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Subject: Re: [Sip] 3840 Device ID -  SIP Capababilties
From: Cullen Jennings <fluffy@cisco.com>
To: <ksrini@motorola.com>, "sip@ietf.org" <sip@ietf.org>
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On 12/28/04 1:40 PM, "Srini Krishnamoorthy" <ksrini@motorola.com> wrote:

> 
> Dear All,
> 
>  Some time last year there was a discussion about Device IDs and
>  SIP capabilities - what was the consensus on that?
> 
>  Was a instance-id or a device-id included? Or is it addressed
>  in some other piece of work?  I couldn't find any text in
>  RFC3840 (SIP Cababilities) on this subject
> 
>  Can any one help?
> 
>  Here is link to that the original discussion:
>    http://www1.ietf.org/mail-archive/web/sip/current/msg08369.html
> 
> 
> Thanks very much

The http://www.ietf.org/internet-drafts/draft-ietf-sip-gruu-02.txt draft
defines +sip.instance in the contact headers. I think everyone agrees this
is the way to deal with this issue.



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From sip-bounces@ietf.org  Wed Jan  5 00:20:33 2005
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Thread-Topic: a question on tags when INVITE is challenged
Thread-Index: AcTy5XsAi9bzp435TOqNKDTkTCH4kg==
From: "Wang, Jin-Gen" <Jin-Gen.Wang@Level3.com>
To: <sip@ietf.org>
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Subject: [Sip] a question on tags when INVITE is challenged
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We are running into an inter-op problem in SIP. It involves the =
interpretation of RFC 3261. Hopefully we can get some input from the =
list. Here is the call flow:

UAC                                UAS
  |     ----- INVITE (F1) ------>  |
  |     <--- 401 (F2) ------------   |
  |     ------ ACK (F3) ---------> |
  |     ------ INVITE (F4) ------> |
  |     ...............................  |
  |    <----- 200 (F5) ----------   |
  |    ------- ACK (F6) -------->  |

where INVITE (F1) has no credential and INVITE (F4) has credential.=20

The vendor 1 believes INVITE (F1) and INVITE (F4) should have same From =
tag and 401 (F2) and 200 (F5) should have same To tag. The following =
quote from section 22.2 in RFC 3261 indicates F4 is an updated request.

"When a UAC resubmits a request with its credentials after receiving a
   401 (Unauthorized) or 407 (Proxy Authentication Required) response,
   it MUST increment the CSeq header field value as it would normally
   when sending an updated request."

It implicitly indicates that INVITE (F1) and INVITE (F4) are within the =
same dialog for a same call even though the dialog is created after =
INVITE (F4).

The vendor 2 believes INVITE (F1) and INVITE (F4) should have different =
>From tag and/or 401 (F2) and 200 (F5) should have different To tags =
because of the quote from section 19.3 in RFC 3261:

"When a tag is generated by a UA for insertion into a request or
   response, it MUST be globally unique and cryptographically random
   with at least 32 bits of randomness.  A property of this selection
   requirement is that a UA will place a different tag into the From
   header of an INVITE than it would place into the To header of the
   response to the same INVITE.  This is needed in order for a UA to
   invite itself to a session, a common case for "hairpinning" of calls
   in PSTN gateways.  Similarly, two INVITEs for different calls will
   have different From tags, and two responses for different calls will
   have different To tags."

Which vendor interprets the tag requirements correctly?

Thanks.

Jin-Gen Wang
Level3 Communications Inc.
1025 Eldorado Blvd.
Broomfield, CO 80021


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


From sip-bounces@ietf.org  Wed Jan  5 01:10:32 2005
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From: "Nataraju A B" <bnataraju@kodiaknetworks.com>
To: "'Wang, Jin-Gen'" <Jin-Gen.Wang@Level3.com>, <sip@ietf.org>
Subject: RE: [Sip] a question on tags when INVITE is challenged
Date: Wed, 5 Jan 2005 11:18:54 +0530
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Comments inline...

