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From: vvarun kumar <arunvv@yahoo.com>
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HI,
I am ArunKumar a software engineer working on SIP.
Please clarify my problem that Iam facing in SIP.
The problem can be small or easy to solve but I am
unable to find solution.

The problem is When I am calling from client to Server
like thru SJPhone software(which resembles Phone in
daily use) in Windows to a software based in Linux.
I am able to recieve trying , ringing message that has
been generated by server but Iam not able to get ACK
or OK message when Iam capturing thru Ethreal
software.

How can I debug and how I can get the OK/ACK message
and when I am calling from other end point I am
getting segementation fault.What could the reason and
How to debug it so that I can get ACK/OK message from
both ends.

Thanks&regards,
Arunkumar.


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From exim@www1.ietf.org  Fri Jan  2 06:45:01 2004
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Date: Tue, 23 Dec 2003 16:42:27 +0530
From: Chirag Dhruv <chirag@einfochips.com>
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Subject: [Sip] sip standard related query
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Hi

I am new to SIP and in the process of feasibility study for SIP. I went 
through a lot of material on the web, and failed to understand if there 
is a common standard for SIP protocol or not. I understand that RFC2543 
is the latest starndard available till now, and if I implement that 
standard in my device, there is no need to implement other standards.

Looking for a response.

Thanks and Regards,
Chirag.


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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: Fri, 02 Jan 2004 17:17:13 +0530
From: shambhu <shambhudr@huawei.com>
Subject: RE: [Sip] sip standard related query
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Hi Chirag,

RFC 2543 has been obsoleted by RFC 3261. rfc 3261 specifies base protocol.
There are some drafts and rfcs extending the same for specific features.

Regards,
Shambhu

Regards,
Shambhu


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Chirag
Dhruv
Sent: Tuesday, 23 December 2003 4:42 PM
To: sip@ietf.org; dromasca@avaya.com
Subject: [Sip] sip standard related query


Hi

I am new to SIP and in the process of feasibility study for SIP. I went
through a lot of material on the web, and failed to understand if there
is a common standard for SIP protocol or not. I understand that RFC2543
is the latest starndard available till now, and if I implement that
standard in my device, there is no need to implement other standards.

Looking for a response.

Thanks and Regards,
Chirag.


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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 exim@www1.ietf.org  Fri Jan  2 11:23:48 2004
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Date: Fri, 02 Jan 2004 16:10:57 +0000
From: "Elwell, John" <john.elwell@siemens.com>
To: SIP@ietf.org
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Subject: [Sip] Use of P-Asserted-Identity in a response
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There seem to be some differences of opinion concerning the use of
P-Asserted-Identity (PAI) in a response. In particular, this is delaying the
progress of draft-ietf-sipping-qsig2sip, which makes use of PAI in a 200 OK
to convey the connected party identity received from the QSIG network. The
same situation is likely to be encountered when interworking with other
legacy signalling systems.

It is my opinion that PAI can be used in a 200 OK to assert the identity of
the connect party. Moreover, it can be inserted by a gateway UAS and, unless
subject to privacy, can be passed on even to untrusted entities including
the UAC.

My evidence for assuming this is as follows:

1. RFC 3325: In section 5, 2nd paragraph it states:
"If the proxy receives a message (request or response) from a node that it
trusts, it can use the information in the P-Asserted-Identity header field,
if any, as if it had authenticated the user itself."
There is nowhere that forbids the use of P-Asserted-ID in a response,
although there are places in the RFC that mention procedures for a request
without mentioning a response. In many places it talks about "message",
which, according to the text above, means request or response.

2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
calling user, PAI in a 180 response the ringing user and PAI in 200 OK the
user who answered the call. RFC 3325 is supposed to meet the requirement of
RFC3324.

I have seen a number of arguments against the use of PAI in a response:

a) It has been suggested that a PAI in a response represents the identity of
the entity that issued the request. This does not hold water. It clearly
would be in contradiction to clause 5 of RFC3324 and there is nowhere in
RFC3324 or RFC3325 that suggests echoing in a response the PAI in the
request.

b)It has been suggested that there would need to be corresponding privacy
requirements for identity information in a response. However, RFC 3323 does
indeed address privacy in responses. See last paragraph of section 3:
"It may be desirable to restrict identity information on both requests and
responses.  Initially, it might seem unusual to suggest that a response has
privacy concerns - presumably, the originator of the request knows who they
were attempting to contact, so the identity of the respondent can hardly be
confidential.  However, some personal information in responses (such as the
contact address at which the respondent is currently registered) is subject
to privacy concerns and can be addressed by these mechanisms."
Also see section 4.2, which states:
"...this document defines a new SIP header, Privacy, that can be used to
specify privacy handling for requests and responses."

c)It has been argued that using PAI in a response may cause an existing
implementation to do the wrong thing with respect to privacy, e.g., if
privacy is required by the caller, a proxy may apply this to the connected
user's identity in the response as well. I don't think this is a likely
scenario in view of RFC 3323's coverage of privacy in requests and
responses.

d)Also it has been suggested that an existing UAS may not take care to
request privacy in a response, not realising that a proxy might add a PAI.
Again, since RFC 3323 covers privacy in both requests and responses, any UA
implementation that supports privacy in requests should also support privacy
in responses.

e) It has been pointed out that a proxy cannot reject a response if the PAI
is wrong. I don't believe this matters - it should be sufficient for the
proxy just to strip out a PAI that is wrong.

In conclusion, I see evidence that PAI can be used in a response to assert
the ID of the connected party and no solid evidence to the contrary.

John Elwell (john.elwell@siemens.com)

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



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From: Sanjay Sinha <sanjsinh@cisco.com>
Organization: Cisco Systems
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Subject: [Sip] Subscribe dialog and target refresh request
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Hi,

For a dialog initiated with SUBSCRIBE, can a subscription for a new 
event on that dialog also be a target refresh request?

- Sanjay.



_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Date: Sat, 03 Jan 2004 14:13:24 -0500
From: David Oran <oran@cisco.com>
To: "Elwell, John" <john.elwell@siemens.com>, SIP@ietf.org
Subject: Re: [Sip] Use of P-Asserted-Identity in a response
Message-ID: <2147483647.1073139204@[10.32.245.157]>
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These arguments are good ones.
Even if they weren't good I'd still be in favor of what John proposes, 
because it's eminently reasonable and provides desperately needed 
functionality for which we currently have no alternative machinery.

Dave.

--On Friday, January 2, 2004 4:10 PM +0000 "Elwell, John" 
<john.elwell@siemens.com> wrote:

> There seem to be some differences of opinion concerning the use of
> P-Asserted-Identity (PAI) in a response. In particular, this is delaying
> the progress of draft-ietf-sipping-qsig2sip, which makes use of PAI in a
> 200 OK to convey the connected party identity received from the QSIG
> network. The same situation is likely to be encountered when interworking
> with other legacy signalling systems.
>
> It is my opinion that PAI can be used in a 200 OK to assert the identity
> of the connect party. Moreover, it can be inserted by a gateway UAS and,
> unless subject to privacy, can be passed on even to untrusted entities
> including the UAC.
>
> My evidence for assuming this is as follows:
>
> 1. RFC 3325: In section 5, 2nd paragraph it states:
> "If the proxy receives a message (request or response) from a node that it
> trusts, it can use the information in the P-Asserted-Identity header
> field, if any, as if it had authenticated the user itself."
> There is nowhere that forbids the use of P-Asserted-ID in a response,
> although there are places in the RFC that mention procedures for a request
> without mentioning a response. In many places it talks about "message",
> which, according to the text above, means request or response.
>
> 2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
> calling user, PAI in a 180 response the ringing user and PAI in 200 OK the
> user who answered the call. RFC 3325 is supposed to meet the requirement
> of RFC3324.
>
> I have seen a number of arguments against the use of PAI in a response:
>
> a) It has been suggested that a PAI in a response represents the identity
> of the entity that issued the request. This does not hold water. It
> clearly would be in contradiction to clause 5 of RFC3324 and there is
> nowhere in RFC3324 or RFC3325 that suggests echoing in a response the PAI
> in the request.
>
> b)It has been suggested that there would need to be corresponding privacy
> requirements for identity information in a response. However, RFC 3323
> does indeed address privacy in responses. See last paragraph of section 3:
> "It may be desirable to restrict identity information on both requests and
> responses.  Initially, it might seem unusual to suggest that a response
> has privacy concerns - presumably, the originator of the request knows
> who they were attempting to contact, so the identity of the respondent
> can hardly be confidential.  However, some personal information in
> responses (such as the contact address at which the respondent is
> currently registered) is subject to privacy concerns and can be addressed
> by these mechanisms."
> Also see section 4.2, which states:
> "...this document defines a new SIP header, Privacy, that can be used to
> specify privacy handling for requests and responses."
>
> c)It has been argued that using PAI in a response may cause an existing
> implementation to do the wrong thing with respect to privacy, e.g., if
> privacy is required by the caller, a proxy may apply this to the connected
> user's identity in the response as well. I don't think this is a likely
> scenario in view of RFC 3323's coverage of privacy in requests and
> responses.
>
> d)Also it has been suggested that an existing UAS may not take care to
> request privacy in a response, not realising that a proxy might add a PAI.
> Again, since RFC 3323 covers privacy in both requests and responses, any
> UA implementation that supports privacy in requests should also support
> privacy in responses.
>
> e) It has been pointed out that a proxy cannot reject a response if the
> PAI is wrong. I don't believe this matters - it should be sufficient for
> the proxy just to strip out a PAI that is wrong.
>
> In conclusion, I see evidence that PAI can be used in a response to assert
> the ID of the connected party and no solid evidence to the contrary.
>
> John Elwell (john.elwell@siemens.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



From exim@www1.ietf.org  Sat Jan  3 19:31:51 2004
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To: David Oran <oran@cisco.com>
CC: "Elwell, John" <john.elwell@siemens.com>, SIP@ietf.org
Subject: Re: [Sip] Use of P-Asserted-Identity in a response
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I would also support this usage.

David Oran wrote:

> These arguments are good ones.
> Even if they weren't good I'd still be in favor of what John proposes, 
> because it's eminently reasonable and provides desperately needed 
> functionality for which we currently have no alternative machinery.
> 
> Dave.
> 
> --On Friday, January 2, 2004 4:10 PM +0000 "Elwell, John" 
> <john.elwell@siemens.com> wrote:
> 
>> There seem to be some differences of opinion concerning the use of
>> P-Asserted-Identity (PAI) in a response. In particular, this is delaying
>> the progress of draft-ietf-sipping-qsig2sip, which makes use of PAI in a
>> 200 OK to convey the connected party identity received from the QSIG
>> network. The same situation is likely to be encountered when interworking
>> with other legacy signalling systems.
>>
>> It is my opinion that PAI can be used in a 200 OK to assert the identity
>> of the connect party. Moreover, it can be inserted by a gateway UAS and,
>> unless subject to privacy, can be passed on even to untrusted entities
>> including the UAC.
>>
>> My evidence for assuming this is as follows:
>>
>> 1. RFC 3325: In section 5, 2nd paragraph it states:
>> "If the proxy receives a message (request or response) from a node 
>> that it
>> trusts, it can use the information in the P-Asserted-Identity header
>> field, if any, as if it had authenticated the user itself."
>> There is nowhere that forbids the use of P-Asserted-ID in a response,
>> although there are places in the RFC that mention procedures for a 
>> request
>> without mentioning a response. In many places it talks about "message",
>> which, according to the text above, means request or response.
>>
>> 2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
>> calling user, PAI in a 180 response the ringing user and PAI in 200 OK 
>> the
>> user who answered the call. RFC 3325 is supposed to meet the requirement
>> of RFC3324.
>>
>> I have seen a number of arguments against the use of PAI in a response:
>>
>> a) It has been suggested that a PAI in a response represents the identity
>> of the entity that issued the request. This does not hold water. It
>> clearly would be in contradiction to clause 5 of RFC3324 and there is
>> nowhere in RFC3324 or RFC3325 that suggests echoing in a response the PAI
>> in the request.
>>
>> b)It has been suggested that there would need to be corresponding privacy
>> requirements for identity information in a response. However, RFC 3323
>> does indeed address privacy in responses. See last paragraph of 
>> section 3:
>> "It may be desirable to restrict identity information on both requests 
>> and
>> responses.  Initially, it might seem unusual to suggest that a response
>> has privacy concerns - presumably, the originator of the request knows
>> who they were attempting to contact, so the identity of the respondent
>> can hardly be confidential.  However, some personal information in
>> responses (such as the contact address at which the respondent is
>> currently registered) is subject to privacy concerns and can be addressed
>> by these mechanisms."
>> Also see section 4.2, which states:
>> "...this document defines a new SIP header, Privacy, that can be used to
>> specify privacy handling for requests and responses."
>>
>> c)It has been argued that using PAI in a response may cause an existing
>> implementation to do the wrong thing with respect to privacy, e.g., if
>> privacy is required by the caller, a proxy may apply this to the 
>> connected
>> user's identity in the response as well. I don't think this is a likely
>> scenario in view of RFC 3323's coverage of privacy in requests and
>> responses.
>>
>> d)Also it has been suggested that an existing UAS may not take care to
>> request privacy in a response, not realising that a proxy might add a 
>> PAI.
>> Again, since RFC 3323 covers privacy in both requests and responses, any
>> UA implementation that supports privacy in requests should also support
>> privacy in responses.
>>
>> e) It has been pointed out that a proxy cannot reject a response if the
>> PAI is wrong. I don't believe this matters - it should be sufficient for
>> the proxy just to strip out a PAI that is wrong.
>>
>> In conclusion, I see evidence that PAI can be used in a response to 
>> assert
>> the ID of the connected party and no solid evidence to the contrary.
>>
>> John Elwell (john.elwell@siemens.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
> 


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



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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "Elwell, John" <john.elwell@siemens.com>, SIP@ietf.org
Subject: RE: [Sip] Use of P-Asserted-Identity in a response
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RFC 3325 section 9.1 and section 9.2 both extend table 2 of RFC 3261 with the new header.

Both entires leave the "where" column empty.

The key to RFC 3261 table 2 states:

      An empty entry in the "where" column indicates that the header
           field may be present in all requests and responses.

I would therefore conclude from the existing specification that both headers may appear in any response to BYE, INV, OPT, SUB, NOT, REF, i.e. those methods that either create a dialog, or are used outside a dialog.

As MESSAGE appeared after RFC 3325 I would also assume that it is allowed in the MESSAGE method, in both requests and all responses.

Note that these tables do not restrict it to 18x and 2xx responses, although for ISDN mapping purposes those responses are the useful ones. It is perfectly valid in 3xx, 4xx and 5xx responses.

Any assertion of identity in responses (and its value) is of course dependent on the specification of privacy by the UAS and on the requirements of the first proxy in the trust domain.

This usage corresponds to what has been specified in 3GPP TS 24.229 and I believe in ITU-T Q.1912.5.

regards

Keith 

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: 02 January 2004 16:11
> To: SIP@ietf.org
> Subject: [Sip] Use of P-Asserted-Identity in a response
> 
> 
> There seem to be some differences of opinion concerning the use of
> P-Asserted-Identity (PAI) in a response. In particular, this 
> is delaying the
> progress of draft-ietf-sipping-qsig2sip, which makes use of 
> PAI in a 200 OK
> to convey the connected party identity received from the QSIG 
> network. The
> same situation is likely to be encountered when interworking 
> with other
> legacy signalling systems.
> 
> It is my opinion that PAI can be used in a 200 OK to assert 
> the identity of
> the connect party. Moreover, it can be inserted by a gateway 
> UAS and, unless
> subject to privacy, can be passed on even to untrusted 
> entities including
> the UAC.
> 
> My evidence for assuming this is as follows:
> 
> 1. RFC 3325: In section 5, 2nd paragraph it states:
> "If the proxy receives a message (request or response) from a 
> node that it
> trusts, it can use the information in the P-Asserted-Identity 
> header field,
> if any, as if it had authenticated the user itself."
> There is nowhere that forbids the use of P-Asserted-ID in a response,
> although there are places in the RFC that mention procedures 
> for a request
> without mentioning a response. In many places it talks about 
> "message",
> which, according to the text above, means request or response.
> 
> 2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
> calling user, PAI in a 180 response the ringing user and PAI 
> in 200 OK the
> user who answered the call. RFC 3325 is supposed to meet the 
> requirement of
> RFC3324.
> 
> I have seen a number of arguments against the use of PAI in a 
> response:
> 
> a) It has been suggested that a PAI in a response represents 
> the identity of
> the entity that issued the request. This does not hold water. 
> It clearly
> would be in contradiction to clause 5 of RFC3324 and there is 
> nowhere in
> RFC3324 or RFC3325 that suggests echoing in a response the PAI in the
> request.
> 
> b)It has been suggested that there would need to be 
> corresponding privacy
> requirements for identity information in a response. However, 
> RFC 3323 does
> indeed address privacy in responses. See last paragraph of section 3:
> "It may be desirable to restrict identity information on both 
> requests and
> responses.  Initially, it might seem unusual to suggest that 
> a response has
> privacy concerns - presumably, the originator of the request 
> knows who they
> were attempting to contact, so the identity of the respondent 
> can hardly be
> confidential.  However, some personal information in 
> responses (such as the
> contact address at which the respondent is currently 
> registered) is subject
> to privacy concerns and can be addressed by these mechanisms."
> Also see section 4.2, which states:
> "...this document defines a new SIP header, Privacy, that can 
> be used to
> specify privacy handling for requests and responses."
> 
> c)It has been argued that using PAI in a response may cause 
> an existing
> implementation to do the wrong thing with respect to privacy, e.g., if
> privacy is required by the caller, a proxy may apply this to 
> the connected
> user's identity in the response as well. I don't think this 
> is a likely
> scenario in view of RFC 3323's coverage of privacy in requests and
> responses.
> 
> d)Also it has been suggested that an existing UAS may not take care to
> request privacy in a response, not realising that a proxy 
> might add a PAI.
> Again, since RFC 3323 covers privacy in both requests and 
> responses, any UA
> implementation that supports privacy in requests should also 
> support privacy
> in responses.
> 
> e) It has been pointed out that a proxy cannot reject a 
> response if the PAI
> is wrong. I don't believe this matters - it should be 
> sufficient for the
> proxy just to strip out a PAI that is wrong.
> 
> In conclusion, I see evidence that PAI can be used in a 
> response to assert
> the ID of the connected party and no solid evidence to the contrary.
> 
> John Elwell (john.elwell@siemens.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: Rene Bartsch <ml@bartschnet.de>
To: SIP@ietf.org
Date: Sun, 4 Jan 2004 17:43:45 +0100
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Subject: [Sip] Emergency Calls with SIP?
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Hi,

currently I'm faced with the problem of emergency calls.

As you need to gather the location of the caller to connect to the nearest 
emergency center when 911, 110, 112, ... is dialed, SIP needs some way for 
determining location.

My idea is to store a location-tag (e.g. address or GPS-data) in the 
SIP-phone, which is transmitted to the SIP-provider in case an emergency 
number is dialed.
That way the SIP-provider can determine the appropriate emergency center by 
country and zip/area code transmitting three addresses to the emergency 
center:

location, SIP-account holder and IP.

With location you can send help directly (e.g. ambulance). In case of wrong 
location or misuse you can investigate the caller by account-holder address 
and IP.

Are there any RFCs or Drafts for a emergency-call use-case with SIP?

Rene Bartsch

_______________________________________________
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Subject: Re: [Sip] Emergency Calls with SIP?
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Henning Schulrinne wrote a number of relevant drafts.  The topic of 
connection from a SIP phone to a legacy emergency call centre was 
explored in draft-taylor-sipping-emerg-scen-01.txt, which also gives the 
references to Henning's drafts.

Rene Bartsch wrote:

> Hi,
> 
> currently I'm faced with the problem of emergency calls.
> 
> As you need to gather the location of the caller to connect to the nearest 
> emergency center when 911, 110, 112, ... is dialed, SIP needs some way for 
> determining location.
> 
> My idea is to store a location-tag (e.g. address or GPS-data) in the 
> SIP-phone, which is transmitted to the SIP-provider in case an emergency 
> number is dialed.
> That way the SIP-provider can determine the appropriate emergency center by 
> country and zip/area code transmitting three addresses to the emergency 
> center:
> 
> location, SIP-account holder and IP.
> 
> With location you can send help directly (e.g. ambulance). In case of wrong 
> location or misuse you can investigate the caller by account-holder address 
> and IP.
> 
> Are there any RFCs or Drafts for a emergency-call use-case with SIP?
> 
> Rene Bartsch
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/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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Regards,
Nataraju A.B.=20


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Sanjay
Sinha
Sent: Saturday, January 03, 2004 12:28 AM
To: sip@ietf.org
Subject: [Sip] Subscribe dialog and target refresh request

Hi,

For a dialog initiated with SUBSCRIBE, can a subscription for a new=20
event on that dialog also be a target refresh request?

[Natraj] No, it's not a Target refresh request. Target refresh could
only happen either through INVITE or UPDATE requests..=20

- Sanjay.



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Subject: RE: [Sip] Subscribe dialog and target refresh request
Date: Mon, 5 Jan 2004 13:43:05 +0200
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Hi

> For a dialog initiated with SUBSCRIBE, can a subscription for a new=20
> event on that dialog also be a target refresh request?
>=20
> [Natraj] No, it's not a Target refresh request. Target refresh could
> only happen either through INVITE or UPDATE requests..=20
>=20

There is a bug in RFC3265. Bug description =
(http://bugs.sipit.net/sipwg/show_bug.cgi?id=3D699) says:

"NOTIFY and SUBSCRIBE are target refresh requests

...but the text never says this explicitly."

Haven't seen this bug closed yet... I remember some discussion on this =
on this a while (a year or so) ago, so you might want to check the =
archives for that.

		Jussi

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



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Subject: Re: [Sip] Re: SIPIT 14 Registration
From: Robert Sparks <rsparks@dynamicsoft.com>
To: sip-implementors@cs.columbia.edu
Cc: sip@ietf.org
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SIPIT 14 Registration will close soon.

The event will be held  9-February through 13-February 2004.
 
The last day to register is 16-January-2004
(The last day to secure a room at the event is 9-January-2004)


Registration information is available at:
http://www.etsi.org/plugtests/SIPIT14.htm	

RjS


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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Jussi.Turunen@nokia.com
CC: nataraju.alilaghatta@wipro.com, sanjsinh@cisco.com, sip@ietf.org
Subject: Re: [Sip] Subscribe dialog and target refresh request
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Yes, SUB and NOT are target refresh requests.

However, the question that was asked, I think, had to do with 
establishing a second subscription in the context of a dialog created 
by a first subscription. In such a case, a SUBSCRIBE or NOTIFY would 
be a target refresh request, but what state does it update? For 
example, if a SUBSCRIBE refresh for the second subscription contains a 
new Contact, does that contact apply to both subscriptions, or just 
the second one? I think its both, because the remote target is part of 
the dialog state, and there is but one dialog here.

However, I think its a much better idea to send a second subscription 
on a different dialog, rather than trying to reuse an existing dialog.

-Jonathan R.

Jussi.Turunen@nokia.com wrote:

> Hi
> 
> 
>>For a dialog initiated with SUBSCRIBE, can a subscription for a new 
>>event on that dialog also be a target refresh request?
>>
>>[Natraj] No, it's not a Target refresh request. Target refresh could
>>only happen either through INVITE or UPDATE requests.. 
>>
> 
> 
> There is a bug in RFC3265. Bug description (http://bugs.sipit.net/sipwg/show_bug.cgi?id=699) says:
> 
> "NOTIFY and SUBSCRIBE are target refresh requests
> 
> ...but the text never says this explicitly."
> 
> Haven't seen this bug closed yet... I remember some discussion on this on this a while (a year or so) ago, so you might want to check the archives for that.
> 
> 		Jussi
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/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
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.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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CC: Jussi.Turunen@nokia.com, nataraju.alilaghatta@wipro.com,
        sanjsinh@cisco.com, sip@ietf.org
Subject: Re: [Sip] Subscribe dialog and target refresh request
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Jonathan Rosenberg wrote:
> Yes, SUB and NOT are target refresh requests.
> 
> However, the question that was asked, I think, had to do with 
> establishing a second subscription in the context of a dialog created by 
> a first subscription. In such a case, a SUBSCRIBE or NOTIFY would be a 
> target refresh request, but what state does it update? For example, if a 
> SUBSCRIBE refresh for the second subscription contains a new Contact, 
> does that contact apply to both subscriptions, or just the second one? I 
> think its both, because the remote target is part of the dialog state, 
> and there is but one dialog here.

I agree.

> However, I think its a much better idea to send a second subscription on 
> a different dialog, rather than trying to reuse an existing dialog.

I think this actually depends on intent. If the subscriptions are 
intended to be independent then I agree. OTOH, if it is important that 
the subscriptions always be associated with the same UA even as that 
moves around, then there is an advantage to using a single dialog.

Its easier to find such cases in the sharing of a dialog between an 
invite and a subscribe than it is between two subscribes.

	Paul

> -Jonathan R.
> 
> Jussi.Turunen@nokia.com wrote:
> 
>> Hi
>>
>>
>>> For a dialog initiated with SUBSCRIBE, can a subscription for a new 
>>> event on that dialog also be a target refresh request?
>>>
>>> [Natraj] No, it's not a Target refresh request. Target refresh could
>>> only happen either through INVITE or UPDATE requests..
>>
>>
>>
>> There is a bug in RFC3265. Bug description 
>> (http://bugs.sipit.net/sipwg/show_bug.cgi?id=699) says:
>>
>> "NOTIFY and SUBSCRIBE are target refresh requests
>>
>> ...but the text never says this explicitly."
>>
>> Haven't seen this bug closed yet... I remember some discussion on this 
>> on this a while (a year or so) ago, so you might want to check the 
>> archives for that.
>>
>>         Jussi
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/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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Date: Mon, 05 Jan 2004 18:00:14 -0600
To: Rene Bartsch <ml@bartschnet.de>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] Emergency Calls with SIP?
Cc: SIP@ietf.org
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 <200401041743.45713.ml@bartschnet.de>
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At 04:22 PM 1/4/2004 -0500, Tom Taylor wrote:
>Henning Schulrinne wrote a number of relevant drafts.  The topic of 
>connection from a SIP phone to a legacy emergency call centre was explored 
>in draft-taylor-sipping-emerg-scen-01.txt, which also gives the references 
>to Henning's drafts.

see:
http://www.ietf.org/internet-drafts/draft-polk-sipping-location-requirements-01.txt

containing a scenario just like I read yours below.

Comments are appreciated

A new version of the above ID will be generated before the Seoul meeting.

BTW - the SIPPING mailing list has had a few threads recently on this ID.


>Rene Bartsch wrote:
>
>>Hi,
>>currently I'm faced with the problem of emergency calls.
>>As you need to gather the location of the caller to connect to the 
>>nearest emergency center when 911, 110, 112, ... is dialed, SIP needs 
>>some way for determining location.
>>My idea is to store a location-tag (e.g. address or GPS-data) in the 
>>SIP-phone, which is transmitted to the SIP-provider in case an emergency 
>>number is dialed.
>>That way the SIP-provider can determine the appropriate emergency center 
>>by country and zip/area code transmitting three addresses to the 
>>emergency center:
>>location, SIP-account holder and IP.
>>With location you can send help directly (e.g. ambulance). In case of 
>>wrong location or misuse you can investigate the caller by account-holder 
>>address and IP.
>>Are there any RFCs or Drafts for a emergency-call use-case with SIP?
>>Rene Bartsch
>>_______________________________________________


cheers,
James

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


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Tue, 06 Jan 2004 02:08:17 -0500
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CC: Jussi.Turunen@nokia.com, nataraju.alilaghatta@wipro.com,
        sanjsinh@cisco.com, sip@ietf.org
Subject: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target
 refresh request
References: <A1C8946F2F77D5479A9233E8BF447B18193242@esebe009.ntc.nokia.com> <3FF9DB9E.60702@dynamicsoft.com> <3FF9F36D.5060604@cisco.com>
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inline.

Paul Kyzivat wrote:


>> However, I think its a much better idea to send a second subscription 
>> on a different dialog, rather than trying to reuse an existing dialog.
> 
> 
> I think this actually depends on intent. If the subscriptions are 
> intended to be independent then I agree. OTOH, if it is important that 
> the subscriptions always be associated with the same UA even as that 
> moves around, then there is an advantage to using a single dialog.
> 
> Its easier to find such cases in the sharing of a dialog between an 
> invite and a subscribe than it is between two subscribes.

The GRUU would allow you to keep both dialogs attached to a particular 
UA instance. Should the UA move around, it would require you to 
generate a target refresh request on each dialog to update the Contact.

This, however, leads me into thinking a bit on the lifecycle 
management for GRUUs. Currently, GRUU is bound to a single 
registration, where the registration is defined by the Contact URI. If 
the UA should "move", in the sense that it acquires a new IP address, 
this will be a new Contact URI, and therefore, a new GRUU. It would be 
nice if, instead, the GRUU were truly bound to the UA instance, and 
that even if the UA instance changed IP addresses, the GRUU could 
remain unchanged. In such a case, if the UA moves, there is actually 
no need to issue any target refresh requests. The UA would simply 
re-register, update the binding of its GRUU to its new IP address, and 
existing calls would be correctly routed.

Indeed, this has the interesting benefit that, even if both UA in a 
call should move at the same time, the call would still be able to 
continue, since such updates are never propagated to the peer in the 
call.

For this to work, it would require that the registration have a way of 
identifying the UA instance. This goes back to the instance identifier 
we had (but removed) from callee-caps. If we added it, we could then 
associate the GRUU with this ID, rather than the Contact URI. In other 
words, if the client registered thusly:

REGISTER
To: sip:user@example.com
Contact: sip:1.2.3.4;+instance-id="9asdd9a887sdasd=asd888a"
Supported: gruu

the registrar would allocate a GRUU, and associate it with that 
instance ID:

200 OK
Contact: 
sip:1.2.3.4;+instance-id="9asdd9a887sdasd=asd888a";gruu=sip:f4@example.com

Now, if the UA should update its registration because it moves:

REGISTER
To: sip:user@example.com
Contact: sip:5.6.7.8;+instance-id="9asdd9a887sdasd=asd888a"
Contact: sip:1.2.3.4;expires=0

the registrar would return the *same* gruu for the new registration:

200 OK
Contact: 
sip:5.6.7.8;+instance-id="9asdd9a887sdasd=asd888a";gruu=sip:f4@example.com


It is still possible for the registrar to not store additional state 
if GRUU is defined this way. The GRUU URI would be computed thusly:

sip:<E(salt,instance-id)>@example.com

That is, the GRUU now encodes the instance ID. When a request is 
received with a GRUU, the registrar extracts the instance ID. It then 
looks up the registration corresponding to that instance ID, and 
obtains its Contact URI. This doesnt require additional state per se, 
but does require that the registration data be structured to allow 
lookups based on the instance-id.

Thoughts?

-Jonathan R.
> 
>     Paul
> 
>> -Jonathan R.
>>
>> Jussi.Turunen@nokia.com wrote:
>>
>>> Hi
>>>
>>>
>>>> For a dialog initiated with SUBSCRIBE, can a subscription for a new 
>>>> event on that dialog also be a target refresh request?
>>>>
>>>> [Natraj] No, it's not a Target refresh request. Target refresh could
>>>> only happen either through INVITE or UPDATE requests..
>>>
>>>
>>>
>>>
>>> There is a bug in RFC3265. Bug description 
>>> (http://bugs.sipit.net/sipwg/show_bug.cgi?id=699) says:
>>>
>>> "NOTIFY and SUBSCRIBE are target refresh requests
>>>
>>> ...but the text never says this explicitly."
>>>
>>> Haven't seen this bug closed yet... I remember some discussion on 
>>> this on this a while (a year or so) ago, so you might want to check 
>>> the archives for that.
>>>
>>>         Jussi
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/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
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Date: Tue, 06 Jan 2004 02:36:56 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sip] Updated GRUU spec
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Folks,

I've just submitted an update to the GRUU spec. It is now a working 
item of the SIP working group. Until it appears in the archives, you 
can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-sip-gruu-00.txt

The changes from the previous version are:

* incorporates the content of the requirements document, which will no 
  longer be proceeding as a separate document

* updates the discussion on registrar behavior to clarify behavior

* removes the 128 bit restriction on the grid

The only open issue, which I have discussed in another thread, is 
whether or not to associate the gruu to the Contact URI or instance ID.

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



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Subject: RE: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target refresh request
Date: Tue, 6 Jan 2004 08:58:59 -0000
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To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
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Comments [in-line]

>-----Original Message-----
>From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>Sent: 06 January 2004 07:08
>To: Paul Kyzivat
>Cc: Jussi.Turunen@nokia.com; nataraju.alilaghatta@wipro.com;
>sanjsinh@cisco.com; sip@ietf.org
>Subject: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and
target
>refresh request
>
>inline.
>
>Paul Kyzivat wrote:
>
>
>>> However, I think its a much better idea to send a second
subscription
>>> on a different dialog, rather than trying to reuse an existing
dialog.
>>
>>
>> I think this actually depends on intent. If the subscriptions are
>> intended to be independent then I agree. OTOH, if it is important
that
>> the subscriptions always be associated with the same UA even as that
>> moves around, then there is an advantage to using a single dialog.
>>
>> Its easier to find such cases in the sharing of a dialog between an
>> invite and a subscribe than it is between two subscribes.
>
>The GRUU would allow you to keep both dialogs attached to a particular
>UA instance. Should the UA move around, it would require you to
>generate a target refresh request on each dialog to update the Contact.
>
>This, however, leads me into thinking a bit on the lifecycle
>management for GRUUs. Currently, GRUU is bound to a single
>registration, where the registration is defined by the Contact URI. If
>the UA should "move", in the sense that it acquires a new IP address,
>this will be a new Contact URI, and therefore, a new GRUU. It would be
>nice if, instead, the GRUU were truly bound to the UA instance, and
>that even if the UA instance changed IP addresses, the GRUU could
>remain unchanged. In such a case, if the UA moves, there is actually
>no need to issue any target refresh requests. The UA would simply
>re-register, update the binding of its GRUU to its new IP address, and
>existing calls would be correctly routed.
>
>Indeed, this has the interesting benefit that, even if both UA in a
>call should move at the same time, the call would still be able to
>continue, since such updates are never propagated to the peer in the
>call.
>
>For this to work, it would require that the registration have a way of
>identifying the UA instance. This goes back to the instance identifier
>we had (but removed) from callee-caps. If we added it, we could then
>associate the GRUU with this ID, rather than the Contact URI. In other
>words, if the client registered thusly:
>
>REGISTER
>To: sip:user@example.com
>Contact: sip:1.2.3.4;+instance-id=3D"9asdd9a887sdasd=3Dasd888a"
>Supported: gruu
>
>the registrar would allocate a GRUU, and associate it with that
>instance ID:
>
>200 OK
>Contact:
>sip:1.2.3.4;+instance-id=3D"9asdd9a887sdasd=3Dasd888a";gruu=3Dsip:f4@examp=
le.
com
>
>Now, if the UA should update its registration because it moves:
>
>REGISTER
>To: sip:user@example.com
>Contact: sip:5.6.7.8;+instance-id=3D"9asdd9a887sdasd=3Dasd888a"
>Contact: sip:1.2.3.4;expires=3D0
>
>the registrar would return the *same* gruu for the new registration:
>
>200 OK
>Contact:
>sip:5.6.7.8;+instance-id=3D"9asdd9a887sdasd=3Dasd888a";gruu=3Dsip:f4@examp=
le.
com
>
>
>It is still possible for the registrar to not store additional state
>if GRUU is defined this way. The GRUU URI would be computed thusly:
>
>sip:<E(salt,instance-id)>@example.com
>
>That is, the GRUU now encodes the instance ID. When a request is
>received with a GRUU, the registrar extracts the instance ID. It then
>looks up the registration corresponding to that instance ID, and
>obtains its Contact URI. This doesnt require additional state per se,
>but does require that the registration data be structured to allow
>lookups based on the instance-id.
>
>Thoughts?

[Chris Boulton] Jonathan - I really like this.  It makes the GRUU
mechanism extremely powerful and increases usability and flexibility.
As long as the rule set for this functionality is strong (e.g. ensuring
that any client wishing to alter a binding for a GRUU terminates the
existing binding) I think it should be included.=20=20=20

Chris.


>-Jonathan R.
>>
>>     Paul
>>
>>> -Jonathan R.
>>>
>>> Jussi.Turunen@nokia.com wrote:
>>>
>>>> Hi
>>>>
>>>>
>>>>> For a dialog initiated with SUBSCRIBE, can a subscription for a
new
>>>>> event on that dialog also be a target refresh request?
>>>>>
>>>>> [Natraj] No, it's not a Target refresh request. Target refresh
could
>>>>> only happen either through INVITE or UPDATE requests..
>>>>
>>>>
>>>>
>>>>
>>>> There is a bug in RFC3265. Bug description
>>>> (http://bugs.sipit.net/sipwg/show_bug.cgi?id=3D699) says:
>>>>
>>>> "NOTIFY and SUBSCRIBE are target refresh requests
>>>>
>>>> ...but the text never says this explicitly."
>>>>
>>>> Haven't seen this bug closed yet... I remember some discussion on
>>>> this on this a while (a year or so) ago, so you might want to check
>>>> the archives for that.
>>>>
>>>>         Jussi
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/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
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.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: "Tolga Asveren" <asveren@ulticom.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Subscribe dialog and target refresh request
Date: Tue, 6 Jan 2004 09:15:03 -0500
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one question/comment below...

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Paul
> Kyzivat
> Sent: Monday, January 05, 2004 6:30 PM
> To: Jonathan Rosenberg
> Cc: Jussi.Turunen@nokia.com; nataraju.alilaghatta@wipro.com;
> sanjsinh@cisco.com; sip@ietf.org
> Subject: Re: [Sip] Subscribe dialog and target refresh request
>
>
>
>
> Jonathan Rosenberg wrote:
> > Yes, SUB and NOT are target refresh requests.
> >
> > However, the question that was asked, I think, had to do with
> > establishing a second subscription in the context of a dialog
> created by
> > a first subscription. In such a case, a SUBSCRIBE or NOTIFY would be a
> > target refresh request, but what state does it update? For
> example, if a
> > SUBSCRIBE refresh for the second subscription contains a new Contact,
> > does that contact apply to both subscriptions, or just the
> second one? I
> > think its both, because the remote target is part of the dialog state,
> > and there is but one dialog here.
>
> I agree.
>
> > However, I think its a much better idea to send a second
> subscription on
> > a different dialog, rather than trying to reuse an existing dialog.
>
> I think this actually depends on intent. If the subscriptions are
> intended to be independent then I agree. OTOH, if it is important that
> the subscriptions always be associated with the same UA even as that
> moves around, then there is an advantage to using a single dialog.
[TOLGA]Maybe it is not an issue totally black and white but I prefer to
think that such associations are better done on application level, rather
than relying them to be part of the same dialogue. I personally like the
approach, to use dialogue construct more for on-the-wire SIP protocol
procedures. Even for SUBSCRIBE in an INVITE initiated dialogue, IMO, those
are two different signalling relationships, asociated on the application
level, rather than on SIP signalling level -I am not saying it is wrong to
send them on the same dialogue, just I think, one won't loose much, if they
are sent on different one-. For that particular target refresh issue, there
is maybe a minor advantage of sending only one target refresh message
instead of two, if both SUBSCRIBESs (or INVITe and SUBSCRIBE) are sent as
part of the same dialogue, but IMO this is insignifact.
>
> Its easier to find such cases in the sharing of a dialog between an
> invite and a subscribe than it is between two subscribes.
>
> 	Paul
>
> > -Jonathan R.
> >
> > Jussi.Turunen@nokia.com wrote:
> >
> >> Hi
> >>
> >>
> >>> For a dialog initiated with SUBSCRIBE, can a subscription for a new
> >>> event on that dialog also be a target refresh request?
> >>>
> >>> [Natraj] No, it's not a Target refresh request. Target refresh could
> >>> only happen either through INVITE or UPDATE requests..
> >>
> >>
> >>
> >> There is a bug in RFC3265. Bug description
> >> (http://bugs.sipit.net/sipwg/show_bug.cgi?id=699) says:
> >>
> >> "NOTIFY and SUBSCRIBE are target refresh requests
> >>
> >> ...but the text never says this explicitly."
> >>
> >> Haven't seen this bug closed yet... I remember some discussion on this
> >> on this a while (a year or so) ago, so you might want to check the
> >> archives for that.
> >>
> >>         Jussi
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/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
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Subject: Re: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target
 refresh request
References: <A1C8946F2F77D5479A9233E8BF447B18193242@esebe009.ntc.nokia.com> <3FF9DB9E.60702@dynamicsoft.com> <3FF9F36D.5060604@cisco.com> <3FFA5EE1.1050302@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> inline.
> 
> Paul Kyzivat wrote:
> 
> 
>>> However, I think its a much better idea to send a second subscription 
>>> on a different dialog, rather than trying to reuse an existing dialog.
>>
>> I think this actually depends on intent. If the subscriptions are 
>> intended to be independent then I agree. OTOH, if it is important that 
>> the subscriptions always be associated with the same UA even as that 
>> moves around, then there is an advantage to using a single dialog.
>>
>> Its easier to find such cases in the sharing of a dialog between an 
>> invite and a subscribe than it is between two subscribes.
> 
> The GRUU would allow you to keep both dialogs attached to a particular 
> UA instance. Should the UA move around, it would require you to generate 
> a target refresh request on each dialog to update the Contact.

Yes, this is another way to solve that problem. I am not especially 
attached to using shared dialogs this way once all the machinery to do 
otherwise is in place. OTOH, I don't think shared dialogs are going 
away, so they remain *a* way to solve this or related problems.

> This, however, leads me into thinking a bit on the lifecycle management 
> for GRUUs. Currently, GRUU is bound to a single registration, where the 
> registration is defined by the Contact URI. If the UA should "move", in 
> the sense that it acquires a new IP address, this will be a new Contact 
> URI, and therefore, a new GRUU. It would be nice if, instead, the GRUU 
> were truly bound to the UA instance, and that even if the UA instance 
> changed IP addresses, the GRUU could remain unchanged. In such a case, 
> if the UA moves, there is actually no need to issue any target refresh 
> requests. The UA would simply re-register, update the binding of its 
> GRUU to its new IP address, and existing calls would be correctly routed.
> 
> Indeed, this has the interesting benefit that, even if both UA in a call 
> should move at the same time, the call would still be able to continue, 
> since such updates are never propagated to the peer in the call.

This seems to presume that address changes of this sort are common, and 
a frequent source of target refreshes.

I don't have any useful data on the subject, but it seems likely to me 
that other reasons for target refreshes would be at least as 
significant. For instance load balancing.

> For this to work, it would require that the registration have a way of 
> identifying the UA instance. This goes back to the instance identifier 
> we had (but removed) from callee-caps. If we added it, we could then 
> associate the GRUU with this ID, rather than the Contact URI. In other 
> words, if the client registered thusly:
> 
> REGISTER
> To: sip:user@example.com
> Contact: sip:1.2.3.4;+instance-id="9asdd9a887sdasd=asd888a"
> Supported: gruu
> 
> the registrar would allocate a GRUU, and associate it with that instance 
> ID:
> 
> 200 OK
> Contact: 
> sip:1.2.3.4;+instance-id="9asdd9a887sdasd=asd888a";gruu=sip:f4@example.com
> 
> Now, if the UA should update its registration because it moves:
> 
> REGISTER
> To: sip:user@example.com
> Contact: sip:5.6.7.8;+instance-id="9asdd9a887sdasd=asd888a"
> Contact: sip:1.2.3.4;expires=0
> 
> the registrar would return the *same* gruu for the new registration:
> 
> 200 OK
> Contact: 
> sip:5.6.7.8;+instance-id="9asdd9a887sdasd=asd888a";gruu=sip:f4@example.com
> 
> 
> It is still possible for the registrar to not store additional state if 
> GRUU is defined this way. The GRUU URI would be computed thusly:
> 
> sip:<E(salt,instance-id)>@example.com
> 
> That is, the GRUU now encodes the instance ID. When a request is 
> received with a GRUU, the registrar extracts the instance ID. It then 
> looks up the registration corresponding to that instance ID, and obtains 
> its Contact URI. This doesnt require additional state per se, but does 
> require that the registration data be structured to allow lookups based 
> on the instance-id.
> 
> Thoughts?

In spite of my comment above, I think I like the general approach. 
However there may still be details to work out. Notably, your stateless 
algorithm won't work right if a UA uses the same instance id when 
registering with multiple AORs managed by the same registrar.

(It could be fixed by using: sip:<E(salt,instance-id,AOR)>@example.com.)

Going this way will also require careful specification of instance-id, 
but that is a useful exercise in any case. It will be important to 
specify what the uniqueness properties are for instance-ids, and to 
investigate what the implications are when they are violated. (Since it 
is easy to predict that they will be violated.)

The implications on registrar data storage and retrieval are 
significant, but in practice most registrars will require some alternate 
access path like this in any case.

	Paul


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Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target refresh request
Date: Tue, 6 Jan 2004 10:25:14 -0500
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Thread-Topic: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target refresh request
Thread-Index: AcPUJWXja5Y3y8BIS1mq255uvfkDnwAQkKLA
From: "Roy, Radhika R, ALABS" <rrroy@att.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: <Jussi.Turunen@nokia.com>, <nataraju.alilaghatta@wipro.com>,
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Hi, Jonathan:

I agree. In fact, this should be the basic principle to address the
mobility in the application layer. That is, the URL (e.g.,
rrroy@att.com) will remain the same no matter where the use moves,
however, the IP address (layer 3) or addresses of the other lower layers
may change.

The change in lower "physical" addresses is called as the "change in
point of contact." The point of contact at different (lower) layer may
change for the mobile user (specially in the case of the terminal
mobility), but its impact may or may not propagate in all upper layers.

If the URL would have been kept same under all circumstances, it would
have been the most elegant solution especially to address the problems
of the terminal mobility.

br,
Radhika

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Tuesday, January 06, 2004 2:08 AM
To: Paul Kyzivat
Cc: Jussi.Turunen@nokia.com; nataraju.alilaghatta@wipro.com;
sanjsinh@cisco.com; sip@ietf.org
Subject: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and
target refresh request


inline.

Paul Kyzivat wrote:


>> However, I think its a much better idea to send a second subscription

>> on a different dialog, rather than trying to reuse an existing
dialog.
>=20
>=20
> I think this actually depends on intent. If the subscriptions are=20
> intended to be independent then I agree. OTOH, if it is important that

> the subscriptions always be associated with the same UA even as that=20
> moves around, then there is an advantage to using a single dialog.
>=20
> Its easier to find such cases in the sharing of a dialog between an=20
> invite and a subscribe than it is between two subscribes.

The GRUU would allow you to keep both dialogs attached to a particular=20
UA instance. Should the UA move around, it would require you to=20
generate a target refresh request on each dialog to update the Contact.

