000 people and =20=

> during a busy hour makes 10 calls per second that are to an address =20=

> external to the domain. Of these only 2 are not whitelists. If the =20
> enterprise proxy was willing to spend say 3GHz worth of CPU at =20
> computing challenges for these 2cps, well that is a pretty hefty =20
> amount of hash that could be required.
>
> Another ways is perhaps all the UAC are willing to do about the amount =
=20
> of hashing that 1 second of G.729 compression would take. It's hard to =
=20
> imagine that most UAs (even large GWs) could not do this. This would =20=

> limit the rate of attack of a single computer to in the 100s of =20
> messages per second. This is a lot less hashing that the example above =
=20
> but contrast this to what is possible today. Today I can send in the =20=

> order of 100,000 message per second. Increasing the cost of an attack =20=

> by 1000 is good. It slows it down and give you time to trace the =20
> source and cut it off before it is everywhere. It makes is less worth =20=

> while to advertise products that a very very small percentage of the =20=

> receivers would buy.
>
>  As discussed in the SPAM draft, you would always want to use white =20=

> lists first then if the person was not on the white list fallback to =20=

> something else like this.

Unstated in either document is the assumption that every legitimate =20
caller and every spammer has, at their disposal, one and only one =20
machine that has the CPU capacity of a modern off-the-shelf PC -- "1 =20
Pentium Unit".  If a legitimate user has less, or a spammer has more, I =20=

don't see hashcash being very valuable.

If a legitimate caller has less than 1 Pentium Unit, it seems that =20
person is SOL -- they either upgrade their UA, upgrade their SIP proxy =20=

to solve the puzzle for them, buy puzzle-solving MIPS from someone =20
else, or simply not complete a call to someone with a difficult =20
challenge?

I do agree that hashcash makes spamming more expensive, but =20
significantly more expensive.  Today, email spammers own botnets which =20=

are used to send email spam.  Tomorrow, SIP spammers can modify those =20=

same botnets to solve hashcash puzzles _and_ send (SIP or email) spam =20=

-- afterall, it isn't computationally-expensive to send email spam =20
(it's mostly bandwidth), so each PC can do a computational puzzle in =20
addition to its normal spam duties.  And the PCs that a spammer 'owns', =20=

which can't send spam due to a clueful ISP, can still solve hashcash =20
puzzles on behalf of a SIP spammer.

And ven without a botnet of PCs, a spammer can buy a workstation with =20=

substantially more than 1 Pentium Unit.  And I expect spammers will =20
write software running on off-the-shelf speciality hardware, such as =20
graphics accelerators, to solve hashcash puzzles faster than 1 Pentium =20=

Unit.

-d


>  Cullen
>
>
>  On 2/24/05 11:30 AM, "Francois Audet" <audet@nortel.com> wrote:
>
>
> We may need to explain some of the limitations of this. It seems to me =
=20
> that this mechanism would not be terribly appropriate for applications =
=20
> that aggregate a large number of clients onto a single platform. For =20=

> example, this may be too computationally intensive for a large PSTN =20=

> Gateway, or even a large Proxy.
>
> Any ideas of how a UAS would make the decision to send the 419 without =
=20
> burdening the UAC? Should there be a supported/require header for this =
=20
> functionality, so that a UAC could "opt out"?
>
>
>  =A0
> -----Original Message-----
> From: =A0sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf =
Of =20
> Cullen =A0Jennings
> Sent: Tuesday, February 22, 2005 16:46
> To: =A0sip@ietf.org
> Subject: [Sip] =A0draft-jennings-sip-hashcash-01
>
>
>  I updated =A0this draft on "SIP Computational Puzzles" so it has =
enough =20
> detail to implement =A0it. (Previous version did not). It is in the =20=

> drafts directory and also in HTML =A0form at:
>
> http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-=20=

> hashcash-01.html
>
>  Cullen
>
>
>
> _______________________________________________
>  Sip mailing list =A0https://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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_______________________________________________
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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_______________________________________________
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
----------agnfvesrvgwxsqwfgizk--




From sip-bounces@ietf.org  Sun Feb 27 13:51:38 2005
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Message-ID: <422217B4.2050502@softarmor.com>
Date: Sun, 27 Feb 2005 12:55:48 -0600
From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Internet Draft Format Checking Tool -- idnits
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Wonder if your I-D "has all the right stuff?"