Best Regards,
Nataraju A.B.
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Wang, Jin-Gen
Sent: Wednesday, January 05, 2005 10:45 AM
To: sip@ietf.org
Subject: [Sip] a question on tags when INVITE is challenged

We are running into an inter-op problem in SIP. It involves the
interpretation of RFC 3261. Hopefully we can get some input from the
list. Here is the call flow:

UAC                                UAS
  |     ----- INVITE (F1) ------>  |
  |     <--- 401 (F2) ------------   |
  |     ------ ACK (F3) ---------> |
  |     ------ INVITE (F4) ------> |
  |     ...............................  |
  |    <----- 200 (F5) ----------   |
  |    ------- ACK (F6) -------->  |

where INVITE (F1) has no credential and INVITE (F4) has credential. 

The vendor 1 believes INVITE (F1) and INVITE (F4) should have same From
tag and 401 (F2) and 200 (F5) should have same To tag. The following
quote from section 22.2 in RFC 3261 indicates F4 is an updated request.

"When a UAC resubmits a request with its credentials after receiving a
   401 (Unauthorized) or 407 (Proxy Authentication Required) response,
   it MUST increment the CSeq header field value as it would normally
   when sending an updated request."

It implicitly indicates that INVITE (F1) and INVITE (F4) are within the
same dialog for a same call even though the dialog is created after
INVITE (F4).

[ABN] this might be confusing text... may needs to be modified...

The vendor 2 believes INVITE (F1) and INVITE (F4) should have different
>From tag and/or 401 (F2) and 200 (F5) should have different To tags
because of the quote from section 19.3 in RFC 3261:

"When a tag is generated by a UA for insertion into a request or
   response, it MUST be globally unique and cryptographically random
   with at least 32 bits of randomness.  A property of this selection
   requirement is that a UA will place a different tag into the From
   header of an INVITE than it would place into the To header of the
   response to the same INVITE.  This is needed in order for a UA to
   invite itself to a session, a common case for "hairpinning" of calls
   in PSTN gateways.  Similarly, two INVITEs for different calls will
   have different From tags, and two responses for different calls will
   have different To tags."

Which vendor interprets the tag requirements correctly?

[ABN] I feel the vendor 2 interprets correctly...
Once the failure response received for an INVITE request and that
transaction is closed the session will also be deleted since there are
no other dialog exists as part of the same session. Hence the
application (no the user) has to initiate a new session with new from
tag to send the INVITE (F4) with challenge response...

Thanks.

Jin-Gen Wang
Level3 Communications Inc.
1025 Eldorado Blvd.
Broomfield, CO 80021


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


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


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HI,All Friends,
    We take the liberty of writing to you with a view to building bussine=
ss relations with your firm.     We are a manufacturer of IP PHONE,ATA at=
 china, Our ip phone=A3=ACATA --YWH100,YWH10 and YGW10 series are well kn=
own for its first-quality,good voice ,high stability ,low cost.=20
Now we have CE and FCC ,at the same time we own export liscense!=20
If you want to know more informations in details, please contact with us =
or visit our website:=20
http://www.yntx.com
=20
=20
Wishing you the best of business.
Carol
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D
Contact person:Carol=20
MSN:Carol_chang8009@hotmail.com=20
Email:ync@vip.163.com=20
YahooID:Carol_chang8009=20
ICQ:200931135=20
Tel:8603717657239=20
http://www.yntx.com
Zhengzhou Yuneng Communication Equipment Co.Ltd.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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