This, however, leads me into thinking a bit on the lifecycle=20
management for GRUUs. Currently, GRUU is bound to a single=20
registration, where the registration is defined by the Contact URI. If=20
the UA should "move", in the sense that it acquires a new IP address,=20
this will be a new Contact URI, and therefore, a new GRUU. It would be=20
nice if, instead, the GRUU were truly bound to the UA instance, and=20
that even if the UA instance changed IP addresses, the GRUU could=20
remain unchanged. In such a case, if the UA moves, there is actually=20
no need to issue any target refresh requests. The UA would simply=20
re-register, update the binding of its GRUU to its new IP address, and=20
existing calls would be correctly routed.

Indeed, this has the interesting benefit that, even if both UA in a=20
call should move at the same time, the call would still be able to=20
continue, since such updates are never propagated to the peer in the=20
call.

For this to work, it would require that the registration have a way of=20
identifying the UA instance. This goes back to the instance identifier=20
we had (but removed) from callee-caps. If we added it, we could then=20
associate the GRUU with this ID, rather than the Contact URI. In other=20
words, if the client registered thusly:

REGISTER
To: sip:user@example.com
Contact: sip:1.2.3.4;+instance-id=3D"9asdd9a887sdasd=3Dasd888a"
Supported: gruu

the registrar would allocate a GRUU, and associate it with that=20
instance ID:

200 OK
Contact:=20
sip:1.2.3.4;+instance-id=3D"9asdd9a887sdasd=3Dasd888a";gruu=3Dsip:f4@exam=
ple.c
om

Now, if the UA should update its registration because it moves:

REGISTER
To: sip:user@example.com
Contact: sip:5.6.7.8;+instance-id=3D"9asdd9a887sdasd=3Dasd888a"
Contact: sip:1.2.3.4;expires=3D0

the registrar would return the *same* gruu for the new registration:

200 OK
Contact:=20
sip:5.6.7.8;+instance-id=3D"9asdd9a887sdasd=3Dasd888a";gruu=3Dsip:f4@exam=
ple.c
om


It is still possible for the registrar to not store additional state=20
if GRUU is defined this way. The GRUU URI would be computed thusly:

sip:<E(salt,instance-id)>@example.com

That is, the GRUU now encodes the instance ID. When a request is=20
received with a GRUU, the registrar extracts the instance ID. It then=20
looks up the registration corresponding to that instance ID, and=20
obtains its Contact URI. This doesnt require additional state per se,=20
but does require that the registration data be structured to allow=20
lookups based on the instance-id.

Thoughts?

-Jonathan R.
>=20
>     Paul
>=20
>> -Jonathan R.
>>
>> Jussi.Turunen@nokia.com wrote:
>>
>>> Hi
>>>
>>>
>>>> For a dialog initiated with SUBSCRIBE, can a subscription for a new

>>>> event on that dialog also be a target refresh request?
>>>>
>>>> [Natraj] No, it's not a Target refresh request. Target refresh
could
>>>> only happen either through INVITE or UPDATE requests..
>>>
>>>
>>>
>>>
>>> There is a bug in RFC3265. Bug description=20
>>> (http://bugs.sipit.net/sipwg/show_bug.cgi?id=3D699) says:
>>>
>>> "NOTIFY and SUBSCRIBE are target refresh requests
>>>
>>> ...but the text never says this explicitly."
>>>
>>> Haven't seen this bug closed yet... I remember some discussion on=20
>>> this on this a while (a year or so) ago, so you might want to check=20
>>> the archives for that.
>>>
>>>         Jussi
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/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
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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

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From: "George Foti (QC/EMC)" <george.foti@ericsson.com>
To: "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>, sip@ietf.org
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Subject: [Sip] Clarification  on Presence Use Case in draft-ietf-sip-gruu-00
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Hi,

The presence use case (section 4.3) indicates that the GRUU, representing the contact URI in the tuple information included in the presence document, needs to be generated by a presence agent.

I am unclear as to why this needs to be the case.

The UA can publish a GRUU it acquired from a registrar to the presence agent.
The registrar will be consulted at call routing to make sure that it gets routed to the proper UA associated with the GRUU.

Did I misunderstand something ?

Thanks/gf 


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Date: Tue, 6 Jan 2004 15:48:41 -0500 (EST)
From: "Ana Elisa P. Goulart" <ana@ece.gatech.edu>
To: sip@ietf.org
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Subject: [Sip] Question about the beginning of the media session
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Hi,
I have a basic SIP question about the actual beginning of the media
session following an INVITE transaction. In section 13.1 (Initiating a
Session), the RFC says that "a 2xx response to an INVITE establishes a
session".
My question is whether this applies to the time the UAC receives the final
2xx response or the time the UAS sends the 2xx response or both.
In other words, from the UAS's perspective, is it correct to assume that
the actual media transmission can begin after the 2xx response
is sent (before the ACK is received) ?

Thanks a lot,
Ana

--
Ana Elisa P. Goulart
The School of Electrical and Computer Engineering at Georgia Tech
Atlanta, GA  30332-0250
E-Mail: ana@ece.gatech.edu, gt8722a@prism.gatech.edu

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        nataraju.alilaghatta@wipro.com, sanjsinh@cisco.com, sip@ietf.org
Subject: Re: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target
 refresh request
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Roy, Radhika R, ALABS wrote:

> Hi, Jonathan:
> 
> I agree. In fact, this should be the basic principle to address the
> mobility in the application layer. That is, the URL (e.g.,
> rrroy@att.com) will remain the same no matter where the use moves,
> however, the IP address (layer 3) or addresses of the other lower layers
> may change.
> 
> The change in lower "physical" addresses is called as the "change in
> point of contact." The point of contact at different (lower) layer may
> change for the mobile user (specially in the case of the terminal
> mobility), but its impact may or may not propagate in all upper layers.
> 
> If the URL would have been kept same under all circumstances, it would
> have been the most elegant solution especially to address the problems
> of the terminal mobility.

Right. The principal value of this approach would be to allow for 
application layer SIP mobility. Effectively, the Contact URI/GRUU 
would join the AOR as a URI that is independent of the point of 
attachment of the UA.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Paul Kyzivat wrote:

> 

>>
>> The GRUU would allow you to keep both dialogs attached to a particular 
>> UA instance. Should the UA move around, it would require you to 
>> generate a target refresh request on each dialog to update the Contact.
> 
> 
> Yes, this is another way to solve that problem. I am not especially 
> attached to using shared dialogs this way once all the machinery to do 
> otherwise is in place. OTOH, I don't think shared dialogs are going 
> away, so they remain *a* way to solve this or related problems.

Well, I would prefer one way rather than two, but we've already beaten 
this horse a bit...

> 
>> This, however, leads me into thinking a bit on the lifecycle 
>> management for GRUUs. Currently, GRUU is bound to a single 
>> registration, where the registration is defined by the Contact URI. If 
>> the UA should "move", in the sense that it acquires a new IP address, 
>> this will be a new Contact URI, and therefore, a new GRUU. It would be 
>> nice if, instead, the GRUU were truly bound to the UA instance, and 
>> that even if the UA instance changed IP addresses, the GRUU could 
>> remain unchanged. In such a case, if the UA moves, there is actually 
>> no need to issue any target refresh requests. The UA would simply 
>> re-register, update the binding of its GRUU to its new IP address, and 
>> existing calls would be correctly routed.
>>
>> Indeed, this has the interesting benefit that, even if both UA in a 
>> call should move at the same time, the call would still be able to 
>> continue, since such updates are never propagated to the peer in the 
>> call.
> 
> 
> This seems to presume that address changes of this sort are common, and 
> a frequent source of target refreshes.

Right. In particular I was thinking about application layer mobility. 
For WiFi SIP phones, this would be particularly attractive, I think.

> 
> I don't have any useful data on the subject, but it seems likely to me 
> that other reasons for target refreshes would be at least as 
> significant. For instance load balancing.

Sure. There too, you get interesting benefits from using GRUUs that 
are not bound to a specific IP address. You could migrate 
subscriptions from one server to another without needing to inform the 
watcher; you simply change the routing logic for the gruu to instead 
route to the server that will now handle the subscription (of course, 
you would have to have taken care of transferring subscription state 
in all cases).

> 
>> For this to work, it would require that the registration have a way of 
>> identifying the UA instance. This goes back to the instance identifier 
>> we had (but removed) from callee-caps. If we added it, we could then 
>> associate the GRUU with this ID, rather than the Contact URI. In other 
>> words, if the client registered thusly:
>>
>> REGISTER
>> To: sip:user@example.com
>> Contact: sip:1.2.3.4;+instance-id="9asdd9a887sdasd=asd888a"
>> Supported: gruu
>>
>> the registrar would allocate a GRUU, and associate it with that 
>> instance ID:
>>
>> 200 OK
>> Contact: 
>> sip:1.2.3.4;+instance-id="9asdd9a887sdasd=asd888a";gruu=sip:f4@example.com 
>>
>>
>> Now, if the UA should update its registration because it moves:
>>
>> REGISTER
>> To: sip:user@example.com
>> Contact: sip:5.6.7.8;+instance-id="9asdd9a887sdasd=asd888a"
>> Contact: sip:1.2.3.4;expires=0
>>
>> the registrar would return the *same* gruu for the new registration:
>>
>> 200 OK
>> Contact: 
>> sip:5.6.7.8;+instance-id="9asdd9a887sdasd=asd888a";gruu=sip:f4@example.com 
>>
>>
>>
>> It is still possible for the registrar to not store additional state 
>> if GRUU is defined this way. The GRUU URI would be computed thusly:
>>
>> sip:<E(salt,instance-id)>@example.com
>>
>> That is, the GRUU now encodes the instance ID. When a request is 
>> received with a GRUU, the registrar extracts the instance ID. It then 
>> looks up the registration corresponding to that instance ID, and 
>> obtains its Contact URI. This doesnt require additional state per se, 
>> but does require that the registration data be structured to allow 
>> lookups based on the instance-id.
>>
>> Thoughts?
> 
> 
> In spite of my comment above, I think I like the general approach. 
> However there may still be details to work out. Notably, your stateless 
> algorithm won't work right if a UA uses the same instance id when 
> registering with multiple AORs managed by the same registrar.
> 
> (It could be fixed by using: sip:<E(salt,instance-id,AOR)>@example.com.)

This is much better. Another benefit of doing this is that you could 
extract the AOR, look up the user, and then just do a search of the 
existing contacts to see which one matched the instance-id. It would 
mean that one could add support for gruus, and be stateless, without 
changing the existing schemas for the back-end data stores (I am 
making some assumptions about how the schema looks).

> 
> Going this way will also require careful specification of instance-id, 
> but that is a useful exercise in any case. It will be important to 
> specify what the uniqueness properties are for instance-ids, and to 
> investigate what the implications are when they are violated. (Since it 
> is easy to predict that they will be violated.)

We can also be sure that there will be cases where the same UA 
instance uses different instance IDs over time. This is particularly 
likely with software agents.

> 
> The implications on registrar data storage and retrieval are 
> significant, but in practice most registrars will require some alternate 
> access path like this in any case.

See above - I think with your proposal they can be minimized.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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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		: Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-sip-callee-caps-03.txt
	Pages		: 45
	Date		: 2004-1-6
	
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Subject: [Sip] Protocol Action: 'Indicating User Agent Capabilities in the
 Session Initiation Protocol (SIP)' to Proposed Standard
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The IESG has approved following document:

- 'Indicating User Agent Capabilities in the Session Initiation Protocol 
(SIP) '
   <draft-ietf-sip-callee-caps-03.txt> as a Proposed Standard

This document is the product of the Session Initiation Protocol Working 
Group. 

The IESG contact persons are Allison Mankin and Jon Peterson.

Technical Summary

  This specification defines mechanisms by which a Session Initiation
   Protocol (SIP) user agent can convey its capabilities and
   characteristics to other user agents and to the registrar for its
   domain. This information is conveyed as parameters of the Contact
   header field.  It may be used by a proxy for call routing. The parameter
   design is based on RFC 2506, media feature tags.  The specification
   creates a SIP tree registry parallel to the IETF tree registry from
   RFC 2506.  The syntax of the parameters is based on RFC 2533.

   Strong considerations regarding the privacy and data
   integrity of the information are discussed by the document.
 
Working Group Summary
 
   The working group took a lot of care and review developing this design.
    There was a mid-course design review from an Applications area
    standpoint that resulted in the advice to work with the RFC 2533,
    2506 approach, which has proved to be very constructive.  The WG
    supported the advancement strongly, after thorough review.
 
Protocol Quality
 
   The document was reviewed for the IESG by Allison Mankin.  The
    Applications mid-course reviews were by Patrik Faltstrom and Ted
    Hardie.


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	Title		: Session Initiation Protocol (SIP) Extension for Event 
			  State Publication
	Author(s)	: A. Niemi
	Filename	: draft-ietf-sip-publish-02.txt
	Pages		: 39
	Date		: 2004-1-7
	
This document describes an extension to the Session Initiation
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is targeted at the publication of presence information.
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	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-00.txt
	Pages		: 20
	Date		: 2004-1-7
	
Several applications of the Session Initiation Protocol (SIP) require
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by anyone on the Internet to route a call to that specific UA
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Date: Wed, 07 Jan 2004 23:04:41 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sip] Re: Clarification  on Presence Use Case in draft-ietf-sip-gruu-00
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inline.

George Foti (QC/EMC) wrote:

> Hi,
> 
> The presence use case (section 4.3) indicates that the GRUU,
> representing the contact URI in the tuple information included in
> the presence document, needs to be generated by a presence agent.
> 
> I am unclear as to why this needs to be the case.
> 
> The UA can publish a GRUU it acquired from a registrar to the
> presence agent. The registrar will be consulted at call routing to
> make sure that it gets routed to the proper UA associated with the
> GRUU.
> 
> Did I misunderstand something ?

It depends on how the composition is done. There may be cases where 
the PA needs to construct a URI that was no supplied by the UA. In 
other cases, a published GRUU would suffice. I've clarified. The text 
now reads:

It is interesting to note that the GRUU may need to be
constructed by a presence agent, depending on how the presence document is
computed by the server.



I think its probably a good idea for the presence document use cases 
draft being done in SIMPLE to talk about publishing presence documents 
with gruus.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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Comments inline
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Wednesday, January 07, 2004 11:05 PM
> To: George Foti (QC/EMC)
> Cc: sip@ietf.org
> Subject: Re: Clarification on Presence Use Case in
> draft-ietf-sip-gruu-00
> 
> 
> inline.
> 
> George Foti (QC/EMC) wrote:
> 
> > Hi,
> > 
> > The presence use case (section 4.3) indicates that the GRUU,
> > representing the contact URI in the tuple information included in
> > the presence document, needs to be generated by a presence agent.
> > 
> > I am unclear as to why this needs to be the case.
> > 
> > The UA can publish a GRUU it acquired from a registrar to the
> > presence agent. The registrar will be consulted at call routing to
> > make sure that it gets routed to the proper UA associated with the
> > GRUU.
> > 
> > Did I misunderstand something ?
> 
> It depends on how the composition is done. There may be cases where 
> the PA needs to construct a URI that was no supplied by the UA. In 
> other cases, a published GRUU would suffice. I've clarified. The text 
> now reads:
> 
> It is interesting to note that the GRUU may need to be
> constructed by a presence agent, depending on how the 
> presence document is
> computed by the server.
> 
I like that better.
> 
> 
> I think its probably a good idea for the presence document use cases 
> draft being done in SIMPLE to talk about publishing presence 
> documents 
> with gruus.
> 
It is definitely needed to be added in there.
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 

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From: "Suha Cubukcuoglu" <suhac@nplusone.net>
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Subject: [Sip] 1xx Provisional Responses
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Hi,

In the RFC for SIP (#3261) it is stated that 1xx provisional responses
are not transmitted reliably and never cause the client to send an ACK
message. SIP clients use UDP to send messages across networks.
Therefore, for example, '180 Ringing' message may or may reach the
calling party without the called party recognizing whether the message
has reached its destination. This has adverse effects in terms of call
setup and usability. If the 180 Ringing message is lost in the network
while the called party is ringing, the calling party will have no idea
what state the call is in at any time (e.g. one could think they dialed
a wrong number, or its taking too long to contact the other end, etc.).
'180 Ringing' message, in our opinion, should be treated as a critical
message just like INVITE and 200 OK or the called party must send
multiple '180 Ringing' messages, instead of just one, to make sure the
other end will receive the message one way or other. 

Could you please tell us your comments about this?

Thanks very much.

Regards,

Suha Cubukcuoglu
nplusone
t: +44 131 514 2000
d: +44 131 514 4037

-- 

This email and any files transmitted with it are confidential and intended
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Use sipping@ietf.org for new developments on the application of sip



From exim@www1.ietf.org  Thu Jan  8 10:32:40 2004
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Subject: RE: [Sip] 1xx Provisional Responses
Date: Thu, 8 Jan 2004 17:18:48 +0200
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB701797595@esebe019.ntc.nokia.com>
Thread-Topic: [Sip] 1xx Provisional Responses
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Cc: <ednet-telco@mailman.ednet.co.uk>
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You can certainly send 180 responses periodically until you send a final =
response.

If you really want to send provisional responses reliably, then please =
look at RFC3262.

Regards,
Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Suha Cubukcuoglu
> Sent: 08.January.2004 17:06
> To: sip@ietf.org
> Cc: edNET Telco
> Subject: [Sip] 1xx Provisional Responses
>=20
>=20
> Hi,
>=20
> In the RFC for SIP (#3261) it is stated that 1xx provisional responses
> are not transmitted reliably and never cause the client to send an ACK
> message. SIP clients use UDP to send messages across networks.
> Therefore, for example, '180 Ringing' message may or may reach the
> calling party without the called party recognizing whether the message
> has reached its destination. This has adverse effects in terms of call
> setup and usability. If the 180 Ringing message is lost in the network
> while the called party is ringing, the calling party will have no idea
> what state the call is in at any time (e.g. one could think=20
> they dialed
> a wrong number, or its taking too long to contact the other=20
> end, etc.).
> '180 Ringing' message, in our opinion, should be treated as a critical
> message just like INVITE and 200 OK or the called party must send
> multiple '180 Ringing' messages, instead of just one, to make sure the
> other end will receive the message one way or other.=20
>=20
> Could you please tell us your comments about this?
>=20
> Thanks very much.
>=20
> Regards,
>=20
> Suha Cubukcuoglu
> nplusone
> t: +44 131 514 2000
> d: +44 131 514 4037
>=20
> --=20
>=20
> This email and any files transmitted with it are confidential=20
> and intended
> solely for the use of the individual or entity to whom they=20
> are addressed.
> If you have received this email in error please notify the sender. Any
> offers or quotation of service are subject to formal specification.
> Errors and omissions excepted.  Please note that any views or opinions
> presented in this email are solely those of the author and do not
> necessarily represent those of edNET or lightershade ltd. Finally, the
> recipient should check this email and any attachments for the=20
> presence of
> viruses.  edNET and lightershade ltd accepts no liability for=20
> any damage
> caused by any virus transmitted by this email.
>=20
> --=20
> --=20
> Virus scanned by edNET.
>=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

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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: "Greg Scallan" <spider@tellme.com>
To: <suhac@ednet.co.uk>, <sip@ietf.org>
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Subject: RE: [Sip] 1xx Provisional Responses
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See RFC 3262 (http://www.ietf.org/rfc/rfc3262.txt), Reliability of =
Provisional Responses in the Session Initiation Protocol (SIP).

greg

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Suha
Cubukcuoglu
Sent: Thursday, January 08, 2004 10:06 AM
To: sip@ietf.org
Cc: edNET Telco
Subject: [Sip] 1xx Provisional Responses


Hi,

In the RFC for SIP (#3261) it is stated that 1xx provisional responses
are not transmitted reliably and never cause the client to send an ACK
message. SIP clients use UDP to send messages across networks.
Therefore, for example, '180 Ringing' message may or may reach the
calling party without the called party recognizing whether the message
has reached its destination. This has adverse effects in terms of call
setup and usability. If the 180 Ringing message is lost in the network
while the called party is ringing, the calling party will have no idea
what state the call is in at any time (e.g. one could think they dialed
a wrong number, or its taking too long to contact the other end, etc.).
'180 Ringing' message, in our opinion, should be treated as a critical
message just like INVITE and 200 OK or the called party must send
multiple '180 Ringing' messages, instead of just one, to make sure the
other end will receive the message one way or other.=20

Could you please tell us your comments about this?

Thanks very much.

Regards,

Suha Cubukcuoglu
nplusone
t: +44 131 514 2000
d: +44 131 514 4037

--=20

This email and any files transmitted with it are confidential and =
intended
solely for the use of the individual or entity to whom they are =
addressed.
If you have received this email in error please notify the sender. Any
offers or quotation of service are subject to formal specification.
Errors and omissions excepted.  Please note that any views or opinions
presented in this email are solely those of the author and do not
necessarily represent those of edNET or lightershade ltd. Finally, the
recipient should check this email and any attachments for the presence =
of
viruses.  edNET and lightershade ltd accepts no liability for any damage
caused by any virus transmitted by this email.

--=20
--=20
Virus scanned by edNET.

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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 exim@www1.ietf.org  Thu Jan  8 10:48:42 2004
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From: "Reddy, Muralidhar" <MReddy@telogy.com>
To: "'suhac@ednet.co.uk'" <suhac@ednet.co.uk>, sip@ietf.org
Cc: edNET Telco <ednet-telco@mailman.ednet.co.uk>
Subject: RE: [Sip] 1xx Provisional Responses
Date: Thu, 8 Jan 2004 10:30:39 -0500 
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Hello,

    We too had lot of discussion regarding when calling party should start
ring back tone (RBT)?
1. Trigger RBT as soon as calling party sends Invite message. But Phone
starts ringing for any number dialed.
Or

2. Trigger RBT only when you receive 180 Ringing. As 180 is optional
message, calling party can get connected without receiving 180 ringing
message.
Or

3. Trigger RBT when calling part receives 100 trying from the Proxy which is
always true in the network. 

There is no strict rule defined in RFC regarding when calling party should
start RBT.

Murali

-----Original Message-----
From: Suha Cubukcuoglu [mailto:suhac@nplusone.net] 
Sent: Thursday, January 08, 2004 10:06 AM
To: sip@ietf.org
Cc: edNET Telco
Subject: [Sip] 1xx Provisional Responses

Hi,

In the RFC for SIP (#3261) it is stated that 1xx provisional responses
are not transmitted reliably and never cause the client to send an ACK
message. SIP clients use UDP to send messages across networks.
Therefore, for example, '180 Ringing' message may or may reach the
calling party without the called party recognizing whether the message
has reached its destination. This has adverse effects in terms of call
setup and usability. If the 180 Ringing message is lost in the network
while the called party is ringing, the calling party will have no idea
what state the call is in at any time (e.g. one could think they dialed
a wrong number, or its taking too long to contact the other end, etc.).
'180 Ringing' message, in our opinion, should be treated as a critical
message just like INVITE and 200 OK or the called party must send
multiple '180 Ringing' messages, instead of just one, to make sure the
other end will receive the message one way or other. 

Could you please tell us your comments about this?

Thanks very much.

Regards,

Suha Cubukcuoglu
nplusone
t: +44 131 514 2000
d: +44 131 514 4037

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Thu Jan  8 10:48:42 2004
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Date: Thu, 8 Jan 2004 10:28:43 -0500 (EST)
From: Srinivasan Krishnamoorthy <ksrini@motorola.com>
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To: suhac@ednet.co.uk
cc: sip@ietf.org, edNET Telco <ednet-telco@mailman.ednet.co.uk>
Subject: Re: [Sip] 1xx Provisional Responses
In-Reply-To: <003a01c3d5f8$f326fc50$e3e714d4@Azopardi>
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Not sure if you looked at RFC3262 - specifies an extension
(100rel) to support reliability of Provisional Responses.

-Srini

On Thu, 8 Jan 2004, Suha Cubukcuoglu wrote:

suhac>Hi,
suhac>
suhac>In the RFC for SIP (#3261) it is stated that 1xx provisional responses
suhac>are not transmitted reliably and never cause the client to send an ACK
suhac>message. SIP clients use UDP to send messages across networks.
suhac>Therefore, for example, '180 Ringing' message may or may reach the
suhac>calling party without the called party recognizing whether the message
suhac>has reached its destination. This has adverse effects in terms of call
suhac>setup and usability. If the 180 Ringing message is lost in the network
suhac>while the called party is ringing, the calling party will have no idea
suhac>what state the call is in at any time (e.g. one could think they dialed
suhac>a wrong number, or its taking too long to contact the other end, etc.).
suhac>'180 Ringing' message, in our opinion, should be treated as a critical
suhac>message just like INVITE and 200 OK or the called party must send
suhac>multiple '180 Ringing' messages, instead of just one, to make sure the
suhac>other end will receive the message one way or other. 
suhac>
suhac>Could you please tell us your comments about this?
suhac>
suhac>Thanks very much.
suhac>
suhac>Regards,
suhac>
suhac>Suha Cubukcuoglu
suhac>nplusone
suhac>t: +44 131 514 2000
suhac>d: +44 131 514 4037
suhac>
suhac>


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From exim@www1.ietf.org  Fri Jan  9 04:25:02 2004
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From: vvarun kumar <arunvv@yahoo.com>
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Hi Jonathan,
I am a S/W Engr working on SIP.
I am new to SIP so please guide me as I have seen
solving these type of answers by u in IETF
mailarchive.
So I am mailing to u regarding this issue so that I
can make a step ahead in my project.

The problem is I am able to register with registrar
and get the 200(OK) after registering.
But when Iam trying to register with redirect server
using aliases I am not able to register.I am using
OPAL (PWLIB and OPenH323) software running in LINUX.
Can u suggest How I can register the redirectserver
using alias.

Thank u,
Regards,
Arunkumar,
Actinium Networks.


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_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Jonathan Rosenberg" <jdrosen@bell-labs.com>,
        "Toip list" <toip@snowshore.com>
Cc: <sip@ietf.org>
Date: Thu, 8 Jan 2004 00:10:29 +0100
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Subject: [Sip] Text well covered in : I-D ACTION:draft-ietf-sip-callee-caps-03.txt
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It is good to see the text medium listed among other media in the
callee-caps specification.
The spec provides an opportunity to declare capabilities for the real tim=
e
conversational media
of video, text and audio to be known at SIP registry time.

I hope this will stimulate to smart applications, invoking transcoding
services when needed, selecting the proper terminal among a group of call=
ed
ones on the same address etc.

I also hope that other editors and authors will remember to include real
time conversational text as a natural component in other specs for SIP ba=
sed
real time services. It is so important that this medium is included from =
the
beginning, and not introduced as a cumbersome afterthought as happened to=
o
many times before.

Thanks

Gunnar Hellstr=F6m
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: owner-ietf-announce@ietf.org
[mailto:owner-ietf-announce@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Wednesday, January 07, 2004 3:03 PM
To: IETF-Announce:
Cc: sip@ietf.org
Subject: I-D ACTION:draft-ietf-sip-callee-caps-03.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Protocol Working Grou=
p
of the IETF.

	Title		: Indicating User Agent Capabilities in the Session Initiation
Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-sip-callee-caps-03.txt
	Pages		: 45
	Date		: 2004-1-6

This specification defines mechanisms by which a Session Initiation
Protocol (SIP) user agent can convey its capabilities and
characteristics to other user agents. These capabilities are conveyed
as parameters of the Contact header field.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-callee-caps-03.txt

To remove yourself from the IETF Announcement list, send a message to
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e.

Internet-Drafts are also available by anonymous FTP. Login with the usern=
ame
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
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A list of Internet-Drafts directories can be found in
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-----------------------------------------------------------------------
The IESG has approved following document:

- 'Indicating User Agent Capabilities in the Session Initiation Protocol
(SIP) '
   <draft-ietf-sip-callee-caps-03.txt> as a Proposed Standard

This document is the product of the Session Initiation Protocol Working
Group.

The IESG contact persons are Allison Mankin and Jon Peterson.

Technical Summary

  This specification defines mechanisms by which a Session Initiation
   Protocol (SIP) user agent can convey its capabilities and
   characteristics to other user agents and to the registrar for its
   domain. This information is conveyed as parameters of the Contact
   header field.  It may be used by a proxy for call routing. The paramet=
er
   design is based on RFC 2506, media feature tags.  The specification
   creates a SIP tree registry parallel to the IETF tree registry from
   RFC 2506.  The syntax of the parameters is based on RFC 2533.

   Strong considerations regarding the privacy and data
   integrity of the information are discussed by the document.

Working Group Summary

   The working group took a lot of care and review developing this design.
    There was a mid-course design review from an Applications area
    standpoint that resulted in the advice to work with the RFC 2533,
    2506 approach, which has proved to be very constructive.  The WG
    supported the advancement strongly, after thorough review.

Protocol Quality

   The document was reviewed for the IESG by Allison Mankin.  The
    Applications mid-course reviews were by Patrik Faltstrom and Ted
    Hardie.



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_______________________________________________
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Subject: RE: [Sip] Re:Redirect server in SIP
Date: Fri, 9 Jan 2004 15:57:11 +0530
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Hi Arun

It is better if u put such questions on sip-implementors or openH323
mailing lists.=20
Answering to ur question, I suggest u go through articles on SIP and RFC
3261 to get a good understaning on SIP and also through openH323
documentation to get a understanding on portable windows library (pwlib)
and h323 code.

then u should be in a position to solve the problems you are facing.

Regards
Ranjit





-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of vvarun
kumar
Sent: Monday, January 05, 2004 2:18 PM
To: sip@ietf.org
Subject: [Sip] Re:Redirect server in SIP


Hi Jonathan,
I am a S/W Engr working on SIP.
I am new to SIP so please guide me as I have seen
solving these type of answers by u in IETF
mailarchive.
So I am mailing to u regarding this issue so that I
can make a step ahead in my project.

The problem is I am able to register with registrar
and get the 200(OK) after registering.
But when Iam trying to register with redirect server
using aliases I am not able to register.I am using
OPAL (PWLIB and OPenH323) software running in LINUX.
Can u suggest How I can register the redirectserver
using alias.

Thank u,
Regards,
Arunkumar,
Actinium Networks.


__________________________________
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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
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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



From exim@www1.ietf.org  Fri Jan  9 08:08:45 2004
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: SIP WG <sip@ietf.org>
Date: Fri, 9 Jan 2004 12:55:56 -0000 
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Subject: [Sip] Server Location and trying requests bug?
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All,

I believe there is an issue with Server Location that makes it impossible to
reroute requests successfully in the following scenario.

If the request reaches the server, but the resulting response does not reach
the client (perhaps a NAT blocked its path) then Server Location processing
cannot be used to find a different route via which the request can be
completed and result in a successful response to the client.

Could people please take a look at this, see if they have already
encountered
it and it so, let me know if they have suggestions for a solution?

At SIPit13 we encountered the following problem.

- Send a request using UDP
- Request reached the server but the response could not get back for some
  reason
- Client timed out and then tried TCP, changing the branch
- New request reached the server which complained because the request did 
  appear to increment the CSeq.

These were specifically REGISTER requests but I believe this affects other
requests, especially those in-dialog and INVITEs.

Clearly the server was operating correctly and so there was some discussion
as
to whether the client was at fault. RFC3261 only states the following
(RFC3261, 8.1.2, p42).

   The UAC SHOULD follow the procedures defined in [4] for stateful
   elements, trying each address until a server is contacted.  Each try
   constitutes a new transaction, and therefore each carries a different
   topmost Via header field value with a new branch parameter.
   Furthermore, the transport value in the Via header field is set to
   whatever transport was determined for the target server.

However it was decided that in fact the CSeq of the transaction should also
have been incremented by the client when sending the new transaction over
TCP.

But now consider what would happen if the CSeq were updated in this manner
but there was an intervening proxy.

- Client sends request which reaches proxy
- Proxy forwards on using UDP
- Request reaches server but response cannot get back
- Proxy times out UDP transaction and now tries TCP, changing the branch and
  the CSeq
- New request reaches the server, which is happy and responds
- Response now reaches proxy but the original request had Cseq X and the
  response has CSeq (X+1)

Now the proxy might be able to massage the response Cseq back to X but then
we
have the strange situation that the client believes the next CSeq should be
(X+1) whilst the server believes it should be (X+2).  Clearly, a subsequent
in-dialog reINVITE from the client will fail because the server does not
believe the Cseq has been incremented.

Clearly, without incrementing Cseq, we will hit the "response not returned,
subsequent request fails" problem in both the UAC/UAS and UAC/proxy/UAS
cases.
But incrementing the CSeq only fixes the UAC/UAS case and introduces more
problems in the UAC/proxy/UAS case.  For this reason we believe the
discussed
solution is wrong.

At the moment, this seems to be a problem that RFC3261 fails to address and
we
have yet to determine a simple fix.

Thanks,
Paul DS.

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




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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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Message-Id: <200401091402.i09E2eIr066494@smtp.eweka.nl>
From: "Arnoud van Wijk" <a.vwijk@viataal.nl>
To: "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        "'Jonathan Rosenberg'" <jdrosen@bell-labs.com>,
        "'Toip list'" <toip@snowshore.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Text well covered in : I-D ACTION:draft-ietf-sip-callee-caps-03.txt
Date: Fri, 9 Jan 2004 14:38:34 +0100
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Maybe it is a good idea to hold a small presentation reminding everybody =
at
the IETF that real time conversational text is a natural component like
video and audio.

Which WG would be the best?

I am open to suggestions.


Drs. Arnoud A. T. van Wijk=20
Viataal=20
Research & Development=20
Afdeling RDS=20
Theerestraat 42=20
5271 GD Sint-Michielsgestel=20
The Netherlands.=20
Mobile: +31651921948=20
International text telephone: +31735588408=20

-----Original Message-----
From: owner-toip@snowshore.com [mailto:owner-toip@snowshore.com] On =
Behalf
Of Gunnar Hellstrom
Sent: donderdag 8 januari 2004 0:10
To: Jonathan Rosenberg; Toip list
Cc: sip@ietf.org
Subject: [Sip] Text well covered in : I-D
ACTION:draft-ietf-sip-callee-caps-03.txt

It is good to see the text medium listed among other media in the
callee-caps specification.
The spec provides an opportunity to declare capabilities for the real =
time
conversational media
of video, text and audio to be known at SIP registry time.

I hope this will stimulate to smart applications, invoking transcoding
services when needed, selecting the proper terminal among a group of =
called
ones on the same address etc.

I also hope that other editors and authors will remember to include real
time conversational text as a natural component in other specs for SIP =
based
real time services. It is so important that this medium is included from =
the
beginning, and not introduced as a cumbersome afterthought as happened =
too
many times before.

Thanks

Gunnar Hellstr=F6m
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: owner-ietf-announce@ietf.org
[mailto:owner-ietf-announce@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Wednesday, January 07, 2004 3:03 PM
To: IETF-Announce:
Cc: sip@ietf.org
Subject: I-D ACTION:draft-ietf-sip-callee-caps-03.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Protocol Working =
Group
of the IETF.

	Title		: Indicating User Agent Capabilities in the Session
Initiation
Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-sip-callee-caps-03.txt
	Pages		: 45
	Date		: 2004-1-6

This specification defines mechanisms by which a Session Initiation
Protocol (SIP) user agent can convey its capabilities and
characteristics to other user agents. These capabilities are conveyed
as parameters of the Contact header field.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-callee-caps-03.txt

To remove yourself from the IETF Announcement list, send a message to
ietf-announce-request with the word unsubscribe in the body of the =
message.

Internet-Drafts are also available by anonymous FTP. Login with the =
username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sip-callee-caps-03.txt".

A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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In the body type:
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NOTE:	The mail server at ietf.org can return the document in
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-----------------------------------------------------------------------
The IESG has approved following document:

- 'Indicating User Agent Capabilities in the Session Initiation Protocol
(SIP) '
   <draft-ietf-sip-callee-caps-03.txt> as a Proposed Standard

This document is the product of the Session Initiation Protocol Working
Group.

The IESG contact persons are Allison Mankin and Jon Peterson.

Technical Summary

  This specification defines mechanisms by which a Session Initiation
   Protocol (SIP) user agent can convey its capabilities and
   characteristics to other user agents and to the registrar for its
   domain. This information is conveyed as parameters of the Contact
   header field.  It may be used by a proxy for call routing. The =
parameter
   design is based on RFC 2506, media feature tags.  The specification
   creates a SIP tree registry parallel to the IETF tree registry from
   RFC 2506.  The syntax of the parameters is based on RFC 2533.

   Strong considerations regarding the privacy and data
   integrity of the information are discussed by the document.

Working Group Summary

   The working group took a lot of care and review developing this =
design.
    There was a mid-course design review from an Applications area
    standpoint that resulted in the advice to work with the RFC 2533,
    2506 approach, which has proved to be very constructive.  The WG
    supported the advancement strongly, after thorough review.

Protocol Quality

   The document was reviewed for the IESG by Allison Mankin.  The
    Applications mid-course reviews were by Patrik Faltstrom and Ted
    Hardie.




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



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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [Sip] Server Location and trying requests bug?
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inline.

Paul D.Smith wrote:

> All,
> 
> I believe there is an issue with Server Location that makes it impossible to
> reroute requests successfully in the following scenario.
> 
> If the request reaches the server, but the resulting response does not reach
> the client (perhaps a NAT blocked its path) then Server Location processing
> cannot be used to find a different route via which the request can be
> completed and result in a successful response to the client.

I dont see why. The client (UAC or proxy) cannot differentiate the 
case where a request fails to reach the server, and the response fails 
to reach the client. Both manifest themselves as failure to receive a 
response.

> At SIPit13 we encountered the following problem.
> 
> - Send a request using UDP
> - Request reached the server but the response could not get back for some
>   reason
> - Client timed out and then tried TCP, changing the branch
> - New request reached the server which complained because the request did 
>   appear to increment the CSeq.

The server should be prepared to receive requests with CSeq more than 
one higher. RFC3261 explicitly discusses this.

> 
> These were specifically REGISTER requests but I believe this affects other
> requests, especially those in-dialog and INVITEs.
> 
> Clearly the server was operating correctly and so there was some discussion
> as
> to whether the client was at fault. RFC3261 only states the following
> (RFC3261, 8.1.2, p42).
> 
>    The UAC SHOULD follow the procedures defined in [4] for stateful
>    elements, trying each address until a server is contacted.  Each try
>    constitutes a new transaction, and therefore each carries a different
>    topmost Via header field value with a new branch parameter.
>    Furthermore, the transport value in the Via header field is set to
>    whatever transport was determined for the target server.
> 
> However it was decided that in fact the CSeq of the transaction should also
> have been incremented by the client when sending the new transaction over
> TCP.

It should, though in rfc3261 (as opposed to 2543), it should actually 
work in either case.

> 
> But now consider what would happen if the CSeq were updated in this manner
> but there was an intervening proxy.
> 
> - Client sends request which reaches proxy
> - Proxy forwards on using UDP
> - Request reaches server but response cannot get back
> - Proxy times out UDP transaction and now tries TCP, changing the branch and
>   the CSeq

It should NOT have done that. The proxy should not touch the CSeq. 
Only the branch ID parameter.

> - New request reaches the server, which is happy and responds
> - Response now reaches proxy but the original request had Cseq X and the
>   response has CSeq (X+1)
> 
> Now the proxy might be able to massage the response Cseq back to X but then
> we
> have the strange situation that the client believes the next CSeq should be
> (X+1) whilst the server believes it should be (X+2).  Clearly, a subsequent
> in-dialog reINVITE from the client will fail because the server does not
> believe the Cseq has been incremented.

As I mentioned above, the server should be prepared to receive 
requests with CSeq one or more greater than the last received.

> 
> Clearly, without incrementing Cseq, we will hit the "response not returned,
> subsequent request fails" problem in both the UAC/UAS and UAC/proxy/UAS
> cases.
> But incrementing the CSeq only fixes the UAC/UAS case and introduces more
> problems in the UAC/proxy/UAS case.  For this reason we believe the
> discussed
> solution is wrong.
> 
> At the moment, this seems to be a problem that RFC3261 fails to address and
> we
> have yet to determine a simple fix.

The problem is in the server. From 12.2.2:

> If the remote sequence number is empty, it MUST be set to the value
>    of the sequence number in the CSeq header field value in the request.
>    If the remote sequence number was not empty, but the sequence number
>    of the request is lower than the remote sequence number, the request
>    is out of order and MUST be rejected with a 500 (Server Internal
>    Error) response.  If the remote sequence number was not empty, and
>    the sequence number of the request is greater than the remote
>    sequence number, the request is in order.  It is possible for the
>    CSeq sequence number to be higher than the remote sequence number by
>    more than one.  This is not an error condition, and a UAS SHOULD be
>    prepared to receive and process requests with CSeq values more than
>    one higher than the previous received request.  The UAS MUST then set
>    the remote sequence number to the value of the sequence number in the
>    CSeq header field value in the request.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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From exim@www1.ietf.org  Mon Jan 12 11:12:59 2004
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: SIP WG <sip@ietf.org>
Subject: RE: [Sip] Server Location and trying requests bug?
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Jonathan,

I have commented on your responses below, but it seems that I have
failed to get across the problems that we encountered.  I will
therefore attempt to recouch the problem using some ASCII-art flows.
My comments on your response are couched in terms of my new examples
so please read this section before looking at my in-line comments.

There are two scenarios which I am considering.

1. UAC/UAS "direct" (i.e. not intervening SIP proxy)
2. UAC/proxy/UAS

Note that in both cases there is a NAT (or similar) between the UAS and the
SIP entity (UAC or proxy) to which it is attached.  I've shown this as a
NAT below.  This entity is not SIP aware but can cause problems with
responses
failing to traverse back towards the UAC.

I have shown requests as RQ[c,b] where c and b represent CSeq and
branches respectively:

- RQ[1,1] and RQ[1,2] have the same CSeq but different branches
- RQ[1,1] and RQ[2,2] have different CSeq and different branches.

Scenario 1.a (we hit this at SIPit13)

UAC                 NAT                 UAS

RQ[1,1] ----------->   RQ[1,1]---------->
                       <---------2XX[1,1]
RQ[1,2] ----------->   RQ[1,2]---------->
<------------5XX[1,2]  <---------5XX[1,2]

What has happened here is that the UAS has received the request, processed
it successfully but the response could not reach the UAC.  Server
(re)Location
then kicked in on the UAC and found a different route (say TCP instead of
UDP)
and the request again reached the UAS.  But the CSeq has not been increased
so the UAS correctly rejects the request.

After discussions with various people at SIPit13, it seemed a reasonable fix
to have the UAC increment the CSeq during Server (re)Location which results
in the following...

Scenario 1.b - Allow Server (re)Location to increment Cseq

UAC                 NAT                 UAS

RQ[1,1] ----------->   RQ[1,1]--------->
                       <--------2XX[1,1]
RQ[2,2] ----------->   RQ[2,2]--------->
<------------2XX[2,2]  <--------2XX[2,2]

This seemed to solve the problem until we considered proxies.  If we
assume that incrementing Cseq is correct we end up with the following.

Scenario 1.2

UAC                 Proxy               NAT                 UAS

RQ[1,1] ----------->     RQ[1,1]------->   RQ[1,1]--------->
                                           <---------2XX[1,1]
                         RQ[2,2]------->   RQ[2,2]---------->
<---------------????     <------2XX[2,2]   <---------2XX[2,2]

But now what happens to the response?  We cannot mangle the CSeq
before the proxy returns it otherwise the next Cseq the UAC would
expect to send would be 2 whilst the UAS would expect a Cseq of 3.

So, as you correctly say, I have to conclude that it must be wrong
for Server (re)Location to increment the Cseq.  But how, therefore,
do we solve the problem of scenario 1.a?

I've commented on your specific feedback in line below, referencing
the scenarios given above.

I look forward to reading your comments,
Paul DS.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: 11 January 2004 14:59
To: SIP (E-mail)
Subject: Re: [Sip] Server Location and trying requests bug?


inline.

Paul D.Smith wrote:

> All,
> 
> I believe there is an issue with Server Location that makes it impossible
to
> reroute requests successfully in the following scenario.
> 
> If the request reaches the server, but the resulting response does not
reach
> the client (perhaps a NAT blocked its path) then Server Location
processing
> cannot be used to find a different route via which the request can be
> completed and result in a successful response to the client.

I dont see why. The client (UAC or proxy) cannot differentiate the 
case where a request fails to reach the server, and the response fails 
to reach the client. Both manifest themselves as failure to receive a 
response.

[PDS] Hopefully scenario 1.a has explained this.  The issue is that the
      UAC and UAS now have different ideas of the state of the dialog or
      call i.e. the next Cseq that the UAC should send on a request.

> At SIPit13 we encountered the following problem.
> 
> - Send a request using UDP
> - Request reached the server but the response could not get back for some
>   reason
> - Client timed out and then tried TCP, changing the branch
> - New request reached the server which complained because the request did 
>   appear to increment the CSeq.

The server should be prepared to receive requests with CSeq more than 
one higher. RFC3261 explicitly discusses this.

[PDS] That is not the problem.  In scenario 1.a, the UAS believes that the
      UAC sent a new request that _didn't_ increment the Cseq.

> 
> These were specifically REGISTER requests but I believe this affects other
> requests, especially those in-dialog and INVITEs.
> 
> Clearly the server was operating correctly and so there was some
discussion
> as
> to whether the client was at fault. RFC3261 only states the following
> (RFC3261, 8.1.2, p42).
> 
>    The UAC SHOULD follow the procedures defined in [4] for stateful
>    elements, trying each address until a server is contacted.  Each try
>    constitutes a new transaction, and therefore each carries a different
>    topmost Via header field value with a new branch parameter.
>    Furthermore, the transport value in the Via header field is set to
>    whatever transport was determined for the target server.
> 
> However it was decided that in fact the CSeq of the transaction should
also
> have been incremented by the client when sending the new transaction over
> TCP.

It should, though in rfc3261 (as opposed to 2543), it should actually 
work in either case.

[PDS] I disagree.  As per scenario 1.a, the UAS is quite correct to reject
      what it believes is a new request that has not incremented the CSeq.


> 
> But now consider what would happen if the CSeq were updated in this manner
> but there was an intervening proxy.
> 
> - Client sends request which reaches proxy
> - Proxy forwards on using UDP
> - Request reaches server but response cannot get back
> - Proxy times out UDP transaction and now tries TCP, changing the branch
and
>   the CSeq

It should NOT have done that. The proxy should not touch the CSeq. 
Only the branch ID parameter.

[PDS] Agreed.  But this means that scenario 2 fails.  Also, if you are
implying
      that the "real" UAC can increment the Cseq but the "proxy" UAC cannot,
then
      the UAC side of proxy and the "real" UAC are now inconsistent.

> - New request reaches the server, which is happy and responds
> - Response now reaches proxy but the original request had Cseq X and the
>   response has CSeq (X+1)
> 
> Now the proxy might be able to massage the response Cseq back to X but
then
> we
> have the strange situation that the client believes the next CSeq should
be
> (X+1) whilst the server believes it should be (X+2).  Clearly, a
subsequent
> in-dialog reINVITE from the client will fail because the server does not
> believe the Cseq has been incremented.

As I mentioned above, the server should be prepared to receive 
requests with CSeq one or more greater than the last received.