Well, I don't know an easy way to tell you that.


But the idnits tool, available at:

http://ietf.levkowetz.com/tools/idnits/


may tell you whether your draft is MISSING some of the right stuff. It 
would probably be a good idea to start giving all drafts a whack with 
this before sending them in.

This would help to have prevented a couple of SIP drafts that were 
recenly rejected by the internet-drafts reviewer from being rejected. We 
know those folks are busy, so let's try and help them out in the future.


Thanks,

--
dean

_______________________________________________
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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  Mon Feb 28 10:01:42 2005
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To: sip-implementors@cs.columbia.edu, SIP WG <sip@ietf.org>
From: Robert Sparks <rjsparks@nostrum.com>
Date: Mon, 28 Feb 2005 09:00:54 -0600
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Subject: [Sip] SIPIT 16 registration deadline
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The special rates for rooms at SIPIT 16 expire March 4th (this Friday).
Please book your rooms before then. If you plan to attend and
have not yet registered for the event itself, please do so today.
Information is available at http://www.sipit.net and 
http://sipit16.jasomi.com

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


From sip-bounces@ietf.org  Mon Feb 28 11:00:54 2005
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From: Dean Willis <dean.willis@softarmor.com>
Date: Mon, 28 Feb 2005 09:55:54 -0600
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Cc: Rohan Mahy <rohan@ekabal.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Subject: [Sip] Agenda for SIP at IETF 62
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Here's the current agenda for the SIP working group meeting. Note that 
we're moving a couple of things (certs, conection management, e2m) from 
the from the draft SIPPING agenda to SIP.

A pretty hyperlinked version is available at:

http://www.softarmor.com/sipwg/meets/ietf62/agenda.html



Agenda, SIP WG, IETF 62

Monday, 1930-2200
1930 Agenda Bash and Status

1945 GRUU Remaining Issues (Knowing AOR, Rewriting by SBC, etc.)
      Jonathan Rosenberg
      draft-ietf-sip-gruu-03.txt

2015 Non-Invite Transactions
     Robert Sparks
     draft-sparks-sip-nit-future-01.txt

2030 REFER Extensions
     Orit Levin
     draft-ietf-sip-refer-with-norefersub-01.txt
     draft-ietf-sip-refer-with-feature-param-01.txt

2045 Identity -- Request
      Jon Peterson
      draft-ietf-sip-identity-04.txt

2100 Identity -- Response
     Jon Peterson
     draft-peterson-message-identity-00.txt

2115 SIP Resource Priority
     James Polk
     draft-ietf-sip-resource-priority-06.txt

2200 Close


Tuesday, 1300-1500
1300 Agenda Bash

1310 SIP Interaction Framework
     Jonathan Rosenberg
     draft-ietf-sipping-app-interaction-framework-04.txt

1330 Connection Reuse
      Rohan Mahy
      draft-ietf-sip-connect-reuse-03

1350 Management of Outbound Connections
     Cullen Jennings
     draft-jennings-sipping-outbound-01.txt

1405 Using Certificates with SIP
     Cullen Jennings
     draft-ietf-sipping-certs-01.txt

1420 End-to-middle security
     Kumiko Ono
     draft-ono-sipping-end2middle-security-04.txt

1440 Extension Negotiation
     Volker Hilt
     draft-hilt-sip-ext-neg-00.txt

1500 Close



_______________________________________________
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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  Mon Feb 28 12:11:33 2005
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From: Idnani Ajaykumar-AIDNANI1 <Ajaykumar.Idnani@motorola.com>
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Hi All,
 
I was reading section 17.1.1.2 of RFC 3261 and I had some basic questions, so I thought I will ask the experts.
 
1. What is the UAC supposed to do when it is in the Proceeding state and it does not receive any final response? The text in section 17.1.1.2 does talk about this case. Also when I look at the flow in Firgure 5, there seems to be no timer attached to get out of the 'Proceeding' state. There is one attached to 'Calling' (Timer B) and 'Completed' (Timer D) states, but none to 'Proceeding' state. So are we saying that once you get into Proceeding state and the other end fails for some reason, the UAC is stuck.
 