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BORDER-LEFT: #1010ff 2px solid">
<DIV>HI,All Friends,</DIV>
<DIV>&nbsp;&nbsp;&nbsp; We take the liberty of writing to you with a view=
 to building bussiness relations with your firm.=20
<DIV><FONT size=3D2><FONT size=3D2>&nbsp;&nbsp; &nbsp;We are&nbsp;a manuf=
acturer of IP PHONE,ATA at china, Our ip phone=A3=ACATA --YWH100,YWH10 an=
d YGW10 series are well known for its first-quality,good voice ,high stab=
ility ,low cost. <BR>Now we have CE  and FCC ,at the same time we own exp=
ort liscense! <BR>If you want to know more informations in details, pleas=
e contact with&nbsp;us or visit our website: <BR><A href=3D"http://www.yn=
tx.com/">http://www.yntx.com</A></FONT></DIV>
<DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>Wishing you the best of business.<BR>Carol<BR>=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D<BR>Contact person:Carol <BR>MSN:Carol_chang8009@hotmail.com <BR>Em=
ail:ync@vip.163.com <BR>YahooID:Carol_chang8009 <BR>ICQ:200931135 <BR>Tel=
:8603717657239 <BR><A href=3D"http://www.yntx.com/">http://www.yntx.com</=
A><BR>Zhengzhou Yuneng Communication Equipment Co.Ltd.<BR>=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D </FONT></DIV></FONT></DIV></DIV>
<P><BR>
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HI,All Friends,
     We take the liberty of writing to you with a view to building bussin=
ess relations with your firm.      We are a manufacturer of IP PHONE,ATA =
at china, Our ip phone=A3=ACATA --YWH100,YWH10 and YGW10 series are well =
known for its first-quality,good voice ,high stability ,low cost.=20
Now we have CE and FCC ,at the same time we own export liscense!=20
If you want to know more informations in details, please contact with us =
or visit our website:=20
http://www.yntx.com
=20
=20
Wishing you the best of business.
Carol
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D
Contact person:Carol=20
MSN:Carol_chang8009@hotmail.com=20
Email:ync@vip.163.com=20
YahooID:Carol_chang8009=20
ICQ:200931135=20
Tel:8603717657239=20
http://www.yntx.com
Zhengzhou Yuneng Communication Equipment Co.Ltd.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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






---------------------------------
Do You Yahoo!?
=D7=A2=B2=E1=CA=C0=BD=E7=D2=BB=C1=F7=C6=B7=D6=CA=B5=C4=D1=C5=BB=A2=C3=E2=B7=
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<DIV>
<DIV>HI,All Friends,</DIV>
<DIV>&nbsp;&nbsp;&nbsp;&nbsp; We take the liberty of writing to you with =
a view to building bussiness relations with your firm.=20
<DIV><FONT size=3D2><FONT size=3D2>&nbsp;&nbsp;&nbsp; &nbsp;We are&nbsp;a=
 manufacturer of IP PHONE,ATA at china, Our ip phone=A3=ACATA --YWH100,YW=
H10 and YGW10 series are well known for its first-quality,good voice ,hig=
h stability ,low cost. <BR>Now we have CE  and FCC ,at the same time we o=
wn export liscense! <BR>If you want to know more informations in details,=
 please contact with&nbsp;us or visit our website: <BR><A href=3D"http://=
www.yntx.com/" target=3D_blank><FONT color=3D#003399>http://www.yntx.com<=
/FONT></A></FONT></DIV>
<DIV>
<DIV><FONT color=3D#003399></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#003399></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Wishing you the best of business.<BR>Carol<BR>=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D<BR>Contact person:Carol <BR>MSN:Carol_chang8009@hotmail.com <BR>Em=
ail:ync@vip.163.com <BR>YahooID:Carol_chang8009 <BR>ICQ:200931135 <BR>Tel=
:8603717657239 <BR><A href=3D"http://www.yntx.com/" target=3D_blank><FONT=
 color=3D#003399>http://www.yntx.com</FONT></A><BR>Zhengzhou Yuneng Commu=
nication Equipment Co.Ltd.<BR>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D </FONT></DIV=
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_______________________________________________
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Date: Wed, 5 Jan 2005 07:20:20 -0800 (PST)
From: Vijay Gaur <vgaur_tech@yahoo.com>
Subject: RE: [Sip] a question on tags when INVITE is challenged
To: Nataraju A B <bnataraju@kodiaknetworks.com>,
        "'Wang, Jin-Gen'" <Jin-Gen.Wang@Level3.com>, sip@ietf.org
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Hi Wang,
  Vendor1's SIP implementation should not expect same To-tag which is present in 401. There is no dialog established by 401 response. New INVITE with credentials is totaly different request. In fact few SIP implementations might send the same From tag of original INVITE. 
 
You can see similar call flows in the draft below:
http://www.iptel.org/info/players/ietf/callflows/draft-ietf-sipping-basic-call-flows-02.txt
 
Thanks,
Vijay Gaur

Nataraju A B <bnataraju@kodiaknetworks.com> wrote:
Comments inline...