[PDS] As above, this is not the problem that we hit.

> 
> Clearly, without incrementing Cseq, we will hit the "response not
returned,
> subsequent request fails" problem in both the UAC/UAS and UAC/proxy/UAS
> cases.
> But incrementing the CSeq only fixes the UAC/UAS case and introduces more
> problems in the UAC/proxy/UAS case.  For this reason we believe the
> discussed
> solution is wrong.
> 
> At the moment, this seems to be a problem that RFC3261 fails to address
and
> we
> have yet to determine a simple fix.

The problem is in the server. From 12.2.2:

> If the remote sequence number is empty, it MUST be set to the value
>    of the sequence number in the CSeq header field value in the request.
>    If the remote sequence number was not empty, but the sequence number
>    of the request is lower than the remote sequence number, the request
>    is out of order and MUST be rejected with a 500 (Server Internal
>    Error) response.  If the remote sequence number was not empty, and
>    the sequence number of the request is greater than the remote
>    sequence number, the request is in order.  It is possible for the
>    CSeq sequence number to be higher than the remote sequence number by
>    more than one.  This is not an error condition, and a UAS SHOULD be
>    prepared to receive and process requests with CSeq values more than
>    one higher than the previous received request.  The UAS MUST then set
>    the remote sequence number to the value of the sequence number in the
>    CSeq header field value in the request.

[PDS] As above, this is not the problem we are hitting.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.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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Date: Mon, 12 Jan 2004 12:24:08 -0500 (EST)
From: Srinivasan Krishnamoorthy <ksrini@motorola.com>
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Subject: [Sip] Registering Multiple 'Contacts'
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Hi,

  Section 10.2.4 of RFC3261 'Refreshing Bindings' says:

  "The UA then issues a REGISTER request for each of its bindings 
  before the expiration interval has elapsed. It MAY combine 
  several updates into one REGISTER request."

  The above seems to indicate that the UAC may send: 
  (forgive the Syntax)

        REGISTER 
        To: user@operator.net
        Contact: sip:user1@10.0.0.1;expires=3600
        
        200 OK
        
        REGISTER 
        To: user@operator.net
        Contact: sip:user1@192.0.0.1;expires=3600
        
        200 OK

  This, according to RFC3261, will register both Contact1 and Contact2 
  for user@operator.net. (The server will then use the q parameter
  of the contact for priority among contacts)

  Take a case where a SIP UAC power-cycles or moves without sending 
  an explicit de-REGISTER.  Assume it gets a new IP-address everytime 
  - therefore the 'Contact' will be a new URI every time it sends a 
  REGISTER after a power-cycle.

  If the power-cycle occured before the 'registration expiration interval' 
  elapsed - the Server will essentially be adding 'multiple' bindings
  for this user. Is this correct interpretation?

  Now, is the UAclient mandated to 'deREGISTER' all of its previous contacts 
  before sending its new REGISTER?  Or risk adding multiple 'contacts' 
  at the Server?

  How will the Server prevent against mis-behaved clients in this case?


Thank you very much for any help.
-Srini


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Subject: Re: [Sip] Registering Multiple 'Contacts'
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Srinivasan Krishnamoorthy wrote:
> Hi,
> 
>   Section 10.2.4 of RFC3261 'Refreshing Bindings' says:
> 
>   "The UA then issues a REGISTER request for each of its bindings 
>   before the expiration interval has elapsed. It MAY combine 
>   several updates into one REGISTER request."
> 
>   The above seems to indicate that the UAC may send: 
>   (forgive the Syntax)
> 
>         REGISTER 
>         To: user@operator.net
>         Contact: sip:user1@10.0.0.1;expires=3600
>         
>         200 OK
>         
>         REGISTER 
>         To: user@operator.net
>         Contact: sip:user1@192.0.0.1;expires=3600
>         
>         200 OK
> 
>   This, according to RFC3261, will register both Contact1 and Contact2 
>   for user@operator.net. (The server will then use the q parameter
>   of the contact for priority among contacts)

Yes.

>   Take a case where a SIP UAC power-cycles or moves without sending 
>   an explicit de-REGISTER.  Assume it gets a new IP-address everytime 
>   - therefore the 'Contact' will be a new URI every time it sends a 
>   REGISTER after a power-cycle.
> 
>   If the power-cycle occured before the 'registration expiration interval' 
>   elapsed - the Server will essentially be adding 'multiple' bindings
>   for this user. Is this correct interpretation?

Yes

>   Now, is the UAclient mandated to 'deREGISTER' all of its previous contacts 
>   before sending its new REGISTER? 

Its not mandated to do so, but that would probably be a wise thing to do.

 > Or risk adding multiple 'contacts'
>   at the Server?

That is the alternative.

>   How will the Server prevent against mis-behaved clients in this case?

The expiration time bounds the problem, but doesn't eliminate it.
Assume the UAC power cycles, forgets its old address, gets a new one and 
registers it. It will then presumably continue to refresh the new 
address, but not the old one. So the old registration will continue to 
be active until it expires. During that time requests will probably be 
forked to it and the new address, but the ones to the old address will fail.

	Paul


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



From exim@www1.ietf.org  Mon Jan 12 17:07:45 2004
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Date: Mon, 12 Jan 2004 23:01:09 +0100
From: Rene Bartsch <ml@bartschnet.de>
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Subject: [Sip] Suggestion to locate nearest ERC-center by ENUM
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Hi,

for emergency calls we have the problem to locate the nearest ERC-center.

As most SIP-phones will have a numeric pad only, we'll have to use ENUM 
anyway. This brought me to the idea to use ENUM for allocating the SIP 
or WhatEverProtocol address of the nearest ERC-center.

The idea is simply:

Just using NAPTR-RR and LOC-RR.

The ITU would run the top-layer ERC-ENUM-domain 1.1.9.e164.arpa. The 
Zone-file would have a LOC-RR and a NAPTR-RR pointing to each national 
ERC-domain (e.g. 1.1.9.1.e164.arpa for the US, 1.1.9.9.4.e164.arpa for 
Germany, 1.1.9.4.4.e164.arpa for UK, 1.1.9.3.3.e164.arpa for France, 
1.1.9.3.4.e164.arpa for Austria, etc.)

Every national regulator would run the national ERC-ENUM-domain in the 
same shape, either pointing directly to the adresses of the ERC-centers 
or having additional sub-layers of ERC-ENUM-domains for e.g. federal 
states, districts, etc.

This would be quite flexible and could also be used for any other 
ERC-service/communication protocol.

The SIP-phones would just use their geographic coordinates (either 
manually stored in fixed phones or e.g. evaluated by GPS for cell 
phones, ...) and do a DNS-lookup when 911 is dialed to find the nearest 
LOC-RR and the corresponding NAPTR-RR for the ERC-center.

Comments?

Rene Bartsch


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From exim@www1.ietf.org  Mon Jan 12 20:03:47 2004
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Cc: Dean Willis <dean.willis@softarmor.com>, aki.niemi@nokia.com,
        rohan@cisco.com
To: sip@ietf.org
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Subject: [Sip] WGLC for PUBLISH
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Hi,

I'd like to begin Working Group Last Call for PUBLISH:

	http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-02.txt

This WGLC will end on January 28, 2004.

thanks,
-rohan


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rene Bartsch'" <ml@bartschnet.de>, sip@ietf.org
Subject: RE: [Sip] Suggestion to locate nearest ERC-center by ENUM
Date: Tue, 13 Jan 2004 16:20:58 -0500
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I'm working on a proposal to use the DNS somewhat like this.
This is a little premature, but Rene brought it up, so....

The fundamental difference between what I will propose, and your
proposal is that the phone number mapping to geographic area
breaks down after the country code (for example, in North America,
there is no relationship between the "area code" (first 3 digits
in a phone number) and ERC service boundaries.  Also, there is
no agreement on the "number" for emergency calls, and in fact,
there is an active draft that should be ready for WGLC soon
that proposes that the SIP URI for an emergency call is
sos@<localdomain>.  Therefore, using ENUM like structure isn't
very usefull.  

As you point out, in most countries, there are one or more levels of 
"delegation" on ERC organization that match political subdivisions.  
For example, in the U.S., ERCs are organized by the states, and 
in some cases, there is further delegation to counties.
This varies by country.

In my immediate area, the ERC is called "NEWCOM" and is in Allegheny
County, in Pennsylvania.  I propose that the service boundary of
this ERC be found in the DNS at something like
newcom.allegheny.pa.us.sos.arpa
There are some details to be worked out.

My proposal will be more complex than this, because I would include
much more information in the DNS.  For example, the boundary of
Room 235 on the 5th floor of the Wigit suite at 123 Main Street in 
Pittsburgh PA might be found in the DNS at 
235.5.wigit.123.main.pittsburgh.allegheny.pa.us.sos.arpa 

In this proposal, there is a VERY BIG ASSUMPTION.  The assumption 
is that the DNS is used for PUBLISHING the data, and is not 
actually used in the processing of an emergency call.  Rather, 
I envision that spiders will periodically walk the DNS tree and 
extract the boundary information, storing it in a local database 
organized to make the operations more efficient.

Boundary information would be available at intermediate nodes in 
this proposal (that is, the boundary of Allegheny County could 
be found at allegheny.pa.us.sos.arpa).  It is probably the case 
that the boundary information below the street address level 
would be encrypted if the building owner did not want to make 
interior layout information public.

The data in the DNS would then allow a conversion between civil
(room/floor/building/street address) and geo (lat/lon/altitide).
It would also replace what in the U.S. is called the "Master
Street Address Guide". 

In operation, phones will learn their location, either civil
or geo.  They pass that information on a call to sos@<localdomain>.
A proxy routing an emergency call converts a civil to a geo using
the (previously spidered) DNS sos.arpa data if necessary, and 
intersects the geo with the ERC service boundaries to find the 
serving ERC.  The entry in the DNS contains a URI for the ERC, 
and the proxy routes the call to that ERC.  The ERC gets
location in civil or geo form.  It validates a civil by verifying
that the address exists in the sos.arpa domain, or converts a geo 
to a (valid) civil for dispatch.

You could also imagine a link between sos.arpa and e164.arpa, 
whereby for a fixed phone, there was a "domain name" in sos.arpa 
stored in the terminal e164.arpa entry that mapped a phone number 
to a civil location.  This too might be encrypted.  You could
use such a mechanism to replace existing "ALI" databases that
convert a caller phone number to a location for legacy phone
systems.

Note that the delegation mechanism in the DNS is very appropriate
for this purpose:
	An international organization (possibly ITU) delegates the
		iso country code in sos.arpa to a national agency
	The national agency delegates to lower layer political entities
		(states in my example)
	The (say) state agency, delegates to any lower level agencies
		which in turn delegate to lower level agencies.  In my
		example, PA delegates to Allegheny County, which would
		delegate newcom.allegheny.pa.us.sos.arpa to NEWCOM.  It
		would delegate pittsburgh.allegheny.pa.us to the City of
		Pittsburgh.  The City would create entries for all of
		its streets, and for all of the buildings on those
		streets.
	The City (township, village,...) would delegate to building
		owners
	Building owners would (if appropriate) delegate to tenants
	Tenants would enter floor/room data

An entity at some level might decide that it would not actually
delegate lower levels, but create an administrative mechanism to
populate the lower levels. 

More later.

Brian


-----Original Message-----
From: Rene Bartsch [mailto:ml@bartschnet.de]
Sent: Monday, January 12, 2004 5:01 PM
To: sip@ietf.org
Subject: [Sip] Suggestion to locate nearest ERC-center by ENUM


Hi,

for emergency calls we have the problem to locate the nearest ERC-center.

As most SIP-phones will have a numeric pad only, we'll have to use ENUM 
anyway. This brought me to the idea to use ENUM for allocating the SIP 
or WhatEverProtocol address of the nearest ERC-center.

The idea is simply:

Just using NAPTR-RR and LOC-RR.

The ITU would run the top-layer ERC-ENUM-domain 1.1.9.e164.arpa. The 
Zone-file would have a LOC-RR and a NAPTR-RR pointing to each national 
ERC-domain (e.g. 1.1.9.1.e164.arpa for the US, 1.1.9.9.4.e164.arpa for 
Germany, 1.1.9.4.4.e164.arpa for UK, 1.1.9.3.3.e164.arpa for France, 
1.1.9.3.4.e164.arpa for Austria, etc.)

Every national regulator would run the national ERC-ENUM-domain in the 
same shape, either pointing directly to the adresses of the ERC-centers 
or having additional sub-layers of ERC-ENUM-domains for e.g. federal 
states, districts, etc.

This would be quite flexible and could also be used for any other 
ERC-service/communication protocol.

The SIP-phones would just use their geographic coordinates (either 
manually stored in fixed phones or e.g. evaluated by GPS for cell 
phones, ...) and do a DNS-lookup when 911 is dialed to find the nearest 
LOC-RR and the corresponding NAPTR-RR for the ERC-center.

Comments?

Rene Bartsch


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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Date: Tue, 13 Jan 2004 23:35:56 +0100
From: Rene Bartsch <ml@bartschnet.de>
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Rosen, Brian schrieb:
> I'm working on a proposal to use the DNS somewhat like this.
> This is a little premature, but Rene brought it up, so....
> 
> The fundamental difference between what I will propose, and your
> proposal is that the phone number mapping to geographic area
> breaks down after the country code (for example, in North America,
> there is no relationship between the "area code" (first 3 digits
> in a phone number) and ERC service boundaries.  Also, there is
> no agreement on the "number" for emergency calls,

The ITU should just delegate ENUM-911 for ERCs world-wide.


  and in fact,
> there is an active draft that should be ready for WGLC soon
> that proposes that the SIP URI for an emergency call is
> sos@<localdomain>.  Therefore, using ENUM like structure isn't
> very usefull.  
> 

But there is a problem:

Most SIP-phones will habe numeric pads, but no alpanumeric 80-key pads.
So we cannot use an addressing out of the decimal range - no letters, no 
"@"!

And there's no problem with the number mapping to the geographic area as 
the geographic is not determined by the area prefixes but by the 
LOC-records.

I think my explanation was a bit misunderstandable.

When you do an ERC,

1.) you dial 911 at your phone
2.) the phone looks up the ENUM 1.1.9.e164.arpa
3.) the phone checks it's geographic data against the LOC-records listed 
in the 1.1.9.e164.arpa-zone and finds the nearest LOC-entry (in your 
case it would be a NAPTR-RR pointing to the ERC-ENUM of the US - 
1.1.9.1.e164.arpa)
4.) the phone looks up the ENUM 1.1.9.1.e164.arpa
5.) the phone checks it's geographic data against the LOC-records listed 
in the 1.1.9.1.e164.arpa-zone and finds the nearest LOC-entry (in your 
case it would be a NAPTR-RR pointing to the ERC-ENUM-domain of 
Pennsylvania - 1.1.9.[WhatEverYouWantLevel1].1.e164.arpa)
6.) the phone looks up the ENUM 1.1.9.[WhatEverYouWantLevel1].1.e164.arpa
7.) the phone checks it's geographic data against the LOC-records listed 
in the 1.1.9.[WhatEverYouWantLevel1].1.e164.arpa-zone and finds the 
nearest LOC-entry (in your case it would be a NAPTR-RR pointing to the 
ERC-ENUM of Allegheny County 1.1.9.[WhatEverYouWantLevel2].1.e164.arpa)
.
.
.
n.) the phone looks up the ENUM 1.1.9.[LastLayer].1.e164.arpa
n+1.) the phone checks it's geographic data against the LOC-records 
listed in the 1.1.9.[LastLayer].1.e164.arpa-zone and finds the nearest 
LOC-entry (in your case it would be a NAPTR-RR pointing to the 
SIP-address of NEWCOM)
n+2.) the phone connects to the ERC transmitting it geographical data 
(longitude, latitude, altitude).
n+3.) the ERC looks up the Address/Room, etc. based on the received 
geographical data and static land office data
n+4.) ambulance, police or whatever is send

It's a quite easy, reliable, flexible and user-friendly solution. Only 
the top-level ENUM 1.1.9.e164.arpa has to be fix, in the lower levels 
you can use ANY domain-name (but I'd suggest to use ENUMs with local 
prefixes to allow users to directly dial their local area ERC-ENUM in 
case upper DNS-brances break down by a nuclear war ;-) ).


> My proposal will be more complex than this, because I would include
> much more information in the DNS.  For example, the boundary of
> Room 235 on the 5th floor of the Wigit suite at 123 Main Street in 
> Pittsburgh PA might be found in the DNS at 
> 235.5.wigit.123.main.pittsburgh.allegheny.pa.us.sos.arpa 
> 
> In this proposal, there is a VERY BIG ASSUMPTION.  The assumption 
> is that the DNS is used for PUBLISHING the data, and is not 
> actually used in the processing of an emergency call.  Rather, 
> I envision that spiders will periodically walk the DNS tree and 
> extract the boundary information, storing it in a local database 
> organized to make the operations more efficient.
> 
> Boundary information would be available at intermediate nodes in 
> this proposal (that is, the boundary of Allegheny County could 
> be found at allegheny.pa.us.sos.arpa).  It is probably the case 
> that the boundary information below the street address level 
> would be encrypted if the building owner did not want to make 
> interior layout information public.
> 
> The data in the DNS would then allow a conversion between civil
> (room/floor/building/street address) and geo (lat/lon/altitide).
> It would also replace what in the U.S. is called the "Master
> Street Address Guide". 
> 
> In operation, phones will learn their location, either civil
> or geo.  They pass that information on a call to sos@<localdomain>.
> A proxy routing an emergency call converts a civil to a geo using
> the (previously spidered) DNS sos.arpa data if necessary, and 
> intersects the geo with the ERC service boundaries to find the 
> serving ERC.  The entry in the DNS contains a URI for the ERC, 
> and the proxy routes the call to that ERC.  The ERC gets
> location in civil or geo form.  It validates a civil by verifying
> that the address exists in the sos.arpa domain, or converts a geo 
> to a (valid) civil for dispatch.

what about privacy? If you put the address data of a person into DNS 
anyone can look it up - No chance to get european countries into the boat!

> 
> You could also imagine a link between sos.arpa and e164.arpa, 
> whereby for a fixed phone, there was a "domain name" in sos.arpa 
> stored in the terminal e164.arpa entry that mapped a phone number 
> to a civil location.  This too might be encrypted.  You could
> use such a mechanism to replace existing "ALI" databases that
> convert a caller phone number to a location for legacy phone
> systems.
> 
> Note that the delegation mechanism in the DNS is very appropriate
> for this purpose:
> 	An international organization (possibly ITU) delegates the
> 		iso country code in sos.arpa to a national agency
> 	The national agency delegates to lower layer political entities
> 		(states in my example)
> 	The (say) state agency, delegates to any lower level agencies
> 		which in turn delegate to lower level agencies.  In my
> 		example, PA delegates to Allegheny County, which would
> 		delegate newcom.allegheny.pa.us.sos.arpa to NEWCOM.  It
> 		would delegate pittsburgh.allegheny.pa.us to the City of
> 		Pittsburgh.  The City would create entries for all of
> 		its streets, and for all of the buildings on those
> 		streets.
> 	The City (township, village,...) would delegate to building
> 		owners
> 	Building owners would (if appropriate) delegate to tenants
> 	Tenants would enter floor/room data
> 
> An entity at some level might decide that it would not actually
> delegate lower levels, but create an administrative mechanism to
> populate the lower levels. 
> 

And such a complex system will never work reliable. If you register all 
that phones and flats the data will never be up-to-date. Do you really 
think users will update DNS here and something else here all the time?

Rene Bartsch


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rene Bartsch'" <ml@bartschnet.de>,
        "Rosen, Brian"
	 <Brian.Rosen@marconi.com>, sip@ietf.org
Subject: RE: [Sip] Suggestion to locate nearest ERC-center by ENUM
Date: Tue, 13 Jan 2004 18:00:58 -0500
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Rene

9-1-1 is a valid number (or service invocation) in many dialing plans.
In fact, the delegation of 1.9.e164.arpa would be to India, since it
owns the +91 country code.

A phone with a numeric pad only would dial the "local" emergency number.
That would be translated by a routing proxy to the universal emergency
URI (sos@localdomain).  The draft discusses this.

A LOC entry is a single point.  You need a boundary.  In fact, you need
a list of polygons.  The "nearest" ERC is not correct.  ERCs have defined
service boundaries.  You must intersect the location (a point) with the
boundary to determine which ERC service area the point is located.
I'll expand this later, but there are circumstances where there might be
more than one ERC serving a given location.  This is even more obvious
with levels above the ERC; the boundary for the U.S. and Canada for example,
cannot be reduced to a point where you can determine which side of the
border you are on by comparing distance from a single point to the
target location.

Your outline of walking the tree differs in detail from my proposal,
but it is similar.  I start at sos.arpa.  You start at 1.1.9.e164.arpa.
I use 2 character ISO country codes.  You use e.164 numeric country codes.
You don't identify the form of "whatever you want".  I use political
subdivisions that actually do delegation for lower levels.  You suffix
all entries with 1.1.9, I'm working on a couple different ways of
identifying
ERC entries.  Otherwise, the BASIC idea is the same, the details are
quite different.

Brian



-----Original Message-----
From: Rene Bartsch [mailto:ml@bartschnet.de]
Sent: Tuesday, January 13, 2004 5:36 PM
To: Rosen, Brian; sip@ietf.org
Subject: Re: [Sip] Suggestion to locate nearest ERC-center by ENUM


Rosen, Brian schrieb:
> I'm working on a proposal to use the DNS somewhat like this.
> This is a little premature, but Rene brought it up, so....
> 
> The fundamental difference between what I will propose, and your
> proposal is that the phone number mapping to geographic area
> breaks down after the country code (for example, in North America,
> there is no relationship between the "area code" (first 3 digits
> in a phone number) and ERC service boundaries.  Also, there is
> no agreement on the "number" for emergency calls,

The ITU should just delegate ENUM-911 for ERCs world-wide.


  and in fact,
> there is an active draft that should be ready for WGLC soon
> that proposes that the SIP URI for an emergency call is
> sos@<localdomain>.  Therefore, using ENUM like structure isn't
> very usefull.  
> 

But there is a problem:

Most SIP-phones will habe numeric pads, but no alpanumeric 80-key pads.
So we cannot use an addressing out of the decimal range - no letters, no 
"@"!

And there's no problem with the number mapping to the geographic area as 
the geographic is not determined by the area prefixes but by the 
LOC-records.

I think my explanation was a bit misunderstandable.

When you do an ERC,

1.) you dial 911 at your phone
2.) the phone looks up the ENUM 1.1.9.e164.arpa
3.) the phone checks it's geographic data against the LOC-records listed 
in the 1.1.9.e164.arpa-zone and finds the nearest LOC-entry (in your 
case it would be a NAPTR-RR pointing to the ERC-ENUM of the US - 
1.1.9.1.e164.arpa)
4.) the phone looks up the ENUM 1.1.9.1.e164.arpa
5.) the phone checks it's geographic data against the LOC-records listed 
in the 1.1.9.1.e164.arpa-zone and finds the nearest LOC-entry (in your 
case it would be a NAPTR-RR pointing to the ERC-ENUM-domain of 
Pennsylvania - 1.1.9.[WhatEverYouWantLevel1].1.e164.arpa)
6.) the phone looks up the ENUM 1.1.9.[WhatEverYouWantLevel1].1.e164.arpa
7.) the phone checks it's geographic data against the LOC-records listed 
in the 1.1.9.[WhatEverYouWantLevel1].1.e164.arpa-zone and finds the 
nearest LOC-entry (in your case it would be a NAPTR-RR pointing to the 
ERC-ENUM of Allegheny County 1.1.9.[WhatEverYouWantLevel2].1.e164.arpa)
.
.
.
n.) the phone looks up the ENUM 1.1.9.[LastLayer].1.e164.arpa
n+1.) the phone checks it's geographic data against the LOC-records 
listed in the 1.1.9.[LastLayer].1.e164.arpa-zone and finds the nearest 
LOC-entry (in your case it would be a NAPTR-RR pointing to the 
SIP-address of NEWCOM)
n+2.) the phone connects to the ERC transmitting it geographical data 
(longitude, latitude, altitude).
n+3.) the ERC looks up the Address/Room, etc. based on the received 
geographical data and static land office data
n+4.) ambulance, police or whatever is send

It's a quite easy, reliable, flexible and user-friendly solution. Only 
the top-level ENUM 1.1.9.e164.arpa has to be fix, in the lower levels 
you can use ANY domain-name (but I'd suggest to use ENUMs with local 
prefixes to allow users to directly dial their local area ERC-ENUM in 
case upper DNS-brances break down by a nuclear war ;-) ).


> My proposal will be more complex than this, because I would include
> much more information in the DNS.  For example, the boundary of
> Room 235 on the 5th floor of the Wigit suite at 123 Main Street in 
> Pittsburgh PA might be found in the DNS at 
> 235.5.wigit.123.main.pittsburgh.allegheny.pa.us.sos.arpa 
> 
> In this proposal, there is a VERY BIG ASSUMPTION.  The assumption 
> is that the DNS is used for PUBLISHING the data, and is not 
> actually used in the processing of an emergency call.  Rather, 
> I envision that spiders will periodically walk the DNS tree and 
> extract the boundary information, storing it in a local database 
> organized to make the operations more efficient.
> 
> Boundary information would be available at intermediate nodes in 
> this proposal (that is, the boundary of Allegheny County could 
> be found at allegheny.pa.us.sos.arpa).  It is probably the case 
> that the boundary information below the street address level 
> would be encrypted if the building owner did not want to make 
> interior layout information public.
> 
> The data in the DNS would then allow a conversion between civil
> (room/floor/building/street address) and geo (lat/lon/altitide).
> It would also replace what in the U.S. is called the "Master
> Street Address Guide". 
> 
> In operation, phones will learn their location, either civil
> or geo.  They pass that information on a call to sos@<localdomain>.
> A proxy routing an emergency call converts a civil to a geo using
> the (previously spidered) DNS sos.arpa data if necessary, and 
> intersects the geo with the ERC service boundaries to find the 
> serving ERC.  The entry in the DNS contains a URI for the ERC, 
> and the proxy routes the call to that ERC.  The ERC gets
> location in civil or geo form.  It validates a civil by verifying
> that the address exists in the sos.arpa domain, or converts a geo 
> to a (valid) civil for dispatch.

what about privacy? If you put the address data of a person into DNS 
anyone can look it up - No chance to get european countries into the boat!

> 
> You could also imagine a link between sos.arpa and e164.arpa, 
> whereby for a fixed phone, there was a "domain name" in sos.arpa 
> stored in the terminal e164.arpa entry that mapped a phone number 
> to a civil location.  This too might be encrypted.  You could
> use such a mechanism to replace existing "ALI" databases that
> convert a caller phone number to a location for legacy phone
> systems.
> 
> Note that the delegation mechanism in the DNS is very appropriate
> for this purpose:
> 	An international organization (possibly ITU) delegates the
> 		iso country code in sos.arpa to a national agency
> 	The national agency delegates to lower layer political entities
> 		(states in my example)
> 	The (say) state agency, delegates to any lower level agencies
> 		which in turn delegate to lower level agencies.  In my
> 		example, PA delegates to Allegheny County, which would
> 		delegate newcom.allegheny.pa.us.sos.arpa to NEWCOM.  It
> 		would delegate pittsburgh.allegheny.pa.us to the City of
> 		Pittsburgh.  The City would create entries for all of
> 		its streets, and for all of the buildings on those
> 		streets.
> 	The City (township, village,...) would delegate to building
> 		owners
> 	Building owners would (if appropriate) delegate to tenants
> 	Tenants would enter floor/room data
> 
> An entity at some level might decide that it would not actually
> delegate lower levels, but create an administrative mechanism to
> populate the lower levels. 
> 

And such a complex system will never work reliable. If you register all 
that phones and flats the data will never be up-to-date. Do you really 
think users will update DNS here and something else here all the time?

Rene Bartsch

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From: Rene Bartsch <ml@bartschnet.de>
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Rosen, Brian schrieb:
> Rene
> 
> 9-1-1 is a valid number (or service invocation) in many dialing plans.
> In fact, the delegation of 1.9.e164.arpa would be to India, since it
> owns the +91 country code.

Ok, we just cut off India ;-)
(Should have checked the dialing plans ...)

So we'll have to find another, world wide ERC-number or use national 
ones with country prefix. But looking at internationalization and people 
crossing the world by plane I'd prefer one world-wide ERC-number. And 
the ITU has the possibilites to realize that.

> 
> A phone with a numeric pad only would dial the "local" emergency number.
> That would be translated by a routing proxy to the universal emergency
> URI (sos@localdomain).  The draft discusses this.

And what if the phone cannot reach a proxy? There should be a 
Peer2Peer-fallback in such a case. In that situation the translation of 
numerics to alphanumerics can only be done by ENUM.

Our national regulator (Germany) will never allow ERC-systems dependent 
on one proxy without redundancy!

> 
> A LOC entry is a single point.  You need a boundary.  In fact, you need
> a list of polygons.

And how do you define a polygon? If I do remind well from my 
math-lessons by several points ...

> The "nearest" ERC is not correct.  ERCs have defined
> service boundaries.  You must intersect the location (a point) with the
> boundary to determine which ERC service area the point is located.
> I'll expand this later, but there are circumstances where there might be
> more than one ERC serving a given location.  This is even more obvious
> with levels above the ERC; the boundary for the U.S. and Canada for example,
> cannot be reduced to a point where you can determine which side of the
> border you are on by comparing distance from a single point to the
> target location.

A polygon of LOCs ...

> 
> Your outline of walking the tree differs in detail from my proposal,
> but it is similar.  I start at sos.arpa.  You start at 1.1.9.e164.arpa.
> I use 2 character ISO country codes.  You use e.164 numeric country codes.

To be discussed. But I'd prefer ENUM as starting-point to avoid problems 
with dialing by numeric pads.

> You don't identify the form of "whatever you want".  I use political
> subdivisions that actually do delegation for lower levels.  You suffix
> all entries with 1.1.9, I'm working on a couple different ways of
> identifying
> ERC entries.  Otherwise, the BASIC idea is the same, the details are
> quite different.

It's my goal to set up a flexible, common layer based system independent 
from political issues - you never know how long political/national 
borders are kept (no Sowjet Union anymore, Germany, France, Great 
Britain, ... may become EU and maybe Iraq the 53rd federal state of the 
US ;-)

My suggestion allows to use any domains below the top-layer domain.

Rene


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From: "wendy" <xyxie@udtech.com.cn>
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Subject: [Sip] SIP DTMF
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Hello all, 

Who can tell me whether there is a RFC or I-D on DTMF in SIP which is still not expired?
I remember someone proposed using SIP INFO method for DTMF one or two years ago.
Is the DTMF relay implemented only by RTP now?


Regards,
Wendy

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Subject: RE: [Sip] SIP DTMF
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Wendy,

There was a discussion in the list some time back and it was decided
that SIP-INFO method should not be used to=20
Send the DTMF info.

It was proposed that it is better to you kpxml for sending DTMF tones.

Please look in to the SIP-Kpxml draft for sending DTMF tomes in-midcall
processing and all....=20


Regards,
Sreeram

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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=3D=3D=3D
Sreeram Kanumuri                Plot No 72, Keonics Electronics City
			              Hosur Main Road
Wipro Technologies              Bangalore -29 INDIA
sreeram.kanumuri@wipro.com      Phone: 8520408 Ex:3208
skanumury@yahoo.com             ESN  :
www.wipro.com                   Mobile:
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=3D=3D=3D=3D


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of wendy
Sent: Wednesday, January 14, 2004 8:08 AM
To: sip@ietf.org
Cc: sipping@ietf.org
Subject: [Sip] SIP DTMF


Hello all,=20

Who can tell me whether there is a RFC or I-D on DTMF in SIP which is
still not expired? I remember someone proposed using SIP INFO method for
DTMF one or two years ago. Is the DTMF relay implemented only by RTP
now?


Regards,
Wendy

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rene Bartsch'" <ml@bartschnet.de>, sip@ietf.org
Subject: RE: [Sip] Suggestion to locate nearest ERC-center by ENUM
Date: Wed, 14 Jan 2004 09:14:55 -0500
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> So we'll have to find another, world wide ERC-number or use national 
> ones with country prefix. But looking at internationalization 
> and people 
> crossing the world by plane I'd prefer one world-wide ERC-number. And 
> the ITU has the possibilites to realize that.
This is WAY beyond anything the IETF can do, and I think is not
practical at this time.  If you want to take such an effort on, please 
go ahead and let us know if you suceed.  In the mean time, we will have
to deal with current reality, which is that there is no universal number.

> 
> > 
> > A phone with a numeric pad only would dial the "local" 
> emergency number.
> > That would be translated by a routing proxy to the 
> universal emergency
> > URI (sos@localdomain).  The draft discusses this.
> 
> And what if the phone cannot reach a proxy? There should be a 
> Peer2Peer-fallback in such a case. In that situation the 
> translation of 
> numerics to alphanumerics can only be done by ENUM.
You are dreaming, I think.  Very few phones will know how to do
an ENUM lookup.  It is not practical for a phone to do the
calculations involved in determining the correct psap.
Therefore, you will depend on a proxy to do it.  There could
be more than one (actually, I personally think that EVERY proxy
server should be capable of handling an emergency call, even
if normally they are short-circuit routed to a proxy that is
designated for emergency calls.

> 
> Our national regulator (Germany) will never allow ERC-systems 
> dependent 
> on one proxy without redundancy!
Redundancy will be required, I agree

> 
> > 
> > A LOC entry is a single point.  You need a boundary.  In 
> fact, you need
> > a list of polygons.
> 
> And how do you define a polygon? If I do remind well from my 
> math-lessons by several points ...
Yes, of course.  There are many details.  We need a new RR to be
able to have a list of polygons.  We need to have the argument about
the format chosen for LOC.  I don't think this is a big problem,
but it will need new DNS machinery.

> 
> > The "nearest" ERC is not correct.  ERCs have defined
> > service boundaries.  You must intersect the location (a 
> point) with the
> > boundary to determine which ERC service area the point is located.
> > I'll expand this later, but there are circumstances where 
> there might be
> > more than one ERC serving a given location.  This is even 
> more obvious
> > with levels above the ERC; the boundary for the U.S. and 
> Canada for example,
> > cannot be reduced to a point where you can determine which 
> side of the
> > border you are on by comparing distance from a single point to the
> > target location.
> 
> A polygon of LOCs ...
A list of a list of LOCS (or something equivalent)

> 
> > 
> > Your outline of walking the tree differs in detail from my proposal,
> > but it is similar.  I start at sos.arpa.  You start at 
> 1.1.9.e164.arpa.
> > I use 2 character ISO country codes.  You use e.164 numeric 
> country codes.
> 
> To be discussed. But I'd prefer ENUM as starting-point to 
> avoid problems 
> with dialing by numeric pads.
I agree that a phone with a numeric pad only has to be able to reliably
place an emergency call.  I don't think we can use ENUM to do that.

> 
> > You don't identify the form of "whatever you want".  I use political
> > subdivisions that actually do delegation for lower levels.  
> You suffix
> > all entries with 1.1.9, I'm working on a couple different ways of
> > identifying
> > ERC entries.  Otherwise, the BASIC idea is the same, the details are
> > quite different.
> 
> It's my goal to set up a flexible, common layer based system 
> independent 
> from political issues - you never know how long political/national 
> borders are kept (no Sowjet Union anymore, Germany, France, Great 
> Britain, ... may become EU and maybe Iraq the 53rd federal 
> state of the 
> US ;-)
Yes, but you have to deal with the fact that emergency response is a
government function, and is organized by different governments in different
ways.  My proposal will be acceptable only if the civil format is acceptable
to every government.  There has been work on this by Henning Schultzrinne.
I do think that we can have a multiple level system where the exact
meaning of each level is not standardized, but the process for delegation
and populating is.  That would allow any country to organize its civil
location formats as it prefers, and still allow a phone manufactured by any
vendor work anywhere, and be able to reliably make emergency calls anywhere.

I think that is what you propose, and we are just having a minor semantic
difficulty.  In my example, you don't need to know that "allegheny" is
a county.  Its just a level in the DNS tree, where there are levels above
and below it.
> 
> My suggestion allows to use any domains below the top-layer domain.
> 
Which I agree with.

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Subject: Re: [Sip] Suggestion to locate nearest ERC-center by ENUM
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Rosen, Brian schrieb:
> This is WAY beyond anything the IETF can do, and I think is not
> practical at this time.  If you want to take such an effort on, please 
> go ahead and let us know if you suceed.  In the mean time, we will have
> to deal with current reality, which is that there is no universal number.
> 

What if they set up a RFC for a inernational ERC-number? I think 
regulators will adapt it, even if it will last years.

> You are dreaming, I think.  Very few phones will know how to do
> an ENUM lookup.

The ENUM-drafts delegate the DNS-lookup to the phones. This is 
reasonable as it will decrease load for the proxies and increase 
redundancy as DNS is spread over the world. And it's just some lines of 
code in the firmware. Simon Morlat has just implemented ENUM into 
Linphone within a few days.

And without dreaming and visions we'd still live in trees ... ;-)

> It is not practical for a phone to do the
> calculations involved in determining the correct psap.
> Therefore, you will depend on a proxy to do it.  There could
> be more than one (actually, I personally think that EVERY proxy
> server should be capable of handling an emergency call, even
> if normally they are short-circuit routed to a proxy that is
> designated for emergency calls.
> 

This would mean a phone must be capable to handle several proxies.

> 
> Yes, of course.  There are many details.  We need a new RR to be
> able to have a list of polygons.  We need to have the argument about
> the format chosen for LOC.  I don't think this is a big problem,
> but it will need new DNS machinery.
> 

Just a firmware-/software update on the DNS machinery, if really necessary.

But maybe we can define a geographical area based on one point and a 
exponential function. That function could just be added to the 
text-field with coordinates of the standard LOC-entry.

> 
> I agree that a phone with a numeric pad only has to be able to reliably
> place an emergency call.  I don't think we can use ENUM to do that.
> 

Why not? DNS has proven being the most reliable addressing system in 
world for 15 years. And setting up a zone-file is quite simple.

> 
> Yes, but you have to deal with the fact that emergency response is a
> government function, and is organized by different governments in different
> ways.  My proposal will be acceptable only if the civil format is acceptable
> to every government.  There has been work on this by Henning Schultzrinne.
> I do think that we can have a multiple level system where the exact
> meaning of each level is not standardized, but the process for delegation
> and populating is.  That would allow any country to organize its civil
> location formats as it prefers, and still allow a phone manufactured by any
> vendor work anywhere, and be able to reliably make emergency calls anywhere.
> 
> I think that is what you propose, and we are just having a minor semantic
> difficulty.  In my example, you don't need to know that "allegheny" is
> a county.  Its just a level in the DNS tree, where there are levels above
> and below it.

In my example, "alleghny" is just a unspecified level, too.

> 
>>My suggestion allows to use any domains below the top-layer domain.
>>
> 
> Which I agree with.

The difference between your ideas and mine is just the way of 
addressing. I want to use ENUM only for reasons of reliability and 
redundancy and combine the caller's data only in the ERC-center for 
reasons of privacy, while you want to use proxies which need excessive 
encryption to ensure privacy and a more complex SIP-protocol, which 
really isn't "Simple" anymore. And all data of each human being will be 
published in DNS - a SPAM-friendly world I don't want to live in!

I've learned to use simple solutions as complex solutions are quite fragile.

Rene


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rene Bartsch'" <ml@bartschnet.de>, sip@ietf.org
Subject: RE: [Sip] Suggestion to locate nearest ERC-center by ENUM
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> > This is WAY beyond anything the IETF can do, and I think is not
> > practical at this time.  If you want to take such an effort 
> on, please 
> > go ahead and let us know if you suceed.  In the mean time, 
> we will have
> > to deal with current reality, which is that there is no 
> universal number.
> > 
> 
> What if they set up a RFC for a inernational ERC-number? I think 
> regulators will adapt it, even if it will last years.
Because the world numbering plan is "owned" by the ITU, and any attempt
to define anything in the e.164 area is going to be impossibly hard.
I don't think the IETF would even try to do this, and I don't think
we have to.

Actually, I think that there is no 3 digit number that would not
be in use in some country for a purpose other than emergency calls.

> 
> > You are dreaming, I think.  Very few phones will know how to do
> > an ENUM lookup.
> 
> The ENUM-drafts delegate the DNS-lookup to the phones. This is 
> reasonable as it will decrease load for the proxies and increase 
> redundancy as DNS is spread over the world. And it's just 
> some lines of 
> code in the firmware. Simon Morlat has just implemented ENUM into 
> Linphone within a few days.
I don't think the drafts talk about who does the lookup.
Most of the systems I know do not have ENUM In the phones, it is
in routing proxies.  I don't see that changing because there isn't
any benefit.  We will have to change the way phones work to
provide emergency calls, so in theory we could do it, but I
don't think it's reasonable, or necessary.

I also don't think that numbers are the right way forward for
any kind of address, but of course it will be aa LONG time
before numeric-only dialing will be obsolete.

> 
> And without dreaming and visions we'd still live in trees ... ;-)
> 
> > It is not practical for a phone to do the
> > calculations involved in determining the correct psap.
> > Therefore, you will depend on a proxy to do it.  There could
> > be more than one (actually, I personally think that EVERY proxy
> > server should be capable of handling an emergency call, even
> > if normally they are short-circuit routed to a proxy that is
> > designated for emergency calls.
> > 
> 
> This would mean a phone must be capable to handle several proxies.
Which is a heck of a lot easier than intersecting a point in a series
of polygons - in code, execution time, and probability of getting it
right in corner cases.

> 
> > 
> > Yes, of course.  There are many details.  We need a new RR to be
> > able to have a list of polygons.  We need to have the argument about
> > the format chosen for LOC.  I don't think this is a big problem,
> > but it will need new DNS machinery.
> > 
> 
> Just a firmware-/software update on the DNS machinery, if 
> really necessary.
Yes, I think this is relatively easy, although difficult to get 
agreement on.

> 
> But maybe we can define a geographical area based on one point and a 
> exponential function. That function could just be added to the 
> text-field with coordinates of the standard LOC-entry.
No, the ERC service area is not defined by any function, it is defined
by a political process and described by an irregular physical boundary.
There is no way to use any simple function to choose the ERC.  You need
to know the service boundary polygons of each ERC and intersect your
location with them until you find the one that you are inside of.

There are circumstances in which you will find that you are in more than
one service area, but I'll leave that problem for a little later.
> 
> > 
> > I agree that a phone with a numeric pad only has to be able 
> to reliably
> > place an emergency call.  I don't think we can use ENUM to do that.
> > 
> 
> Why not? DNS has proven being the most reliable addressing system in 
> world for 15 years. And setting up a zone-file is quite simple.
I'm happy to use the DNS.  I don't think you can use e164.arpa as the
root.

Would you agree that the only difference between your proposal and 
mine comes down to two things:
1. The root of the tree you want to see is <dialing prefix>.e164.arpa
and the root I propose is <iso country code>.sos.arpa?
2. The terminal nodes in the tree you propose is something like 1.1.9.
while the terminal node I propose is something like newcom (or any name
for the psap).  I'm still discussing my proposal with some DNS folks.
It may be that the DNS entry is newcom.psap.allegheny.pa.us.sos.arpa,
or it may be that the newcom.allegheny.pa.us.arpa is somehow marked
as a psap entry.

I think that you can't really take advantage of the fact that the
terminal part of your proposal is the emergency calling number,
because the middle of the tree is not ENUM.  If you branch off
the real ENUM tree, then it doesn't matter, much, what the
terminal part of the entry looks like - it's a new definition
of a part of the DNS.  The name doesn't matter to the phone;
it has to find all of the service boundaries in the tree, and
find the intersection of the point in the polygon.

A complication we have to deal with is that in some areas,
the psap is located at different depths of the tree. So, for
example, you might have a statewide ERC center, but a city
ERC for the largest few cities.  This means either that
the service boundary has holes in it, or that you need a
"most specific" kind of search of the tree (and maybe both).

I don't think that your proposal is any simpler than mine.
I think I have merely thought out exactly how the data in the
DNS gets populated, and how you deal with geo as well as civil
location.

Take, for example, the interior layout information.
Right now, in most places I know about, the way that is kept is
that the fire marshall carries a big box of paper in his truck.
The way that location to a building or a room is communicated between
an enterprise and the psap involves a bunch of forms, two or three
intermediaries and, sometimes, a fax, that maps a phony phone number
to a room/floor/building string.  

I'm proposing a mechanism that we know will scale well to delegate
and populate a database that contains the information needed to 
respond.  Consider, for example, that EVERY CALL needs both a 
geo and a civil location:

You need a geo to find the ERC service boundary, as I have explained.
You need a civil to dispatch a responder.

Therefore, we need a good source of civil to geo translation.
We need that to match what is called a "Master Street Address Guide".
We need that to work down to a room on a floor in a building.

Not every phone will have geo, not every phone will have civil, but
every phone will, we propose, have one or the other.  That means
we need a universal, reliable conversion database between civil
and geo.
> 
> > 
> > Yes, but you have to deal with the fact that emergency response is a
> > government function, and is organized by different 
> governments in different
> > ways.  My proposal will be acceptable only if the civil 
> format is acceptable
> > to every government.  There has been work on this by 
> Henning Schultzrinne.
> > I do think that we can have a multiple level system where the exact
> > meaning of each level is not standardized, but the process 
> for delegation
> > and populating is.  That would allow any country to 
> organize its civil
> > location formats as it prefers, and still allow a phone 
> manufactured by any
> > vendor work anywhere, and be able to reliably make 
> emergency calls anywhere.
> > 
> > I think that is what you propose, and we are just having a 
> minor semantic
> > difficulty.  In my example, you don't need to know that 
> "allegheny" is
> > a county.  Its just a level in the DNS tree, where there 
> are levels above
> > and below it.
> 
> In my example, "alleghny" is just a unspecified level, too.
> 
> > 
> >>My suggestion allows to use any domains below the top-layer domain.
> >>
> > 
> > Which I agree with.
> 
> The difference between your ideas and mine is just the way of 
> addressing. I want to use ENUM only for reasons of reliability and 
> redundancy and combine the caller's data only in the ERC-center for 
> reasons of privacy, while you want to use proxies which need 
> excessive 
> encryption to ensure privacy and a more complex SIP-protocol, which 
> really isn't "Simple" anymore. And all data of each human 
> being will be 
> published in DNS - a SPAM-friendly world I don't want to live in!
As above, I think we have the same mechanism to address, up to the
ERC boundary location.  I propose to go beyond that to solve another
real problem.

The proxies don't need to decrypt any data to route a call.  The
ERC would have to decrypt data below building level in order to
dispatch to a specific place in a building.  I think that is 
quite reasonable (and the decrypting can be done in advance).  I
propose to use the DNS to publish the data because the delegation
mechanism matches what we need so well.  Having the civil to geo
translation database to street address level will be usefull for
many other purposes as well.  

I think the SIP part of the problem is in the eye of the beholder.
I don't think UAs will implement ENUM, and I think proxies will
route emergency calls.  You think phones will implement ENUM
and phones will be able to directly route calls.  The SIP messaging
is the same, right?  Its only where routing decisions are made.