Or is the figure inaccurate and it should really show a Timer B associated with one of the arrows coming out of that state? In which case my questions would be does a provisional response reset this timer? Otherwise an INVITE transaction would be constrained to 32secs assuming a T1 of 500ms (i.e. a call should go from 'Calling' to 'Completed' in 32secs!
 
Appreciate your responses.
 
Thank you.
- Ajay  

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From: "Francois Audet" <audet@nortel.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>, "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] draft-jennings-sip-hashcash-01
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Yeah, a white list approach might work. 
 
The practical scenario that I am thinking about is with a typical VoIP
service provider.
 
Today, the vast majority of calls from subscribers terminate on gateways
(owned by the operator of the service).  These calls are not to a specific
"user" but really to a gateway, using presumably something like the
tel-URI-in-SIP-URI approach. In this case then, the white list is not to a
whole URI but to "something" that is related to the gateway itself. This is
different from the IM-scenario you quote. 
 
Now, if we only have to worry about one set of gateways (like the ones owned
by the service provider), I guess you could base the white list on the
domain name or find some other solution. It may be more difficult when
gateways are scattered all over the place (like in an enterprise for
example, or a federation of enterprise, or when it terminates to another
service provider). 
 
I believe we should make sure that we describe white list properly before we
publish the puzzle mechanism. I have a fear that we could define something
where the people who make the SIP end-devices have an incentive to do
something that is good for them (i.e., limit vulnerability to spam) but
could be very problematic for the Gateways (or Proxies). Somehow we have to
be able to mandate a white list capabily (or some other mechanism).
Otherwise software upgrades could kill the operator of the service.
 
PS: Note that this is a recurring problem with gateways (I think certs has
the same issue). 

-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com] 
Sent: Friday, February 25, 2005 08:03
To: Audet, Francois [SC100:9T51:EXCH]; sip@ietf.org
Subject: Re: [Sip] draft-jennings-sip-hashcash-01



The idea would be to have a very skewed amount of computation to help rate
limit attacks. Whatever amount of computation is "reasonable" for a valid
client to do, an attacker is going to be able to apply a lot more
computation to the problem. However, this still limits the number of
messages the attacker can do and thus raises the cost. I discuss this some
what in the SPAM draft. 

This could all happen in different ways. Imagine a case where my enterprise
proxy compute the puzzles on behalf of my phone when I call another domain.
Say in a large enterprise, it has 10,000 people and during a busy hour makes
10 calls per second that are to an address external to the domain. Of these
only 2 are not whitelists. If the enterprise proxy was willing to spend say
3GHz worth of CPU at computing challenges for these 2cps, well that is a
pretty hefty amount of hash that could be required. 

Another ways is perhaps all the UAC are willing to do about the amount of
hashing that 1 second of G.729 compression would take. It's hard to imagine
that most UAs (even large GWs) could not do this. This would limit the rate
of attack of a single computer to in the 100s of messages per second. This
is a lot less hashing that the example above but contrast this to what is
possible today. Today I can send in the order of 100,000 message per second.
Increasing the cost of an attack by 1000 is good. It slows it down and give
you time to trace the source and cut it off before it is everywhere. It
makes is less worth while to advertise products that a very very small
percentage of the receivers would buy. 

As discussed in the SPAM draft, you would always want to use white lists
first then if the person was not on the white list fallback to something
else like this. 

Cullen


On 2/24/05 11:30 AM, "Francois Audet" <audet@nortel.com> wrote:



We may need to explain some of the limitations of this. It seems to me that
this mechanism would not be terribly appropriate for applications that
aggregate a large number of clients onto a single platform. For example,
this may be too computationally intensive for a large PSTN Gateway, or even
a large Proxy.

Any ideas of how a UAS would make the decision to send the 419 without
burdening the UAC? Should there be a supported/require header for this
functionality, so that a UAC could "opt out"?