Best Regards,
Nataraju A.B.
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Wang, Jin-Gen
Sent: Wednesday, January 05, 2005 10:45 AM
To: sip@ietf.org
Subject: [Sip] a question on tags when INVITE is challenged

We are running into an inter-op problem in SIP. It involves the
interpretation of RFC 3261. Hopefully we can get some input from the
list. Here is the call flow:

UAC UAS
| ----- INVITE (F1) ------> |
| <--- 401 (F2) ------------ |
| ------ ACK (F3) ---------> |
| ------ INVITE (F4) ------> |
| ............................... |
| <----- 200 (F5) ---------- |
| ------- ACK (F6) --------> |

where INVITE (F1) has no credential and INVITE (F4) has credential. 

The vendor 1 believes INVITE (F1) and INVITE (F4) should have same From
tag and 401 (F2) and 200 (F5) should have same To tag. The following
quote from section 22.2 in RFC 3261 indicates F4 is an updated request.

"When a UAC resubmits a request with its credentials after receiving a
401 (Unauthorized) or 407 (Proxy Authentication Required) response,
it MUST increment the CSeq header field value as it would normally
when sending an updated request."

It implicitly indicates that INVITE (F1) and INVITE (F4) are within the
same dialog for a same call even though the dialog is created after
INVITE (F4).

[ABN] this might be confusing text... may needs to be modified...

The vendor 2 believes INVITE (F1) and INVITE (F4) should have different
>From tag and/or 401 (F2) and 200 (F5) should have different To tags
because of the quote from section 19.3 in RFC 3261:

"When a tag is generated by a UA for insertion into a request or
response, it MUST be globally unique and cryptographically random
with at least 32 bits of randomness. A property of this selection
requirement is that a UA will place a different tag into the From
header of an INVITE than it would place into the To header of the
response to the same INVITE. This is needed in order for a UA to
invite itself to a session, a common case for "hairpinning" of calls
in PSTN gateways. Similarly, two INVITEs for different calls will
have different From tags, and two responses for different calls will
have different To tags."

Which vendor interprets the tag requirements correctly?

[ABN] I feel the vendor 2 interprets correctly...
Once the failure response received for an INVITE request and that
transaction is closed the session will also be deleted since there are
no other dialog exists as part of the same session. Hence the
application (no the user) has to initiate a new session with new from
tag to send the INVITE (F4) with challenge response...

Thanks.

Jin-Gen Wang
Level3 Communications Inc.
1025 Eldorado Blvd.
Broomfield, CO 80021


_______________________________________________
Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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


Vijay Gaur
It's how you deal with failure that determines how you achieve success.
Au revoir, Adios, Namaste
		