I don't think there is any more data on individual humans in my
proposal than yours.  There is a more boundary information
for the interior of buildings.  In some jurisdictions, this
is public data already.  In most it is not, which is why it needs
to be protected.  I think the mechanism to do that is simple,
and is only between the publisher (building owner or tenant
for example) and the emergency response providers.
> 
> I've learned to use simple solutions as complex solutions are 
> quite fragile.
Agree

But you have to solve the whole problem, not just a part of it.

In emergency calls, there are 4 basic things you have to do:
1. Recognize a call as an emergency call
2. Route the call to the correct ERC
3. Deliver location of the caller
4. Provide a useable call-back identifier for the call

The ERCs would also like to be able to prevent the UA from hanging
up the call, but I don't think we can give that to them.

Brian


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From exim@www1.ietf.org  Wed Jan 14 20:17:00 2004
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Thread-Topic: Question regarding RFC 3265.
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From: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
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Subject: [Sip] Question regarding RFC 3265.
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Hi:

Per section 3.3.2 of RFC 3265, a NOTIFY request is considered failed if =
a response times out and that if the subscription was installed using =
soft-state (SUBSCRIBE), the notifier should remove the subscription. The =
reason for this being the subscriber could have crashed or disappeared =
for ever.=20
While the above mentioned approach works well where the reasons provided =
in the RFC are the real reasons; it has an undesirable side effect in =
scenarios where a NOTIFY timesout due to a temporary network outage, an =
intermediate proxy crash, et al.  It results in a subscriber not getting =
status until such time he/she attempts to reSUBSCRIBE. Thus applications =
using the same like MWI, presence, dialog state break down until such =
period...... Wanted to know if this issue was considered??

Regards,
Venkatesh



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Subject: RE: [Sip] Question regarding RFC 3265.
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> Per section 3.3.2 of RFC 3265, a NOTIFY request 
> is considered failed if a response times out and 
> that if the subscription was installed using 
> soft-state (SUBSCRIBE), the notifier 
> should remove the subscription. The reason for this 
> being the subscriber could have crashed or disappeared 
> for ever. 
> While the above mentioned approach works well where the 
> reasons provided in the RFC are the real reasons; it has an 
> undesirable side effect in scenarios where a NOTIFY timesout 
> due to a temporary network outage, an intermediate proxy 
> crash, et al.  It results in a subscriber not getting status 
> until such time he/she attempts to reSUBSCRIBE. Thus 
> applications using the same like MWI, presence, dialog state 
> break down until such period...... Wanted to know if this 
> issue was considered??

Yes.  The same concept is associated with the 
session-timer and rfc3261; a 408 type situation
triggers the release of a call.

The strength is currently only a SHOULD.
If you want to avoid some customer
complaints, you might not want to apply
the SHOULD to every event package.
You potentially can satisfy
the "undue strain on a network" aspects
of 3.2.2 by allowing a 408 type situation
to trigger a temporary throttling mechanism 
instead of removing the subscription.


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Subject: Re: [Sip] re: WGLC: resource priority
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In a message dated 12/31/2003 4:46:50 PM Eastern Standard Time, 
rohan@cisco.com writes:


> 1. There is no normative description of the behavior of SIP devices 
> which receive a particular priority value.  Some specifications needs 
> to either specify detailed behavior for specific priority values, or 
> the detailed semantics of those values.  I think this belongs in the 
> IANA registration of each namespace, but those references are currently 
> very thin.  I made some specific comments to this end in the 
> appropriate sections.
> 

[MAP] I believe this document should be limited to the definition of the data 
(header) and leave detailed behaviour and procedures to other documents that 
define specific services/capabilities which use it. It should not specify 
detailed behaviour for specific priority values. The references to IANA 
registration are thin because they only describe what IANA needs to deal with - 
assigning an ordered set of names for the priority levels for each "namespace".

> 
> 3. Finally, the document shows that proxies can modify, insert, and 
> delete the RP header. I don't think this is a good idea. Where 
> cooperating UAs and proxies need to agree on an RP value, I believe the 
> session policy mechanism can be used.  A proxy can reject the message 
> for policy reasons (use a lower/higher/specific resource-value) and the 
> UA can retry the request with these new values.
> 


[MAP] While it may not be normal for a proxy to modify or delete an RP 
header, there may be services defined using this header that are based on proxies 
doing exactly that. For example, an originator tries to place a call using a 
high priority header. The proxy authenticates, determines tha the user is only 
authorized a lower priority level, automatically lowers it by changing the RP 
header, and forwards the call on its way, without requiring the UA to retry the 
call. Other documents would describe various operations, possibly including 
some cases in which a proxy can not modify the RP header.

Mike Pierce



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 12/31/2003 4:46:50 PM Eastern Standard Time, rohan@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">1. There is no normative de=
scription of the behavior of SIP devices=20
<BR>which receive a particular priority value. &nbsp;Some specifications nee=
ds=20
<BR>to either specify detailed behavior for specific priority values, or=20
<BR>the detailed semantics of those values. &nbsp;I think this belongs in th=
e=20
<BR>IANA registration of each namespace, but those references are currently=20
<BR>very thin. &nbsp;I made some specific comments to this end in the=20
<BR>appropriate sections.
<BR></BLOCKQUOTE>
<BR>
<BR>[MAP] I believe this document should be limited to the definition of the=
 data (header) and leave detailed behaviour and procedures to other document=
s that define specific services/capabilities which use it. It should not spe=
cify detailed behaviour for specific priority values. The references to IANA=
 registration are thin because they only describe what IANA needs to deal wi=
th - assigning an ordered set of names for the priority levels for each "nam=
espace".
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>3. Finally, the document shows that proxies can modify, insert, and=20
<BR>delete the RP header. I don't think this is a good idea. Where=20
<BR>cooperating UAs and proxies need to agree on an RP value, I believe the=20
<BR>session policy mechanism can be used. &nbsp;A proxy can reject the messa=
ge=20
<BR>for policy reasons (use a lower/higher/specific resource-value) and the=20
<BR>UA can retry the request with these new values.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>[MAP] While it may not be normal for a proxy to modify or delete an RP h=
eader, there may be services defined using this header that are based on pro=
xies doing exactly that. For example, an originator tries to place a call us=
ing a high priority header. The proxy authenticates, determines tha the user=
 is only authorized a lower priority level, automatically lowers it by chang=
ing the RP header, and forwards the call on its way, without requiring the U=
A to retry the call. Other documents would describe various operations, poss=
ibly including some cases in which a proxy can not modify the RP header.
<BR>
<BR>Mike Pierce
<BR>
<BR></FONT></HTML>

--part1_128.3964bb9e.2d37542c_boundary--

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From: Bilajbegovic Damir <damir.bilajbegovic@siemens.com>
To: sip@ietf.org
Subject: [Sip] Two IP addresses (c= lines) in invite
Date: Thu, 15 Jan 2004 09:41:08 +0100
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    If I sent a INVITE message with two IP addresses (i.e. user has two IP
addresses and on one it should receive messages) what happens?
Is it valid and how the terminal or proxy should react on this. (there are
c= lines )
And one more additional question: If one c= is IPv6 and another line is c=
IPv4 and colleen (B) party is IPv4 what does is do? Will B party (or B
proxy) reject the message, replay with deleted misunderstood lines, or will
put the port numbers on 0.  
The message invite is down below (I think it should look like this- is it
good?) 
 Thanks in advance.  
 
            Damir Bilajbegovic
 
INVITE tel:+1-212-555-2222 SIP/2.0
 
  ............ 
 
v=0
o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd
s=-
c=IN IP6 5555::aaa:bbb:ccc:ddd 
t=0 0
m=video 3400 RTP/AVP 98 99
b=AS:75
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos none remote sendrecv
a=rtpmap:98 H263
a=fmtp:98 profile-level-id=0
a=rtpmap:99 MP4V-ES
 
 
c=IN IP6 5555::aaa:bbb:ccc:eee   
t=0 0
m=video 3400 RTP/AVP 98 99
b=AS:75
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos none remote sendrecv
a=rtpmap:98 H263
a=fmtp:98 profile-level-id=0
a=rtpmap:99 MP4V-ES

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Date: Thu, 15 Jan 2004 18:10:55 +0900
From: OKUMURA Shinji <shin@softfront.co.jp>
Subject: Re: [Sip] WGLC for PUBLISH
To: Rohan Mahy <rohan@cisco.com>
Cc: sip@ietf.org, Dean Willis <dean.willis@softarmor.com>, aki.niemi@nokia.com
Organization: SOFTFRONT
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$B1|B<$G$9!#(B
$B$*Hh$lMM$G$9!#(B

$B0J2<$N%a!<%k$r1QLu$7$F$/$@$5$$!#(B

-----------------------------------------------------------------------
[1] 12. Examples$B$N(B
	M1$B$N(B
	      Contact: <sip:watcher@example.com>
	$B$O(B
	      Contact: <sip:watcher@10.0.0.1>
	
	M2$B$N(B
	      Contact: <sip:pa@example.com>
	$B$O(B
	      Contact: <sip:pa.example.com>

	M3$B$N(B
	      NOTIFY sip:presentity@example.com SIP/2.0
	      Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
	      To: <sip:watcher@example.com>;tag=12341234
	      From: <sip:presentity@example.com>;tag=abcd1234
	      Call-ID: 12345678@10.0.0.1
	      CSeq: 1 NOTIFY
				.....
	$B$O(B
	      NOTIFY sip:watcher@10.0.0.1 SIP/2.0
	      Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
	      To: <sip:watcher@example.com>;tag=12341234
	      From: <sip:presentity@example.com>;tag=abcd1234
	      Call-ID: 12345678@10.0.0.1
	      CSeq: 1 NOTIFY
	      Contact: <sip:pa.example.com>
				.....
	$B$G$O$J$$$+!#(B

[2] 12. Examples$B$NNc$O(B
       PUA <---> PA <---> WATCHER
$B$NB>$K(B
       PUA <---> proxy <---> PA
$B$H$$$&Nc$bM_$7$$!#(B
$BNc$($P(B
       PUA <-----> proxy <------> PA
        | M1 PUBLISH |            |
        |----------->| M2 PUBLISH |
        |            |----------->|
        |            | M3 200 OK  |
        | M4 200 OK  |<-----------|
        |<-----------|            |

	M1
      PUBLISH sip:pa@example.com SIP/2.0
      Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
      From: <sip:presentity@example.com>;tag=1234wxyz
      To: <pres:presentity@example.com>
      Call-ID: 81818181@pua.example.com
      CSeq: 1 PUBLISH
      Max-Forwards: 70
      Expires: 3600
      Event: presence
      Content-Type: application/pidf+xml
      Content-Length: ...

	M2
      PUBLISH sip:pa.example.com SIP/2.0
      Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKpx123
      Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
      From: <sip:presentity@example.com>;tag=1234wxyz
      To: <pres:presentity@example.com>
      Call-ID: 81818181@pua.example.com
      CSeq: 1 PUBLISH
      Max-Forwards: 70
      Expires: 3600
      Event: presence
      Content-Type: application/pidf+xml
      Content-Length: ...

	M3
      SIP/2.0 200 OK
      Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKpx123
      Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
      From: <sip:presentity@example.com>;tag=1234wxyz
      To: <pres:presentity@example.com>;tag=1a2b3c4d
      Call-ID: 81818181@pua.example.com
      CSeq: 1 PUBLISH
      SIP-ETag: dx200xyz
      Expires: 1800
      Content-Length: 0

	M4
      SIP/2.0 200 OK
      Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
      From: <sip:presentity@example.com>;tag=1234wxyz
      To: <pres:presentity@example.com>;tag=1a2b3c4d
      Call-ID: 81818181@pua.example.com
      CSeq: 1 PUBLISH
      SIP-ETag: dx200xyz
      Expires: 1800
      Content-Length: 0

[3] Request-URI$B$O%a%C%;!<%8$N08@h$H$7$F;H$$!"%W%l%<%s%9%j%=!<%9$N(B
	$B;XDj$O(BTo$B%X%C%@$G9T$&$h$&$K$7$?J}$,NI$$$H;W$&!#(B
	REGISTER$B$K$h$k%m%1!<%7%g%sEPO?$HF1$8F0:n$KJ}$,$o$+$j$d$9$$!#(B
	$B$3$NE@$O(BRFC3265$B$K$D$$$F$bF1$8$3$H$,8@$($k!#(B
	$B$J$<$J$i!"(BRFC3256$B$K$O(B

	|3.1.5. Proxy SUBSCRIBE Behavior
	|  Proxies need no additional behavior beyond that described in SIP [1]
	|  to support SUBSCRIBE.

$B$H$"$k$+$i$G$"$k!#(B

[4]$B%W%l%<%s%9%j%=!<%9$r$"$i$o$9(BURI$B$O(Bsip-URI$B$G$O$J$/!"(B
pres-URI(draft-ietf-impp-pres-04)$B$,NI$$$H;W$&!#(B

$B7k6I(B[3][4]$B$r9g$o$;$k$H0J2<$N$h$&$J(BURI$B$N;H$$J}$,NI$$$H;W$&!#(B
+------------+-----------------------------+-----------------------------+-----------------------------+
|            | PUBLISH                     | SUBSCRIBE                   | NOTIFY                      |
+------------+-----------------------------+-----------------------------+-----------------------------+
|Request-URI | URI of PA                   | URI of PA                   | Contact-URI of SUBSCRIBE    |
|            | sip:pa@example.com          | sip:pa@example.com          | sip:watcher.example.com     |
+------------+-----------------------------+-----------------------------+-----------------------------+
|To-URI      | URI of presence resource    | URI of presence resource    | From-URI of SUBSCRIBE       |
|            | pres:presentity@example.com | pres:presentity@example.com | sip:watcher@example.com     |
+------------+-----------------------------+-----------------------------+-----------------------------+
|From-URI    | URI of presentity           | URI of watcher              | To-URI of SUBSCRIBE         |
|            | sip:presentity@example.com  | sip:watcher@example.com     | pres:presentity@example.com |
+------------+-----------------------------+-----------------------------+-----------------------------+


Rohan Mahy wrote in <C3094B8F-4563-11D8-B291-0003938AF740@cisco.com>
>Hi,
>
>I'd like to begin Working Group Last Call for PUBLISH:
>
>	http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-02.txt
>
>This WGLC will end on January 28, 2004.
>
>thanks,
>-rohan
___________________________________________________________
$B1|B<(B $B?-Fs(B ($B3t(B)$B%=%U%H%U%m%s%H(B $B")(B060-0009 $BKL(B9$B>r@>(B15$BCzL\(B28-195
tel:+81-11-623-1003             mailto:shin@softfront.co.jp
fax:+81-11-623-1005                sip:shin@softfront.co.jp

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From exim@www1.ietf.org  Thu Jan 15 04:57:00 2004
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Date: Thu, 15 Jan 2004 10:53:38 +0100 (CET)
From: Bernie Hoeneisen <bhoeneis@switch.ch>
To: Bilajbegovic Damir <damir.bilajbegovic@siemens.com>
cc: sip@ietf.org
Subject: Re: [Sip] Two IP addresses (c= lines) in invite
In-Reply-To: <F4E9A9AFE620F94FA8FD4C69F39BE4B302B57ED4@zagh102a.zag.siemens.hr>
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Hi Damir!

Have you already checked this in RFC 2327 and RFC 3266?

  http://www.ietf.org/rfc/rfc2327 (SDP)
  http://www.ietf.org/rfc/rfc3266 (IPv6 extensions to SDP)

  (Note, that in RFC 3266 there is a typo in section 1, where it refers to
  RFC 2328. It should be RFC 2327.)

According to RFC 2327 (section 6) you can have either one connection 'c= '
line for the whole session or then serveral connection 'c= ' lines per
media 'm= ' line.  In the lattter the connection 'c= ' lines must appear
after the media 'm= ' lines.

I just wonder, how many SIP implementations out in the world do handle
this as standardized...

cheers,
 Bernie



On Thu, 15 Jan 2004, Bilajbegovic Damir wrote:

>     If I sent a INVITE message with two IP addresses (i.e. user has two IP
> addresses and on one it should receive messages) what happens?
> Is it valid and how the terminal or proxy should react on this. (there are
> c= lines )
> And one more additional question: If one c= is IPv6 and another line is c=
> IPv4 and colleen (B) party is IPv4 what does is do? Will B party (or B
> proxy) reject the message, replay with deleted misunderstood lines, or will
> put the port numbers on 0.
> The message invite is down below (I think it should look like this- is it
> good?)
>  Thanks in advance.
>
>             Damir Bilajbegovic
>
> INVITE tel:+1-212-555-2222 SIP/2.0
>
>   ............
>
> v=0
> o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd
> s=-
> c=IN IP6 5555::aaa:bbb:ccc:ddd
> t=0 0
> m=video 3400 RTP/AVP 98 99
> b=AS:75
> a=curr:qos local none
> a=curr:qos remote none
> a=des:qos mandatory local sendrecv
> a=des:qos none remote sendrecv
> a=rtpmap:98 H263
> a=fmtp:98 profile-level-id=0
> a=rtpmap:99 MP4V-ES
>
>
> c=IN IP6 5555::aaa:bbb:ccc:eee
> t=0 0
> m=video 3400 RTP/AVP 98 99
> b=AS:75
> a=curr:qos local none
> a=curr:qos remote none
> a=des:qos mandatory local sendrecv
> a=des:qos none remote sendrecv
> a=rtpmap:98 H263
> a=fmtp:98 profile-level-id=0
> a=rtpmap:99 MP4V-ES
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Thu, 15 Jan 2004 19:49:09 +0900
From: OKUMURA Shinji <shin@softfront.co.jp>
Subject: Re: [Sip] WGLC for PUBLISH
To: Rohan Mahy <rohan@cisco.com>
Cc: sip@ietf.org, Dean Willis <dean.willis@softarmor.com>, aki.niemi@nokia.com
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Hi,

[1] I believe it requires following changes in some headers of
    the illustrated example in section 12.

    In M1,   Contact: <sip:watcher@example.com> should be rewritten as
             Contact: <sip:watcher@10.0.0.1>

    In M2,   Contact: <sip:pa@example.com> should be
             Contact: <sip:pa.example.com>

    In M3,    NOTIFY sip:presentity@example.com SIP/2.0
              Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
              To: <sip:watcher@example.com>;tag=12341234
              From: <sip:presentity@example.com>;tag=abcd1234
              Call-ID: 12345678@10.0.0.1
              CSeq: 1 NOTIFY
              ....

              should be

              NOTIFY sip:watcher@10.0.0.1 SIP/2.0
              Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
              To: <sip:watcher@example.com>;tag=12341234
              From: <sip:presentity@example.com>;tag=abcd1234
              Call-ID: 12345678@10.0.0.1
              CSeq: 1 NOTIFY
              Contact: <sip:pa.example.com>
              ....
 
[2] It could be helpful if the following example (2) is added in 
    the document as an addition to the only example (1) illustrated 
    in section 12.

    (1) PUA <---> PA <---> WATCHER

    (2) PUA <---> proxy <---> PA

        For example:

        PUA <-----> proxy <------> PA
         |            |            |
         | M1 PUBLISH |            |
         |----------->| M2 PUBLISH |
         |            |----------->|
         |            | M3 200 OK  |
         | M4 200 OK  |<-----------|
         |<-----------|            |
 
        M1:
            PUBLISH sip:pa@example.com SIP/2.0
            Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
            From: <sip:presentity@example.com>;tag=1234wxyz
            To: <pres:presentity@example.com>
            Call-ID: 81818181@pua.example.com
            CSeq: 1 PUBLISH
            Max-Forwards: 70
            Expires: 3600
            Event: presence
            Content-Type: application/pidf+xml
            Content-Length: ...
 
        M2:
            PUBLISH sip:pa.example.com SIP/2.0
            Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKpx123
            Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
            From: <sip:presentity@example.com>;tag=1234wxyz
            To: <pres:presentity@example.com>
            Call-ID: 81818181@pua.example.com
            CSeq: 1 PUBLISH
            Max-Forwards: 70
            Expires: 3600
            Event: presence
            Content-Type: application/pidf+xml
            Content-Length: ...
 
        M3:
            SIP/2.0 200 OK
            Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKpx123
            Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
            From: <sip:presentity@example.com>;tag=1234wxyz
            To: <pres:presentity@example.com>;tag=1a2b3c4d
            Call-ID: 81818181@pua.example.com
            CSeq: 1 PUBLISH
            SIP-ETag: dx200xyz
            Expires: 1800
            Content-Length: 0
 
        M4:
            SIP/2.0 200 OK
            Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
            From: <sip:presentity@example.com>;tag=1234wxyz
            To: <pres:presentity@example.com>;tag=1a2b3c4d
            Call-ID: 81818181@pua.example.com
            CSeq: 1 PUBLISH
            SIP-ETag: dx200xyz
            Expires: 1800
            Content-Length: 0
 
[3] It could be better if the "Request-URI" is used as the message's
    target address and the 'Presence Resource' is specified using "To"
    header.

    It would be easy for understanding if REGISTER's location 
    registration operation like mechanism is considered.

    The same point could also be said about RFC3265. This is because,

        |3.1.5. Proxy SUBSCRIBE Behavior
        |  Proxies need no additional behavior beyond that described in SIP [1]
        |  to support SUBSCRIBE.

[4] For expressing 'Presence Resource', the pres-URI(draft-ietf-impp-pres-04)
    would be better than that of the sip-URI.

    Finally, when [3] & [4] are considered together, the various usages 
    of URI can be expressed as listed below.

+------------+-----------------------------+-----------------------------+-----------------------------+
|            | PUBLISH                     | SUBSCRIBE                   | NOTIFY                      |
+------------+-----------------------------+-----------------------------+-----------------------------+
|Request-URI | URI of PA                   | URI of PA                   | Contact-URI of SUBSCRIBE    |
|            | sip:pa@example.com          | sip:pa@example.com          | sip:watcher.example.com     |
+------------+-----------------------------+-----------------------------+-----------------------------+
|To-URI      | URI of presence resource    | URI of presence resource    | From-URI of SUBSCRIBE       |
|            | pres:presentity@example.com | pres:presentity@example.com | sip:watcher@example.com     |
+------------+-----------------------------+-----------------------------+-----------------------------+
|From-URI    | URI of presentity           | URI of watcher              | To-URI of SUBSCRIBE         |
|            | sip:presentity@example.com  | sip:watcher@example.com     | pres:presentity@example.com |
+------------+-----------------------------+-----------------------------+-----------------------------+

Regards,
shinji

Rohan Mahy wrote in <C3094B8F-4563-11D8-B291-0003938AF740@cisco.com>
>Hi,
>
>I'd like to begin Working Group Last Call for PUBLISH:
>
>	http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-02.txt
>
>This WGLC will end on January 28, 2004.
>
>thanks,
>-rohan
_________________________________________________
OKUMURA Shinji        E-mail:shin@softfront.co.jp

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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 exim@www1.ietf.org  Thu Jan 15 07:08:57 2004
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From: Bilajbegovic Damir <damir.bilajbegovic@siemens.com>
To: Bernie Hoeneisen <bhoeneis@switch.ch>
Cc: sip@ietf.org
Subject: RE: [Sip] Two IP addresses (c= lines) in invite
Date: Thu, 15 Jan 2004 12:56:40 +0100
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Hi,
     Yes I looked for it and I did not see the answer. 
It answers about hawing the lines "c= ..." but it doesn't say how it reacts
when this message is not valid.

	Regards  Damir Bilajbegovic
-----Original Message-----
From: Bernie Hoeneisen [mailto:bhoeneis@switch.ch] 
Sent: Thursday, January 15, 2004 10:54 AM
To: Bilajbegovic Damir
Cc: sip@ietf.org
Subject: Re: [Sip] Two IP addresses (c= lines) in invite


Hi Damir!

Have you already checked this in RFC 2327 and RFC 3266?

  http://www.ietf.org/rfc/rfc2327 (SDP)
  http://www.ietf.org/rfc/rfc3266 (IPv6 extensions to SDP)

  (Note, that in RFC 3266 there is a typo in section 1, where it refers to
  RFC 2328. It should be RFC 2327.)

According to RFC 2327 (section 6) you can have either one connection 'c= '
line for the whole session or then serveral connection 'c= ' lines per media
'm= ' line.  In the lattter the connection 'c= ' lines must appear after the
media 'm= ' lines.

I just wonder, how many SIP implementations out in the world do handle this
as standardized...

cheers,
 Bernie



On Thu, 15 Jan 2004, Bilajbegovic Damir wrote:

>     If I sent a INVITE message with two IP addresses (i.e. user has
> two IP addresses and on one it should receive messages) what happens? 
> Is it valid and how the terminal or proxy should react on this. (there 
> are c= lines ) And one more additional question: If one c= is IPv6 and 
> another line is c= IPv4 and colleen (B) party is IPv4 what does is do? 
> Will B party (or B
> proxy) reject the message, replay with deleted misunderstood lines, or 
> will put the port numbers on 0. The message invite is down below (I 
> think it should look like this- is it
> good?)
>  Thanks in advance.
>
>             Damir Bilajbegovic
>
> INVITE tel:+1-212-555-2222 SIP/2.0
>
>   ............
>
> v=0
> o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd
> s=-
> c=IN IP6 5555::aaa:bbb:ccc:ddd
> t=0 0
> m=video 3400 RTP/AVP 98 99
> b=AS:75
> a=curr:qos local none
> a=curr:qos remote none
> a=des:qos mandatory local sendrecv
> a=des:qos none remote sendrecv
> a=rtpmap:98 H263
> a=fmtp:98 profile-level-id=0
> a=rtpmap:99 MP4V-ES
>
>
> c=IN IP6 5555::aaa:bbb:ccc:eee
> t=0 0
> m=video 3400 RTP/AVP 98 99
> b=AS:75
> a=curr:qos local none
> a=curr:qos remote none
> a=des:qos mandatory local sendrecv
> a=des:qos none remote sendrecv
> a=rtpmap:98 H263
> a=fmtp:98 profile-level-id=0
> a=rtpmap:99 MP4V-ES
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/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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Date: Thu, 15 Jan 2004 14:00:59 +0100 (CET)
From: Bernie Hoeneisen <bhoeneis@switch.ch>
To: Bilajbegovic Damir <damir.bilajbegovic@siemens.com>
cc: sip@ietf.org
Subject: RE: [Sip] Two IP addresses (c= lines) in invite
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Hi Damir!

What about:  400 Bad Request

RFC 3261, 21.4.1: "400 Bad Request
                   The request could not be understood due to
                   malformed syntax. ..."

But I have a feeling, that is is not what you are looking for.
Maybe you'd ask the sip-implementors list for current practice on this
issue:  http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

Regards,
 Bernie

On Thu, 15 Jan 2004, Bilajbegovic Damir wrote:

> Hi,
>      Yes I looked for it and I did not see the answer.
> It answers about hawing the lines "c= ..." but it doesn't say how it reacts
> when this message is not valid.
>
> 	Regards  Damir Bilajbegovic
> -----Original Message-----
> From: Bernie Hoeneisen [mailto:bhoeneis@switch.ch]
> Sent: Thursday, January 15, 2004 10:54 AM
> To: Bilajbegovic Damir
> Cc: sip@ietf.org
> Subject: Re: [Sip] Two IP addresses (c= lines) in invite
>
>
> Hi Damir!
>
> Have you already checked this in RFC 2327 and RFC 3266?
>
>   http://www.ietf.org/rfc/rfc2327 (SDP)
>   http://www.ietf.org/rfc/rfc3266 (IPv6 extensions to SDP)
>
>   (Note, that in RFC 3266 there is a typo in section 1, where it refers to
>   RFC 2328. It should be RFC 2327.)
>
> According to RFC 2327 (section 6) you can have either one connection 'c= '
> line for the whole session or then serveral connection 'c= ' lines per media
> 'm= ' line.  In the lattter the connection 'c= ' lines must appear after the
> media 'm= ' lines.
>
> I just wonder, how many SIP implementations out in the world do handle this
> as standardized...
>
> cheers,
>  Bernie
>
>
>
> On Thu, 15 Jan 2004, Bilajbegovic Damir wrote:
>
> >     If I sent a INVITE message with two IP addresses (i.e. user has
> > two IP addresses and on one it should receive messages) what happens?
> > Is it valid and how the terminal or proxy should react on this. (there
> > are c= lines ) And one more additional question: If one c= is IPv6 and
> > another line is c= IPv4 and colleen (B) party is IPv4 what does is do?
> > Will B party (or B
> > proxy) reject the message, replay with deleted misunderstood lines, or
> > will put the port numbers on 0. The message invite is down below (I
> > think it should look like this- is it
> > good?)
> >  Thanks in advance.
> >
> >             Damir Bilajbegovic
> >
> > INVITE tel:+1-212-555-2222 SIP/2.0
> >
> >   ............
> >
> > v=0
> > o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd
> > s=-
> > c=IN IP6 5555::aaa:bbb:ccc:ddd
> > t=0 0
> > m=video 3400 RTP/AVP 98 99
> > b=AS:75
> > a=curr:qos local none
> > a=curr:qos remote none
> > a=des:qos mandatory local sendrecv
> > a=des:qos none remote sendrecv
> > a=rtpmap:98 H263
> > a=fmtp:98 profile-level-id=0
> > a=rtpmap:99 MP4V-ES
> >
> >
> > c=IN IP6 5555::aaa:bbb:ccc:eee
> > t=0 0
> > m=video 3400 RTP/AVP 98 99
> > b=AS:75
> > a=curr:qos local none
> > a=curr:qos remote none
> > a=des:qos mandatory local sendrecv
> > a=des:qos none remote sendrecv
> > a=rtpmap:98 H263
> > a=fmtp:98 profile-level-id=0
> > a=rtpmap:99 MP4V-ES
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/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 exim@www1.ietf.org  Thu Jan 15 08:53:42 2004
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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: <sip@ietf.org>
Subject: [Sip] re: WGLC: resource priority 
Date: Thu, 15 Jan 2004 08:50:34 -0500
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My apologies for the tardy comments on this thread.

> From: Rohan Mahy <rohan@cisco.com> 
> Date: Wed, 31 Dec 2003 13:42:01 -0800
>
> 1. There is no normative description of the behavior of SIP devices 
> which receive a particular priority value. Some specifications needs
> to either specify detailed behavior for specific priority values, or
> the detailed semantics of those values. I think this belongs in the
> IANA registration of each namespace, but those references are
currently 
> very thin. I made some specific comments to this end in the
appropriate 
> sections.

I'll agree with Mike Pierce on this.  The header is just a vessel
designed to facilitate carrying a variety of values.  Specifications and
descriptions of behavior should not be in this draft.

> include a registration template (for example)
>
> Namespace:
> Description:
> Organization Responsible for Change Control of this Namespace:
> Priority values (ordered from least priority to greatest priority):
> Default priority:
> Preempt or Prioritize:
> Document Containing Normative Description:
> Section Containing Normative Description:
> Additional security requirements:

I'm not thrilled about a registration template in this document at this
time.  My feeling is that should be addressed by subsequent document(s)
since the form of the registration is outside the scope of the header.  

As an example, I would take issue with line three above "Organization
responsible for Change Control".  On the surface, this implies
versioning of the namespace, which we don't have in the R-P header
format, and continued control by that organization.  Outside of who is
right or wrong on that point of discussion, it seems to me that its out
of scope of the definition of the header.

> would also consider including Section B.1 as section 9.4, B.2 -> 9.5, 
> B.3 -> 9.6

Given what I've stated above, I'd like section B to stay outside of the
body of the draft.  The items listed in that section are only an example
to consider.

Also, my preference would be to remove section A.  I think its helpful
for the drafts initial introduction to the group, but in the end, it's a
bit of overkill for the final document.

Regards,

-ken


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Date: Thu, 15 Jan 2004 09:08:51 -0800
Subject: Re: [Sip] re: WGLC: resource priority
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Hi Mike,

comments inline...

On Wednesday, January 14, 2004, at 06:25 PM, Mpierce1@aol.com wrote:
> In a message dated 12/31/2003 4:46:50 PM Eastern Standard Time,=20
> rohan@cisco.com writes:
>
> 1. There is no normative description of the behavior of SIP devices
> which receive a particular priority value. =A0Some specifications =
needs
> to either specify detailed behavior for specific priority values, or
> the detailed semantics of those values. =A0I think this belongs in the
> IANA registration of each namespace, but those references are =
currently
> very thin. =A0I made some specific comments to this end in the
> appropriate sections.
>
> [MAP] I believe this document should be limited to the definition of=20=

> the data (header) and leave detailed behaviour and procedures to other=20=

> documents that define specific services/capabilities which use it. It=20=

> should not specify detailed behaviour for specific priority values.=20
> The references to IANA registration are thin because they only=20
> describe what IANA needs to deal with - assigning an ordered set of=20
> names for the priority levels for each "namespace".

How will an *implementer* determine correct behavior without a=20
normative description or reference?  This kind of reference is not=20
unusual in IANA registrations.

> 3. Finally, the document shows that proxies can modify, insert, and
> delete the RP header. I don't think this is a good idea. Where
> cooperating UAs and proxies need to agree on an RP value, I believe =
the
> session policy mechanism can be used. =A0A proxy can reject the =
message
> for policy reasons (use a lower/higher/specific resource-value) and =
the
> UA can retry the request with these new values.
>
> [MAP] While it may not be normal for a proxy to modify or delete an RP=20=

> header, there may be services defined using this header that are based=20=

> on proxies doing exactly that. For example, an originator tries to=20
> place a call using a high priority header. The proxy authenticates,=20
> determines tha the user is only authorized a lower priority level,=20
> automatically lowers it by changing the RP header, and forwards the=20
> call on its way, without requiring the UA to retry the call. Other=20
> documents would describe various operations, possibly including some=20=

> cases in which a proxy can not modify the RP header.

How is this any different from any of the other proxy modifications=20
(ex: modifying SDP) which we've decided to avoid by using session=20
policy?  Cooperating user agents can easily find out what the policy is=20=

and would only very rarely have to retry a call (no extra round trips=20
for most calls).  Non-cooperating user agents have their non-policy=20
conforming requests rejected.

thanks,
-rohan


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Hi Mike,


comments inline...


On Wednesday, January 14, 2004, at 06:25 PM, Mpierce1@aol.com wrote:

<excerpt><fontfamily><param>Arial</param><smaller>In a message dated
12/31/2003 4:46:50 PM Eastern Standard Time, rohan@cisco.com writes:


</smaller></fontfamily>1. There is no normative description of the
behavior of SIP devices

which receive a particular priority value. =A0Some specifications needs

to either specify detailed behavior for specific priority values, or

the detailed semantics of those values. =A0I think this belongs in the

IANA registration of each namespace, but those references are currently

very thin. =A0I made some specific comments to this end in the

appropriate sections.


[MAP] I believe this document should be limited to the definition of
the data (header) and leave detailed behaviour and procedures to other
documents that define specific services/capabilities which use it. It
should not specify detailed behaviour for specific priority values.
The references to IANA registration are thin because they only
describe what IANA needs to deal with - assigning an ordered set of
names for the priority levels for each "namespace".

</excerpt>

How will an *implementer* determine correct behavior without a
normative description or reference?  This kind of reference is not
unusual in IANA registrations.


<excerpt>3. Finally, the document shows that proxies can modify,
insert, and

delete the RP header. I don't think this is a good idea. Where

cooperating UAs and proxies need to agree on an RP value, I believe the

session policy mechanism can be used. =A0A proxy can reject the message

for policy reasons (use a lower/higher/specific resource-value) and the

UA can retry the request with these new values.


=
<fontfamily><param>Arial</param><color><param>0000,0000,0000</param><small=
er>[MAP]
While it may not be normal for a proxy to modify or delete an RP
header, there may be services defined using this header that are based
on proxies doing exactly that. For example, an originator tries to
place a call using a high priority header. The proxy authenticates,
determines tha the user is only authorized a lower priority level,
automatically lowers it by changing the RP header, and forwards the
call on its way, without requiring the UA to retry the call. Other
documents would describe various operations, possibly including some
cases in which a proxy can not modify the RP header.

</smaller></color></fontfamily></excerpt>

How is this any different from any of the other proxy modifications
(ex: modifying SDP) which we've decided to avoid by using session
policy?  Cooperating user agents can easily find out what the policy
is and would only very rarely have to retry a call (no extra round
trips for most calls).  Non-cooperating user agents have their
non-policy conforming requests rejected.


thanks,

-rohan



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Subject: Re: [Sip] re: WGLC: resource priority
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Regarding IANA Considerations:

In a message dated 12/31/2003 4:46:50 PM Eastern Standard Time, 
rohan@cisco.com writes:


> include a registration template  (for example)
> 
> Namespace:
> Description:
> Organization Responsible for Change Control of this Namespace:
> Priority values (ordered from least priority to greatest priority):
> Default priority:
> Preempt or Prioritize:
> Document Containing Normative Description:
> Section Containing Normative Description:
> Additional security requirements:
> 
> Namespace: q735
> Description: ITU Q.735.3 Multi-level precedence and preemption in SS7
> Organization Responsible for Change Control of this Namespace: ITU-T
> Priority values (ordered from least priority to greatest priority): 4, 
> 3, 2, 1, 0
> Default priority: 4
> Preempt or Prioritize: prioritize
> Document Containing Normative Description: ITU-T Q.735.3 [3]
> Section Containing Normative Description: ??
> Additional security requirements: none
> 
> Namespace: dsrn
> Description: United States Defense Red Switched Network
> Organization Responsible for Change Control of this Namespace: US 
> Department of Defense?
> Priority values (ordered from least priority to greatest priority):
>    routine, priority, immediate, flash, flash-override, 
> flash-override-override
> Default priority: routine
> Preempt or Prioritize: preempt
> Document Containing Normative Description: FIPS XXXX ??  [?]
> Section Containing Normative Description: ??
> Additional security requirements: All implementations must follow the 
> certificate management guidelines specified in xxxx [?]
> 


I presume that, if these namespaces are registered by inclusion in this RFC, 
then change control continues to be under control of the IETF, by revision of 
this RFC. (In fact, the text indicates that changes "should not" occur.)

Other namespaces registered later are under change control of whoever 
registers them. (It isn't clear to me who can register new namespaces, and how.)

The IANA considerations and registration information should only contain what 
is needed for the registration. "Preempt or Prioritize" is not needed as part 
of the registration. It is a detail to be described in other RFCs or 
documents for the particular application. For example, some service might require that 
either be done depending on other conditions. In addition, there may be 
preferential treatments that would not be classified as either "preempt" or 
"prioritize", as described in my draft on the subject. Note that Section 2 of this 
draft lists 4 treatments, some of which I would not classify as being either 
"preempt" or "prioritize".

The security requirements depend on the application, so the registration 
should not make normative statements about what "MUST" be done.

Further, in Section 12, I believe it is necessary that the registration 
indicate the relative precedence levels (as shown in the above suggested template), 
not "may". It doesn't seem to make sense to not require specifying order, but 
yet require a default.

I continue to believe that every namespace must have at least two priority 
values defined. The statements in Section 3 indicate otherwise. A namespace with 
only one value (which therefore must be the default) would be useless.

Mike Pierce



--part1_139.2a12df5e.2d38242a_boundary
Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>Regarding IAN=
A Considerations:
<BR>
<BR>In a message dated 12/31/2003 4:46:50 PM Eastern Standard Time, rohan@ci=
sco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">include a registration temp=
late &nbsp;(for example)
<BR>
<BR>Namespace:
<BR>Description:
<BR>Organization Responsible for Change Control of this Namespace:
<BR>Priority values (ordered from least priority to greatest priority):
<BR>Default priority:
<BR>Preempt or Prioritize:
<BR>Document Containing Normative Description:
<BR>Section Containing Normative Description:
<BR>Additional security requirements:
<BR>
<BR>Namespace: q735
<BR>Description: ITU Q.735.3 Multi-level precedence and preemption in SS7
<BR>Organization Responsible for Change Control of this Namespace: ITU-T
<BR>Priority values (ordered from least priority to greatest priority): 4,=20
<BR>3, 2, 1, 0
<BR>Default priority: 4
<BR>Preempt or Prioritize: prioritize
<BR>Document Containing Normative Description: ITU-T Q.735.3 [3]
<BR>Section Containing Normative Description: ??
<BR>Additional security requirements: none
<BR>
<BR>Namespace: dsrn
<BR>Description: United States Defense Red Switched Network
<BR>Organization Responsible for Change Control of this Namespace: US=20
<BR>Department of Defense?
<BR>Priority values (ordered from least priority to greatest priority):
<BR> &nbsp;&nbsp;routine, priority, immediate, flash, flash-override,=20
<BR>flash-override-override
<BR>Default priority: routine
<BR>Preempt or Prioritize: preempt
<BR>Document Containing Normative Description: FIPS XXXX ?? &nbsp;[?]
<BR>Section Containing Normative Description: ??
<BR>Additional security requirements: All implementations must follow the=20
<BR>certificate management guidelines specified in xxxx [?]
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>I presume that, if these namespaces are registered by inclusion in this=20=
RFC, then change control continues to be under control of the IETF, by revis=
ion of this RFC. (In fact, the text indicates that changes "should not" occu=
r.)
<BR>
<BR>Other namespaces registered later are under change control of whoever re=
gisters them. (It isn't clear to me who can register new namespaces, and how=
.)
<BR>
<BR>The IANA considerations and registration information should only contain=
 what is needed for the registration. "Preempt or Prioritize" is not needed=20=
as part of the registration. It is a detail to be described in other RFCs or=
 documents for the particular application. For example, some service might r=
equire that either be done depending on other conditions. In addition, there=
 may be preferential treatments that would not be classified as either "pree=
mpt" or "prioritize", as described in my draft on the subject. Note that Sec=
tion 2 of this draft lists 4 treatments, some of which I would not classify=20=
as being either "preempt" or "prioritize".
<BR>
<BR>The security requirements depend on the application, so the registration=
 should not make normative statements about what "MUST" be done.
<BR>
<BR>Further, in Section 12, I believe it is necessary that the registration=20=
indicate the relative precedence levels (as shown in the above suggested tem=
plate), not "may". It doesn't seem to make sense to not require specifying o=
rder, but yet require a default.
<BR>
<BR>I continue to believe that every namespace must have at least two priori=
ty values defined. The statements in Section 3 indicate otherwise. A namespa=
ce with only one value (which therefore must be the default) would be useles=
s.
<BR>
<BR>Mike Pierce
<BR>
<BR></FONT></HTML>

--part1_139.2a12df5e.2d38242a_boundary--

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Use sip-implementors@cs.columbia.edu for questions on current sip
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From exim@www1.ietf.org  Thu Jan 15 14:30:48 2004
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From: "Paul Long" <plong@packetizer.com>
To: <sip@ietf.org>
Subject: RE: [Sip] SIP DTMF
Date: Thu, 15 Jan 2004 13:21:56 -0600
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Sreeram,

I can't find a "kpxml" draft on the IETF site or via Google--anywhere.
Can you provide a link to it?

Also, from my experience, the preferred method of DTMF carriage is RFC
2833, followed by in-band audio, and then various proprietary
INFO-method schemes. What is the standing of RFC 2833 vis-=E0-vis =
"kpxml?"

Paul

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
sreeram.kanumuri@wipro.com
Sent: Tuesday, January 13, 2004 9:06 PM
To: xyxie@udtech.com.cn; sip@ietf.org
Cc: sipping@ietf.org
Subject: RE: [Sip] SIP DTMF

Wendy,

There was a discussion in the list some time back and it was decided
that SIP-INFO method should not be used to=20
Send the DTMF info.

It was proposed that it is better to you kpxml for sending DTMF tones.

Please look in to the SIP-Kpxml draft for sending DTMF tomes in-midcall
processing and all....=20


Regards,
Sreeram

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D
Sreeram Kanumuri                Plot No 72, Keonics Electronics City
			              Hosur Main Road
Wipro Technologies              Bangalore -29 INDIA
sreeram.kanumuri@wipro.com      Phone: 8520408 Ex:3208
skanumury@yahoo.com             ESN  :
www.wipro.com                   Mobile:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of wendy
Sent: Wednesday, January 14, 2004 8:08 AM
To: sip@ietf.org
Cc: sipping@ietf.org
Subject: [Sip] SIP DTMF


Hello all,=20

Who can tell me whether there is a RFC or I-D on DTMF in SIP which is
still not expired? I remember someone proposed using SIP INFO method for
DTMF one or two years ago. Is the DTMF relay implemented only by RTP
now?


Regards,
Wendy


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Subject: Re: [Sip] re: WGLC: resource priority
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In a message dated 1/15/2004 12:07:39 PM Eastern Standard Time, 
rohan@cisco.com writes:


> How will an *implementer* determine correct behavior without a 
> normative description or reference?  This kind of reference is not 
> unusual in IANA registrations.
> 

[MAP] I think the proper relationship of documents is that the implementor 
will be given a requirements document for equipment to be purchased. (Might be 
another RFC or could be a document produced by the entity that is going to 
purchase the equipement - like the DOD.) That document would point back to the 
RP-header RFC for the definition of that part of the protocol, just like it would 
point to lots of other documents.

The RP-header document does not have to make forward references to those 
documents which might come later or that give all the procedural requirements.

In a message dated 1/15/2004 12:07:39 PM Eastern Standard Time, 
rohan@cisco.com writes:


> How is this any different from any of the other proxy modifications 
> (ex: modifying SDP) which we've decided to avoid by using session 
> policy?  Cooperating user agents can easily find out what the policy is 
> and would only very rarely have to retry a call (no extra round trips 
> for most calls).  Non-cooperating user agents have their non-policy 
> conforming requests rejected.
> 


[MAP] I understand all this, but it doesn't mean that the RP-header spec 
should mandate that Proxies can not change or insert the header, which I believe 
is what you were suggesting. I feel there are legitimate cases in which it 
could do this. For example, if the originating user does not enter a header, the 
proxy uses the default, and might insert the header with the default value. Or, 
if the user includes a header with a priority values higher than what they 
are allowed, the proxy may simply lower the value to what they are allowed and 
forward the call. The RP-header definition should not forbid this.

This is not an issue of "cooperating user agents" finding out what the policy 
is. The user (person) pushes a button and the UA (telephone) codes an INVITE 
based on what buttons they pushed. Some networks/service providers may not 
want to reject the call and make the user try it again. It may not be the UA but 
the person who controls the retry or "knows" the policy.