 
-----Original Message-----
From:  sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]
<mailto:sip-bounces@ietf.org]>  On Behalf Of Cullen  Jennings
Sent: Tuesday, February 22, 2005 16:46
To:  sip@ietf.org
Subject: [Sip]  draft-jennings-sip-hashcash-01


I updated  this draft on "SIP Computational Puzzles" so it has enough detail
to implement  it. (Previous version did not). It is in the drafts directory
and also in HTML  form at:

http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcash
-01.html
<http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcas
h-01.html> 

Cullen




  _____  

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<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 size=2>Yeah, 
a&nbsp;white list approach might work. </FONT></SPAN></DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 size=2>The 
practical scenario that I am thinking about is with a typical VoIP service 
provider.</FONT></SPAN></DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 size=2>Today, 
the vast majority of calls&nbsp;from subscribers terminate on gateways (owned by 
the operator of the service). &nbsp;These calls are not to a specific "user" but 
really to a gateway, using presumably something like the tel-URI-in-SIP-URI 
approach. In this case then, the white list is not to a whole URI but to 
"something" that is related to the gateway itself. This is different from the 
IM-scenario you quote. </FONT></SPAN></DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 size=2>Now, 
if we only have to worry about one set of gateways (like the ones owned by the 
service provider), I guess you could base the white list on the domain name or 
find some other solution. It may be more difficult when gateways are scattered 
all over the place (like in an enterprise for example, or a federation of 
enterprise, or when it terminates to another service provider). 
</FONT></SPAN></DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 size=2>I 
believe we should make sure that we describe white list properly before we 
publish the puzzle mechanism. I have a fear that we could define something where 
the people who make the SIP end-devices have an incentive to do something that 
is good for them (i.e., limit vulnerability to spam) but could be very 
problematic for the Gateways (or Proxies). Somehow we have to be able to mandate 
a white list capabily (or some other mechanism). Otherwise software upgrades 
could kill the operator of the service.</FONT></SPAN></DIV>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=760215818-28022005>
<DIV><SPAN class=760215818-28022005><FONT face=Arial color=#800000 size=2>PS: 
Note that this is a recurring problem with gateways (I think certs has the same 
issue). </FONT></SPAN></DIV></SPAN></DIV>
<BLOCKQUOTE 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT 
  face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> Cullen Jennings 
  [mailto:fluffy@cisco.com] <BR><B>Sent:</B> Friday, February 25, 2005 
  08:03<BR><B>To:</B> Audet, Francois [SC100:9T51:EXCH]; 
  sip@ietf.org<BR><B>Subject:</B> Re: [Sip] 
  draft-jennings-sip-hashcash-01<BR><BR></FONT></DIV><FONT 
  face="Verdana, Helvetica, Arial"><SPAN style="FONT-SIZE: 12px"><BR>The idea 
  would be to have a very skewed amount of computation to help rate limit 
  attacks. Whatever amount of computation is "reasonable" for a valid client to 
  do, an attacker is going to be able to apply a lot more computation to the 
  problem. However, this still limits the number of messages the attacker can do 
  and thus raises the cost. I discuss this some what in the SPAM draft. 
  <BR><BR>This could all happen in different ways. Imagine a case where my 
  enterprise proxy compute the puzzles on behalf of my phone when I call another 
  domain. Say in a large enterprise, it has 10,000 people and during a busy hour 
  makes 10 calls per second that are to an address external to the domain. Of 
  these only 2 are not whitelists. If the enterprise proxy was willing to spend 
  say 3GHz worth of CPU at computing challenges for these 2cps, well that is a 
  pretty hefty amount of hash that could be required. <BR><BR>Another ways is 
  perhaps all the UAC are willing to do about the amount of hashing that 1 
  second of G.729 compression would take. It's hard to imagine that most UAs 
  (even large GWs) could not do this. This would limit the rate of attack of a 
  single computer to in the 100s of messages per second. This is a lot less 
  hashing that the example above but contrast this to what is possible today. 
  Today I can send in the order of 100,000 message per second. Increasing the 
  cost of an attack by 1000 is good. It slows it down and give you time to trace 
  the source and cut it off before it is everywhere. It makes is less worth 
  while to advertise products that a very very small percentage of the receivers 
  would buy. <BR><BR>As discussed in the SPAM draft, you would always want to 
  use white lists first then if the person was not on the white list fallback to 
  something else like this. <BR><BR>Cullen<BR><BR><BR>On 2/24/05 11:30 AM, 
  "Francois Audet" &lt;audet@nortel.com&gt; wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><SPAN style="FONT-SIZE: 12px"><FONT color=#800000><FONT 
    face=Arial>We may need to explain some of the limitations of this. It seems 
    to me that this mechanism would not be terribly appropriate for applications 
    that aggregate a large number of clients onto a single platform. For 
    example, this may be too computationally intensive for a large PSTN Gateway, 
    or even a large Proxy.<BR></FONT></FONT><FONT 
    face="Verdana, Helvetica, Arial"><BR></FONT><FONT color=#800000><FONT 
    face=Arial>Any ideas of how a UAS would make the decision to send the 419 
    without burdening the UAC? Should there be a supported/require header for 
    this functionality, so that a UAC could "opt out"?<BR></FONT></FONT></SPAN>
    <BLOCKQUOTE><SPAN style="FONT-SIZE: 12px"><FONT 
      face="Verdana, Helvetica, Arial"><BR>&nbsp;<BR></FONT><FONT 
      face=Tahoma>-----Original Message-----<BR><B>From:</B> 
      &nbsp;sip-bounces@ietf.org [<A 
      href="mailto:sip-bounces@ietf.org]">mailto:sip-bounces@ietf.org]</A> <B>On 
      Behalf Of </B>Cullen &nbsp;Jennings<BR><B>Sent:</B> Tuesday, February 22, 
      2005 16:46<BR><B>To:</B> &nbsp;sip@ietf.org<BR><B>Subject:</B> [Sip] 
      &nbsp;draft-jennings-sip-hashcash-01<BR><BR></FONT><FONT 
      face="Verdana, Helvetica, Arial"><BR>I updated &nbsp;this draft on "SIP 
      Computational Puzzles" so it has enough detail to implement &nbsp;it. 
      (Previous version did not). It is in the drafts directory and also in HTML 
      &nbsp;form at:<BR><BR><A 
      href="http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcash-01.html">http://scm.sipfoundry.org/rep/ietf-drafts/fluffy/draft-jennings-sip-hashcash-01.html</A><BR><BR>Cullen<BR></FONT></SPAN></BLOCKQUOTE><SPAN 
    style="FONT-SIZE: 12px"><FONT face="Verdana, Helvetica, Arial"><BR>
    <HR align=center width="95%" SIZE=3>
    </FONT></SPAN><FONT size=2><FONT face="Monaco, Courier New"><SPAN 
    style="FONT-SIZE: 10px">_______________________________________________<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 
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From sip-bounces@ietf.org  Mon Feb 28 15:35:39 2005
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, sip <sip@ietf.org>
Subject: RE: [Sip] Draft defining the Accept-Disposition header field
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Could you leverage RFC3459?

> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> Sent: Wednesday, January 12, 2005 7:36 AM
> To: sip
> Subject: [Sip] Draft defining the Accept-Disposition header field
> 
> 
> Folks,
> 
> I have submitted a draft that defines the Accept-Disposition header 
> field. The usage of this header field would be similar to the 
> usage of 
> Accept, Accept-Encoding, or Accept-Language, but it would indicate 
> support for disposition types.
> 
>
http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept-disposition-0
0.txt

Your comments are welcome.

Thanks,

Gonzalo

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From sip-bounces@ietf.org  Mon Feb 28 15:42:47 2005
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From: Eric Burger <eburger@brooktrout.com>
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Subject: RE: [Sip] Draft defining the Accept-Disposition header field
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Inline.

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Friday, January 21, 2005 4:00 PM
> To: gonzalo.camarillo@ericsson.com
> Cc: sip@ietf.org; Paul H Kyzivat
> Subject: Re: [Sip] Draft defining the Accept-Disposition header field
> 
[snip]
> 
> > The same thing with uri-list bodies. A conference server may support
> > uri-list bodies for some reason, but may not support the
> > 'recipient-list' disposition type. The disposition type is 
> what tells me
> > that the conference server supports (or does not support) 
> the feature I
> > want to use. (So far there are two disposition types associated with
> > uri-list bodies and may be more in the future.)
> >
> 
> I guess I am wondering if using the MIME dispositions is 
> really the right
> way to basically tell the UA what you want it to do with a 
> given SIP command
> - what does recipient-list mean in the context of email or 
> other users of
> MIME?
[snip]
It doesn't have to make any sense whatsoever (SIP vs. E-Mail).  What is
important is the disposition is well defined for the given media type.

I would offer that it might be better to have different names than "inline"
or "render", so it is clear we mean what we mean.

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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
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