---------------------------------
Do you Yahoo!?
 Take Yahoo! Mail with you! Get it on your mobile phone.
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<DIV>Hi Wang,</DIV>
<DIV>&nbsp; Vendor1's SIP implementation should not expect same To-tag which is present in 401. There is no dialog established by 401 response. New INVITE with credentials is totaly different request. In fact few SIP implementations might send the same From tag of original INVITE. </DIV>
<DIV>&nbsp;</DIV>
<DIV>You can see similar call flows in the draft below:</DIV>
<DIV><A href="http://www.iptel.org/info/players/ietf/callflows/draft-ietf-sipping-basic-call-flows-02.txt">http://www.iptel.org/info/players/ietf/callflows/draft-ietf-sipping-basic-call-flows-02.txt</A></DIV>
<DIV>&nbsp;</DIV>
<DIV>Thanks,</DIV>
<DIV>Vijay Gaur<BR><BR><B><I>Nataraju A B &lt;bnataraju@kodiaknetworks.com&gt;</I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Comments inline...<BR><BR>Best Regards,<BR>Nataraju A.B.<BR>-----Original Message-----<BR>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of<BR>Wang, Jin-Gen<BR>Sent: Wednesday, January 05, 2005 10:45 AM<BR>To: sip@ietf.org<BR>Subject: [Sip] a question on tags when INVITE is challenged<BR><BR>We are running into an inter-op problem in SIP. It involves the<BR>interpretation of RFC 3261. Hopefully we can get some input from the<BR>list. Here is the call flow:<BR><BR>UAC UAS<BR>| ----- INVITE (F1) ------&gt; |<BR>| &lt;--- 401 (F2) ------------ |<BR>| ------ ACK (F3) ---------&gt; |<BR>| ------ INVITE (F4) ------&gt; |<BR>| ............................... |<BR>| &lt;----- 200 (F5) ---------- |<BR>| ------- ACK (F6) --------&gt; |<BR><BR>where INVITE (F1) has no credential and INVITE (F4) has credential. <BR><BR>The vendor 1 believes INVITE (F1) and INVITE (F4)!
  should
 have same From<BR>tag and 401 (F2) and 200 (F5) should have same To tag. The following<BR>quote from section 22.2 in RFC 3261 indicates F4 is an updated request.<BR><BR>"When a UAC resubmits a request with its credentials after receiving a<BR>401 (Unauthorized) or 407 (Proxy Authentication Required) response,<BR>it MUST increment the CSeq header field value as it would normally<BR>when sending an updated request."<BR><BR>It implicitly indicates that INVITE (F1) and INVITE (F4) are within the<BR>same dialog for a same call even though the dialog is created after<BR>INVITE (F4).<BR><BR>[ABN] this might be confusing text... may needs to be modified...<BR><BR>The vendor 2 believes INVITE (F1) and INVITE (F4) should have different<BR>&gt;From tag and/or 401 (F2) and 200 (F5) should have different To tags<BR>because of the quote from section 19.3 in RFC 3261:<BR><BR>"When a tag is generated by a UA for insertion into a request or<BR>response, it MUST be globally unique and
 cryptographically random<BR>with at least 32 bits of randomness. A property of this selection<BR>requirement is that a UA will place a different tag into the From<BR>header of an INVITE than it would place into the To header of the<BR>response to the same INVITE. This is needed in order for a UA to<BR>invite itself to a session, a common case for "hairpinning" of calls<BR>in PSTN gateways. Similarly, two INVITEs for different calls will<BR>have different From tags, and two responses for different calls will<BR>have different To tags."<BR><BR>Which vendor interprets the tag requirements correctly?<BR><BR>[ABN] I feel the vendor 2 interprets correctly...<BR>Once the failure response received for an INVITE request and that<BR>transaction is closed the session will also be deleted since there are<BR>no other dialog exists as part of the same session. Hence the<BR>application (no the user) has to initiate a new session with new from<BR>tag to send the INVITE (F4) with challenge
 response...<BR><BR>Thanks.<BR><BR>Jin-Gen Wang<BR>Level3 Communications Inc.<BR>1025 Eldorado Blvd.<BR>Broomfield, CO 80021<BR><BR><BR>_______________________________________________<BR>Sip mailing list https://www1.ietf.org/mailman/listinfo/sip<BR>This list is for NEW development of the core SIP Protocol<BR>Use sip-implementors@cs.columbia.edu for questions on current sip<BR>Use sipping@ietf.org for new developments on the application of sip<BR><BR><BR>_______________________________________________<BR>Sip mailing list https://www1.ietf.org/mailman/listinfo/sip<BR>This list is for NEW development of the core SIP Protocol<BR>Use sip-implementors@cs.columbia.edu for questions on current sip<BR>Use sipping@ietf.org for new developments on the application of sip<BR></BLOCKQUOTE><BR><BR>Vijay Gaur<br>It's how you deal with failure that determines how you achieve success.<br>Au revoir, Adios, Namaste<p>
		<hr size=1>Do you Yahoo!?<br> 
<a href="http://us.rd.yahoo.com/mail_us/taglines/mobile/*http://mobile.yahoo.com/maildemo">Take Yahoo! Mail with you!</a> Get it on your mobile phone.
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From: "Brett Tate" <brett@broadsoft.com>
To: "'Shan Lu'" <shanlu@sentito.com>
Subject: RE: [Sip] RFC 3261: 32 bit CSeq number
Date: Wed, 5 Jan 2005 12:18:23 -0500
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> But Section 8.1.1.5 does apply to in-dialog=20
> request -- for the side that acted as UAS=20
> in dialog creation.=20
>=20
> 12.2.1.1/RFC2161
>=20
> "If the local sequence number is empty,=20
> an initial value MUST be chosen using the=20
> guidelines of Section 8.1.1.5."

I agree.  It was discussed some within the following email if you would =
like
to provide comments.

http://www1.ietf.org/mail-archive/web/sip/current/msg10905.html


> Did I miss something?