Mike Pierce


--part1_1d2.17ec8e80.2d3847aa_boundary
Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 1/15/2004 12:07:39 PM Eastern Standard Time, rohan@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">How will an *implementer* d=
etermine correct behavior without a=20
<BR>normative description or reference? &nbsp;This kind of reference is not=20
<BR>unusual in IANA registrations.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>[MAP] I think the proper relationship of documents is that the implement=
or will be given a requirements document for equipment to be purchased. (Mig=
ht be another RFC or could be a document produced by the entity that is goin=
g to purchase the equipement - like the DOD.) That document would point back=
 to the RP-header RFC for the definition of that part of the protocol, just=20=
like it would point to lots of other documents.
<BR>
<BR>The RP-header document does not have to make forward references to those=
 documents which might come later or that give all the procedural requiremen=
ts.
<BR>
<BR>In a message dated 1/15/2004 12:07:39 PM Eastern Standard Time, rohan@ci=
sco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">How is this any different f=
rom any of the other proxy modifications=20
<BR>(ex: modifying SDP) which we've decided to avoid by using session=20
<BR>policy? &nbsp;Cooperating user agents can easily find out what the polic=
y is=20
<BR>and would only very rarely have to retry a call (no extra round trips=20
<BR>for most calls). &nbsp;Non-cooperating user agents have their non-policy=
=20
<BR>conforming requests rejected.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>[MAP] I understand all this, but it doesn't mean that the RP-header spec=
 should mandate that Proxies can not change or insert the header, which I be=
lieve is what you were suggesting. I feel there are legitimate cases in whic=
h it could do this. For example, if the originating user does not enter a he=
ader, the proxy uses the default, and might insert the header with the defau=
lt value. Or, if the user includes a header with a priority values higher th=
an what they are allowed, the proxy may simply lower the value to what they=20=
are allowed and forward the call. The RP-header definition should not forbid=
 this.
<BR>
<BR>This is not an issue of "cooperating user agents" finding out what the p=
olicy is. The user (person) pushes a button and the UA (telephone) codes an=20=
INVITE based on what buttons they pushed. Some networks/service providers ma=
y not want to reject the call and make the user try it again. It may not be=20=
the UA but the person who controls the retry or "knows" the policy.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

--part1_1d2.17ec8e80.2d3847aa_boundary--

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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: Re: [Sip] re: WGLC: resource priority
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From: Rohan Mahy <rohan@cisco.com>
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On Thursday, January 15, 2004, at 09:13 AM, Mpierce1@aol.com wrote:
> I presume that, if these namespaces are registered by inclusion in 
> this RFC, then change control continues to be under control of the 
> IETF, by revision of this RFC. (In fact, the text indicates that 
> changes "should not" occur.)
> Other namespaces registered later are under change control of whoever 
> registers them. (It isn't clear to me who can register new namespaces, 
> and how.)

Please do not confuse the change control of the *registration* with the 
change control of the *normative behavior*.  Only the change control of 
the registration is under the control of the IETF. Obviously if you 
change the normative behavior in a separate document, you need to 
change the registration to point at the changes behavior.

Recall that the IANA registration serves two purposes: 1) to prevent 
namespace conflicts, and 2) to allows implementors to find the 
documents relevant to any namespaces which interest them.

> The IANA considerations and registration information should only 
> contain what is needed for the registration. "Preempt or Prioritize" 
> is not needed as part of the registration. It is a detail to be 
> described in other RFCs or documents for the particular application. 
> For example, some service might require that either be done depending 
> on other conditions. In addition, there may be preferential treatments 
> that would not be classified as either "preempt" or "prioritize", as 
> described in my draft on the subject. Note that Section 2 of this 
> draft lists 4 treatments, some of which I would not classify as being 
> either "preempt" or "prioritize".

Can you provide an existence proof of a separate type of treatment?

> The security requirements depend on the application, so the 
> registration should not make normative statements about what "MUST" be 
> done.

That should be purely at the discretion of the registrar of each 
namespace.  If additional security requirements are warranted then they 
can certainly be included.  If there are no additional requirements or 
requirements are deferred to other documents then the registration 
should indicate that.

> Further, in Section 12, I believe it is necessary that the 
> registration indicate the relative precedence levels (as shown in the 
> above suggested template), not "may". It doesn't seem to make sense to 
> not require specifying order, but yet require a default.
> I continue to believe that every namespace must have at least two 
> priority values defined. The statements in Section 3 indicate 
> otherwise. A namespace with only one value (which therefore must be 
> the default) would be useless.

agreed to both.

thanks,
-rohan


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On Thursday, January 15, 2004, at 09:13 AM, Mpierce1@aol.com wrote:

<excerpt><fontfamily><param>Arial</param><color><param>0000,0000,0000</param><smaller>I
presume that, if these namespaces are registered by inclusion in this
RFC, then change control continues to be under control of the IETF, by
revision of this RFC. (In fact, the text indicates that changes
"should not" occur.)

Other namespaces registered later are under change control of whoever
registers them. (It isn't clear to me who can register new namespaces,
and how.)

</smaller></color></fontfamily></excerpt>

Please do not confuse the change control of the *registration* with
the change control of the *normative behavior*.  Only the change
control of the registration is under the control of the IETF.
Obviously if you change the normative behavior in a separate document,
you need to change the registration to point at the changes behavior.


Recall that the IANA registration serves two purposes: 1) to prevent
namespace conflicts, and 2) to allows implementors to find the
documents relevant to any namespaces which interest them.


<excerpt><fontfamily><param>Arial</param><color><param>0000,0000,0000</param><smaller>The
IANA considerations and registration information should only contain
what is needed for the registration. "Preempt or Prioritize" is not
needed as part of the registration. It is a detail to be described in
other RFCs or documents for the particular application. For example,
some service might require that either be done depending on other
conditions. In addition, there may be preferential treatments that
would not be classified as either "preempt" or "prioritize", as
described in my draft on the subject. Note that Section 2 of this
draft lists 4 treatments, some of which I would not classify as being
either "preempt" or "prioritize".

</smaller></color></fontfamily></excerpt>

Can you provide an existence proof of a separate type of treatment?


<excerpt><fontfamily><param>Arial</param><color><param>0000,0000,0000</param><smaller>The
security requirements depend on the application, so the registration
should not make normative statements about what "MUST" be done.

</smaller></color></fontfamily></excerpt>

That should be purely at the discretion of the registrar of each
namespace.  If additional security requirements are warranted then
they can certainly be included.  If there are no additional
requirements or requirements are deferred to other documents then the
registration should indicate that.


<excerpt><fontfamily><param>Arial</param><color><param>0000,0000,0000</param><smaller>Further,
in Section 12, I believe it is necessary that the registration
indicate the relative precedence levels (as shown in the above
suggested template), not "may". It doesn't seem to make sense to not
require specifying order, but yet require a default.

I continue to believe that every namespace must have at least two
priority values defined. The statements in Section 3 indicate
otherwise. A namespace with only one value (which therefore must be
the default) would be useless.

</smaller></color></fontfamily></excerpt>

agreed to both.


thanks,

-rohan



--Apple-Mail-10--747681949--


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Use sip-implementors@cs.columbia.edu for questions on current sip
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To: Paul Kyzivat <pkyzivat@cisco.com>
Cc: ksrini@motorola.com, sip@ietf.org
Subject: Re: [Sip] Registering Multiple 'Contacts'
References: <Pine.LNX.4.21.0401121159440.23050-100000@bulb.corp.mot.com>
	<4002E0B9.8030203@cisco.com>
Organization: Pingtel Corp. http://www.pingtel.com/
From: Scott Lawrence <slawrence@pingtel.com>
Date: Thu, 15 Jan 2004 14:56:03 -0500
In-Reply-To: <4002E0B9.8030203@cisco.com> (Paul Kyzivat's message of "Mon,
 12 Jan 2004 13:00:25 -0500")
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Srinivasan Krishnamoorthy wrote:

>>   Section 10.2.4 of RFC3261 'Refreshing Bindings' says:
>>   "The UA then issues a REGISTER request for each of its bindings
>> before the expiration interval has elapsed. It MAY combine   several
>> updates into one REGISTER request."
>>   The above seems to indicate that the UAC may send:   (forgive the
>> Syntax)
>>         REGISTER         To: user@operator.net
>>         Contact: sip:user1@10.0.0.1;expires=3600
>>                 200 OK
>>         REGISTER         To: user@operator.net
>>         Contact: sip:user1@192.0.0.1;expires=3600
>>                 200 OK
>>   This, according to RFC3261, will register both Contact1 and Contact2
>> for user@operator.net. (The server will then use the q parameter
>>   of the contact for priority among contacts)

Paul Kyzivat <pkyzivat@cisco.com> writes:

> Yes.

My reading is that the results should differ depending on the Call-Id
used in those two REGISTER requests:  

  - If they have different Call-Id values, then the server should
    register both contacts.

  - If they have the same Call-Id value, then the second REGISTER
    should cause the server to replace the first contact with the
    second, so that there would be only

If correct, this adds an additional alternative for the UA - if it can
remember its Call-Id values across contact changes, it can make sure
that old registrations are replaced even before they expire.

-- 
Scott Lawrence        
  Pingtel Corp.   


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Date: Thu, 15 Jan 2004 12:10:07 -0800
Subject: Re: [Sip] re: WGLC: resource priority
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On Thursday, January 15, 2004, at 11:44 AM, Mpierce1@aol.com wrote:

> In a message dated 1/15/2004 12:07:39 PM Eastern Standard Time,=20
> rohan@cisco.com writes:
>
>
> How will an *implementer* determine correct behavior without a
> normative description or reference? =A0This kind of reference is not
> unusual in IANA registrations.
>
> [MAP] I think the proper relationship of documents is that the=20
> implementor will be given a requirements document for equipment to be=20=

> purchased. (Might be another RFC or could be a document produced by=20
> the entity that is going to purchase the equipement - like the DOD.)=20=

> That document would point back to the RP-header RFC for the definition=20=

> of that part of the protocol, just like it would point to lots of=20
> other documents.
>
> The RP-header document does not have to make forward references to=20
> those documents which might come later or that give all the procedural=20=

> requirements.

I believe that the IANA *registration* does need to have such a forward=20=

reference and that the IESG will tend to agree with me.

> In a message dated 1/15/2004 12:07:39 PM Eastern Standard Time,=20
> rohan@cisco.com writes:
>
> How is this any different from any of the other proxy modifications
> (ex: modifying SDP) which we've decided to avoid by using session
> policy? =A0Cooperating user agents can easily find out what the policy =
is
> and would only very rarely have to retry a call (no extra round trips
> for most calls). =A0Non-cooperating user agents have their non-policy
> conforming requests rejected.
>
> [MAP] I understand all this, but it doesn't mean that the RP-header=20
> spec should mandate that Proxies can not change or insert the header,=20=

> which I believe is what you were suggesting. I feel there are=20
> legitimate cases in which it could do this. For example, if the=20
> originating user does not enter a header, the proxy uses the default,=20=

> and might insert the header with the default value. Or, if the user=20
> includes a header with a priority values higher than what they are=20
> allowed, the proxy may simply lower the value to what they are allowed=20=

> and forward the call. The RP-header definition should not forbid this.
>
> This is not an issue of "cooperating user agents" finding out what the=20=

> policy is. The user (person) pushes a button and the UA (telephone)=20
> codes an INVITE based on what buttons they pushed. Some=20
> networks/service providers may not want to reject the call and make=20
> the user try it again. It may not be the UA but the person who=20
> controls the retry or "knows" the policy.

The UA should know the policy.  There is no technical reason it can't=20
know or collect the policy and then use the appropriate policy. That's=20=

what the session policy mechanism is for. However, there are a number=20
of technical problems related to security we get when proxies start to=20=

insert or modify RP.

thanks,
-rohan=

--Apple-Mail-14--746556426
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On Thursday, January 15, 2004, at 11:44 AM, Mpierce1@aol.com wrote:


<excerpt><fontfamily><param>Arial</param><smaller>In a message dated
1/15/2004 12:07:39 PM Eastern Standard Time, rohan@cisco.com writes:



</smaller></fontfamily>How will an *implementer* determine correct
behavior without a

normative description or reference? =A0This kind of reference is not

unusual in IANA registrations.


=
<fontfamily><param>Arial</param><color><param>0000,0000,0000</param><small=
er>[MAP]
I think the proper relationship of documents is that the implementor
will be given a requirements document for equipment to be purchased.
(Might be another RFC or could be a document produced by the entity
that is going to purchase the equipement - like the DOD.) That
document would point back to the RP-header RFC for the definition of
that part of the protocol, just like it would point to lots of other
documents.


The RP-header document does not have to make forward references to
those documents which might come later or that give all the procedural
requirements.

</smaller></color></fontfamily></excerpt>

I believe that the IANA *registration* does need to have such a
forward reference and that the IESG will tend to agree with me.


=
<excerpt><fontfamily><param>Arial</param><color><param>0000,0000,0000</par=
am><smaller>In
a message dated 1/15/2004 12:07:39 PM Eastern Standard Time,
rohan@cisco.com writes:


How is this any different from any of the other proxy modifications

(ex: modifying SDP) which we've decided to avoid by using session

policy? =A0Cooperating user agents can easily find out what the policy =
is

and would only very rarely have to retry a call (no extra round trips

for most calls). =A0Non-cooperating user agents have their non-policy

conforming requests rejected.


[MAP] I understand all this, but it doesn't mean that the RP-header
spec should mandate that Proxies can not change or insert the header,
which I believe is what you were suggesting. I feel there are
legitimate cases in which it could do this. For example, if the
originating user does not enter a header, the proxy uses the default,
and might insert the header with the default value. Or, if the user
includes a header with a priority values higher than what they are
allowed, the proxy may simply lower the value to what they are allowed
and forward the call. The RP-header definition should not forbid this.


This is not an issue of "cooperating user agents" finding out what the
policy is. The user (person) pushes a button and the UA (telephone)
codes an INVITE based on what buttons they pushed. Some
networks/service providers may not want to reject the call and make
the user try it again. It may not be the UA but the person who
controls the retry or "knows" the policy.

</smaller></color></fontfamily></excerpt>

The UA should know the policy.  There is no technical reason it can't
know or collect the policy and then use the appropriate policy. That's
what the session policy mechanism is for. However, there are a number
of technical problems related to security we get when proxies start to
insert or modify RP.


thanks,

-rohan=

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From exim@www1.ietf.org  Thu Jan 15 16:24:39 2004
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From: Srinivasan Krishnamoorthy <ksrini@motorola.com>
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Subject: Re: [Sip] Registering Multiple 'Contacts'
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>> How will the Server prevent against mis-behaved clients in this case?

>The expiration time bounds the problem, but doesn't eliminate it.
>Assume the UAC power cycles, forgets its old address, gets a new one and registers it. It will then presumably continue to refresh the new address, but not the old one. So the old registration will continue to be active until it expires. During that time requests will probably be forked to it and the new address, but the ones to the old address will fail.
>
>Paul

Another case I could think of is a DHCP environment - where a mobile-ua gets 
different ipaddresses on every power-cycle.

In this case we may have to make sure the DHCP lease timers are longer than
registration duration in the network - or else we could be terminating
a call to an old-IP - that has possibly been granted to another UA that
registered in the interim???

it looks like nothing can be done at the server to 'eliminate' the problem.

Thanks
-Srini



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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		: Session Timers in the Session Initiation Protocol (SIP)
	Author(s)	: S. Donovan, J. Rosenberg
	Filename	: draft-ietf-sip-session-timer-13.txt
	Pages		: 36
	Date		: 2004-1-15
	
This document defines an extension to the Session Initiation 
Protocol (SIP).  This extension allows for a periodic refresh of SIP 
sessions through a re-INVITE or UPDATE request. The refresh allows 
both user agents and proxies to determine if the SIP session is 
still active. The extension defines two new header fields, Session-
Expires, which conveys the lifetime of the session, and Min-SE, 
which conveys the minimum allowed value for the session timer.

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To: "'Rohan Mahy'" <rohan@cisco.com>, <Mpierce1@aol.com>
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Subject: RE: [Sip] re: WGLC: resource priority
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>> That should be purely at the discretion of the registrar of each 
>> namespace. If additional security requirements are warranted then 
>> they can certainly be included. If there are no additional 
>> requirements or requirements are deferred to other documents then 
>> the registration should indicate that.
>
> Further, in Section 12, I believe it is necessary that the
registration 
> indicate the relative precedence levels (as shown in the above
suggested 
> template), not "may". It doesn't seem to make sense to not require 
> specifying order, but yet require a default.
> I continue to believe that every namespace must have at least two 
> priority values defined. The statements in Section 3 indicate
otherwise. 
> A namespace with only one value (which therefore must be the default) 
> would be useless.
>
> agreed to both.

I disagree on the latter part that every namespace must have two
"priority" values.  If one has a binary "on/off" NameSpace Value, and
the "off" means that the message is to be treated the same as if the R-P
never existed, then I don't see why an additional Value must exist.  Its
counter to the age old practice of be conservative in what you send.

An example of such a binary condition is if I wanted to have a namespace
that mapped to the binary NS/EP codepoint of T1.631.

-ken



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Subject: Re: [Sip] re: WGLC: resource priority
To: Janet P Gunn <jgunn6@csc.com>
Cc: Mpierce1@aol.com, rohan@cisco.com, sip@ietf.org
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                      Janet P Gunn                                                                                              
                      /FED/CSC                 To:      <Mpierce1@aol.com>                                                      
                                               cc:      rohan@cisco.com, sip@ietf.org                                           
                      01/15/2004 12:16         Subject: Re: [Sip] re: WGLC: resource priority(Document link: Janet P Gunn)      
                      PM                                                                                                        
                                                                                                                                
                                                                                                                                








I support Mike on this one.

The Resource Priority Header is the "protocol", not the "policy".  What
policy will be evoked in a particular Administrative Domain, by a
particular RPH namespace:value will be the subject of an SLA between
relevant parties.  Restrictions on the extent of the policy  (for instance
that it not be the instrument for DoS attacks) are already in the document.

Janet


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                      <Mpierce1                                                                                                 
                      @aol.com>                To:      <rohan@cisco.com>, <sip@ietf.org>                                       
                                               cc:                                                                              
                      01/14/2004 09:25         Subject: Re: [Sip] re: WGLC: resource priority                                   
                      PM                                                                                                        
                                                                                                                                
                                                                                                                                




In a message dated 12/31/2003 4:46:50 PM Eastern Standard Time,
rohan@cisco.com writes:




1. There is no normative description of the behavior of SIP devices
which receive a particular priority value.  Some specifications needs
to either specify detailed behavior for specific priority values, or
the detailed semantics of those values.  I think this belongs in the
IANA registration of each namespace, but those references are currently
very thin.  I made some specific comments to this end in the
appropriate sections.




[MAP] I believe this document should be limited to the definition of the
data (header) and leave detailed behaviour and procedures to other
documents that define specific services/capabilities which use it. It
should not specify detailed behaviour for specific priority values. The
references to IANA registration are thin because they only describe what
IANA needs to deal with - assigning an ordered set of names for the
priority levels for each "namespace".




3. Finally, the document shows that proxies can modify, insert, and
delete the RP header. I don't think this is a good idea. Where
cooperating UAs and proxies need to agree on an RP value, I believe the
session policy mechanism can be used.  A proxy can reject the message
for policy reasons (use a lower/higher/specific resource-value) and the
UA can retry the request with these new values.





[MAP] While it may not be normal for a proxy to modify or delete an RP
header, there may be services defined using this header that are based on
proxies doing exactly that. For example, an originator tries to place a
call using a high priority header. The proxy authenticates, determines tha
the user is only authorized a lower priority level, automatically lowers it
by changing the RP header, and forwards the call on its way, without
requiring the UA to retry the call. Other documents would describe various
operations, possibly including some cases in which a proxy can not modify
the RP header.

Mike Pierce









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To: slawrence@pingtel.com, pkyzivat@cisco.com
Cc: ksrini@motorola.com, sip@ietf.org
Subject: Re: [Sip] Registering Multiple 'Contacts'
From: Takuya Sawada <tu-sawada@kddi.com>
References: <Pine.LNX.4.21.0401121159440.23050-100000@bulb.corp.mot.com>
	<4002E0B9.8030203@cisco.com>
	<vheku1f0ks.fsf@sukothai.pingtel.com>
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Hi,

> 
> Srinivasan Krishnamoorthy wrote:
> 
> >>   Section 10.2.4 of RFC3261 'Refreshing Bindings' says:
> >>   "The UA then issues a REGISTER request for each of its bindings
> >> before the expiration interval has elapsed. It MAY combine   several
> >> updates into one REGISTER request."
> >>   The above seems to indicate that the UAC may send:   (forgive the
> >> Syntax)
> >>         REGISTER         To: user@operator.net
> >>         Contact: sip:user1@10.0.0.1;expires=3600
> >>                 200 OK
> >>         REGISTER         To: user@operator.net
> >>         Contact: sip:user1@192.0.0.1;expires=3600
> >>                 200 OK
> >>   This, according to RFC3261, will register both Contact1 and Contact2
> >> for user@operator.net. (The server will then use the q parameter
> >>   of the contact for priority among contacts)
> 
> Paul Kyzivat <pkyzivat@cisco.com> writes:
> 
> > Yes.
> 
> My reading is that the results should differ depending on the Call-Id
> used in those two REGISTER requests:  
> 
>   - If they have different Call-Id values, then the server should
>     register both contacts.
> 
>   - If they have the same Call-Id value, then the second REGISTER
>     should cause the server to replace the first contact with the
>     second, so that there would be only
> 
In my reading of RFC 3261 section 10.3 step 7, Contact URI will be added
to the list irrelevant to Call-ID value if its value is new to the list.

> If correct, this adds an additional alternative for the UA - if it can
> remember its Call-Id values across contact changes, it can make sure
> that old registrations are replaced even before they expire.
> 
If it can remember any value in the previous power-cycle, it may remember
Contact URI and then can remove it with expires=0 explicitly.

Note that assumeing it uses the same "fixed" Call-ID across power-cycle,
it must remember "dynamic" CSeq value in old power-cycle.

Regards,
Takuya

> -- 
> Scott Lawrence        
>   Pingtel Corp.   
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/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: +81-3-6678-0285
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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Date: Thu, 15 Jan 2004 20:03:08 EST
Subject: Re: [Sip] re: WGLC: resource priority (3)
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In a message dated 1/15/2004 5:12:56 PM Eastern Standard Time, 
carlberg@g11.org.uk writes:


> 
> I disagree on the latter part that every namespace must have two
> "priority" values.  If one has a binary "on/off" NameSpace Value, and
> the "off" means that the message is to be treated the same as if the R-P
> never existed, then I don't see why an additional Value must exist.  Its
> counter to the age old practice of be conservative in what you send.
> 
> An example of such a binary condition is if I wanted to have a namespace
> that mapped to the binary NS/EP codepoint of T1.631.
> 
> 


[MAP] You're going to have to explain how that could work, considering that 
the draft states:

Every namespace must have a default level.
The default is assumed in the absence of a priority level (in a message).

That means that, if there is only one priority value in the namespace (such 
as "on") registered with IANA, it must be the default, and any INVITE that does 
not contain an RP-header is treated as if it contained the default value 
("on"). How would it be possible to signal the "off" value?

If one wants a "binary" on/off, then why not just define two values: ON and 
OFF with OFF as the default? Then either can be signalled, and the absence of 
the RP-header means OFF.

I think the age old practice of being conservative in what you send applies 
to what we say to each other, not what signaling is sent by telecommunications 
devices :>}

Explain how one would signal the NS/EP codepoint case with only one value 
(which is the default).

Mike Pierce




--part1_1d4.1891a8a9.2d38924c_boundary
Content-Type: text/html; charset="US-ASCII"
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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 1/15/2004 5:12:56 PM Eastern Standard Time, carlberg@g11.org.uk writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>I disagree on the latter part that every namespace must have two
<BR>"priority" values. &nbsp;If one has a binary "on/off" NameSpace Value, a=
nd
<BR>the "off" means that the message is to be treated the same as if the R-P
<BR>never existed, then I don't see why an additional Value must exist. &nbs=
p;Its
<BR>counter to the age old practice of be conservative in what you send.
<BR>
<BR>An example of such a binary condition is if I wanted to have a namespace
<BR>that mapped to the binary NS/EP codepoint of T1.631.
<BR>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>[MAP] You're going to have to explain how that could work, considering t=
hat the draft states:
<BR>
<BR>Every namespace must have a default level.
<BR>The default is assumed in the absence of a priority level (in a message)=
.
<BR>
<BR>That means that, if there is only one priority value in the namespace (s=
uch as "on") registered with IANA, it must be the default, and any INVITE th=
at does not contain an RP-header is treated as if it contained the default v=
alue ("on"). How would it be possible to signal the "off" value?
<BR>
<BR>If one wants a "binary" on/off, then why not just define two values: ON=20=
and OFF with OFF as the default? Then either can be signalled, and the absen=
ce of the RP-header means OFF.
<BR>
<BR>I think the age old practice of being conservative in what you send appl=
ies to what we say to each other, not what signaling is sent by telecommunic=
ations devices :&gt;}
<BR>
<BR>Explain how one would signal the NS/EP codepoint case with only one valu=
e (which is the default).
<BR>
<BR>Mike Pierce
<BR>
<BR>
<BR></FONT></HTML>

--part1_1d4.1891a8a9.2d38924c_boundary--

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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: ksrini@motorola.com, sip@ietf.org
Cc: pkyzivat@cisco.com
Subject: Re: [Sip] Registering Multiple 'Contacts'
From: Takuya Sawada <tu-sawada@kddi.com>
References: <vheku1f0ks.fsf@sukothai.pingtel.com>
	<Pine.LNX.4.21.0401151503070.26805-100000@bulb.corp.mot.com>
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Hi,
> 
> >> How will the Server prevent against mis-behaved clients in this case?
> 
> >The expiration time bounds the problem, but doesn't eliminate it.
> >Assume the UAC power cycles, forgets its old address, gets a new one and registers it. It will then 
presumably continue to refresh the new address, but not the old one. So the old registration will continue 
to be active until it expires. During that time requests will probably be forked to it and the new address, 
but the ones to the old address will fail.
> >
> >Paul
> 
> Another case I could think of is a DHCP environment - where a mobile-ua gets 
> different ipaddresses on every power-cycle.
> 
> In this case we may have to make sure the DHCP lease timers are longer than
> registration duration in the network - or else we could be terminating
> a call to an old-IP - that has possibly been granted to another UA that
> registered in the interim???
> 
I think that SIP UA which use dynamic IP address SHOULD obey the behaviour 
described above.
Keep DHCP lease timer > SIP registration timer. (any timer used in app level)

> it looks like nothing can be done at the server to 'eliminate' the problem.
> 
I agree.
As a SIP UAS, it can reject the request which has other Request-URI other than
its registered Contact URI with 404, or just ignore such a request.
This can eliminate or reduce the possibility of wrong call termination to the UAS.
Here I assume that user part of Contact URI is unique enough to avoid accidental
collision.

Regards,
Takuya

> Thanks
> -Srini
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/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: +81-3-6678-0285
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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Date: Thu, 15 Jan 2004 17:46:52 -0800
Subject: Re: [Sip] re: WGLC: resource priority
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Cc: <Mpierce1@aol.com>, <sip@ietf.org>
To: "Ken Carlberg" <carlberg@g11.org.uk>
From: Rohan Mahy <rohan@cisco.com>
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Ken,

The binary namespace you described has TWO values.  (ON and OFF)

thanks,
-rohan


On Thursday, January 15, 2004, at 02:09 PM, Ken Carlberg wrote:

>
>>> That should be purely at the discretion of the registrar of each
>>> namespace. If additional security requirements are warranted then
>>> they can certainly be included. If there are no additional
>>> requirements or requirements are deferred to other documents then
>>> the registration should indicate that.
>>
>> Further, in Section 12, I believe it is necessary that the
> registration
>> indicate the relative precedence levels (as shown in the above
> suggested
>> template), not "may". It doesn't seem to make sense to not require
>> specifying order, but yet require a default.
>> I continue to believe that every namespace must have at least two
>> priority values defined. The statements in Section 3 indicate
> otherwise.
>> A namespace with only one value (which therefore must be the default)
>> would be useless.
>>
>> agreed to both.
>
> I disagree on the latter part that every namespace must have two
> "priority" values.  If one has a binary "on/off" NameSpace Value, and
> the "off" means that the message is to be treated the same as if the 
> R-P
> never existed, then I don't see why an additional Value must exist.  
> Its
> counter to the age old practice of be conservative in what you send.
>
> An example of such a binary condition is if I wanted to have a 
> namespace
> that mapped to the binary NS/EP codepoint of T1.631.
>
> -ken
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Subject: Re: [Sip] re: WGLC: resource priority
To: <Mpierce1@aol.com>
Cc: rohan@cisco.com, sip@ietf.org
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I support Mike on this one.

The Resource Priority Header is the "protocol", not the "policy".  What
policy will be evoked in a particular Administrative Domain, by a
particular RPH namespace:value will be the subject of an SLA between
relevant parties.  Restrictions on the extent of the policy  (for instance
that it not be the instrument for DoS attacks) are already in the document.

Janet


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                      <Mpierce1                                                                                                  
                      @aol.com>                To:      <rohan@cisco.com>, <sip@ietf.org>                                        
                                               cc:                                                                               
                      01/14/2004 09:25         Subject: Re: [Sip] re: WGLC: resource priority                                    
                      PM                                                                                                         
                                                                                                                                 
                                                                                                                                 




In a message dated 12/31/2003 4:46:50 PM Eastern Standard Time,
rohan@cisco.com writes:




1. There is no normative description of the behavior of SIP devices
which receive a particular priority value.  Some specifications needs
to either specify detailed behavior for specific priority values, or
the detailed semantics of those values.  I think this belongs in the
IANA registration of each namespace, but those references are currently
very thin.  I made some specific comments to this end in the
appropriate sections.




[MAP] I believe this document should be limited to the definition of the
data (header) and leave detailed behaviour and procedures to other
documents that define specific services/capabilities which use it. It
should not specify detailed behaviour for specific priority values. The
references to IANA registration are thin because they only describe what
IANA needs to deal with - assigning an ordered set of names for the
priority levels for each "namespace".




3. Finally, the document shows that proxies can modify, insert, and
delete the RP header. I don't think this is a good idea. Where
cooperating UAs and proxies need to agree on an RP value, I believe the
session policy mechanism can be used.  A proxy can reject the message
for policy reasons (use a lower/higher/specific resource-value) and the
UA can retry the request with these new values.





[MAP] While it may not be normal for a proxy to modify or delete an RP
header, there may be services defined using this header that are based on
proxies doing exactly that. For example, an originator tries to place a
call using a high priority header. The proxy authenticates, determines tha
the user is only authorized a lower priority level, automatically lowers it
by changing the RP header, and forwards the call on its way, without
requiring the UA to retry the call. Other documents would describe various
operations, possibly including some cases in which a proxy can not modify
the RP header.

Mike Pierce







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From: "Jesske, R" <R.Jesske@t-com.net>
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Dear all,
in 1.3 Ensuring the Privacy of History_Info it is proposed to use a priv-value of Session or Header level privacy. I would like to propose a own privacy level for the history info, because the Information shows also information about the network. With regard to the provider operating a network some of the providers don't like to transmit information about call history. Or a redirecting party don't like to show the redirecting address.
Therefore it should be possible to delete the history info at domain boundaries based on the privacy-value.

I propose the priv-value   =   "history_info"  for the whole History-Info header.

In addition a option would be useful where only parts (e.g Indes1.1) of the history info are restricted. That could be done with a privacy statement within the History-Info .

Best Regards

Roland


Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T332-2
Section T33; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:      +49 6151 83-4577 
email:   r.jesske@t-com.net




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Subject: RE: [Sip] WGLC for PUBLISH
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Hi Shinji,

Thanks for the comments. Answers inline.

 > -----Original Message-----
 > From: ext OKUMURA Shinji [mailto:shin@softfront.co.jp]
 > Sent: 15 January, 2004 12:49
 > To: Rohan Mahy
 > Cc: sip@ietf.org; Dean Willis; Niemi Aki (Nokia-M/Helsinki)
 > Subject: Re: [Sip] WGLC for PUBLISH
 >=20
 >=20
 > Hi,
 >=20
 > [1] I believe it requires following changes in some headers of
 >     the illustrated example in section 12.
 >=20
 >     In M1,   Contact: <sip:watcher@example.com> should be=20
 > rewritten as
 >              Contact: <sip:watcher@10.0.0.1>
 >
 >=20
 >     In M2,   Contact: <sip:pa@example.com> should be
 >              Contact: <sip:pa.example.com>
 >
 >     In M3,    NOTIFY sip:presentity@example.com SIP/2.0
 >               Via: SIP/2.0/UDP pa.example.com;branch=3Dz9hG4bK8sdf2
 >               To: <sip:watcher@example.com>;tag=3D12341234
 >               From: <sip:presentity@example.com>;tag=3Dabcd1234
 >               Call-ID: 12345678@10.0.0.1
 >               CSeq: 1 NOTIFY
 >               ....
 >=20
 >               should be
 >=20
 >               NOTIFY sip:watcher@10.0.0.1 SIP/2.0
 >               Via: SIP/2.0/UDP pa.example.com;branch=3Dz9hG4bK8sdf2
 >               To: <sip:watcher@example.com>;tag=3D12341234
 >               From: <sip:presentity@example.com>;tag=3Dabcd1234
 >               Call-ID: 12345678@10.0.0.1
 >               CSeq: 1 NOTIFY
 >               Contact: <sip:pa.example.com>
 >               ....

All true. Also, since the Via contains a domain name, I will add a =
"received" param to the Via in the response.

It's very good that you pointed out these mistakes in the examples. I =
swear, I've looked at them a few times before, but I guess my internal =
SIP parser is so well built that it never gave me an error message. ;-)

I'll fix these for the next revision.
 =20
 > [2] It could be helpful if the following example (2) is added in=20
 >     the document as an addition to the only example (1) illustrated=20
 >     in section 12.

I'm a little reluctant to adding this. There are excellent SIP call =
flows already documented elsewhere (see RFC 3665, e.g., Section 3.2), =
and to this particular method, adding a proxy in between the PUA and the =
PA works no  different from how this works for other methods (MESSAGE, =
INVITE). I.e., the proxy rewriting the Request-URI follows the same =
logic as the other methods would.

However, since this issue has come up before, it might be a good idea to =
add a proxy between the PUA and PA. Any other views on this?=20

 >     (1) PUA <---> PA <---> WATCHER
 >=20
 >     (2) PUA <---> proxy <---> PA
 >=20
 >         For example:
 >=20
 >         PUA <-----> proxy <------> PA
 >          |            |            |
 >          | M1 PUBLISH |            |
 >          |----------->| M2 PUBLISH |
 >          |            |----------->|
 >          |            | M3 200 OK  |
 >          | M4 200 OK  |<-----------|
 >          |<-----------|            |
 > =20
 >         M1:
 >             PUBLISH sip:pa@example.com SIP/2.0
 >             Via: SIP/2.0/UDP pua.example.com;branch=3Dz9hG4bK652hsge
 >             From: <sip:presentity@example.com>;tag=3D1234wxyz
 >             To: <pres:presentity@example.com>
 >             Call-ID: 81818181@pua.example.com
 >             CSeq: 1 PUBLISH
 >             Max-Forwards: 70
 >             Expires: 3600
 >             Event: presence
 >             Content-Type: application/pidf+xml
 >             Content-Length: ...

Note that the Request-URI needs to be 'sip:presentity@example.com'. =
PUBLISH is specifically addressed to the resource, not the presence =
agent.
 =20
 >         M2:
 >             PUBLISH sip:pa.example.com SIP/2.0
 >             Via: SIP/2.0/UDP proxy.example.com;branch=3Dz9hG4bKpx123
 >             Via: SIP/2.0/UDP pua.example.com;branch=3Dz9hG4bK652hsge
 >             From: <sip:presentity@example.com>;tag=3D1234wxyz
 >             To: <pres:presentity@example.com>
 >             Call-ID: 81818181@pua.example.com
 >             CSeq: 1 PUBLISH
 >             Max-Forwards: 70
 >             Expires: 3600
 >             Event: presence
 >             Content-Type: application/pidf+xml
 >             Content-Length: ...

The proxy would rewrite the Request-URI, but probably to something like =
'sip:presentity@pa.example.com'. Note that when the request is received =
by the PA/ESC, it specifically uses the Request-URI to determine what =
resource the PUBLISH is targeted for. This is identical to a SUBSCRIBE.  =


 >         M3:
 >             SIP/2.0 200 OK
 >             Via: SIP/2.0/UDP proxy.example.com;branch=3Dz9hG4bKpx123
 >             Via: SIP/2.0/UDP pua.example.com;branch=3Dz9hG4bK652hsge
 >             From: <sip:presentity@example.com>;tag=3D1234wxyz
 >             To: <pres:presentity@example.com>;tag=3D1a2b3c4d
 >             Call-ID: 81818181@pua.example.com
 >             CSeq: 1 PUBLISH
 >             SIP-ETag: dx200xyz
 >             Expires: 1800
 >             Content-Length: 0
 > =20
 >         M4:
 >             SIP/2.0 200 OK
 >             Via: SIP/2.0/UDP pua.example.com;branch=3Dz9hG4bK652hsge
 >             From: <sip:presentity@example.com>;tag=3D1234wxyz
 >             To: <pres:presentity@example.com>;tag=3D1a2b3c4d
 >             Call-ID: 81818181@pua.example.com
 >             CSeq: 1 PUBLISH
 >             SIP-ETag: dx200xyz
 >             Expires: 1800
 >             Content-Length: 0
 > =20
 > [3] It could be better if the "Request-URI" is used as the message's
 >     target address and the 'Presence Resource' is specified=20
 > using "To"
 >     header.
 >=20
 >     It would be easy for understanding if REGISTER's location=20
 >     registration operation like mechanism is considered.

This is actually how PUBLISH used to work. But having the Request-URI =
point to a PA requires special handling from the proxies and/or a =
specific naming convention for PAs to be able to route the request. So =
we decided to reject the REGISTER-like operation (determining the =
resource by the To-field) in favor of the SUBSCRIBE-like behavior the =
draft currently has.

 >=20
 >     The same point could also be said about RFC3265. This is because,
 >=20
 >         |3.1.5. Proxy SUBSCRIBE Behavior
 >         |  Proxies need no additional behavior beyond that=20
 > described in SIP [1]
 >         |  to support SUBSCRIBE.
 >=20
 > [4] For expressing 'Presence Resource', the=20
 > pres-URI(draft-ietf-impp-pres-04)
 >     would be better than that of the sip-URI.
 >=20
 >     Finally, when [3] & [4] are considered together, the=20
 > various usages=20
 >     of URI can be expressed as listed below.
 >=20
 > +------------+-----------------------------+-----------------
 > ------------+-----------------------------+
 > |            | PUBLISH                     | SUBSCRIBE      =20
 >             | NOTIFY                      |
 > +------------+-----------------------------+-----------------
 > ------------+-----------------------------+
 > |Request-URI | URI of PA                   | URI of PA      =20
 >             | Contact-URI of SUBSCRIBE    |
 > |            | sip:pa@example.com          |=20
 > sip:pa@example.com          | sip:watcher.example.com     |
 > +------------+-----------------------------+-----------------
 > ------------+-----------------------------+
 > |To-URI      | URI of presence resource    | URI of presence=20
 > resource    | From-URI of SUBSCRIBE       |
 > |            | pres:presentity@example.com |=20
 > pres:presentity@example.com | sip:watcher@example.com     |
 > +------------+-----------------------------+-----------------
 > ------------+-----------------------------+
 > |From-URI    | URI of presentity           | URI of watcher =20
 >             | To-URI of SUBSCRIBE         |
 > |            | sip:presentity@example.com  |=20
 > sip:watcher@example.com     | pres:presentity@example.com |
 > +------------+-----------------------------+-----------------
 > ------------+-----------------------------+

Adding to my previous comments, how would the UA find out the address of =
a PA that services the presentity? We can't mandate a specific naming =
convention for PAs in general, which is why the Request-URI is always =
the address of the subscribed/published resource.

Note that one valid location for a PA is in the client device. Because =
of the way PUBLISH and SUBSCRIBE currently work, this requires nothing =
special from a proxy - it simply forwards/redirects these requests to =
the UA like any other request for that AoR.

Cheers,
Aki

 >=20
 > Regards,
 > shinji
 >=20
 > Rohan Mahy wrote in <C3094B8F-4563-11D8-B291-0003938AF740@cisco.com>
 > >Hi,
 > >
 > >I'd like to begin Working Group Last Call for PUBLISH:
 > >
 > >=09
 > http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-02.txt
 > >
 > >This WGLC will end on January 28, 2004.
 > >
 > >thanks,
 > >-rohan
 > _________________________________________________
 > OKUMURA Shinji        E-mail:shin@softfront.co.jp
 >=20

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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: <Mpierce1@aol.com>, <rohan@cisco.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] re: WGLC: resource priority (3)
Date: Fri, 16 Jan 2004 08:19:22 -0500
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> [MAP] You're going to have to explain how that could work, considering

> that the draft states: 
>
> Every namespace must have a default level. 
> The default is assumed in the absence of a priority level (in a 
> message). 
>
> That means that, if there is only one priority value in the 
> namespace (such as "on") registered with IANA, it must be the 
> default, and any INVITE that does not contain an RP-header is 
> treated as if it contained the default value ("on"). 

You've just explained your own request.

> How would it be possible to signal the "off" value? 

"off" = absence of message/signal, ie, the SIP Invite msg is the same as
if the R-P header didn't exist.  If I'm forced to have an "off" R-P
header value, than I'll need to write more code to recognize this, and
I'll do more processing for the label and the subsequent authentication.
And what did I gain in doing this added work compared to not sending the
R-P header Invite?  Nothing.  I'm going to have a fun time explaining
that to my guy who has written code to support the previous versions of
the draft that he is going to write additional code that amounts to no
gain.

In fact, in thinking about this, it will be code that will never be used
in our client using a namespace to support the NS/EP codepoint.  The
trigger for our SIP R-P client is a particular phone number sequence.
Absence of that sequence means that all other calls from that client are
regular SIP messages.  

> I think the age old practice of being conservative in what you send 
> applies to what we say to each other, not what signaling is sent by 
> telecommunications devices :>} 

agreed on the former, but Postel applied it to the latter and drilled it
into many a head, who in turn continued the drilling :-). (note, the
full quote is be liberal in what you accept, and conservative in what
you send, see rfc-760)

> Explain how one would signal the NS/EP codepoint case with only one 
> value (which is the default). 

You already have.  And I've added a case in point where the "off" value
will never see light of day.

As a side note, this issue has been discussed in the past on a different
list, so there is a reason why the draft has taken the position not
requiring at least 2 values per namespace.  My desire is to defend the
current position, but the issue is not critical, so I will not add any
other comments on the matter on the list.

Regards,

-ken

ps, it would be nice to see plain text messages rather than html marked
ones.



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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Subject: RE: [Sip] re: WGLC: resource priority (3)
To: "Ken Carlberg" <carlberg@g11.org.uk>
Cc: Mpierce1@aol.com, rohan@cisco.com, sip@ietf.org
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Just a very small quibble.

You say
"The trigger for our SIP R-P client is a particular phone number sequence.
Absence of that sequence means that all other calls from that client are
regular SIP messages."

That is one trigger, but there are other triggers for the NS/EP code point.
But that doesn't change your point that if the call doesn't "trigger" the
NS/EP code point, the RPH won't be there.

Janet


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                      "Ken Carlberg"                                                                                            
                      <carlberg                To:      <Mpierce1@aol.com>, <rohan@cisco.com>                                   
                      @g11.org.uk>             cc:      <sip@ietf.org>                                                          
                                               Subject: RE: [Sip] re: WGLC: resource priority (3)                               
                      01/16/2004 08:19                                                                                          
                      AM                                                                                                        
                                                                                                                                
                                                                                                                                




> [MAP] You're going to have to explain how that could work, considering

> that the draft states:
>
> Every namespace must have a default level.
> The default is assumed in the absence of a priority level (in a
> message).
>
> That means that, if there is only one priority value in the
> namespace (such as "on") registered with IANA, it must be the
> default, and any INVITE that does not contain an RP-header is
> treated as if it contained the default value ("on").

You've just explained your own request.

> How would it be possible to signal the "off" value?

"off" = absence of message/signal, ie, the SIP Invite msg is the same as
if the R-P header didn't exist.  If I'm forced to have an "off" R-P
header value, than I'll need to write more code to recognize this, and
I'll do more processing for the label and the subsequent authentication.
And what did I gain in doing this added work compared to not sending the
R-P header Invite?  Nothing.  I'm going to have a fun time explaining
that to my guy who has written code to support the previous versions of
the draft that he is going to write additional code that amounts to no
gain.

In fact, in thinking about this, it will be code that will never be used
in our client using a namespace to support the NS/EP codepoint.  The
trigger for our SIP R-P client is a particular phone number sequence.
Absence of that sequence means that all other calls from that client are
regular SIP messages.

> I think the age old practice of being conservative in what you send
> applies to what we say to each other, not what signaling is sent by
> telecommunications devices :>}

agreed on the former, but Postel applied it to the latter and drilled it
into many a head, who in turn continued the drilling :-). (note, the
full quote is be liberal in what you accept, and conservative in what
you send, see rfc-760)

> Explain how one would signal the NS/EP codepoint case with only one
> value (which is the default).

You already have.  And I've added a case in point where the "off" value
will never see light of day.

As a side note, this issue has been discussed in the past on a different
list, so there is a reason why the draft has taken the position not
requiring at least 2 values per namespace.  My desire is to defend the
current position, but the issue is not critical, so I will not add any
other comments on the matter on the list.

Regards,

-ken

ps, it would be nice to see plain text messages rather than html marked
ones.



_______________________________________________
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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





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



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Subject: Re: [Sip] re: WGLC: resource priority (2)
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In a message dated 1/15/2004 3:09:02 PM Eastern Standard Time, 
rohan@cisco.com writes:


I believe that the IANA *registration* does need to have such a forward 
reference and that the IESG will tend to agree with me.


[MAP] Then it's a chicken and egg problem. I can't get the other normative 
reference through as an RFC or an internal DOD spec or an ITU-T Recommendation 
until this one is approved and we can't get this one approved until all the 
other ones it needs to reference are approved.

In a message dated 1/15/2004 3:09:02 PM Eastern Standard Time, 
rohan@cisco.com writes:


The UA should know the policy.  There is no technical reason it can't 
know or collect the policy and then use the appropriate policy. That's 
what the session policy mechanism is for. However, there are a number 
of technical problems related to security we get when proxies start to 
insert or modify RP.


[MAP] I didn't mean that the UA could never know the "policy" and do all 
kinds of things like ensuring that the human user is complying with "policy" 
before it sends an INVITE. Certainly one can be built to do this. But it is equally 
possible to build/operate a UA that does not do these types of things - that 
is, a "dumb device" that expects the human user to know what to do and to 
react properly.

Both are possible, so lets ensure that the definition of the RP-header in 
this RFC allows both.

Mike Pierce

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Subject: Re: [Sip] re: WGLC: resource priority (3)
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OK, now _I_ am confused.

You say "If a message (INVITE) is sent without an RP-header, it should be
treated by
the receiver as if it contained an RP-header with the default value for
that
namespace, that is, "ON"."

But  if the message does not contain the R-P Header, how do I even know
which namespace it (isn't) referring to.  Are you saying that if a  SIP
message contains no R-P Header, it should be assumed to be "the default"
for EVERY registered namespace? Or even for every namespace recognized by
that network? (This doesn't seem sensible, since the default value, and the
associated default behavior, are different for different namespaces.)

I understood the document to say that the default value is used when there
IS a R-P Header, and the header DOES contain the namespace, but DOES NOT
contain the value.

There is no one-to-one mapping between R-P namespaces and IP Administrative
domains.

 I understood that two (or more) R-P Header namespaces (with different
associated policies/behaviors) can be used in the same network, and even in
the same SIP message.


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                      Mpierce1                                                                                                   
                      @aol.com                 To:      carlberg@g11.org.uk, rohan@cisco.com                                     
                      Sent by:                 cc:      sip@ietf.org                                                             
                      sip-admin                Subject: Re: [Sip] re: WGLC: resource priority (3)                                
                                                                                                                                 
                                                                                                                                 
                      01/16/2004 10:58                                                                                           
                      AM                                                                                                         
                                                                                                                                 
                                                                                                                                 




In a message dated 1/16/2004 8:19:35 AM Eastern Standard Time,
carlberg@g11.org.uk writes:


> How would it be possible to signal the "off" value?

"off" = absence of message/signal, ie, the SIP Invite msg is the same as
if the R-P header didn't exist.  If I'm forced to have an "off" R-P
header value, than I'll need to write more code to recognize this, and
I'll do more processing for the label and the subsequent authentication.
And what did I gain in doing this added work compared to not sending the
R-P header Invite?  Nothing.  I'm going to have a fun time explaining
that to my guy who has written code to support the previous versions of
the draft that he is going to write additional code that amounts to no
gain.


[MAP]
We must be talking about two different things or we have completely
different
understandings of what the draft RP-header doc means, which scares me. I
hope
someone else can examine this issue and help us resolve it.

If a namespace is defined with one single value of "ON", then that value
"ON"
must also be designated as the default.

If a message (INVITE) is sent without an RP-header, it should be treated by

the receiver as if it contained an RP-header with the default value for
that
namespace, that is, "ON".

There is then no way to signal any other value (e.g.,"OFF") for this
namespace since no other value is defined in the IANA registration. The
absence of
"ON" is not "OFF" since these are defined in IANA as text strings, not a
boolean.
"OFF" simply does not exist for this namespace.

I don't understand the argument about saving (a trivial) amount of code (to

recognize the "OFF" value). It seems to me that any implementation should
be
able to handle the case of multiple priority values (e.g., 5 for DOD), so
the
code to recognize 2 must already be there.

What I always learned when defining protocols is that it is always best to
be
explicit in what you send, so that the receiver does not have to infer or
guess or assume what was really meant.

Mike Pierce


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Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sip] re: WGLC: resource priority (3)
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In a message dated 1/16/2004 8:19:35 AM Eastern Standard Time, 
carlberg@g11.org.uk writes:


> How would it be possible to signal the "off" value? 

"off" = absence of message/signal, ie, the SIP Invite msg is the same as
if the R-P header didn't exist.  If I'm forced to have an "off" R-P
header value, than I'll need to write more code to recognize this, and
I'll do more processing for the label and the subsequent authentication.
And what did I gain in doing this added work compared to not sending the
R-P header Invite?  Nothing.  I'm going to have a fun time explaining
that to my guy who has written code to support the previous versions of
the draft that he is going to write additional code that amounts to no
gain.


[MAP]
We must be talking about two different things or we have completely different 
understandings of what the draft RP-header doc means, which scares me. I hope 
someone else can examine this issue and help us resolve it.

If a namespace is defined with one single value of "ON", then that value "ON" 
must also be designated as the default.

If a message (INVITE) is sent without an RP-header, it should be treated by 
the receiver as if it contained an RP-header with the default value for that 
namespace, that is, "ON".

There is then no way to signal any other value (e.g.,"OFF") for this 
namespace since no other value is defined in the IANA registration. The absence of 
"ON" is not "OFF" since these are defined in IANA as text strings, not a boolean. 
"OFF" simply does not exist for this namespace.

I don't understand the argument about saving (a trivial) amount of code (to 
recognize the "OFF" value). It seems to me that any implementation should be 
able to handle the case of multiple priority values (e.g., 5 for DOD), so the 
code to recognize 2 must already be there.

What I always learned when defining protocols is that it is always best to be 
explicit in what you send, so that the receiver does not have to infer or 
guess or assume what was really meant.

Mike Pierce


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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: Fri, 16 Jan 2004 09:34:36 -0800
Subject: Re: [Sip] re: WGLC: resource priority (3)
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To: "Ken Carlberg" <carlberg@g11.org.uk>
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Ken,

Just because you normally don't send the value "off" doesn't mean it 
doesn't exist.

thx,
-r



On Friday, January 16, 2004, at 05:19  AM, Ken Carlberg wrote:

>> [MAP] You're going to have to explain how that could work, considering
>
>> that the draft states:
>>
>> Every namespace must have a default level.
>> The default is assumed in the absence of a priority level (in a
>> message).
>>
>> That means that, if there is only one priority value in the
>> namespace (such as "on") registered with IANA, it must be the
>> default, and any INVITE that does not contain an RP-header is
>> treated as if it contained the default value ("on").
>
> You've just explained your own request.
>
>> How would it be possible to signal the "off" value?
>
> "off" = absence of message/signal, ie, the SIP Invite msg is the same 
> as
> if the R-P header didn't exist.  If I'm forced to have an "off" R-P
> header value, than I'll need to write more code to recognize this, and
> I'll do more processing for the label and the subsequent 
> authentication.
> And what did I gain in doing this added work compared to not sending 
> the
> R-P header Invite?  Nothing.  I'm going to have a fun time explaining
> that to my guy who has written code to support the previous versions of
> the draft that he is going to write additional code that amounts to no
> gain.
>
> In fact, in thinking about this, it will be code that will never be 
> used
> in our client using a namespace to support the NS/EP codepoint.  The
> trigger for our SIP R-P client is a particular phone number sequence.
> Absence of that sequence means that all other calls from that client 
> are
> regular SIP messages.
>
>> I think the age old practice of being conservative in what you send
>> applies to what we say to each other, not what signaling is sent by
>> telecommunications devices :>}
>
> agreed on the former, but Postel applied it to the latter and drilled 
> it
> into many a head, who in turn continued the drilling :-). (note, the
> full quote is be liberal in what you accept, and conservative in what
> you send, see rfc-760)
>
>> Explain how one would signal the NS/EP codepoint case with only one
>> value (which is the default).
>
> You already have.  And I've added a case in point where the "off" value
> will never see light of day.
>
> As a side note, this issue has been discussed in the past on a 
> different
> list, so there is a reason why the draft has taken the position not
> requiring at least 2 values per namespace.  My desire is to defend the
> current position, but the issue is not critical, so I will not add any
> other comments on the matter on the list.
>
> Regards,
>
> -ken
>
> ps, it would be nice to see plain text messages rather than html marked
> ones.
>
>


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Date: Fri, 16 Jan 2004 09:47:13 -0800
Subject: Re: [Sip] re: WGLC: resource priority (2)
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From: Rohan Mahy <rohan@cisco.com>
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On Friday, January 16, 2004, at 07:58  AM, Mpierce1@aol.com wrote:
> In a message dated 1/15/2004 3:09:02 PM Eastern Standard Time,
> rohan@cisco.com writes:
>
> I believe that the IANA *registration* does need to have such a forward
> reference and that the IESG will tend to agree with me.
>
> [MAP] Then it's a chicken and egg problem. I can't get the other 
> normative
> reference through as an RFC or an internal DOD spec or an ITU-T 
> Recommendation
> until this one is approved and we can't get this one approved until 
> all the
> other ones it needs to reference are approved.

There is no chicken and egg problem here.  The Resource-Priority header 
and setting up the IANA namespace can happen as soon as this document 
is ready to go.  Any initial namespaces that need to get registered are 
really optional--they can get registered if there is a document to 
reference.  There is no dependency here.

My understanding is that documents already exist which describe the 
behavior expected in the PSTN world for the proposed initial namespace 
registrations.  If they aren't, then at a later time someone can write 
a document explaining the normative behavior and include the IANA 
registration in the document if it is an IETF document, or write a very 
short (~2 page) companion IETF document which just performs the 
registration.


> In a message dated 1/15/2004 3:09:02 PM Eastern Standard Time,
> rohan@cisco.com writes:
>
> The UA should know the policy.  There is no technical reason it can't
> know or collect the policy and then use the appropriate policy. That's
> what the session policy mechanism is for. However, there are a number
> of technical problems related to security we get when proxies start to
> insert or modify RP.
>
> [MAP] I didn't mean that the UA could never know the "policy" and do 
> all
> kinds of things like ensuring that the human user is complying with 
> "policy"
> before it sends an INVITE. Certainly one can be built to do this. But 
> it is equally
> possible to build/operate a UA that does not do these types of things 
> - that
> is, a "dumb device" that expects the human user to know what to do and 
> to
> react properly.

Yes, I agree it is possible to build a dumb UA.  That hurts, so don't 
do that.

> Both are possible, so lets ensure that the definition of the RP-header 
> in
> this RFC allows both.

If you insist on allowing both, this work will have to wait for the 
appropriate mechanisms to secure proxy insertion (Middle to End 
security) which will take many months.  I don't think this is in your 
best interest, or the best interest of the WG or the larger community.

thanks,
-rohan


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Subject: Re: [Sip] re: WGLC: resource priority (3)
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In a message dated 1/16/2004 11:28:02 AM Eastern Standard Time, 
jgunn6@csc.com writes:

[JG]
OK, now _I_ am confused.

You say "If a message (INVITE) is sent without an RP-header, it should be
treated by
the receiver as if it contained an RP-header with the default value for
that
namespace, that is, "ON"."

But  if the message does not contain the R-P Header, how do I even know
which namespace it (isn't) referring to.  Are you saying that if a  SIP
message contains no R-P Header, it should be assumed to be "the default"
for EVERY registered namespace? Or even for every namespace recognized by
that network? (This doesn't seem sensible, since the default value, and the
associated default behavior, are different for different namespaces.)



[MAP] Yes, that is what I'm saying. Namespaces aren't some arbitrary things 
that just may or may not be present at random. Any device that might possibly 
react to the priority value of a namespace (or the default implied by its 
absence), must have an awareness of all the namespaces it needs to be concerned 
with. It must ignore all others, since they are meaningless. For the namespaces 
that it must be concerned with for a call setup, it must have a value to 
associate with each, even if derived by default. For Ken's apparent case where the 
defined value of "ON" means to use the NS/EP codepoint of T1.631, then the 
default (when nothing is specified as in 99.9999% of the calls) is to default to 
"OFF" for this namespace. If a device doesn't understand the need to default 
the namespace to "OFF", then it also could not know what "ON" means. This 
requires defining a namespace with two values, "ON" and "OFF", with "OFF" as the 
default. There is no way to do this with only one value ("ON") defined in the 
IANA registration. I agree with Ken's "be conservative in what you send" idea, 
that an implementation would never send an RP-header containing "OFF" for this 
namespace, but it still must be registered and receivers must be able to 
understand it ("be liberal in what you accept"). As Rohan says "Just because you 
normally don't send the value "off" doesn't mean it doesn't exist."

[JG]

I understood the document to say that the default value is used when there
IS a R-P Header, and the header DOES contain the namespace, but DOES NOT
contain the value.



[MAP] The draft only says "in the absence of the priority value", which you 
could read to mean what you said above. I read it to mean any condition in 
which the priority value is not available, including the absence of the header, 
which I expect to be the way to code the message for the majority of calls that 
want the default value (as explained above). As I read the ABNF, the priority 
value can not be left out if a namespace is specified.

[JG]

There is no one-to-one mapping between R-P namespaces and IP Administrative
domains.

I understood that two (or more) R-P Header namespaces (with different
associated policies/behaviors) can be used in the same network, and even in
the same SIP message.



[MAP] Yes, but the draft intentionally does not talk about the interaction 
between them. Each is treated separately.

Mike Pierce



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 1/16/2004 11:28:02 AM Eastern Standard Time, jgunn6@csc.com writes:
<BR>
<BR>[JG]
<BR>OK, now _I_ am confused.
<BR>
<BR>You say "If a message (INVITE) is sent without an RP-header, it should b=
e
<BR>treated by
<BR>the receiver as if it contained an RP-header with the default value for
<BR>that
<BR>namespace, that is, "ON"."
<BR>
<BR>But &nbsp;if the message does not contain the R-P Header, how do I even=20=
know
<BR>which namespace it (isn't) referring to. &nbsp;Are you saying that if a=20=
&nbsp;SIP
<BR>message contains no R-P Header, it should be assumed to be "the default"
<BR>for EVERY registered namespace? Or even for every namespace recognized b=
y
<BR>that network? (This doesn't seem sensible, since the default value, and=20=
the
<BR>associated default behavior, are different for different namespaces.)
<BR>
<BR>
<BR>
<BR>[MAP] Yes, that is what I'm saying. Namespaces aren't some arbitrary thi=
ngs that just may or may not be present at random. Any device that might pos=
sibly react to the priority value of a namespace (or the default implied by=20=
its absence), must have an awareness of all the namespaces it needs to be co=
ncerned with. It must ignore all others, since they are meaningless. For the=
 namespaces that it must be concerned with for a call setup, it must have a=20=
value to associate with each, even if derived by default. For Ken's apparent=
 case where the defined value of "ON" means to use the NS/EP codepoint of T1=
.631, then the default (when nothing is specified as in 99.9999% of the call=
s) is to default to "OFF" for this namespace. If a device doesn't understand=
 the need to default the namespace to "OFF", then it also could not know wha=
t "ON" means. This requires defining a namespace with two values, "ON" and "=
OFF", with "OFF" as the default. There is no way to do this with only one va=
lue ("ON") defined in the IANA registration. I agree with Ken's "be conserva=
tive in what you send" idea, that an implementation would never send an RP-h=
eader containing "OFF" for this namespace, but it still must be registered a=
nd receivers must be able to understand it ("be liberal in what you accept")=
. As Rohan says "Just because you normally don't send the value "off" doesn'=
t mean it doesn't exist."
<BR>
<BR>[JG]
<BR>
<BR>I understood the document to say that the default value is used when the=
re
<BR>IS a R-P Header, and the header DOES contain the namespace, but DOES NOT
<BR>contain the value.
<BR>
<BR>
<BR>
<BR>[MAP] The draft only says "in the absence of the priority value", which=20=
you could read to mean what you said above. I read it to mean any condition=20=
in which the priority value is not available, including the absence of the h=
eader, which I expect to be the way to code the message for the majority of=20=
calls that want the default value (as explained above). As I read the ABNF,=20=
the priority value can not be left out if a namespace is specified.
<BR>
<BR>[JG]
<BR>
<BR>There is no one-to-one mapping between R-P namespaces and IP Administrat=
ive
<BR>domains.
<BR>
<BR>I understood that two (or more) R-P Header namespaces (with different
<BR>associated policies/behaviors) can be used in the same network, and even=
 in
<BR>the same SIP message.
<BR>
<BR>
<BR>
<BR>[MAP] Yes, but the draft intentionally does not talk about the interacti=
on between them. Each is treated separately.
<BR>
<BR>Mike Pierce
<BR>
<BR></FONT></HTML>

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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
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Date: Fri, 16 Jan 2004 19:13:58 -0500
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Subject: [Sip] Is Retry-After:0 valid?
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(Trying the sip list because I did not
receive any reply concerning my 
sip-implementors posting...)


Is 0 a valid value for the retry-after header?

I assume that it is; and the bnf allows it.
However I guess that it depends upon the meaning 
of "positive" in section 20.33 and if the 
"between" statement of section 14.2 is inclusive.

RFC3261 section 20.33 mentions the following:
 "The value of this field is a positive integer
  number of seconds (in decimal) after the time 
  of the response."

RFC3261 section 14.2 mentions the following:
 "MUST include a Retry-After header field with a
  randomly chosen value of between 0 and 10 seconds."


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Subject: Re: [Sip] SIP DTMF
From: Cullen Jennings <fluffy@cisco.com>
To: Paul Long <plong@packetizer.com>, <sip@ietf.org>
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The draft is
http://www.ietf.org/internet-drafts/draft-burger-sipping-kpml-02.txt

2833 is the preferred method, but if an off media axis device wished to get
notification about key presses, it could. How these all relate is covered i=
n

http://www.softarmor.com/wgdb/docs/
         draft-ietf-sipping-app-interaction-framework-00.txt

Cullen


On 1/15/04 11:21 AM, "Paul Long" <plong@packetizer.com> wrote:

> Sreeram,
>=20
> I can't find a "kpxml" draft on the IETF site or via Google--anywhere.
> Can you provide a link to it?
>=20
> Also, from my experience, the preferred method of DTMF carriage is RFC
> 2833, followed by in-band audio, and then various proprietary
> INFO-method schemes. What is the standing of RFC 2833 vis-=E0-vis "kpxml?"
>=20
> Paul
>=20
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
> sreeram.kanumuri@wipro.com
> Sent: Tuesday, January 13, 2004 9:06 PM
> To: xyxie@udtech.com.cn; sip@ietf.org
> Cc: sipping@ietf.org
> Subject: RE: [Sip] SIP DTMF
>=20
> Wendy,
>=20
> There was a discussion in the list some time back and it was decided
> that SIP-INFO method should not be used to
> Send the DTMF info.
>=20
> It was proposed that it is better to you kpxml for sending DTMF tones.
>=20
> Please look in to the SIP-Kpxml draft for sending DTMF tomes in-midcall
> processing and all....
>=20
>=20
> Regards,
> Sreeram
>=20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Sreeram Kanumuri                Plot No 72, Keonics Electronics City
>              Hosur Main Road
> Wipro Technologies              Bangalore -29 INDIA
> sreeram.kanumuri@wipro.com      Phone: 8520408 Ex:3208
> skanumury@yahoo.com             ESN  :
> www.wipro.com                   Mobile:
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
>=20
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of wendy
> Sent: Wednesday, January 14, 2004 8:08 AM
> To: sip@ietf.org
> Cc: sipping@ietf.org
> Subject: [Sip] SIP DTMF
>=20
>=20
> Hello all,=20
>=20
> Who can tell me whether there is a RFC or I-D on DTMF in SIP which is
> still not expired? I remember someone proposed using SIP INFO method for
> DTMF one or two years ago. Is the DTMF relay implemented only by RTP
> now?
>=20
>=20
> Regards,
> Wendy
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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: Home  Cortes John Alexander <John.Home@TILAB.COM>
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I'm studying the specification 3GPP TS 23.228, and there are concepts =
related to SIP. I would like to know some concepts and clarify some =
terms. The identification of User are defined by some user identities. =
It's defined that the temporary public user identity shall take the form =
of a SIP URL as described in the section 4.3.3.1 (3GPP TS 23.228 V5.9.0 =
(2003-06), "Network Architecture" (Release 5). I looked for this term =
"SIP URL" in the RFC3261 and SIP URL is not mentioned. As far as know, =
user agents are identify for a SIP URI.=20

I would like to know what is the difference between SIP URL and SIP URI?

I would appreciate for your help
Thanks in advance.
John Home=20





=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Subject: RE: [Sip] Is Retry-After:0 valid?
Date: Mon, 19 Jan 2004 11:45:48 +0200
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I don't see why it would be disallowed. It is certainly valid and says =
to retry immediately (after 0 seconds).

I guess the second quote can have the word "inclusive" added.

Regards,
Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> Brett Tate
> Sent: 17.January.2004 02:14
> To: sip-ietf (E-mail)
> Subject: [Sip] Is Retry-After:0 valid?
>=20
>=20
> (Trying the sip list because I did not
> receive any reply concerning my=20
> sip-implementors posting...)
>=20
>=20
> Is 0 a valid value for the retry-after header?
>=20
> I assume that it is; and the bnf allows it.
> However I guess that it depends upon the meaning=20
> of "positive" in section 20.33 and if the=20
> "between" statement of section 14.2 is inclusive.
>=20
> RFC3261 section 20.33 mentions the following:
>  "The value of this field is a positive integer
>   number of seconds (in decimal) after the time=20
>   of the response."
>=20
> RFC3261 section 14.2 mentions the following:
>  "MUST include a Retry-After header field with a
>   randomly chosen value of between 0 and 10 seconds."
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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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Thread-Topic: Comment in draft-ietf-sipping-torture-tests-02.txt
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Subject: [Sip] Comment in draft-ietf-sipping-torture-tests-02.txt
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Hi,
=20
I have a question on the parsing of the following INVITE message given
in section  3.1.1.1 of draft-ietf-sipping-torture-tests-02.txt.
=20
In this message the contact header (m line) is extended over multiple
lines, but the extra line does not precede with at least one SP or HT.
Is this a valid Contact header ?
=20
Section 7.3.1 in RFC 3261 mentions that:=20
=20
"Header fields can be extended over multiple lines by preceding each
extra line with at least one SP or horizontal tab (HT).  The line break
and the whitespace at the beginning of the next line are treated as a
single SP character."
=20
------------------------------
INVITE sip:vivekg@chair-dnrc.example.com;unknownparam SIP/2.0
TO :
 sip:vivekg@chair-dnrc.example.com ;   tag    =3D 1918181833n
from   : "J Rosenberg \\\""       <sip:jdrosen@example.com>
  ;
  tag =3D 98asjd8
MaX-fOrWaRdS: 0068
Call-ID: 0ha0isndaksdj@192.0.2.1
Content-Length   : 151
cseq: 0009
  INVITE
Via  : SIP  /   2.0
 /UDP
    192.0.2.2;branch=3D390skdjuw
s :
NewFangledHeader:   newfangled value
 continued newfangled value
UnknownHeaderWithUnusualValue: ;;,,;;,;
Content-Type: application/sdp
Route:
 <sip:services.example.com;lr;unknownwith=3Dvalue;unknown-no-value>
v:  SIP  / 2.0  / TCP     spindle.example.com   ;
  branch  =3D   z9hG4bK9ikj8  ,
 SIP  /    2.0   / UDP  192.168.255.111   ; branch=3D
 z9hG4bK30239
m:"Quoted string \"\"" <sip:jdrosen@example.com> ; newparam =3D
newvalue ;
  secondparam ; q =3D 0.33
=20
v=3D0
o=3Dmhandley 29739 7272939 IN IP4 192.0.2.3
s=3D-
c=3DIN IP4 192.0.2.4
t=3D0 0
m=3Daudio 492170 RTP/AVP 0 12
m=3Dvideo 3227 RTP/AVP 31
a=3Drtpmap:31 LPC
=20

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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
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<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D305310810-19012004>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D305310810-19012004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D305310810-19012004>I have =
a question on=20
the parsing of the following INVITE message given in section&nbsp;=20
3.1.1.1&nbsp;of =
draft-ietf-sipping-torture-tests-02.txt.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D305310810-19012004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D305310810-19012004>In =
this=20
message&nbsp;the&nbsp;contact header (m line) is extended over multiple =
lines,=20
but the extra line does not precede with at least one SP or HT. Is this =
a valid=20
Contact header ?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D305310810-19012004><FONT face=3DArial =
size=3D2>Section 7.3.1 in RFC=20
3261 mentions that: </FONT></SPAN></DIV>
<DIV><SPAN class=3D305310810-19012004><EM></EM></SPAN>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT size=3D2><EM><SPAN=20
class=3D305310810-19012004>"</SPAN>Header fields can be extended over =
multiple=20
lines by preceding each<SPAN =
class=3D305310810-19012004>&nbsp;</SPAN>extra line=20
with at least one SP or horizontal tab (HT).&nbsp; The line<SPAN=20
class=3D305310810-19012004>&nbsp;</SPAN>break and the whitespace at the =
beginning=20
of the next line are<SPAN class=3D305310810-19012004> </SPAN>treated as =
a single=20
SP character.<SPAN =
class=3D305310810-19012004>"</SPAN></EM></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><EM></EM></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D305310810-19012004><FONT face=3DArial=20
size=3D2>------------------------------</FONT></SPAN></DIV>
<DIV><FONT face=3DArial size=3D2>INVITE=20
sip:vivekg@chair-dnrc.example.com;unknownparam SIP/2.0<BR>TO=20
:<BR>&nbsp;sip:vivekg@chair-dnrc.example.com ;&nbsp;&nbsp; =
tag&nbsp;&nbsp;&nbsp;=20
=3D 1918181833n<BR>from&nbsp;&nbsp; : "J Rosenberg=20
<A>\\\</A>""&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;sip:jdrosen@example.com&gt;<BR>&nbsp; ;<BR>&nbsp; tag =3D=20
98asjd8<BR>MaX-fOrWaRdS: 0068<BR>Call-ID: <A=20
href=3D"mailto:0ha0isndaksdj@192.0.2.1">0ha0isndaksdj@192.0.2.1</A><BR>Co=
ntent-Length&nbsp;&nbsp;=20
: 151<BR>cseq: 0009<BR>&nbsp; INVITE<BR>Via&nbsp; : SIP&nbsp; =
/&nbsp;&nbsp;=20
2.0<BR>&nbsp;/UDP<BR>&nbsp;&nbsp;&nbsp; =
192.0.2.2;branch=3D390skdjuw<BR>s=20
:<BR>NewFangledHeader:&nbsp;&nbsp; newfangled value<BR>&nbsp;continued=20
newfangled value<BR>UnknownHeaderWithUnusualValue: =
;;,,;;,;<BR>Content-Type:=20
application/sdp<BR>Route:<BR>&nbsp;&lt;sip:services.example.com;lr;unknow=
nwith=3Dvalue;unknown-no-value&gt;<BR>v:&nbsp;=20
SIP&nbsp; / 2.0&nbsp; / TCP&nbsp;&nbsp;&nbsp;&nbsp;=20
spindle.example.com&nbsp;&nbsp; ;<BR>&nbsp; branch&nbsp; =3D&nbsp;&nbsp; =

z9hG4bK9ikj8&nbsp; ,<BR>&nbsp;SIP&nbsp; /&nbsp;&nbsp;&nbsp; =
2.0&nbsp;&nbsp; /=20
UDP&nbsp; 192.168.255.111&nbsp;&nbsp; ;=20
branch=3D<BR>&nbsp;z9hG4bK30239<BR>m:"Quoted string \"\""=20
&lt;sip:jdrosen@example.com&gt; ; newparam =3D<BR>newvalue ;<BR>&nbsp; =
secondparam=20
; q =3D 0.33</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>v=3D0<BR>o=3Dmhandley 29739 7272939 IN =
IP4=20
192.0.2.3<BR>s=3D-<BR>c=3DIN IP4 192.0.2.4<BR>t=3D0 0<BR>m=3Daudio =
492170 RTP/AVP 0=20
12<BR>m=3Dvideo 3227 RTP/AVP 31<BR>a=3Drtpmap:31 LPC</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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Subject: RE: [Sip] Comment in draft-ietf-sipping-torture-tests-02.txt
Date: Mon, 19 Jan 2004 17:35:47 +0530
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Thread-Topic: [Sip] Comment in draft-ietf-sipping-torture-tests-02.txt
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=20

=20

Thanks & Regards,
Nataraju A.B.=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Ajit
Katankot (WT01 - TELECOM & INTER-NETWORKING SOLUTIONS)
Sent: Monday, January 19, 2004 4:48 PM
To: sip@ietf.org
Cc: rsparks@dynamicsoft.com
Subject: [Sip] Comment in draft-ietf-sipping-torture-tests-02.txt

=20

Hi,

=20

I have a question on the parsing of the following INVITE message given
in section  3.1.1.1 of draft-ietf-sipping-torture-tests-02.txt.

=20

In this message the contact header (m line) is extended over multiple
lines, but the extra line does not precede with at least one SP or HT.
Is this a valid Contact header ?

=20

 [ABN] this is an INVALID input, the parser should not consider this
header as an invalid header...

=20

Section 7.3.1 in RFC 3261 mentions that:=20

=20

"Header fields can be extended over multiple lines by preceding each
extra line with at least one SP or horizontal tab (HT).  The line break
and the whitespace at the beginning of the next line are treated as a
single SP character."

=20

------------------------------

INVITE sip:vivekg@chair-dnrc.example.com;unknownparam SIP/2.0
TO :
 sip:vivekg@chair-dnrc.example.com ;   tag    =3D 1918181833n
from   : "J Rosenberg \\\""       <sip:jdrosen@example.com>
  ;
  tag =3D 98asjd8
MaX-fOrWaRdS: 0068
Call-ID: 0ha0isndaksdj@192.0.2.1
Content-Length   : 151
cseq: 0009
  INVITE
Via  : SIP  /   2.0
 /UDP
    192.0.2.2;branch=3D390skdjuw
s :
NewFangledHeader:   newfangled value
 continued newfangled value
UnknownHeaderWithUnusualValue: ;;,,;;,;
Content-Type: application/sdp
Route:
 <sip:services.example.com;lr;unknownwith=3Dvalue;unknown-no-value>
v:  SIP  / 2.0  / TCP     spindle.example.com   ;
  branch  =3D   z9hG4bK9ikj8  ,
 SIP  /    2.0   / UDP  192.168.255.111   ; branch=3D
 z9hG4bK30239
m:"Quoted string \"\"" <sip:jdrosen@example.com> ; newparam =3D
newvalue ;
  secondparam ; q =3D 0.33

=20

v=3D0
o=3Dmhandley 29739 7272939 IN IP4 192.0.2.3
s=3D-
c=3DIN IP4 192.0.2.4
t=3D0 0
m=3Daudio 492170 RTP/AVP 0 12
m=3Dvideo 3227 RTP/AVP 31
a=3Drtpmap:31 LPC

=20


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<p><font size=3D2 color=3Dblue face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New";color:blue'>Thanks &amp; Regards,<br>
</span></font><font size=3D2 color=3Dblue face=3D"Courier New"><span
 style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Nataraju A.B.</span></font><font
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<p class=3DMsoNormal><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'>-----Original Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> sip-admin@ietf.org
[mailto:sip-admin@ietf.org] <b><span style=3D'font-weight:bold'>On =
Behalf Of </span></b>Ajit
Katankot (WT01 - TELECOM &amp; INTER-NETWORKING SOLUTIONS)<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> </span></font><font =
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Monday, January
 19, 2004</span></font><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'> </span></font><font size=3D2 face=3DTahoma><span
 style=3D'font-size:10.0pt;font-family:Tahoma'>4:48 =
PM</span></font><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> =
rsparks@dynamicsoft.com<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Comment in
draft-ietf-sipping-torture-tests-02.txt</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I have a question on the parsing of the following =
INVITE
message given in section&nbsp; 3.1.1.1&nbsp;of
draft-ietf-sipping-torture-tests-02.txt.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In this message&nbsp;the&nbsp;contact header (m line) =
is
extended over multiple lines, but the extra line does not precede with =
at least
one SP or HT. Is this a valid Contact header ?</span></font></p>

</div>

<div>

<p class=3DMsoNormal><b><i><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue;font-weight:bold;font-style:italic'>=
&nbsp;</span></font></i></b></p>

<p class=3DMsoNormal><b><i><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue;font-weight:bold;font-style:italic'>=
&nbsp;[ABN]
</span></font></i></b><font color=3Dblue><span style=3D'color:blue'>this =
is an INVALID
input, the parser should not consider this header as an invalid =
header&#8230;</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Section 7.3.1 in RFC 3261 mentions that: =
</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><em><i><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&quot;Header fields can be extended over multiple =
lines by
preceding each&nbsp;extra line with at least one SP or horizontal tab
(HT).&nbsp; The line&nbsp;break and the whitespace at the beginning of =
the next
line are treated as a single SP =
character.&quot;</span></font></i></em></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>INVITE sip:vivekg@chair-dnrc.example.com;unknownparam
SIP/2.0<br>
TO :<br>
&nbsp;sip:vivekg@chair-dnrc.example.com ;&nbsp;&nbsp; =
tag&nbsp;&nbsp;&nbsp; =3D
1918181833n<br>
from&nbsp;&nbsp; : &quot;J Rosenberg
\\\&quot;&quot;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;sip:jdrosen@example.com&gt;<br>
&nbsp; ;<br>
&nbsp; tag =3D 98asjd8<br>
MaX-fOrWaRdS: 0068<br>
Call-ID: <a =
href=3D"mailto:0ha0isndaksdj@192.0.2.1">0ha0isndaksdj@192.0.2.1</a><br>
Content-Length&nbsp;&nbsp; : 151<br>
cseq: 0009<br>
&nbsp; INVITE<br>
Via&nbsp; : SIP&nbsp; /&nbsp;&nbsp; 2.0<br>
&nbsp;/UDP<br>
&nbsp;&nbsp;&nbsp; 192.0.2.2;branch=3D390skdjuw<br>
s :<br>
NewFangledHeader:&nbsp;&nbsp; newfangled value<br>
&nbsp;continued newfangled value<br>
UnknownHeaderWithUnusualValue: ;;,,;;,;<br>
Content-Type: application/sdp<br>
Route:<br>
&nbsp;&lt;sip:services.example.com;lr;unknownwith=3Dvalue;unknown-no-valu=
e&gt;<br>
v:&nbsp; SIP&nbsp; / 2.0&nbsp; / TCP&nbsp;&nbsp;&nbsp;&nbsp;
spindle.example.com&nbsp;&nbsp; ;<br>
&nbsp; branch&nbsp; =3D&nbsp;&nbsp; z9hG4bK9ikj8&nbsp; ,<br>
&nbsp;SIP&nbsp; /&nbsp;&nbsp;&nbsp; 2.0&nbsp;&nbsp; / UDP&nbsp;
192.168.255.111&nbsp;&nbsp; ; branch=3D<br>
&nbsp;z9hG4bK30239<br>
m:&quot;Quoted string \&quot;\&quot;&quot; =
&lt;sip:jdrosen@example.com&gt; ;
newparam =3D<br>
newvalue ;<br>
&nbsp; secondparam ; q =3D 0.33</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>v=3D0<br>
o=3Dmhandley 29739 7272939 IN IP4 192.0.2.3<br>
s=3D-<br>
c=3DIN IP4 192.0.2.4<br>
t=3D0 0<br>
m=3Daudio 492170 RTP/AVP 0 12<br>
m=3Dvideo 3227 RTP/AVP 31<br>
a=3Drtpmap:31 LPC</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

</div>

</body>

</html>

------_=_NextPart_001_01C3DE84.929427AA--

_______________________________________________
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From exim@www1.ietf.org  Mon Jan 19 09:53:30 2004
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	Mon, 19 Jan 2004 08:49:40 -0600
Subject: RE: [Sip] Comment in draft-ietf-sipping-torture-tests-02.txt
From: Robert Sparks <rsparks@dynamicsoft.com>
To: nataraju.alilaghatta@wipro.com
Cc: ajit.katankot@wipro.com, sip@ietf.org
In-Reply-To: <10C4348A1BA43A4FA8D5B767E052CEADE05A13@blr-ec-msg04.wipro.com>
References: <10C4348A1BA43A4FA8D5B767E052CEADE05A13@blr-ec-msg04.wipro.com>
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Date: Mon, 19 Jan 2004 08:49:33 -0600
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This is a SIPPING draft.

This bug has already been discussed on that list and will
be fixed in the next version of the draft (which will appear before
the draft deadline).

RjS

On Mon, 2004-01-19 at 06:05, nataraju.alilaghatta@wipro.com wrote:
> =20
>=20
> =20
>=20
> Thanks & Regards,
> Nataraju A.B.=20
>=20
>=20
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Ajit
> Katankot (WT01 - TELECOM & INTER-NETWORKING SOLUTIONS)
> Sent: Monday, January 19, 2004 4:48 PM
> To: sip@ietf.org
> Cc: rsparks@dynamicsoft.com
> Subject: [Sip] Comment in draft-ietf-sipping-torture-tests-02.txt
>=20
> =20
>=20
> Hi,
>=20
>=20
> =20
>=20
>=20
> I have a question on the parsing of the following INVITE message given
> in section  3.1.1.1 of draft-ietf-sipping-torture-tests-02.txt.
>=20
>=20
> =20
>=20
>=20
> In this message the contact header (m line) is extended over multiple
> lines, but the extra line does not precede with at least one SP or HT.
> Is this a valid Contact header ?
>=20
>=20
> =20
>=20
>  [ABN]this is an INVALID input, the parser should not consider this
> header as an invalid header=E2=80=A6
>=20
> =20
>=20
>=20
> Section 7.3.1 in RFC 3261 mentions that:=20
>=20
>=20
> =20
>=20
>=20
> "Header fields can be extended over multiple lines by preceding
> each extra line with at least one SP or horizontal tab (HT).  The
> line break and the whitespace at the beginning of the next line are
> treated as a single SP character."
>=20
>=20
> =20
>=20
>=20
> ------------------------------
>=20
>=20
> INVITE sip:vivekg@chair-dnrc.example.com;unknownparam SIP/2.0
> TO :
>  sip:vivekg@chair-dnrc.example.com ;   tag    =3D 1918181833n
> from   : "J Rosenberg \\\""       <sip:jdrosen@example.com>
>   ;
>   tag =3D 98asjd8
> MaX-fOrWaRdS: 0068
> Call-ID: 0ha0isndaksdj@192.0.2.1
> Content-Length   : 151
> cseq: 0009
>   INVITE
> Via  : SIP  /   2.0
>  /UDP
>     192.0.2.2;branch=3D390skdjuw
> s :
> NewFangledHeader:   newfangled value
>  continued newfangled value
> UnknownHeaderWithUnusualValue: ;;,,;;,;
> Content-Type: application/sdp
> Route:
>  <sip:services.example.com;lr;unknownwith=3Dvalue;unknown-no-value>
> v:  SIP  / 2.0  / TCP     spindle.example.com   ;
>   branch  =3D   z9hG4bK9ikj8  ,
>  SIP  /    2.0   / UDP  192.168.255.111   ; branch=3D
>  z9hG4bK30239
> m:"Quoted string \"\"" <sip:jdrosen@example.com> ; newparam =3D
> newvalue ;
>   secondparam ; q =3D 0.33
>=20
>=20
> =20
>=20
>=20
> v=3D0
> o=3Dmhandley 29739 7272939 IN IP4 192.0.2.3
> s=3D-
> c=3DIN IP4 192.0.2.4
> t=3D0 0
> m=3Daudio 492170 RTP/AVP 0 12
> m=3Dvideo 3227 RTP/AVP 31
> a=3Drtpmap:31 LPC
>=20
>=20
> =20
>=20
>=20
> Confidentiality Notice
>=20
>=20
> The information contained in this electronic message and any attachment=
s to this message are intended
> for the exclusive use of the addressee(s) and may contain confidential =
or privileged information. If
> you are not the intended recipient, please notify the sender at Wipro o=
r Mailadmin@wipro.com immediately
> and destroy all copies of this message and any attachments.


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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: Mon, 19 Jan 2004 11:04:12 -0500
From: Bert Culpepper <bertculpepper@netscape.net>
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To: Cullen Jennings <fluffy@cisco.com>
CC: Paul Long <plong@packetizer.com>, sip@ietf.org
Subject: Re: [Sip] SIP DTMF
References: <BC30A7A1.2D252%fluffy@cisco.com>
In-Reply-To: <BC30A7A1.2D252%fluffy@cisco.com>
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The SIPPING work group web page on the IETF site has=20
http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-01.txt as=20
the current draft.  This version has significant differences than the=20
one quoted below, one in particular is the use of SUBSCRIBE/NOTIFY for=20
requesting and reporting key presses.

Best regards,
Bert

Cullen Jennings wrote:
> The draft is
> http://www.ietf.org/internet-drafts/draft-burger-sipping-kpml-02.txt
>=20
> 2833 is the preferred method, but if an off media axis device wished to=
 get
> notification about key presses, it could. How these all relate is cover=
ed in
>=20
> http://www.softarmor.com/wgdb/docs/
>          draft-ietf-sipping-app-interaction-framework-00.txt
>=20
> Cullen
>=20
>=20
> On 1/15/04 11:21 AM, "Paul Long" <plong@packetizer.com> wrote:
>=20
>=20
>>Sreeram,
>>
>>I can't find a "kpxml" draft on the IETF site or via Google--anywhere.
>>Can you provide a link to it?
>>
>>Also, from my experience, the preferred method of DTMF carriage is RFC
>>2833, followed by in-band audio, and then various proprietary
>>INFO-method schemes. What is the standing of RFC 2833 vis-=E0-vis "kpxm=
l?"
>>
>>Paul
>>
>>-----Original Message-----
>>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
>>sreeram.kanumuri@wipro.com
>>Sent: Tuesday, January 13, 2004 9:06 PM
>>To: xyxie@udtech.com.cn; sip@ietf.org
>>Cc: sipping@ietf.org
>>Subject: RE: [Sip] SIP DTMF
>>
>>Wendy,
>>
>>There was a discussion in the list some time back and it was decided
>>that SIP-INFO method should not be used to
>>Send the DTMF info.
>>
>>It was proposed that it is better to you kpxml for sending DTMF tones.
>>
>>Please look in to the SIP-Kpxml draft for sending DTMF tomes in-midcall
>>processing and all....
>>
>>
>>Regards,
>>Sreeram
>>
>>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D
>>Sreeram Kanumuri                Plot No 72, Keonics Electronics City
>>             Hosur Main Road
>>Wipro Technologies              Bangalore -29 INDIA
>>sreeram.kanumuri@wipro.com      Phone: 8520408 Ex:3208
>>skanumury@yahoo.com             ESN  :
>>www.wipro.com                   Mobile:
>>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D
>>
>>
>>-----Original Message-----
>>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of wendy
>>Sent: Wednesday, January 14, 2004 8:08 AM
>>To: sip@ietf.org
>>Cc: sipping@ietf.org
>>Subject: [Sip] SIP DTMF
>>
>>
>>Hello all,=20
>>
>>Who can tell me whether there is a RFC or I-D on DTMF in SIP which is
>>still not expired? I remember someone proposed using SIP INFO method fo=
r
>>DTMF one or two years ago. Is the DTMF relay implemented only by RTP
>>now?
>>
>>
>>Regards,
>>Wendy
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Difference between SIP URL and SIP URI
Date: Mon, 19 Jan 2004 12:21:52 -0500
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> I would like to know what is the difference 
> between SIP URL and SIP URI?

There was a naming problem.

RFC 2543 should have defined a SIP URI
instead of SIP URL.  It was corrected
by RFC 3261.  All reference to SIP URL
should be references to SIP URI.

The same problem exists concerning 
RFC 2806's definition of telephone-url.
It should have been telephone-uri.  
It is corrected within the rfc2806bis.


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To: Takuya Sawada <tu-sawada@kddi.com>
CC: slawrence@pingtel.com, ksrini@motorola.com, sip@ietf.org
Subject: Re: [Sip] Registering Multiple 'Contacts'
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I agree with Takuya.

	Paul

Takuya Sawada wrote:
> Hi,
> 
> 
>>Srinivasan Krishnamoorthy wrote:
>>
>>
>>>>  Section 10.2.4 of RFC3261 'Refreshing Bindings' says:
>>>>  "The UA then issues a REGISTER request for each of its bindings
>>>>before the expiration interval has elapsed. It MAY combine   several
>>>>updates into one REGISTER request."
>>>>  The above seems to indicate that the UAC may send:   (forgive the
>>>>Syntax)
>>>>        REGISTER         To: user@operator.net
>>>>        Contact: sip:user1@10.0.0.1;expires=3600
>>>>                200 OK
>>>>        REGISTER         To: user@operator.net
>>>>        Contact: sip:user1@192.0.0.1;expires=3600
>>>>                200 OK
>>>>  This, according to RFC3261, will register both Contact1 and Contact2
>>>>for user@operator.net. (The server will then use the q parameter
>>>>  of the contact for priority among contacts)
>>>
>>Paul Kyzivat <pkyzivat@cisco.com> writes:
>>
>>
>>>Yes.
>>
>>My reading is that the results should differ depending on the Call-Id
>>used in those two REGISTER requests:  
>>
>>  - If they have different Call-Id values, then the server should
>>    register both contacts.
>>
>>  - If they have the same Call-Id value, then the second REGISTER
>>    should cause the server to replace the first contact with the
>>    second, so that there would be only
>>
> 
> In my reading of RFC 3261 section 10.3 step 7, Contact URI will be added
> to the list irrelevant to Call-ID value if its value is new to the list.
> 
> 
>>If correct, this adds an additional alternative for the UA - if it can
>>remember its Call-Id values across contact changes, it can make sure
>>that old registrations are replaced even before they expire.
>>
> 
> If it can remember any value in the previous power-cycle, it may remember
> Contact URI and then can remove it with expires=0 explicitly.
> 
> Note that assumeing it uses the same "fixed" Call-ID across power-cycle,
> it must remember "dynamic" CSeq value in old power-cycle.
> 
> Regards,
> Takuya
> 
> 
>>-- 
>>Scott Lawrence        
>>  Pingtel Corp.   
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/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: +81-3-6678-0285
> 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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Takuya Sawada wrote:
> Hi,
> 
>>>>How will the Server prevent against mis-behaved clients in this case?
>>>
>>>The expiration time bounds the problem, but doesn't eliminate it.
>>>Assume the UAC power cycles, forgets its old address, gets a new one and registers it. It will then 
>>
> presumably continue to refresh the new address, but not the old one. So the old registration will continue 
> to be active until it expires. During that time requests will probably be forked to it and the new address, 
> but the ones to the old address will fail.
> 
>>>Paul
>>
>>Another case I could think of is a DHCP environment - where a mobile-ua gets 
>>different ipaddresses on every power-cycle.
>>
>>In this case we may have to make sure the DHCP lease timers are longer than
>>registration duration in the network - or else we could be terminating
>>a call to an old-IP - that has possibly been granted to another UA that
>>registered in the interim???
>>
> 
> I think that SIP UA which use dynamic IP address SHOULD obey the behaviour 
> described above.
> Keep DHCP lease timer > SIP registration timer. (any timer used in app level)

I think I would go further and say the UA which uses addresses like this 
MUST do so. If the UA only has temporary ownership of the address, then 
it has no business registering it for a time period when it may not own it.

	Paul


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Subject: Re: [Sip] Registering Multiple 'Contacts'
From: Scott Lawrence <slawrence@pingtel.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Cc: Takuya Sawada <tu-sawada@kddi.com>, ksrini@motorola.com, sip@ietf.org
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On Mon, 2004-01-19 at 13:42, Paul Kyzivat wrote:
> I agree with Takuya.

> Takuya Sawada wrote:

> > If it can remember any value in the previous power-cycle, it may remember
> > Contact URI and then can remove it with expires=0 explicitly.
> > 
> > Note that assumeing it uses the same "fixed" Call-ID across power-cycle,
> > it must remember "dynamic" CSeq value in old power-cycle.

I agree with this as well; my point was actually that the correct thing
for the registry in the original inquiry could not actually be
determined based on the messages given, because they did not include the
Call-Id and Cseq values.

Whether or not registrations should be dependant on these values, and
what the right strategy is for a user agent are interesting quesitons
too.



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From: Alex Audu <alex.audu@alcatel.com>
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To: Rohan Mahy <rohan@cisco.com>
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Subject: Re: [Sip] re: WGLC: resource priority
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Hi,

The questions I had while reading this draft were mostly along the points Rohan has listed.
There were attempts to describe behavior (section 4.5 User Agent Client Behavior, 4.6 User
Agent Server Behavior, and 4.6 Proxy Behavior), but these sections were very silent regarding
what actions these elements would take to prevent or ease congestion based on assigned
priority values.

Another issue that needs to be addressed is: what should be the behavior of these devices
to ensure low priority events/requests don't get starved out of processing resources?

I was also a bit confused by the fact that you seemed to be addressing authorization
issues within priority specifications (section 4.3, and 4.6 and 5.0 for example).  I think it
will be better to decouple authorization issues from this document and leave such issues
to drafts devoted to authorzation. Besides, authorization processing should have passed
before prioritization.

Lastly, is 417 (Unknown Resource-Priority) a new error code? I assume it is since it is not
in rfc 3261.   If so, can we make the name more generic,..like Unknown Value (or Parameter)?
This will aid re-use of this code, instead of it being limited to priority handling alone.

Regards,
Alex.


Rohan Mahy wrote:

> Hi,
>
> Below is my review of resource-priority. Detailed comments are below, but I have a few general ones:
>
> 1. There is no normative description of the behavior of SIP devices which receive a particular priority value. Some specifications needs to either specify detailed behavior for specific priority values, or the detailed semantics of those values. I think this belongs in the IANA registration of each namespace, but those references are currently very thin. I made some specific comments to this end in the appropriate sections.
>
> 2. Also, I am curious what the semantics of this header mean in the context of SUB/NOT. Can someone provide an example that makes sense?
>
> 3. Finally, the document shows that proxies can modify, insert, and delete the RP header. I don't think this is a good idea. Where cooperating UAs and proxies need to agree on an RP value, I believe the session policy mechanism can be used. A proxy can reject the message for policy reasons (use a lower/higher/specific resource-value) and the UA can retry the request with these new values.
>
> thanks,
> -rohan
>
> --------------
>
> detailed comments:
>
> put TOC at the beginning
>
> section 2
>
> suggest
> ..telephone circuits, IP bandwidth ^reservations or allocations^...
>
> remove or motivate references to presence
>
> section 3:
>
> " Implementations MAY change the value offered in the request; in some
> environments, the response value is known to be the same as in the
> request." ??? what does that mean ???
>
> BNF is wrong (your current syntax would require a leading comma). do this instead:
>
> Resource-Priority = "Resource-Priority" HCOLON Resource-value
>
> Accept-Resource-Priority = "Accept-Resource-Priority" HCOLON
> [Resource-value *(COMMA Resource-value)]
>
> check consolidated bnf for reuse of terms
>
> also, the case of Resource-value is highly unusual.
>
> reword this, it is awkward:
>
> " We require that even namespaces with only one priority
> value list that value to avoid problems if additional
> priority values are added later."
>
> There may be multiple resource values or, equivalently, multiple
> Resource-Priority header /fields/field values/.
>
> what does it mean to "support" a particular Resource-value?
>
> our current method list is:
> INV ACK CAN BYE REG OPT PRA
> SUB NOT UPD MSG REF INF PUB
>
> what does RP mean in a SUB or NOT?
>
> where is section 4.2?
>
> section 4.3
>
> add comma here please...
> Elements receiving
> requests with namespaces or priority values that they do not
> understand ^,^ act according to the rules in the next section.
>
> section 4.4: you should introduce the resource-priority option tag (just the tag) back in section 3
>
> section 4.7: i don't think a proxy should be able to insert, delete, or modify the RP header. This makes it impossible to use e2e on the RP headers, which I think will be required.
>
> section 9-12:
>
> option-tags are always lowercase by convention
>
> you should add a note to the RFC editor asking for RFCXXXX to be replaced with the RFC number of this document.
>
> please make section 9 be IANA considerations (start with the text in section 12), then under this, create section 9.1 (formerly section 9) to register the new headers, section 9.2 (formerly section 10) to register the resource-priority header, section 11 becomes 9.3.
>
> include a registration template (for example)
>
> Namespace:
> Description:
> Organization Responsible for Change Control of this Namespace:
> Priority values (ordered from least priority to greatest priority):
> Default priority:
> Preempt or Prioritize:
> Document Containing Normative Description:
> Section Containing Normative Description:
> Additional security requirements:
>
> I would also consider including Section B.1 as section 9.4, B.2 -> 9.5, B.3 -> 9.6
> (IANA will be happy if all the work they have to do is consolidated under a single 1st-level section heading).
>
> in any case for the initial registrations, please fill in the template for each one; for example:
>
> Namespace: q735
> Description: ITU Q.735.3 Multi-level precedence and preemption in SS7
> Organization Responsible for Change Control of this Namespace: ITU-T
> Priority values (ordered from least priority to greatest priority): 4, 3, 2, 1, 0
> Default priority: 4
> Preempt or Prioritize: prioritize
> Document Containing Normative Description: ITU-T Q.735.3 [3]
> Section Containing Normative Description: ??
> Additional security requirements: none
>
> Namespace: dsrn
> Description: United States Defense Red Switched Network
> Organization Responsible for Change Control of this Namespace: US Department of Defense?
> Priority values (ordered from least priority to greatest priority):
> routine, priority, immediate, flash, flash-override, flash-override-override
> Default priority: routine
> Preempt or Prioritize: preempt
> Document Containing Normative Description: FIPS XXXX ?? [?]
> Section Containing Normative Description: ??
> Additional security requirements: All implementations must follow the certificate management guidelines specified in xxxx [?]


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From: OKUMURA Shinji <shin@softfront.co.jp>
Subject: Re: [Sip] WGLC for PUBLISH
To: aki.niemi@nokia.com
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Hi Aki,

Thank you for your comments.

You say that R-uri is used to determine what resource the
PUBLISH is targeted for. But R-uri has the possibility to 
be rewritten by proxy.

Even if PUA sends "sip:presentity@example.com",
"sip:presentity@pa.example.com" may reach to PA.

Now my questions are:

a) PUBLISH message MUST NOT pass proxy?

b) proxy MUST NOT rewrite R-uri?

   b-1. the uri MUST NOT be registered?
   b-2. Register-Contact is <sip:presentity@example.com;maddr=pa.example.com> ?
   b-3. original message has Route: <sip:pa.example.com;lr> ?

c) PA MUST determine the target resource ONLY by username of R-uri?

Best regards,


Shinji

aki.niemi@nokia.com wrote in <98C7D2E5BCD2374C9AAE0BCD2E9C1DD9027D92C9@esebe013.ntc.nokia.com>
>Hi Shinji,
>
>Thanks for the comments. Answers inline.
>
> > -----Original Message-----
> > From: ext OKUMURA Shinji [mailto:shin@softfront.co.jp]
> > Sent: 15 January, 2004 12:49
> > To: Rohan Mahy
> > Cc: sip@ietf.org; Dean Willis; Niemi Aki (Nokia-M/Helsinki)
> > Subject: Re: [Sip] WGLC for PUBLISH
> > 
> > 
> > Hi,
> > 
> > [1] I believe it requires following changes in some headers of
> >     the illustrated example in section 12.
> > 
> >     In M1,   Contact: <sip:watcher@example.com> should be 
> > rewritten as
> >              Contact: <sip:watcher@10.0.0.1>
> >
> > 
> >     In M2,   Contact: <sip:pa@example.com> should be
> >              Contact: <sip:pa.example.com>
> >
> >     In M3,    NOTIFY sip:presentity@example.com SIP/2.0
> >               Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
> >               To: <sip:watcher@example.com>;tag=12341234
> >               From: <sip:presentity@example.com>;tag=abcd1234
> >               Call-ID: 12345678@10.0.0.1
> >               CSeq: 1 NOTIFY
> >               ....
> > 
> >               should be
> > 
> >               NOTIFY sip:watcher@10.0.0.1 SIP/2.0
> >               Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
> >               To: <sip:watcher@example.com>;tag=12341234
> >               From: <sip:presentity@example.com>;tag=abcd1234
> >               Call-ID: 12345678@10.0.0.1
> >               CSeq: 1 NOTIFY
> >               Contact: <sip:pa.example.com>
> >               ....
>
>All true. Also, since the Via contains a domain name, I will add a "received" 
>param to the Via in the response.
>
>It's very good that you pointed out these mistakes in the examples. I swear, 
>I've looked at them a few times before, but I guess my internal SIP parser is 
>so well built that it never gave me an error message. ;-)
>
>I'll fix these for the next revision.
>  
> > [2] It could be helpful if the following example (2) is added in 
> >     the document as an addition to the only example (1) illustrated 
> >     in section 12.
>
>I'm a little reluctant to adding this. There are excellent SIP call flows 
>already documented elsewhere (see RFC 3665, e.g., Section 3.2), and to this 
>particular method, adding a proxy in between the PUA and the PA works no  
>different from how this works for other methods (MESSAGE, INVITE). I.e., the 
>proxy rewriting the Request-URI follows the same logic as the other methods 
>would.
>
>However, since this issue has come up before, it might be a good idea to add a 
>proxy between the PUA and PA. Any other views on this? 
>
> >     (1) PUA <---> PA <---> WATCHER
> > 
> >     (2) PUA <---> proxy <---> PA
> > 
> >         For example:
> > 
> >         PUA <-----> proxy <------> PA
> >          |            |            |
> >          | M1 PUBLISH |            |
> >          |----------->| M2 PUBLISH |
> >          |            |----------->|
> >          |            | M3 200 OK  |
> >          | M4 200 OK  |<-----------|
> >          |<-----------|            |
> >  
> >         M1:
> >             PUBLISH sip:pa@example.com SIP/2.0
> >             Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
> >             From: <sip:presentity@example.com>;tag=1234wxyz
> >             To: <pres:presentity@example.com>
> >             Call-ID: 81818181@pua.example.com
> >             CSeq: 1 PUBLISH
> >             Max-Forwards: 70
> >             Expires: 3600
> >             Event: presence
> >             Content-Type: application/pidf+xml
> >             Content-Length: ...
>
>Note that the Request-URI needs to be 'sip:presentity@example.com'. PUBLISH is 
>specifically addressed to the resource, not the presence agent.
>  
> >         M2:
> >             PUBLISH sip:pa.example.com SIP/2.0
> >             Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKpx123
> >             Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
> >             From: <sip:presentity@example.com>;tag=1234wxyz
> >             To: <pres:presentity@example.com>
> >             Call-ID: 81818181@pua.example.com
> >             CSeq: 1 PUBLISH
> >             Max-Forwards: 70
> >             Expires: 3600
> >             Event: presence
> >             Content-Type: application/pidf+xml
> >             Content-Length: ...
>
>The proxy would rewrite the Request-URI, but probably to something like 
>'sip:presentity@pa.example.com'. Note that when the request is received by the 
>PA/ESC, it specifically uses the Request-URI to determine what resource the 
>PUBLISH is targeted for. This is identical to a SUBSCRIBE.  
>
> >         M3:
> >             SIP/2.0 200 OK
> >             Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bKpx123
> >             Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
> >             From: <sip:presentity@example.com>;tag=1234wxyz
> >             To: <pres:presentity@example.com>;tag=1a2b3c4d
> >             Call-ID: 81818181@pua.example.com
> >             CSeq: 1 PUBLISH
> >             SIP-ETag: dx200xyz
> >             Expires: 1800
> >             Content-Length: 0
> >  
> >         M4:
> >             SIP/2.0 200 OK
> >             Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
> >             From: <sip:presentity@example.com>;tag=1234wxyz
> >             To: <pres:presentity@example.com>;tag=1a2b3c4d
> >             Call-ID: 81818181@pua.example.com
> >             CSeq: 1 PUBLISH
> >             SIP-ETag: dx200xyz
> >             Expires: 1800
> >             Content-Length: 0
> >  
> > [3] It could be better if the "Request-URI" is used as the message's
> >     target address and the 'Presence Resource' is specified 
> > using "To"
> >     header.
> > 
> >     It would be easy for understanding if REGISTER's location 
> >     registration operation like mechanism is considered.
>
>This is actually how PUBLISH used to work. But having the Request-URI point to 
>a PA requires special handling from the proxies and/or a specific naming 
>convention for PAs to be able to route the request. So we decided to reject the 
>REGISTER-like operation (determining the resource by the To-field) in favor of 
>the SUBSCRIBE-like behavior the draft currently has.
>
> > 
> >     The same point could also be said about RFC3265. This is because,
> > 
> >         |3.1.5. Proxy SUBSCRIBE Behavior
> >         |  Proxies need no additional behavior beyond that 
> > described in SIP [1]
> >         |  to support SUBSCRIBE.
> > 
> > [4] For expressing 'Presence Resource', the 
> > pres-URI(draft-ietf-impp-pres-04)
> >     would be better than that of the sip-URI.
> > 
> >     Finally, when [3] & [4] are considered together, the 
> > various usages 
> >     of URI can be expressed as listed below.
> > 
> > +------------+-----------------------------+-----------------
> > ------------+-----------------------------+
> > |            | PUBLISH                     | SUBSCRIBE       
> >             | NOTIFY                      |
> > +------------+-----------------------------+-----------------
> > ------------+-----------------------------+
> > |Request-URI | URI of PA                   | URI of PA       
> >             | Contact-URI of SUBSCRIBE    |
> > |            | sip:pa@example.com          | 
> > sip:pa@example.com          | sip:watcher.example.com     |
> > +------------+-----------------------------+-----------------
> > ------------+-----------------------------+
> > |To-URI      | URI of presence resource    | URI of presence 
> > resource    | From-URI of SUBSCRIBE       |
> > |            | pres:presentity@example.com | 
> > pres:presentity@example.com | sip:watcher@example.com     |
> > +------------+-----------------------------+-----------------
> > ------------+-----------------------------+
> > |From-URI    | URI of presentity           | URI of watcher  
> >             | To-URI of SUBSCRIBE         |
> > |            | sip:presentity@example.com  | 
> > sip:watcher@example.com     | pres:presentity@example.com |
> > +------------+-----------------------------+-----------------
> > ------------+-----------------------------+
>
>Adding to my previous comments, how would the UA find out the address of a PA 
>that services the presentity? We can't mandate a specific naming convention for 
>PAs in general, which is why the Request-URI is always the address of the 
>subscribed/published resource.
>
>Note that one valid location for a PA is in the client device. Because of the 
>way PUBLISH and SUBSCRIBE currently work, this requires nothing special from a 
>proxy - it simply forwards/redirects these requests to the UA like any other 
>request for that AoR.
>
>Cheers,
>Aki
>
> > 
> > Regards,
> > shinji
> > 
> > Rohan Mahy wrote in <C3094B8F-4563-11D8-B291-0003938AF740@cisco.com>
> > >Hi,
> > >
> > >I'd like to begin Working Group Last Call for PUBLISH:
> > >
> > >	
> > http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-02.txt
> > >
> > >This WGLC will end on January 28, 2004.
> > >
> > >thanks,
> > >-rohan

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From: "Niemi Aki (Nokia-M/Helsinki)" <aki.niemi@nokia.com>
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To: ext OKUMURA Shinji <shin@softfront.co.jp>
CC: sip@ietf.org
Subject: Re: [Sip] WGLC for PUBLISH
References: <200401151049.AA03044@mail2.softfront.co.jp>	<98C7D2E5BCD2374C9AAE0BCD2E9C1DD9027D92C9@esebe013.ntc.nokia.com> <200401201059.AA03088@mail2.softfront.co.jp>
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Hi Shinji,

The key thing here is that there is nothing special about how a proxy 
determines the next hop target for PUBLISH requests. There are several 
options to doing this:

	- A domain has configured all PUBLISH requests to be
	  forwarded to a PA
	- A PA has explicitly added an address binding for that
	  AOR, possibly using the SIP registration
	- All requests get forwarded to a UA (no matter what the
	  method is) that has an active address binding in the
	  domain's location service for that AoR
	- The request gets targeted using some other, undefined
	  logic

The administrator for that domain just needs to pick a way of doing 
this. So when proxying a PUBLISH request, Section 16 in RFC 3261 applies 
in full.

Cheers,
Aki

ext OKUMURA Shinji wrote:

> Hi Aki,
> 
> Thank you for your comments.
> 
> You say that R-uri is used to determine what resource the PUBLISH is
> targeted for. But R-uri has the possibility to be rewritten by proxy.
> 
> 
> Even if PUA sends "sip:presentity@example.com", 
> "sip:presentity@pa.example.com" may reach to PA.
> 
> Now my questions are:
> 
> a) PUBLISH message MUST NOT pass proxy?
> 
> b) proxy MUST NOT rewrite R-uri?
> 
> b-1. the uri MUST NOT be registered? b-2. Register-Contact is
> <sip:presentity@example.com;maddr=pa.example.com> ? b-3. original
> message has Route: <sip:pa.example.com;lr> ?
> 
> c) PA MUST determine the target resource ONLY by username of R-uri?
> 
> Best regards,
> 
> 
> Shinji
> 
> aki.niemi@nokia.com wrote in
> <98C7D2E5BCD2374C9AAE0BCD2E9C1DD9027D92C9@esebe013.ntc.nokia.com>
> 
>> Hi Shinji,
>> 
>> Thanks for the comments. Answers inline.
>> 
>> 
>>> -----Original Message----- From: ext OKUMURA Shinji
>>> [mailto:shin@softfront.co.jp] Sent: 15 January, 2004 12:49 To:
>>> Rohan Mahy Cc: sip@ietf.org; Dean Willis; Niemi Aki
>>> (Nokia-M/Helsinki) Subject: Re: [Sip] WGLC for PUBLISH
>>> 
>>> 
>>> Hi,
>>> 
>>> [1] I believe it requires following changes in some headers of 
>>> the illustrated example in section 12.
>>> 
>>> In M1,   Contact: <sip:watcher@example.com> should be rewritten
>>> as Contact: <sip:watcher@10.0.0.1>
>>> 
>>> 
>>> In M2,   Contact: <sip:pa@example.com> should be Contact:
>>> <sip:pa.example.com>
>>> 
>>> In M3,    NOTIFY sip:presentity@example.com SIP/2.0 Via:
>>> SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2 To:
>>> <sip:watcher@example.com>;tag=12341234 From:
>>> <sip:presentity@example.com>;tag=abcd1234 Call-ID:
>>> 12345678@10.0.0.1 CSeq: 1 NOTIFY ....
>>> 
>>> should be
>>> 
>>> NOTIFY sip:watcher@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP
>>> pa.example.com;branch=z9hG4bK8sdf2 To:
>>> <sip:watcher@example.com>;tag=12341234 From:
>>> <sip:presentity@example.com>;tag=abcd1234 Call-ID:
>>> 12345678@10.0.0.1 CSeq: 1 NOTIFY Contact: <sip:pa.example.com> 
>>> ....
>> 
>> All true. Also, since the Via contains a domain name, I will add a
>> "received" param to the Via in the response.
>> 
>> It's very good that you pointed out these mistakes in the examples.
>> I swear, I've looked at them a few times before, but I guess my
>> internal SIP parser is so well built that it never gave me an error
>> message. ;-)
>> 
>> I'll fix these for the next revision.
>> 
>> 
>>> [2] It could be helpful if the following example (2) is added in
>>>  the document as an addition to the only example (1) illustrated
>>>  in section 12.
>> 
>> I'm a little reluctant to adding this. There are excellent SIP call
>> flows already documented elsewhere (see RFC 3665, e.g., Section
>> 3.2), and to this particular method, adding a proxy in between the
>> PUA and the PA works no different from how this works for other
>> methods (MESSAGE, INVITE). I.e., the proxy rewriting the
>> Request-URI follows the same logic as the other methods would.
>> 
>> However, since this issue has come up before, it might be a good
>> idea to add a proxy between the PUA and PA. Any other views on
>> this?
>> 
>> 
>>> (1) PUA <---> PA <---> WATCHER
>>> 
>>> (2) PUA <---> proxy <---> PA
>>> 
>>> For example:
>>> 
>>> PUA <-----> proxy <------> PA |            |            | | M1
>>> PUBLISH |            | |----------->| M2 PUBLISH | |
>>> |----------->| |            | M3 200 OK  | | M4 200 OK
>>> |<-----------| |<-----------|            |
>>> 
>>> M1: PUBLISH sip:pa@example.com SIP/2.0 Via: SIP/2.0/UDP
>>> pua.example.com;branch=z9hG4bK652hsge From:
>>> <sip:presentity@example.com>;tag=1234wxyz To:
>>> <pres:presentity@example.com> Call-ID: 81818181@pua.example.com 
>>> CSeq: 1 PUBLISH Max-Forwards: 70 Expires: 3600 Event: presence 
>>> Content-Type: application/pidf+xml Content-Length: ...
>> 
>> Note that the Request-URI needs to be 'sip:presentity@example.com'.
>> PUBLISH is specifically addressed to the resource, not the presence
>> agent.
>> 
>> 
>>> M2: PUBLISH sip:pa.example.com SIP/2.0 Via: SIP/2.0/UDP
>>> proxy.example.com;branch=z9hG4bKpx123 Via: SIP/2.0/UDP
>>> pua.example.com;branch=z9hG4bK652hsge From:
>>> <sip:presentity@example.com>;tag=1234wxyz To:
>>> <pres:presentity@example.com> Call-ID: 81818181@pua.example.com 
>>> CSeq: 1 PUBLISH Max-Forwards: 70 Expires: 3600 Event: presence 
>>> Content-Type: application/pidf+xml Content-Length: ...
>> 
>> The proxy would rewrite the Request-URI, but probably to something
>> like 'sip:presentity@pa.example.com'. Note that when the request is
>> received by the PA/ESC, it specifically uses the Request-URI to
>> determine what resource the PUBLISH is targeted for. This is
>> identical to a SUBSCRIBE.
>> 
>> 
>>> M3: SIP/2.0 200 OK Via: SIP/2.0/UDP
>>> proxy.example.com;branch=z9hG4bKpx123 Via: SIP/2.0/UDP
>>> pua.example.com;branch=z9hG4bK652hsge From:
>>> <sip:presentity@example.com>;tag=1234wxyz To:
>>> <pres:presentity@example.com>;tag=1a2b3c4d Call-ID:
>>> 81818181@pua.example.com CSeq: 1 PUBLISH SIP-ETag: dx200xyz 
>>> Expires: 1800 Content-Length: 0
>>> 
>>> M4: SIP/2.0 200 OK Via: SIP/2.0/UDP
>>> pua.example.com;branch=z9hG4bK652hsge From:
>>> <sip:presentity@example.com>;tag=1234wxyz To:
>>> <pres:presentity@example.com>;tag=1a2b3c4d Call-ID:
>>> 81818181@pua.example.com CSeq: 1 PUBLISH SIP-ETag: dx200xyz 
>>> Expires: 1800 Content-Length: 0
>>> 
>>> [3] It could be better if the "Request-URI" is used as the
>>> message's target address and the 'Presence Resource' is specified
>>>  using "To" header.
>>> 
>>> It would be easy for understanding if REGISTER's location 
>>> registration operation like mechanism is considered.
>> 
>> This is actually how PUBLISH used to work. But having the
>> Request-URI point to a PA requires special handling from the
>> proxies and/or a specific naming convention for PAs to be able to
>> route the request. So we decided to reject the REGISTER-like
>> operation (determining the resource by the To-field) in favor of 
>> the SUBSCRIBE-like behavior the draft currently has.
>> 
>> 
>>> The same point could also be said about RFC3265. This is because,
>>> 
>>> 
>>> |3.1.5. Proxy SUBSCRIBE Behavior |  Proxies need no additional
>>> behavior beyond that described in SIP [1] |  to support
>>> SUBSCRIBE.
>>> 
>>> [4] For expressing 'Presence Resource', the 
>>> pres-URI(draft-ietf-impp-pres-04) would be better than that of
>>> the sip-URI.
>>> 
>>> Finally, when [3] & [4] are considered together, the various
>>> usages of URI can be expressed as listed below.
>>> 
>>> +------------+-----------------------------+----------------- 
>>> ------------+-----------------------------+ |            |
>>> PUBLISH                     | SUBSCRIBE | NOTIFY
>>> | +------------+-----------------------------+----------------- 
>>> ------------+-----------------------------+ |Request-URI | URI of
>>> PA                   | URI of PA | Contact-URI of SUBSCRIBE    | 
>>> |            | sip:pa@example.com          | sip:pa@example.com
>>> | sip:watcher.example.com     | 
>>> +------------+-----------------------------+----------------- 
>>> ------------+-----------------------------+ |To-URI      | URI of
>>> presence resource    | URI of presence resource    | From-URI of
>>> SUBSCRIBE       | |            | pres:presentity@example.com | 
>>> pres:presentity@example.com | sip:watcher@example.com     | 
>>> +------------+-----------------------------+----------------- 
>>> ------------+-----------------------------+ |From-URI    | URI of
>>> presentity           | URI of watcher | To-URI of SUBSCRIBE
>>> | |            | sip:presentity@example.com  | 
>>> sip:watcher@example.com     | pres:presentity@example.com | 
>>> +------------+-----------------------------+----------------- 
>>> ------------+-----------------------------+
>> 
>> Adding to my previous comments, how would the UA find out the
>> address of a PA that services the presentity? We can't mandate a
>> specific naming convention for PAs in general, which is why the
>> Request-URI is always the address of the subscribed/published
>> resource.
>> 
>> Note that one valid location for a PA is in the client device.
>> Because of the way PUBLISH and SUBSCRIBE currently work, this
>> requires nothing special from a proxy - it simply
>> forwards/redirects these requests to the UA like any other request
>> for that AoR.
>> 
>> Cheers, Aki
>> 
>> 
>>> Regards, shinji
>>> 
>>> Rohan Mahy wrote in
>>> <C3094B8F-4563-11D8-B291-0003938AF740@cisco.com>
>>> 
>>>> Hi,
>>>> 
>>>> I'd like to begin Working Group Last Call for PUBLISH:
>>>> 
>>>> 
>>> 
>>> http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-02.txt
>>> 
>>> 
>>>> This WGLC will end on January 28, 2004.
>>>> 
>>>> thanks, -rohan

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Subject: [Sip] Who Should Initiate Re-invite??
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Hi,

I have a question regarding the SDP negotiation.
Consider the following scenario,

	INVITE(w/o SDP)
     -------------------->
	100 trying
     <--------------------
	180 Ringing
     <--------------------	
	200(with SDP, a=inactive)
     <--------------------	
	Ack(with SDP)
     -------------------->

As a=inactive has been used in SDP the RTP session is not established.
As RFC 2327 does not take care of a=inactive, some older implementations
respond with one codec in ACK with sendrecv mode(default). 

1. In this scenario, who should initiate Re-       Invite? 
2. What if both parties do not initiate
   a Re-invite?
3. What if both parties initiate a Re-invite? 
4. When does the call gets terminated?
5. Does this have some interop issues(between     3264,2327 implementations)?

Thanks & Rgds,
Murali

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murali wrote:
> Hi,
> 
> I have a question regarding the SDP negotiation.
> Consider the following scenario,
> 
> 	INVITE(w/o SDP)
>      -------------------->
> 	100 trying
>      <--------------------
> 	180 Ringing
>      <--------------------	
> 	200(with SDP, a=inactive)
>      <--------------------	
> 	Ack(with SDP)
>      -------------------->
> 
> As a=inactive has been used in SDP the RTP session is not established.

What do you mean by "not established"? There should be valid ports, 
though no packets should be flowing. And RTCP should be flowing between 
them.

> As RFC 2327 does not take care of a=inactive, some older implementations
> respond with one codec in ACK with sendrecv mode(default). 

Yes, the issue of backward compatibility hasn't been well addressed. 
Presumably we should expect the party sending the a=inactive to be 
prepared for a peer that doesn't understand and do the right thing.

In this case, in addition to returning sendrecv, the caller will most 
likely begin to send media. The callee had better be prepared to deal 
with that.

> 1. In this scenario, who should initiate Re-       Invite? 

You can't expect much from the caller since it is a now obsolete device. 
If you are going to impose new requirements on it, the first one should 
be to support a=inactive.

It isn't clear that a reinvite is required in this case. The caller is 
happy, and the callee is presumably dealing with the incoming audio it 
doesn't want. But if it wants to stop that, it could send a reinvite 
with c=0.0.0.0 to kill the audio.

> 2. What if both parties do not initiate
>    a Re-invite?

See above.

> 3. What if both parties initiate a Re-invite? 

If they happen concurrently, then it is a glare condition, and is 
covered by 3261. If they don't happen concurrently there is no problem - 
they both get acted upon.

> 4. When does the call gets terminated?

When somebody sends a BYE.

> 5. Does this have some interop issues(between     3264,2327 implementations)?

Yes, clearly there are some backward compatibility issues.

This has been discussed somewhat. If you use a=sendonly, recvonly, or 
inactive, you probably ought to be prepared for the other party not 
understanding. If it doesn't return a corresponding a-line in accord 
with the rules in 3264 then you should probably conclude it doesn't 
understand and act accordingly.

	Paul


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From exim@www1.ietf.org  Tue Jan 20 23:04:49 2004
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Date: Wed, 21 Jan 2004 13:01:53 +0900
From: OKUMURA Shinji <shin@softfront.co.jp>
Subject: Re: [Sip] WGLC for PUBLISH
To: "Niemi Aki (Nokia-M/Helsinki)" <aki.niemi@nokia.com>
Cc: sip@ietf.org
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Hi Aki, 

Thank you for your quick response. 

Now I understand.

The important point I think is that PA determines the target resource 
using the r-uri of PUBLISH message "received by PA". 

In other words, the r-uri sending by PUA and the one received by PA
need not to be same. 

Am I right or missing something? 
 
Cheers,

Shinji

>Hi Shinji,
>
>The key thing here is that there is nothing special about how a proxy
>determines the next hop target for PUBLISH requests. There are several
>options to doing this:
>
>	- A domain has configured all PUBLISH requests to be
>	  forwarded to a PA
>	- A PA has explicitly added an address binding for that
>	  AOR, possibly using the SIP registration
>	- All requests get forwarded to a UA (no matter what the
>	  method is) that has an active address binding in the
>	  domain's location service for that AoR
>	- The request gets targeted using some other, undefined
>	  logic
>
>The administrator for that domain just needs to pick a way of doing
>this. So when proxying a PUBLISH request, Section 16 in RFC 3261 applies
>in full.
>
>Cheers,
>Aki

_______________________________________________
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Hi All,

There was a draft-levy-sip-diversion-06.txt for diversion header processing which had expired and has been removed from the IETF site. A new version of the same draft draft-levy-sip-diversion-08.txt is available from

www.employees.org/~fluffy/ietf/draft-levy-sip-diversion-08.txt 

but not on the IETF draft site.

Moreover, as per one of the mail threads from the archives, it seems that "History-Info header" is the way to go instead of "Diversion header". So, is the "Diversion header" approach being dropped in favour of "History-Info header" and what is the status of the "Diversion header" draft. Can anybody provide information on these aspects.

Thanks

Ajay




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<DIV>
<DIV><FONT size=2>
<P>Hi All,</P>
<P>There was a draft-levy-sip-diversion-06.txt for diversion header processing which had expired and has been removed from the IETF site. A new version of the same draft draft-levy-sip-diversion-08.txt is available from</P>
<P>www.employees.org/~fluffy/ietf/draft-levy-sip-diversion-08.txt </P>
<P>but not on the IETF draft site.</P>
<P>Moreover, as per one of the mail threads&nbsp;from the archives, it seems that "History-Info header" is the way to go instead of "Diversion header". So,&nbsp;is the "Diversion header" approach being dropped in favour of "History-Info header" and&nbsp;what is the status of the "Diversion header" draft. Can anybody provide information on these aspects.</P>
<P>Thanks</P>
<P>Ajay</P></FONT></DIV></DIV><p><hr SIZE=1>
Do you Yahoo!?<br>
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To: ext OKUMURA Shinji <shin@softfront.co.jp>
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Subject: Re: [Sip] WGLC for PUBLISH
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Hi Shinji,

Exactly, that's the way it goes.

Cheers,
Aki

ext OKUMURA Shinji wrote:

> Hi Aki, 
> 
> Thank you for your quick response. 
> 
> Now I understand.
> 
> The important point I think is that PA determines the target resource 
> using the r-uri of PUBLISH message "received by PA". 
> 
> In other words, the r-uri sending by PUA and the one received by PA
> need not to be same. 
> 
> Am I right or missing something? 
>  
> Cheers,
> 
> Shinji
> 
> 
>>Hi Shinji,
>>
>>The key thing here is that there is nothing special about how a proxy
>>determines the next hop target for PUBLISH requests. There are several
>>options to doing this:
>>
>>	- A domain has configured all PUBLISH requests to be
>>	  forwarded to a PA
>>	- A PA has explicitly added an address binding for that
>>	  AOR, possibly using the SIP registration
>>	- All requests get forwarded to a UA (no matter what the
>>	  method is) that has an active address binding in the
>>	  domain's location service for that AoR
>>	- The request gets targeted using some other, undefined
>>	  logic
>>
>>The administrator for that domain just needs to pick a way of doing
>>this. So when proxying a PUBLISH request, Section 16 in RFC 3261 applies
>>in full.
>>
>>Cheers,
>>Aki

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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,
	I feel the grammar of the Timestamp header is buggy.
The grammar being
Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
               [ "." *(DIGIT) ] [ LWS delay ]
delay      =  *(DIGIT) [ "." *(DIGIT) ]

Can allow the following two cases which makes no sense

Timestamp : 1.
Timestamp : 1.0 .

Should'nt a bug be raised against this to update the grammar to

Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
               [ "." 1*(DIGIT) ] [ LWS delay ]
delay      =  1*(DIGIT) [ "." 1*(DIGIT) ]

Thanks for the patient reading.

Cheers,
Prasanna

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Subject: Re: [Sip] re: WGLC: resource priority (1)
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In a message dated 1/15/2004 2:50:15 PM Eastern Standard Time, 
rohan@cisco.com writes:


Please do not confuse the change control of the *registration* with the 
change control of the *normative behavior*.  Only the change control of 
the registration is under the control of the IETF. Obviously if you 
change the normative behavior in a separate document, you need to 
change the registration to point at the changes behavior.

Recall that the IANA registration serves two purposes: 1) to prevent 
namespace conflicts, and 2) to allows implementors to find the 
documents relevant to any namespaces which interest them.

[MAP] That's the whole point I'm trying to get at - to keep the control of 
the "registration" and the control of the "normative behavior" separate. The 
IANA consideration section of this draft, and any "template" that is defined to 
provide information to IANA for the registration, should be limited to what is 
required for the registration. That should not include information about the 
behaviour of other entities that IANA doesn't need to do its job.


In a message dated 1/15/2004 2:50:15 PM Eastern Standard Time, 
rohan@cisco.com writes:

Note that Section 2 of this 
> draft lists 4 treatments, some of which I would not classify as being 
> either "preempt" or "prioritize".

Can you provide an existence proof of a separate type of treatment?

[MAP] I'm reading the word "prioritize" in its literal sense - to put one 
thing in front of another.

One of the preferential treatments that may be applied to a call marked with 
a higher value is to use a higher call acceptance limit. Another is to exempt 
it from controls that may prevent lower valued calls from being carried, just 
as is done today for GETS. I just don't like to use the word "prioritize" to 
describe these types of preferential treatments, whihc have nothing to do with 
putting one call in front of another.

There can be other types of "preferential treatment" besides just "preempt" 
and "prioritize", so the template used to register the RP-header with IANA 
could not be limited to only those two choices. It is not possible to say what all 
the methods might be in the future. Besides, there is no reason for the IANA 
registration to include such information. IANA can not use this information.

The template should be only the following:

Namespace:
Description:
Organization Responsible for Change Control of this Namespace:
Priority values (ordered from least priority to greatest priority):
Default priority:


Mike Pierce

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Prasanna'" <prasanna@huawei.com>, dean.willis@softarmor.com,
        sip@ietf.org, rohan@cisco.com, lavis@huawei.com
Subject: RE: [Sip] Issue with the Timestamp grammar in RFC-3261
Date: Thu, 22 Jan 2004 09:00:29 -0500
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Generally, a number with a trailing decimal period is allowed.
However, I'll agree you need at least one digit in delay.
OTOH, it could be a fractional delay (.3).  So, delay might have
to be

delay = (1*(DIGIT) [ "." *(DIGIT) ]) \ ( *(DIGIT) "." 1*DIGIT )

Brian

> -----Original Message-----
> From: Prasanna [mailto:prasanna@huawei.com]
> Sent: Thursday, January 22, 2004 12:36 AM
> To: dean.willis@softarmor.com; sip@ietf.org; rohan@cisco.com;
> lavis@huawei.com
> Subject: [Sip] Issue with the Timestamp grammar in RFC-3261
> 
> 
> Hi,
> 	I feel the grammar of the Timestamp header is buggy.
> The grammar being
> Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
>                [ "." *(DIGIT) ] [ LWS delay ]
> delay      =  *(DIGIT) [ "." *(DIGIT) ]
> 
> Can allow the following two cases which makes no sense
> 
> Timestamp : 1.
> Timestamp : 1.0 .
> 
> Should'nt a bug be raised against this to update the grammar to
> 
> Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
>                [ "." 1*(DIGIT) ] [ LWS delay ]
> delay      =  1*(DIGIT) [ "." 1*(DIGIT) ]
> 
> Thanks for the patient reading.
> 
> Cheers,
> Prasanna
> 
> *****************************************************************
> This e-mail and its attachments contain confidential information from
> HUAWEI, which is intended only for the person or entity whose 
> address is
> listed above. Any use of the information contained herein in any way
> (including, but not limited to, total or partial disclosure,
> reproduction, or dissemination) by persons other than the intended
> recipient(s) is prohibited. If you receive this e-mail in 
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>  
> 
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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> 

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Subject: Re: [Sip] Two IP addresses (c= lines) in invite
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The SDP below is OK. I point out the following text in RFC2327:

The connection (`c=') and attribute (`a=') information in the
    session-level section applies to all the media of that session unless
    overridden by connection information or an attribute of the same name
    in the media description.  For instance, in the example below, each
    media behaves as if it were given a `recvonly' attribute.

however, you won't get the behavior i think you want, which is to 
offer multiple IP addresses to which a stream can be sent. For within 
thte same IP version, the alt parameter is defined in ICE for this 
purpose:

http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-00.txt

for indicating that a UA is dual stack and can receive using either v4 
or v6, there is the ANAT specification:

http://www.ietf.org/internet-drafts/draft-ietf-mmusic-anat-00.txt

-jonathan R.

Bilajbegovic Damir wrote:

>     If I sent a INVITE message with two IP addresses (i.e. user has two IP
> addresses and on one it should receive messages) what happens?
> Is it valid and how the terminal or proxy should react on this. (there are
> c= lines )
> And one more additional question: If one c= is IPv6 and another line is c=
> IPv4 and colleen (B) party is IPv4 what does is do? Will B party (or B
> proxy) reject the message, replay with deleted misunderstood lines, or will
> put the port numbers on 0.  
> The message invite is down below (I think it should look like this- is it
> good?) 
>  Thanks in advance.  
>  
>             Damir Bilajbegovic
>  
> INVITE tel:+1-212-555-2222 SIP/2.0
>  
>   ............ 
>  
> v=0
> o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd
> s=-
> c=IN IP6 5555::aaa:bbb:ccc:ddd 
> t=0 0
> m=video 3400 RTP/AVP 98 99
> b=AS:75
> a=curr:qos local none
> a=curr:qos remote none
> a=des:qos mandatory local sendrecv
> a=des:qos none remote sendrecv
> a=rtpmap:98 H263
> a=fmtp:98 profile-level-id=0
> a=rtpmap:99 MP4V-ES
>  
>  
> c=IN IP6 5555::aaa:bbb:ccc:eee   
> t=0 0
> m=video 3400 RTP/AVP 98 99
> b=AS:75
> a=curr:qos local none
> a=curr:qos remote none
> a=des:qos mandatory local sendrecv
> a=des:qos none remote sendrecv
> a=rtpmap:98 H263
> a=fmtp:98 profile-level-id=0
> a=rtpmap:99 MP4V-ES
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/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
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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To: Scott Lawrence <slawrence@pingtel.com>
CC: Paul Kyzivat <pkyzivat@cisco.com>, ksrini@motorola.com, sip@ietf.org
Subject: Re: [Sip] Registering Multiple 'Contacts'
References: <Pine.LNX.4.21.0401121159440.23050-100000@bulb.corp.mot.com>	<4002E0B9.8030203@cisco.com> <vheku1f0ks.fsf@sukothai.pingtel.com>
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Scott Lawrence wrote:

> Srinivasan Krishnamoorthy wrote:
> 
> 
>>>  Section 10.2.4 of RFC3261 'Refreshing Bindings' says:
>>>  "The UA then issues a REGISTER request for each of its bindings
>>>before the expiration interval has elapsed. It MAY combine   several
>>>updates into one REGISTER request."
>>>  The above seems to indicate that the UAC may send:   (forgive the
>>>Syntax)
>>>        REGISTER         To: user@operator.net
>>>        Contact: sip:user1@10.0.0.1;expires=3600
>>>                200 OK
>>>        REGISTER         To: user@operator.net
>>>        Contact: sip:user1@192.0.0.1;expires=3600
>>>                200 OK
>>>  This, according to RFC3261, will register both Contact1 and Contact2
>>>for user@operator.net. (The server will then use the q parameter
>>>  of the contact for priority among contacts)
> 
> 
> Paul Kyzivat <pkyzivat@cisco.com> writes:
> 
> 
>>Yes.
> 
> 
> My reading is that the results should differ depending on the Call-Id
> used in those two REGISTER requests:  
> 
>   - If they have different Call-Id values, then the server should
>     register both contacts.
> 
>   - If they have the same Call-Id value, then the second REGISTER
>     should cause the server to replace the first contact with the
>     second, so that there would be only

No, this is not the case. The callid/cseq are used as an ordering 
tool. In the abovoe case, if the Call-ID are the same, both contacts 
are added, and both are marked as having that particular callid, but 
with different cseq. if the Call-id are different, both contacts are 
added, but with different call-id and cseq.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.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: hisham.khartabil@nokia.com
CC: brett@broadsoft.com, sip@ietf.org
Subject: Re: [Sip] Is Retry-After:0 valid?
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Yes, its valid. However the text is incorrect, since formally, 
"positive" most definitely does NOT include zero (see 
http://en2.wikipedia.org/wiki/Positive_number).

I've logged this in bugzilla:
http://bugs.sipit.net/sipwg/show_bug.cgi?id=746

-Jonathan R.

hisham.khartabil@nokia.com wrote:

> I don't see why it would be disallowed. It is certainly valid and says to retry immediately (after 0 seconds).
> 
> I guess the second quote can have the word "inclusive" added.
> 
> Regards,
> Hisham
> 
> 
>>-----Original Message-----
>>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
>>Brett Tate
>>Sent: 17.January.2004 02:14
>>To: sip-ietf (E-mail)
>>Subject: [Sip] Is Retry-After:0 valid?
>>
>>
>>(Trying the sip list because I did not
>>receive any reply concerning my 
>>sip-implementors posting...)
>>
>>
>>Is 0 a valid value for the retry-after header?
>>
>>I assume that it is; and the bnf allows it.
>>However I guess that it depends upon the meaning 
>>of "positive" in section 20.33 and if the 
>>"between" statement of section 14.2 is inclusive.
>>
>>RFC3261 section 20.33 mentions the following:
>> "The value of this field is a positive integer
>>  number of seconds (in decimal) after the time 
>>  of the response."
>>
>>RFC3261 section 14.2 mentions the following:
>> "MUST include a Retry-After header field with a
>>  randomly chosen value of between 0 and 10 seconds."
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/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
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: Re: [Sip] Registering Multiple 'Contacts'
From: Scott Lawrence <slawrence@pingtel.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Paul Kyzivat <pkyzivat@cisco.com>, ksrini@motorola.com, sip@ietf.org
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Earlier, I wrote:

> > My reading is that the results should differ depending on the Call-Id
> > used in those two REGISTER requests:  
> > 
> >   - If they have different Call-Id values, then the server should
> >     register both contacts.
> > 
> >   - If they have the same Call-Id value, then the second REGISTER
> >     should cause the server to replace the first contact with the
> >     second, so that there would be only

On Thu, 2004-01-22 at 10:44, Jonathan Rosenberg wrote:

> No, this is not the case. The callid/cseq are used as an ordering 
> tool. In the abovoe case, if the Call-ID are the same, both contacts 
> are added, and both are marked as having that particular callid, but 
> with different cseq. if the Call-id are different, both contacts are 
> added, but with different call-id and cseq.

I had quite a struggle with this recently; perhaps you could help
clarify what is meant by RFC 3261 sec 10.3 in step 6 when it says:

                                            ...the registrar checks
whether
   the Call-ID agrees with the value stored for each binding.  If
   not, it MUST remove the binding.  

I read that to mean that if my registry contains a binding

   to:        me@example.com
   call-id: 1234@example.org
   cseq:     100
   contact: line1@10.1.1.1

and I receive a REGISTER for:

   to:        me@example.com
   call-id: 5678@example.org
   cseq:     10
   contact: line2@10.1.1.2

the Call-ID does not agree (making Cseq irrelevant), so shouldn't the
registry remove the binding as directed above?  

[Whatever the answer, I believe that 10.3 is too hard to interpret.  If
I can learn what the right behavior is, I'll be happy to try to draft
text]

-- 
Scott Lawrence        
  Pingtel Corp.   



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To: slawrence@pingtel.com
Cc: sip@ietf.org
Subject: Re: [Sip] Registering Multiple 'Contacts'
From: Takuya Sawada <tu-sawada@kddi.com>
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Hi,

> Earlier, I wrote:
> 
> > > My reading is that the results should differ depending on the Call-Id
> > > used in those two REGISTER requests:  
> > > 
> > >   - If they have different Call-Id values, then the server should
> > >     register both contacts.
> > > 
> > >   - If they have the same Call-Id value, then the second REGISTER
> > >     should cause the server to replace the first contact with the
> > >     second, so that there would be only
> 
> On Thu, 2004-01-22 at 10:44, Jonathan Rosenberg wrote:
> 
> > No, this is not the case. The callid/cseq are used as an ordering 
> > tool. In the abovoe case, if the Call-ID are the same, both contacts 
> > are added, and both are marked as having that particular callid, but 
> > with different cseq. if the Call-id are different, both contacts are 
> > added, but with different call-id and cseq.
> 
> I had quite a struggle with this recently; perhaps you could help
> clarify what is meant by RFC 3261 sec 10.3 in step 6 when it says:
> 
>                                             ...the registrar checks
> whether
>    the Call-ID agrees with the value stored for each binding.  If
>    not, it MUST remove the binding.  
> 
This part, step 6, applies in  the case when REGISTER request contains
one Contact field value that contains the special value "*" and 
Expires that value is zero to remove all bindings.
When adding bindings or removing each binding or refresh bindings,
you must follow step 7.

Regards,
Takuya


> I read that to mean that if my registry contains a binding
> 
>    to:        me@example.com
>    call-id: 1234@example.org
>    cseq:     100
>    contact: line1@10.1.1.1
> 
> and I receive a REGISTER for:
> 
>    to:        me@example.com
>    call-id: 5678@example.org
>    cseq:     10
>    contact: line2@10.1.1.2
> 
> the Call-ID does not agree (making Cseq irrelevant), so shouldn't the
> registry remove the binding as directed above?  
> 
> [Whatever the answer, I believe that 10.3 is too hard to interpret.  If
> I can learn what the right behavior is, I'll be happy to try to draft
> text]
> 
> -- 
> Scott Lawrence        
>   Pingtel Corp.   
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/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: +81-3-6678-0285
tu-sawada@kddi.com

_______________________________________________
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From exim@www1.ietf.org  Thu Jan 22 23:25:04 2004
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Date: Fri, 23 Jan 2004 09:51:40 +0530
From: Prasanna <prasanna@huawei.com>
Subject: RE: [Sip] Issue with the Timestamp grammar in RFC-3261
In-reply-to: 
 <313680C9A886D511A06000204840E1CF070B630A@whq-msgusr-02.pit.comms.marconi.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>, dean.willis@softarmor.com,
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I agree to this grammar.  But I have one query?
Any reasons why a number with a trailing decimal period is allowed (it
is allowed in all the decimal numbers like "q" etc.)?
Thanks,
Prasanna

-----Original Message-----
From: Rosen, Brian [mailto:Brian.Rosen@marconi.com] 
Sent: Thursday, January 22, 2004 7:30 PM
To: 'Prasanna'; dean.willis@softarmor.com; sip@ietf.org;
rohan@cisco.com; lavis@huawei.com
Subject: RE: [Sip] Issue with the Timestamp grammar in RFC-3261


Generally, a number with a trailing decimal period is allowed. However,
I'll agree you need at least one digit in delay. OTOH, it could be a
fractional delay (.3).  So, delay might have to be

delay = (1*(DIGIT) [ "." *(DIGIT) ]) \ ( *(DIGIT) "." 1*DIGIT )

Brian

> -----Original Message-----
> From: Prasanna [mailto:prasanna@huawei.com]
> Sent: Thursday, January 22, 2004 12:36 AM
> To: dean.willis@softarmor.com; sip@ietf.org; rohan@cisco.com; 
> lavis@huawei.com
> Subject: [Sip] Issue with the Timestamp grammar in RFC-3261
> 
> 
> Hi,
> 	I feel the grammar of the Timestamp header is buggy.
> The grammar being
> Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
>                [ "." *(DIGIT) ] [ LWS delay ]
> delay      =  *(DIGIT) [ "." *(DIGIT) ]
> 
> Can allow the following two cases which makes no sense
> 
> Timestamp : 1.
> Timestamp : 1.0 .
> 
> Should'nt a bug be raised against this to update the grammar to
> 
> Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
>                [ "." 1*(DIGIT) ] [ LWS delay ]
> delay      =  1*(DIGIT) [ "." 1*(DIGIT) ]
> 
> Thanks for the patient reading.
> 
> Cheers,
> Prasanna
> 
> *****************************************************************
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> _______________________________________________
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> 


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Subject: RE : [Sip] draft-ietf-sip-history-info-01.txt
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From: =?iso-8859-1?Q?PROUVOST_S=E9bastien_FTRD/DAC/ISS?= <sebastien.prouvost@francetelecom.com>
To: "Jesske, R" <R.Jesske@t-com.net>, <sip@ietf.org>,
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I fully support this proposal.=20

The privacy regarding the history-info header relates to the privacy =
policy accorded to the target of the request (before it is retargeted). =
Therefore it shall not depend on the privacy policy accorded to the =
calling UA. The history-info header shall have its own privacy value.=20
Moreover in case the request is retargeted more than once (particularly =
if the targets represent different users), it shall be possible to =
request privacy for a target independently of the others. =20

Regards,
S=E9bastien Prouvost
France T=E9l=E9com R&D/DAC/CAR

-----Message d'origine-----
De : Jesske, R [mailto:R.Jesske@t-com.net]=20
Envoy=E9 : vendredi 16 janvier 2004 10:29
=C0 : sip@ietf.org
Cc : Poetzl, Joachim; Alexeitsev, D
Objet : [Sip] draft-ietf-sip-history-info-01.txt


Dear all,
in 1.3 Ensuring the Privacy of History_Info it is proposed to use a =
priv-value of Session or Header level privacy. I would like to propose a =
own privacy level for the history info, because the Information shows =
also information about the network. With regard to the provider =
operating a network some of the providers don't like to transmit =
information about call history. Or a redirecting party don't like to =
show the redirecting address. Therefore it should be possible to delete =
the history info at domain boundaries based on the privacy-value.

I propose the priv-value   =3D   "history_info"  for the whole =
History-Info header.

In addition a option would be useful where only parts (e.g Indes1.1) of =
the history info are restricted. That could be done with a privacy =
statement within the History-Info .

Best Regards

Roland


Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T332-2
Section T33; Signalling, Gateways and Switching Systems=20
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940=20
Fax:      +49 6151 83-4577=20
email:   r.jesske@t-com.net




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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 exim@www1.ietf.org  Fri Jan 23 14:08:42 2004
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Subject: [Sip] Questions on draft-ietf-sip-session-timer-13 and refresh timers
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I'me reviewing some IESG comments on this draft, and I've found
something I don't understand.

The original comment from Ted was something like "Why do we recommend
refresh at 1/2 the session timer in sections 7 and 8, but require a BYE
at a recommended 1/3St-10s in section 10?"

I understand why we require refresh, and why we suggested refreshing at
1/2 of the session timer value. What I don't understand is the BYE logic
in Section 10.


>From Section 10:

   If no 2xx response to a session refresh request is received before
   the session expiration, the UA SHOULD send a BYE request to terminate
   the session. It SHOULD send this BYE slightly before session
   expiration. The minimum of ten seconds and one third the session
   interval is RECOMMENDED.

   For example, if the session interval is 120 seconds, one third of
   this is 40 seconds. Since the minimum of 10 seconds and 40 seconds
   is 10 seconds, the BYE would be sent 10 seconds before the session
   expires.


I read this as:

A UA has issues a session refresh request, and is waiting around for
that transaction to complete. Unfortunately, it either 1) didn't issue
the request sufficiently early for the transaction to complete before
the session is going to expire, so it needs to abandon the refreshing
transaction and terminate the session before the session expires, 
or 2) the session refresh request failed, but we're going to continue to
use this session for a while before tearing it down before it expires.

It seems more reasonable that the UA should issue the session refresh
request far enough in advance of session expiration that the session
refresh request will complete (or fail with timeout) before the session
expires, and that if the session refresh request fails, the UA should
immediately terminate the session with a BYE.

This implies that the lowest allowable threshold for the negotiated
session timer should be larger than the largest timeout value for a
session refresh request (which could be an INVITE transaction). It also
may place requirements on UAs that have implemented non-standard timer
values to calculate this lowest allowable threshold for the negotiated
session timer value and include this calculated value in their
negotiation by initializing the Min-SE value.

--
Dean






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You are correct. Step 6 is only for Contact: *.

-Jonathan R.

Takuya Sawada wrote:

> Hi,
> 
> 
>>Earlier, I wrote:
>>
>>
>>>>My reading is that the results should differ depending on the Call-Id
>>>>used in those two REGISTER requests:  
>>>>
>>>>  - If they have different Call-Id values, then the server should
>>>>    register both contacts.
>>>>
>>>>  - If they have the same Call-Id value, then the second REGISTER
>>>>    should cause the server to replace the first contact with the
>>>>    second, so that there would be only
>>
>>On Thu, 2004-01-22 at 10:44, Jonathan Rosenberg wrote:
>>
>>
>>>No, this is not the case. The callid/cseq are used as an ordering 
>>>tool. In the abovoe case, if the Call-ID are the same, both contacts 
>>>are added, and both are marked as having that particular callid, but 
>>>with different cseq. if the Call-id are different, both contacts are 
>>>added, but with different call-id and cseq.
>>
>>I had quite a struggle with this recently; perhaps you could help
>>clarify what is meant by RFC 3261 sec 10.3 in step 6 when it says:
>>
>>                                            ...the registrar checks
>>whether
>>   the Call-ID agrees with the value stored for each binding.  If
>>   not, it MUST remove the binding.  
>>
> 
> This part, step 6, applies in  the case when REGISTER request contains
> one Contact field value that contains the special value "*" and 
> Expires that value is zero to remove all bindings.
> When adding bindings or removing each binding or refresh bindings,
> you must follow step 7.
> 
> Regards,
> Takuya
> 
> 
> 
>>I read that to mean that if my registry contains a binding
>>
>>   to:        me@example.com
>>   call-id: 1234@example.org
>>   cseq:     100
>>   contact: line1@10.1.1.1
>>
>>and I receive a REGISTER for:
>>
>>   to:        me@example.com
>>   call-id: 5678@example.org
>>   cseq:     10
>>   contact: line2@10.1.1.2
>>
>>the Call-ID does not agree (making Cseq irrelevant), so shouldn't the
>>registry remove the binding as directed above?  
>>
>>[Whatever the answer, I believe that 10.3 is too hard to interpret.  If
>>I can learn what the right behavior is, I'll be happy to try to draft
>>text]
>>
>>-- 
>>Scott Lawrence        
>>  Pingtel Corp.   
>>
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/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: +81-3-6678-0285
> 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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Prasanna wrote:

> I agree to this grammar.  

I believe the grammar for delay should be identical to the timestamp 
itself, so that:

delay = 1*(DIGIT) [ "." *(DIGIT) ]

I don't see the need to support somethiing like ".3"; since you can 
represent that as "0.3".

I've logged a bug against this:
http://bugs.sipit.net/sipwg/show_bug.cgi?id=748

But I have one query?
> Any reasons why a number with a trailing decimal period is allowed (it
> is allowed in all the decimal numbers like "q" etc.)?

It was inherited from HTTP which did it this way. In hindsight there 
is really no point in supporting multiple different ways of 
representing a rational number, it just makes the implementation more 
complicated. But, its what we have, so be it.


-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Subject: [Sip] querying the current event state of a publication
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In draft-ietf-sip-publish-02, section 5.7 proposes a way to query a 
specific published event state. It recommends subscribing to the AOR 
to learn it. However, it gives this caveat:

  Note that a subscription to the event package will likely deliver
       results of the event composition process of the state agent, which
       may be a subset or a superset of the current published event
       state.

which is another way of saying, that subscribing to the aor doesnt 
really allow you to query for a specific published event state.

If we want this functionality, I have an idea for an alternative 
approach that works much better, triggered by some conversations I had 
with Ben C. recently.

In the response to a PUBLISH request, the ESC returns a URI that 
identifies that particular event state publication (effectively, a URI 
that is bound to the AOR+event+etag of a publication). This URI is 
basically a GRUU. If a user wishes to learn the state of this 
particular event state, they merely subscribe to that URI. Indeed, a 
particularly interestinig way to do this is to define sIP etags as 
URIs, rather than reusing the definition from http. I think that 
having an etag as a URI is far more elegant, as it would allow for 
things like this function in a natural way. The alternative would be 
to possibly use the Contact header field, as the meaning is the same 
as in a 2xx to INVITE - identifying the specific *instance* to which 
you were connected when you contacted the AOR. This makes sense if you 
think of the event state publication URI as a GRUU.

Indeed, one might even then PUBLISH to that URI, and it would have the 
exact result you would expect - that particular event state is 
updated. In such a case the server woould NOT return an etag or 
another URI in the response.

So, as an example, if I want to publish for sip:jdrosen@dynamicsoft.com:

PUBLISH sip:jdrosen@dynamicsoft.com
Event: presence

<body contains cpim-pidf>


the server responds:

SIP 200 oK
SIP-Etag: sip:9adjsss=9888sdahhsd77fggld@dynamicsoft.com


now, I have a handle for the specific event state. To update it:

PUBLISH sip:9adjsss=9888sdahhsd77fggld@dynamicsoft.com
Event: presence

<body contains cpimd-pidf>


This would mean we wouldnt need the SIP-If-match.

Another alternative is to say that we are simply too far along in 
PUBLISH, and the time has passed for major changes as this would be.

Thoughts?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



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enclosed are my comments on -02. Most are minor. There is one major 
question, and I sent a separate note on that.



* the current title is not in line with the guidelines in 
draft-ietf-sip-guiudelines, which says:

  An important decision to be made about the extension is its title.
    The title MUST indicate that the document is an extension to SIP. It
    is RECOMMENDED that the title follow the basic form of "A [summary of
    function] for the Session Initiation Protocol (SIP)", where the
    summary of function is a one to three word description of the


as such, I suggest changing to "An Event State Publication Extension 
to the Session Initiation Protocol (SIP)"

* wording nit in the abstract:

This document describes an extension to the Session Initiation
    Protocol (SIP) for publishing event state used within the framework
    for SIP Event Notification.

is clearer as:

This document describes an extension to the Session Initiation
    Protocol (SIP) for publishing event state used within the SIP 
event framework.

* also in the abstract:

It is not intended to be a general-purpose mechanism
    for transport of arbitrary data, as there are better-suited
    mechanisms for this purpose (FTP, HTTP, etc.)

i suggest dropping the examples (FTP, HTTP) since it may raise some 
alarms that we are asserting that http should be used for all kinds of 
data transport functions.

* In the intro:

> The focus of this specification is to provide a framework for the
>    publication of event state from a user agent to an entity that is

focus implies that it does other things also, but it doesnt. I would 
rather say "this specification provides a framework for the ...."

* also in the intro:

> The first application of this mechanism is the publication of
>    presence state by a presence user agent to a presence compositor,
>    which has a tightly coupled relationship with the presence agent.

the beginning of this sentence doesnt make it clear that this spec 
indeed defines that application. how abouot:

"In addition to defining an event publication framework, this 
specification defines a usage of that framework for the publication of 
presence state [REF draft-ietf-simple-presence]..."

You shouuld add a reference to rfc2778 after "presence user agent", 
and to draft-ietf-simple-presence for the definition of presence 
agent. presence compositor is used here but not defined. Indeed, 
'presence compositor' does not appear in the definitions which follow. 
  As I believe the requiremenets and model will be folded into this 
spec, the definition shouould appear here.

* from the intro:

>  The mechanism described in this document can be extended to support
>    publication of any event state, for which there exists an appropriate
>    event package as defined in [1].

the comma in the midddle is not needed

* from the intro:

> It is not intended to be a
>    general-purpose mechanism for transport of arbitrary data, as there
>    are better-suited mechanisms for this purpose (FTP [6], HTTP [7],
>    etc.)

as above re: ftp, http

* in the definitions section, you offer definitions of soft and hard 
state. These are not the definitions for those terms as they are 
commonly known. I suggest you instead usue the terms "event hard 
state" and "event soft state" to match the scope of the definitions.

* section 3:

>  have multiple UAs or endpoints that publish event state. Each
>    endpoint may publish its own unique state, out of which the event
>    agent generates the composite event state of the resource.

s/event agent/event state compositor

* section 3:

> Through a
>    subscription to that event package,

"that" is a dangling reference, since you have not yet mentioned that 
publications address a resouurce and a specific event package. I 
suggest you make mention of that in the text abvoe

* section 3, para 2;

> In the generic sense, a UAC that publishes event state is labeled an
>    Event Publication Agent (EPA)

"in the generic sense" implies that there is a more specific case 
under which the definition can be understood, which is not the case. I 
suggest removing 'in the generic sense'

* section 3, paragraph 3:

> That is, the steady-state of
>    this event package in the absence of, or in addition to, soft state
>    provided through the PUBLISH mechanism

this makes it sound like steady state is a function of the event 
packae only. the hard state is a function of the resource uRI and 
event package. Indeed, one may think of an event package as merely 
sub-resource within the full resource. I wonder if it wouldn't be 
useful to come up with a term for this - for example, "scoped 
resource" is a combination of a URI and event package, and identifies 
a particular piece of event state.

* section 3, para 3:

> Setting this hard state or
>    configuring the composer policy is out of the scope of this
>    specification.

you use the term composer here; it should be "event state compositor 
policy". I know it sounds like a real nit, but having rigorously 
consistent terminology is really important for new people to pick up 
the spec and understand it. We were religious about this in RFC 3261.

* i suggest moving section 4 to later in the spec. the reason is that 
section 4 basically gives instructions for packages to use publish, 
yet it has not said how publish works. that context will be usefull to 
interpret section 4.

* section 4.2:

> There is no such meaning for the body of a response to a presence
>    publication when the document format used is CPIM PIDF.

this implies that, in general, the meaning of the body in the response 
depends on the body in the request. the previous paragraph says that 
this is rather defined on a package by package basis. I suggest 
rermoving "when the document format used is CPIM PIDF"

* section 4.3:

> 4.3 Multple Sources for Event State

s/Multple/Multiple

* re: table 1, I suggest adding a column which is "Expires value". 
Without this, there is no way to know whether or not a publish with no 
body and a sip-if-match is refreshing or deleting.

* section 5.1:

> Identification of published event state is provided by four pieces of
>    information: Request-URI, event type, and (optionally) an entity-tag
>    and the message body.

I think its just the first three. the message body itself does not 
IDENTIFY the event state - it IS the event state. The first three 
identify it.

* section 5.2

> For determining the type of the published event state, the EPA MUST
>    include a single Event header field in the PUBLISH requests. The

s/in the PUBLISH requests/in PUBLISH requests

> The
>    value of this header field indicates the event package, for which
>    this request is publishing event state.

comma in the middle is not needed

* section 5.2:

> For each successful PUBLISH request, the ESC will generate and assign
>    an entity-tag and return it in the SIP-ETag header field of the 200
>    (OK) response.

only 200 OK? Or 2xx? I thihnk its 2xx.

* section 5.1:

> The presence of a body and the SIP-If-Match header field determine
>    the specific operation that the request is performing, as described
>    in Table 1. These operations are described in more detail in the
>    following sections.

it would probably be helpful though to say one sentence here on what 
each of these mean.

* section 5.1:

> As with any other SIP message, the PUBLISH mechanism MAY use the
>    content indirection mechanism defined in [10].

this statement makes [10] a normative reference, I believe. It is 
listed as informative in the references section.

* section 5.2

> The EPA MAY send subsequent PUBLISH requests to refresh, modify, or
>    remove the event state established by a prior publication and
>    identified by the associated entity-tag. These operations will be
>    described in the following sections.

this paragraph is redundant. you have already said in 5.1 to see the 
sections below on how to do reefresh, modify or removal.

* section 5.3:

> PUBLISH requests SHOULD contain a single Expires header field.

i think MAY is sufficient; its a perfectly common case where the EPA 
jusut doesnt care and wants the server to tell it when to refresh.


* section 5.3:

> The actual validity period of the soft state is defined
>    by local policy at the ESC, although typically the event state is
>    cleared immediately after the publication expires.

this sentence might lead one to believe that an ESC can still keep 
around, and thus use, event state after its actual expiration. That is 
not a good thing; an ESC MUST expire event state after its expiration. 
of course this doesnt mean that the data must physically be removed 
from any dB or memory in the eSC, but rather that the ESC behaves as 
if the data has been removed.

we went through this with registrations. There are serious privacy 
implications for having the eSC in any way extend the lifetime as 
suggested by the client.

Indeed, section 5.3 shouuld say that the ESC won't ever extend the 
lifetime of the publication.

* section 5.3 is out of place. Section 5 is about generating requests. 
the subsections for the most part deal with the specific 4 sub-cases 
you identified in 5.1. However, 5.3 talks about setting expirations, 
which is something that is actually done for the cases in 5.2, 5.4, 
5.5 and 5.6. I would suggest that eaech of those four sections rather 
describe how the Expires header field is constructed for each case. 
That is consistent with having a column in table 1 on expires. Indeed, 
each section should also talk about bodies and etag. Along those 
lines, section 5.2 does not actually say that a body has to be 
present; it shouold be stated given the importance of this fact from 
table 1.

* section 5.4:

> Note that like any other 200 (OK) response to a PUBLISH, also this
>       response will contain a SIP-ETag header field with an entity-tag.

This is a little awkward. Suggest, "Like the 2xx rersponse to an 
initial PUBLISH request, the response to a refresh PUBLISH request 
will contain a SIP-ETag header field with an entity-tag."

* section 5.5:

> If the entity-tag matches previously
>    published event state at the ESC, that event state is replaced by the
>    event state carried in the PUBLISH request, and the EPA receives a
>    200 (OK) response.

I think we need to be really careful with the words here. What you 
have writteen actually disallows partial publication as an extension, 
wherein a UAC could send a publish that modifies a piece of the 
published state (i.e., change the note in tuple with id "3" to "out to 
lunch"). Thats because the text says the state is *replaced* by whats 
in the body. I would prefer to say that the state is updated, and that 
update depends on the content of the body. If the body contains a 
format whose semantics are 'this is the state of the resource' then 
the event state is replaced.

* section 5.7 is out of place, since it has nothing to do with 
constructing publish at all. indeed, since the note basically says 
that this doesnt work, i would suggest removing the section entirely.

It occurs to me that its a huge shame that etags are not URIs. If they 
weree, we would have a rerally neat way to find out the specific state 
of a particular publication - you woould subscribe to that etag. The 
next best thing might be to define a header field in the PUBLISH 
response that identifies this particular published instance. Hmm, 
Contact might even be a sensible choice there... worth discussing. 
I'll send a separate email on the topic.


* I suggest changing section 5.8 to a top level section, section 6, 
and have this be 'processing PUBLISH responses'. it doesnt belong as a 
subsection of 5, since 5 is about generating requests. Also, sectiono 
5.8 should point to 8.1.3 of RfC 3261 and say that those steps apply.

* section 6:

>  PUBLISH requests MUST be processed in the order that they are
>    received. 

isnt this only true for requuests with the same r-uri?

* section 6:

> A client may probe the ESC for the support of PUBLISH using the
>    OPTIONS request defined in SIP [2]. In the response to such an
>    OPTIONS request, the ESC SHOULD include "PUBLISH" to the list of
>    allowed methods in the Allow header field. Also, it SHOULD list the
>    supported event packages in an Allow-Events header field.
> 
>       The "methods" Contact header field parameter may also be used to
>       specifically announce support for PUBLISH messages when
>       registering. (See SIP Capabilities [11] for details on the
>       "methods" parameter).


This bit is out of place here, since the section is about publish 
processing. youo might want to make this section 7, else have section 
6 be "ESC behavior" and then have subsections for PUBLISH and OPTIONS 
processing.

Also, the indented note is not correct. Callee caps says that the 
Allow header should be used.

* sectiono 6:

The ESC inspects the Request-URI to determine whether this
        request is targeted to a resource for which the ESC is
        responsible for maintaining event state. If not, the ESC MUST
        return a 404 (Not Found) response and skip the remaining steps.

this sounds like the publish cant be proxied, which it can. I suppose 
in such a case, its not an ESC at all, and that ESC is particularly 
something that processes the request. This should be clarified I think.

* sectiono 6, steps 2,3,4 duplicate the behavipr specified in RFC 
3261, section 8.2. I suggest referring to that section instead in the 
beginning of section 6.

* the steps in section 6 omit handling for the case where there was no 
SIP-If-Match header field. step 6 says to check for it, and all the 
following steps assume it was there. I would suggest adding a bullet 
to step 6 which says, if there was no SIP-If-Match header field, the 
ESC creates a new etag, and associates it with as-yet empty event 
state. Processing continues in the steps below. This way, you always 
exit step 6 with an etag.

* step 8 of section 6:

>  The ESC processes the published event state, typically contained
>        in the body of the PUBLISH request. If the request contains no
>        body (when it should contain one), 

i think we need to list the conditions for when it should contain one. 
Indeed, since you are always allowed to publish without a body in 
order to refresh or delete, under what conditions is it illegal for 
the body to not be there?

* step 8:

>  *  If present, the ESC stores the event state delivered in the
>           PUBLISH request and identified by the associated entity-tag,
>           replacing any existing event state for that entity-tag.

as above - we need careful wording to allow partial publication

* step 9:

> The ESC returns a 200 (OK) response. The response MUST contain an
>        Expires header 

s/header/header field

* step 9:

> The response MUST also contain a SIP-ETag header field for
>        which the ESC MUST generate and store a locally unique entity-tag
>        for identifying the publication.

You should clarify that it generates a new one in each response, and 
does so for both initial and refreshes. Also, clarify that this etag 
replaces the previous, such that the previous etag is no longer valid 
for identifying the publication.

* step 9:

> After returning the 200 (OK)
>        response, the state agent associated with this ESC may then issue
>        appropriate NOTIFY requests to any watchers of this event state.

terminology confusioon here. Somoetimes, event state is usued to refer 
to the inputs of composition. Other times, the unique output 
associated with the aor. Here, it explicitly refers to the output 
associated with the aor, which may not even actually change 
dependening on comoposition policy.

* good work on section 7; very concise and clear on the differences 
with http.

* section 8:

> As already explained in Section 5.3, choosing the expiration
>       interval for a publication is ultimately the ESC's responsibility,
>       and choosing longer expiration values reduces the rate at which
>       publications are refreshed.


Careful - the ESC can't extend the expiration suggested by the client. 
It can reject one that it thinks is too short, but thats not the same, 
since the client could give up at that point. So, its not under eSC 
control completely.

* section 9 isnt syntax since it defines the headers as well as the 
syntax. I would instead suggest following the pattern in rfc3261, and 
have a section that defines any additional semantics for the headers, 
methods and response codes, but has no syntax. Then, have a separate 
section just for bnf.

* on evil table 2:

              | Allow               |   2xx   |    o    |
               | Allow               |    r    |    o    |

these can be folded together

* the trend in tables 4/5 is to include columns for each method that 
has reached rfc.

* sectiton 11.1:

> Authentication issues are discussed in SIP [2]. The exact methods for
>    creation and manipulation of the ESC authorization policies are
>    outside the scope of this document.

you will get dinged on this by Bellovin, guaranteed. He wants to see 
statements about baseline mandtory to implement mechahnisms. Its fine 
to say that this mechanism is digest, as mandated by rfc3261.

>  To prevent replay attacks, implementations SHOULD require
>    authentication with anti-replay protection. Authentication issues are
>    discussed in SIP [2].

as above.... does this mean digest with next-nonce or are you talking 
sips?

and section 11.4:

> To prevent such attacks, implementations SHOULD, at a minimum,
>    provide integrity protection across the To, From, Event,
>    SIP-If-Match, Route, and Expires headers and the bodies of PUBLISH
>    requests.

as above, he will ask for a mechanism. This requiurement makes the 
choice sips.

Same in 11.5.

* In the examples, I suggest not using net10. You shouold instead use 
something in 192.0.2.0/24.

* i think references 8 and 9 aree normative, since you are mandating a 
particular publication format (pidf) for usage with the presence event 
package.


Thanks,
Jonathan R.





Internet-Drafts@ietf.org wrote:

> 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		: Session Initiation Protocol (SIP) Extension for Event 
> 			  State Publication
> 	Author(s)	: A. Niemi
> 	Filename	: draft-ietf-sip-publish-02.txt
> 	Pages		: 39
> 	Date		: 2004-1-7
> 	
> This document describes an extension to the Session Initiation
> Protocol (SIP) for publishing event state used within the framework
> for SIP Event Notification. The first application of this extension
> is targeted at the publication of presence information.
> The mechanism described in this document can be extended to support
> publication of any event state, for which there exists an appropriate
> event package. It is not intended to be a general-purpose mechanism
> for transport of arbitrary data, as there are better-suited
> mechanisms for this purpose (FTP, HTTP, etc.)
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-02.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-sip-publish-02.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
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> 	
> NOTE:	The mail server at ietf.org can return the document in
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> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From exim@www1.ietf.org  Wed Jan 28 03:01:42 2004
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From: "Aarti Iyengar" <aartii@hotmail.com>
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Subject: [Sip] SIP-T and BICC
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All,
I am trying to understand SIP-T and BICC in detail.
>From my initial reading, both are essentially to seamlessly interface the 
PSTN to broadband networks. SIP-T is specifically for SIP based IP networks, 
whereas BICC is bearer independent and can interface with IP/ATM networks 
well. Apart from that, SIP-T deals with ISUP translation and encapsulation 
while BICC deals with extensions with ISUP for additional bearer support.
BICC CS3 will also enable SIP services to co-exist with BICC based networks.
My questions are -
Are these the only key differences between these protocols which essentially 
are intended perform the same function? If there are others, what are they?
Are there certain situations where one would fit in better than the other, 
and if so what are these?
How well would these interoperate with each other if they have to co-exist?

Also, any pointers to urls/whitepapers on the above (BICC versus SIP-T) 
would be appreciated.

thanks
Aarti

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From: Paul Drew <PD@metaswitch.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>, SIP WG <sip@ietf.org>
Subject: RE: [Sip] New ID related to voicemail and history requirements
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Cullen,

This draft seems to address the same requirements as
http://www.ietf.org/internet-drafts/draft-levy-sip-diversion-07.txt, which
specifies a Diversion header to contain the target and reason parameter.

Is there a preferred working group direction to address this issue?

In addition the Referred-By header specified in draft-ietf-sip-referredby-03
addresses a similar related issue.

Regards,

Paul

------------
Paul Drew
Product Manager
MetaSwitch

mailto: pd@metaswitch.com
http://www.metaswitch.com
Tel: +44 (0) 1244 305220  Fax +44 (0) 1244 312422
MetaSwitch is a division of Data Connection (DCL)


-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com]
Sent: 19 October 2003 10:18
To: sip@ietf.org
Cc: Cullen Jennings
Subject: [Sip] New ID related to voicemail and history requirements



I have submitted a new draft that deals with meeting a subset of the
"history" requirements for sending calls to voicemail systems. Until it
shows up in the archive, you can find it at:

http://www.employees.org/~fluffy/ietf/
       draft-jennings-sip-voicemail-uri-00.txt

Or 

http://www.employees.org/~fluffy/ietf/
       draft-jennings-sip-voicemail-uri-00.html

Cullen


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From: "Niemi Aki (Nokia-M/Helsinki)" <aki.niemi@nokia.com>
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To: ext Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [Sip] querying the current event state of a publication
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Hi,

Inline.

ext Jonathan Rosenberg wrote:
> In draft-ietf-sip-publish-02, section 5.7 proposes a way to query a 
> specific published event state. It recommends subscribing to the AOR to 
> learn it. However, it gives this caveat:
> 
>  Note that a subscription to the event package will likely deliver
>       results of the event composition process of the state agent, which
>       may be a subset or a superset of the current published event
>       state.
> 
> which is another way of saying, that subscribing to the aor doesnt 
> really allow you to query for a specific published event state.
> 
> If we want this functionality, I have an idea for an alternative 
> approach that works much better, triggered by some conversations I had 
> with Ben C. recently.
> 
> In the response to a PUBLISH request, the ESC returns a URI that 
> identifies that particular event state publication (effectively, a URI 
> that is bound to the AOR+event+etag of a publication). This URI is 
> basically a GRUU. If a user wishes to learn the state of this particular 
> event state, they merely subscribe to that URI. Indeed, a particularly 
> interestinig way to do this is to define sIP etags as URIs, rather than 
> reusing the definition from http. I think that having an etag as a URI 
> is far more elegant, as it would allow for things like this function in 
> a natural way. The alternative would be to possibly use the Contact 
> header field, as the meaning is the same as in a 2xx to INVITE - 
> identifying the specific *instance* to which you were connected when you 
> contacted the AOR. This makes sense if you think of the event state 
> publication URI as a GRUU.

But a PUA has to know the state it just published, simply because it was 
the source of that state in the first place. What it can't learn is the 
state of the other active PUAs.

And this is actually something currently missing from PUBLISH. A given 
PUA can't find out about publications other than thiose it itself has 
made. As a consequence, it can't for example, override the publications 
of a zombie PUA (a use case we have discussed in the past at length).

In order to enable a PUA to find out about the *other* publications to a 
given resource, the ESC would have to return not only a pointer to this 
PUAs publication, but to all active publications for that AoR. This is 
very similar to a registrar returning all active address bindings of an AoR.

> Indeed, one might even then PUBLISH to that URI, and it would have the 
> exact result you would expect - that particular event state is updated. 
> In such a case the server woould NOT return an etag or another URI in 
> the response.

When you add the feature I describe above, etags would still be needed 
since it would allow two PUAs to publish to the same instance of state. 
Etags would be needed for versioning in that case.

<snip />

> This would mean we wouldnt need the SIP-If-match.
> 
> Another alternative is to say that we are simply too far along in 
> PUBLISH, and the time has passed for major changes as this would be.

I do think we are quite far along in PUBLISH for adding this sort of 
functionality. However, I see some low-hanging fruit here, and to be 
honest, it has always bothered me a bit that there is no way for a PUA 
to know about the state other PUAs are publishing.

There is a requirement that calls for basically exactly this in 
draft-ietf-simple-publish-reqs-00 and it states:

    14.  PUAs must have a capability that allows them to query for the
         identifiers of all of the segments of presence information that
         have currently been published for a presentity (provided that
         the PUA is authorized to receive this information).

Using the register-like behavior for returning the set of active state 
in the 2xx response to PUBLISH + having this set be a set of GRUUs would 
fulfill the above. The PUA would be able to query the state (using 
SUBSCRIBE), but also override it (using PUBLISH). This is also a 
requireemnt in the reqs draft:

   15.  It must be possible for the publication of presence information
         for a particular segment to overwrite existing information for
         that segment, even if the existing information had been
         published by a different PUA. ...

So I would be willing to add this functionality simply because it would 
make the solution more complete. As far as adding complexity, sure, but 
an ESC whose policy doesn't allow cross PUA state could always just 
refuse to send the publication contacts in the PUBLISH response.

Does this sound reasonable?

Cheers,
Aki

P.S. I added an example message flow to illustrate the above ramblings...

---

PUA publishes its (initial) state:

       PUBLISH sip:presentity@example.com SIP/2.0
       To: <sip:presentity@example.com>
       From: <sip:presentity@example.com>;tag=54321mm
       Expires: 3600
       Event: presence
       ...

ESC returns with an etag for that state, plus all active publications in 
the form of URIs:

       SIP/2.0 200 OK
       To: <sip:presentity@example.com>;tag=effe22aa
       From: <sip:presentity@example.com>;tag=54321mm
       SIP-ETag: qwi982ks
       Expires: 3600
       Contact: sip:kaskjsdh@pa.example.com;etag=qwi982ks;expires=3600
       Contact: sip:jhsk23jj@pa.example.com;etag=hh3j4kll;expires=3229
       Contact: sip:hhfkjhjk@pa.example.com;etag=3ihy4hhk;expires=1200


> Thoughts?
> 
> -Jonathan R.
> 

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From exim@www1.ietf.org  Wed Jan 28 12:41:10 2004
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Date: Wed, 28 Jan 2004 09:37:16 -0800
Subject: Re: [Sip] New ID related to voicemail and history requirements
From: Cullen Jennings <fluffy@cisco.com>
To: Paul Drew <PD@metaswitch.com>, <sip@ietf.org>
Message-ID: <BC3D334C.2E9FD%fluffy@cisco.com>
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Well, it's always hard to speak for the whole WG :-)  The history work has
been adopted as a WG item - the others have not. One of the key things
holding history back is that some people thought an idea using URI was more
appropriate.  I carefully wrote down these URI idea as best I could in
draft-jennings-sip-voicemail-uri so it can be discussed and compared to
history. Then the WG can decide what to do. Options include History, both,
none. 

I could be completely wrong in my next statement but, it is my personal best
guess that the SIP WG is highly unlikely to make draft-levy-sip-diversion a
WG item for a standards track RFC. I think they are more likely to decide
that text encoding was a bad idea, deprecate forking, then deprecate INVITE
and replace with two methods called OFFER and ANSWER. Statements like "over
my dead body" don't bode well for a draft getting WG consensus.

We do need a solution to the set of problems these drafts address. I really
hope we choose one, anything, some time soon.

Cullen




On 1/28/04 3:11 AM, "Paul Drew" <PD@metaswitch.com> wrote:

> Cullen,
> 
> This draft seems to address the same requirements as
> http://www.ietf.org/internet-drafts/draft-levy-sip-diversion-07.txt, which
> specifies a Diversion header to contain the target and reason parameter.
> 
> Is there a preferred working group direction to address this issue?
> 
> In addition the Referred-By header specified in draft-ietf-sip-referredby-03
> addresses a similar related issue.
> 
> Regards,
> 
> Paul
> 
> ------------
> Paul Drew
> Product Manager
> MetaSwitch
> 
> mailto: pd@metaswitch.com
> http://www.metaswitch.com
> Tel: +44 (0) 1244 305220  Fax +44 (0) 1244 312422
> MetaSwitch is a division of Data Connection (DCL)
> 
> 
> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: 19 October 2003 10:18
> To: sip@ietf.org
> Cc: Cullen Jennings
> Subject: [Sip] New ID related to voicemail and history requirements
> 
> 
> 
> I have submitted a new draft that deals with meeting a subset of the
> "history" requirements for sending calls to voicemail systems. Until it
> shows up in the archive, you can find it at:
> 
> http://www.employees.org/~fluffy/ietf/
>      draft-jennings-sip-voicemail-uri-00.txt
> 
> Or 
> 
> http://www.employees.org/~fluffy/ietf/
>      draft-jennings-sip-voicemail-uri-00.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
> 


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Subject: RE: GRUU and UA instances, was: Re: [Sip] Subscribe dialog and target refresh request
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From: "Christian Jansson" <christian.jansson@hotsip.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Paul Kyzivat" <pkyzivat@cisco.com>
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Perhaps a little late, and hopefully not off topic. Comments inline.

> Jonathan R. wrote
> This, however, leads me into thinking a bit on the lifecycle
> management for GRUUs. Currently, GRUU is bound to a single=20
> registration, where the registration is defined by the Contact URI. If

> the UA should "move", in the sense that it acquires a new IP address,=20
> this will be a new Contact URI, and therefore, a new GRUU. It would be
> nice if, instead, the GRUU were truly bound to the UA instance, and=20
> that even if the UA instance changed IP addresses, the GRUU could=20
> remain unchanged. In such a case, if the UA moves, there is actually=20
> no need to issue any target refresh requests. The UA would simply=20
> re-register, update the binding of its GRUU to its new IP=20
> address, and=20
> existing calls would be correctly routed.

But if the ip actually changes, it would be necessary to send an updated
SDP with the new ip in the c=3D line. So your idea won't save you from
having to send a SIP request to the other end every time the ip changes
in an INVITE session. For a non-INVITE session though the idea would
save a lot of request from being sent if the session only contains
signaling.

/ Christian Jansson



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Hi all,
I am doing research on VoIP. 
I have not been able to find methods of doing conferencing in SIP in the RFC3261. 
As per my understanding it is not mentioned that a UA should only use one SIP URI in the request.
The question I have is,
                        
1) Is it possible for a UA to send INVITE request with multiple SIP URIs as in e-mail addresses separated with semi-columns.
2) If possible,what shall the proxy server do with that Invite request, request not valid or other response?
3) If not,then how a conference call is established and controlled ?

Any comments and answers to help would be appreciated,
thanks,

Davinder S Bains
Room: Y3-24, 
School of Computing and Information Technology
University of Western Sydney 
Locked Bag 1797 
Penrith South DC NSW 1797 
Australia 

Ph : 61 2 4736 0862 
Mob: 0425351960
Email : dbains@cit.uws.edu.au 


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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
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Date: Thu, 29 Jan 2004 09:02:04 +0200
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Davinder S Bains <dbains@cit.uws.edu.au>
CC: sip@ietf.org
Subject: Re: [Sip] Multiple URIs in a Request
References: <BasiliX-1.0.4b-1075339453401860bd82949@>
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You may want to read this draft:
http://www.ietf.org/internet-drafts/draft-camarillo-sipping-uri-list-00.txt

Gonzalo

Davinder S Bains wrote:
> Hi all,
> I am doing research on VoIP. 
> I have not been able to find methods of doing conferencing in SIP in the RFC3261. 
> As per my understanding it is not mentioned that a UA should only use one SIP URI in the request.
> The question I have is,
>                         
> 1) Is it possible for a UA to send INVITE request with multiple SIP URIs as in e-mail addresses separated with semi-columns.
> 2) If possible,what shall the proxy server do with that Invite request, request not valid or other response?
> 3) If not,then how a conference call is established and controlled ?
> 
> Any comments and answers to help would be appreciated,
> thanks,
> 
> Davinder S Bains


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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 exim@www1.ietf.org  Thu Jan 29 08:17:55 2004
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To: Davinder S Bains <dbains@cit.uws.edu.au>
CC: sip <sip@ietf.org>
Subject: Re: [Sip] Multiple URIs in a Request
References: <BasiliX-1.0.4b-1075339453401860bd82949@>
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There are several internet drafts by the sipping WG that you may find 
useful, especially:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-conferencing-framework-01.txt

You can find the rest here:
http://www.ietf.org/html.charters/sipping-charter.html

Regards,
Steve


Davinder S Bains wrote:

>Hi all,
>I am doing research on VoIP. 
>I have not been able to find methods of doing conferencing in SIP in the RFC3261. 
>As per my understanding it is not mentioned that a UA should only use one SIP URI in the request.
>The question I have is,
>                        
>1) Is it possible for a UA to send INVITE request with multiple SIP URIs as in e-mail addresses separated with semi-columns.
>2) If possible,what shall the proxy server do with that Invite request, request not valid or other response?
>3) If not,then how a conference call is established and controlled ?
>
>Any comments and answers to help would be appreciated,
>thanks,
>
>Davinder S Bains
>Room: Y3-24, 
>School of Computing and Information Technology
>University of Western Sydney 
>Locked Bag 1797 
>Penrith South DC NSW 1797 
>Australia 
>
>Ph : 61 2 4736 0862 
>Mob: 0425351960
>Email : dbains@cit.uws.edu.au 
>
>
>Scanned by SCIT E-Mail Gateway http://www.cit.uws.edu.au
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>
>  
>


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



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Subject: RE: [Sip] SIP-T and BICC
Date: Fri, 30 Jan 2004 10:51:37 +0530
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Thread-Topic: [Sip] SIP-T and BICC
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Hi
    You can refer RFC 3372 for SIP-T Context and Architecture.=20

Ranjit
Ph: 8520408/16 extn: 2173




-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Aarti
Iyengar
Sent: Wednesday, January 28, 2004 1:28 PM
To: sip@ietf.org
Subject: [Sip] SIP-T and BICC


All,
I am trying to understand SIP-T and BICC in detail.
>From my initial reading, both are essentially to seamlessly interface
the=20
PSTN to broadband networks. SIP-T is specifically for SIP based IP
networks,=20
whereas BICC is bearer independent and can interface with IP/ATM
networks=20
well. Apart from that, SIP-T deals with ISUP translation and
encapsulation=20
while BICC deals with extensions with ISUP for additional bearer
support. BICC CS3 will also enable SIP services to co-exist with BICC
based networks. My questions are - Are these the only key differences
between these protocols which essentially=20
are intended perform the same function? If there are others, what are
they? Are there certain situations where one would fit in better than
the other,=20
and if so what are these?
How well would these interoperate with each other if they have to
co-exist?

Also, any pointers to urls/whitepapers on the above (BICC versus SIP-T)=20
would be appreciated.

thanks
Aarti

_________________________________________________________________
Add photos to your e-mail with MSN 8. Get 2 months FREE*.=20
http://join.msn.com/?page=3Dfeatures/featuredemail


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/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: Aarti Iyengar <aartii@hotmail.com>
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Subject: Re: [Sip] SIP-T and BICC
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The BICC work was originally started to extend ISUP capabilities to ATM networks, 
and this was what BICC CS1 achieved.  The work took on a life of its own, and BICC 
CS2 supports voice over IP networks.  I'm not sure BICC CS3 will ever happen -- it 
seems to have bogged down.  The specific advantage of BICC is that it can take 
advantage of already-installed SS7 infrastructure.

One can use SIP to make connections across ATM networks, using the attributes and 
conventions defined in RFC 3108.  Unfortunately some of the syntax in that 
document (for the m= line) does not conform to the latest SDP syntax as defined in 
draft-ietf-mmusic-sdp-new-15.txt.

The key point of SIP-T is to provide a transition mechanism from a mixed 
legacy/SIP network to a pure SIP network.  If you install a BICC network, you have 
to install a parallel SIP network to handle multimedia and new services.  You face 
a problem when you eventually want to move voice services onto the SIP network to 
achieve savings through network and service integration.

So the brief answer is that the two protocols overlap in functionality, but which 
to use is an interesting strategic decision depending on what you have in your 
network already and what services you plan to offer in the future.

Aarti Iyengar wrote:

> All,
> I am trying to understand SIP-T and BICC in detail.
> 
>> From my initial reading, both are essentially to seamlessly interface the 
> 
> PSTN to broadband networks. SIP-T is specifically for SIP based IP 
> networks, whereas BICC is bearer independent and can interface with 
> IP/ATM networks well. Apart from that, SIP-T deals with ISUP translation 
> and encapsulation while BICC deals with extensions with ISUP for 
> additional bearer support.
> BICC CS3 will also enable SIP services to co-exist with BICC based 
> networks.
> My questions are -
> Are these the only key differences between these protocols which 
> essentially are intended perform the same function? If there are others, 
> what are they?
> Are there certain situations where one would fit in better than the 
> other, and if so what are these?
> How well would these interoperate with each other if they have to co-exist?
> 
> Also, any pointers to urls/whitepapers on the above (BICC versus SIP-T) 
> would be appreciated.
> 
> thanks
> Aarti
> 
> _________________________________________________________________
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> 


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From exim@www1.ietf.org  Fri Jan 30 11:29:38 2004
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: "'brett@broadsoft.com'" <brett@broadsoft.com>
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Subject: [Sip] RE: [Sip-implementors] Via: processing
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Brett, Arlie,

RFC3261 clearly allows a response to be "rerouted" if it can be determined
that the response did not reach the first-attempt target.  For example, if
the request is received over TCP, and the connection has gone and cannot be
reestablished by the time the response is ready to be sent back, RFC3261
states the following

--- from RFC3261, section 18.2.2, 1st bullet.

If that connection attempt fails, the server SHOULD use the procedures in
[4] for servers in order to determine the IP address and port to open the
connection and send the response to.

--- end of cut

RFC3263 then gives an algorithm for attempting to determine alternate IP
addresses to which the response might be sent.  Clearly having an FQDN in
the sent-by will give a chance of success which having an IP address will
not, especially if the sent-by and received IP addresses are the same!

Paul DS.

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: Brett Tate [mailto:brett@broadsoft.com]
Sent: 29 January 2004 00:27
To: SIP (E-mail)
Subject: RE: [Sip-implementors] Via: processing


> I'm looking at the spec for Via: processing.  
> RFC 3261 clearly shows that a Via: line can 
> contain DNS names, as well as textual
> representations of network addresses.
> 
> How many products actually implement support 
> for resolving DNS names provided in Via: lines?  

Not very many support DNS resolution of a
Via entry.  However many support the concept
of adding the Via's received parameter when
it does not match the Via's sent-by address.

> This seems...  well, it seems unnecessary,
> potentially insecure (allowing for easy DDoS 
> and potentially amplification attacks), and 
> just a general nuisance.  Is this kind of
> thing ever actually used for a good purpose?  

Some people have toyed with the idea of
advancing response addresses based upon
rfc3263 during failure situations.

> Does any product do anything *but* just put 
> real network addresses in Via: lines in
> requests?

Yes.

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