The original posting was more associated with the validity of a cseq =
value
between 2**31 and (2**32)-1. =20

Within the above link, it evolved into the potential issues associated =
with
a UAS choosing to return a 400 response when it thinks that the UAC =
provided
a cseq value between 2**31 and (2**32)-1 when it should not have done =
so.



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

	Title		: Suppression of REFER Implicit Subscription
	Author(s)	: O. Levin
	Filename	: draft-ietf-sip-refer-with-norefersub-00.txt
	Pages		: 13
	Date		: 2005-1-5
	
This specification defines the way to suppress an implicit
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Could someone help me interpret RFC 3261 related to the the case when 
the Content-Disposition header is missing from an Invite. 
 
The verbage below clearly specifies that if the handling parameter is missing 
from a Content disposition header, then it should be assumed that the Disposition 
is required.  Extra-polating that, should a missing Content-Disposition header be 
treated as a required Disposition. 
 
Here is the snippet from RFC 3261, section 20.11 about Content Disposition.
 
Several new "disposition-types" of the Content-Disposition header are

defined by SIP. The value "session" indicates that the body part

describes a session, for either calls or early (pre-call) media. The

value "render" indicates that the body part should be displayed or

otherwise rendered to the user. Note that the value "render" is used

rather than "inline" to avoid the connotation that the MIME body is

displayed as a part of the rendering of the entire message (since the

MIME bodies of SIP messages oftentimes are not displayed to users).

For backward-compatibility, if the Content-Disposition header field

is missing, the server SHOULD assume bodies of Content-Type

application/sdp are the disposition "session", while other content

types are "render".


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<DIV><FONT color=#800080><SPAN class=459554513-06012005><FONT size=2><FONT 
face="Century Gothic"><FONT color=#000000><FONT face=Arial>Could someone help me 
interpret RFC 3261 related to the the case when 
</FONT></FONT></FONT></FONT></SPAN></FONT></DIV>
<DIV><FONT><SPAN class=459554513-06012005>the Content-Disposition header is 
missing from an Invite. </SPAN></FONT></DIV>
<DIV><FONT><SPAN class=459554513-06012005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT><SPAN class=459554513-06012005>The verbage below clearly specifies 
that if the handling parameter </SPAN></FONT><FONT><SPAN 
class=459554513-06012005>is missing </SPAN></FONT></DIV>
<DIV><FONT><SPAN class=459554513-06012005>from a Content disposition header, 
then it should be assumed that </SPAN></FONT><FONT><SPAN 
class=459554513-06012005>the Disposition </SPAN></FONT></DIV>
<DIV><FONT><SPAN class=459554513-06012005>is required.&nbsp; Extra-polating 
that, should a&nbsp;missing </SPAN></FONT><FONT><SPAN 
class=459554513-06012005>Content-Disposition header be </SPAN></FONT></DIV>
<DIV><FONT><SPAN class=459554513-06012005>treated as a required 
Disposition.<FONT size=+0><SPAN 
class=459554513-06012005>&nbsp;</SPAN></FONT></SPAN></FONT></DIV>
<DIV><FONT><SPAN class=459554513-06012005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT><SPAN class=459554513-06012005>Here is the snippet from RFC 3261, 
section 20.11 about Content Disposition.</SPAN></FONT></DIV>
<DIV><FONT><SPAN class=459554513-06012005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT><SPAN class=459554513-06012005><FONT face=Courier size=2>
<P align=left>Several new "disposition-types" of the Content-Disposition header 
are</P>
<P align=left>defined by SIP. The value "session" indicates that the body 
part</P>
<P align=left>describes a session, for either calls or early (pre-call) media. 
The</P>
<P align=left>value "render" indicates that the body part should be displayed 
or</P>
<P align=left>otherwise rendered to the user. Note that the value "render" is 
used</P>
<P align=left>rather than "inline" to avoid the connotation that the MIME body 
is</P>
<P align=left>displayed as a part of the rendering of the entire message (since 
the</P>
<P align=left>MIME bodies of SIP messages oftentimes are not displayed to 
users).</P>
<P align=left>For backward-compatibility, if the Content-Disposition header 
field</P>
<P align=left>is missing, the server SHOULD assume bodies of Content-Type</P>
<P align=left>application/sdp are the disposition "session", while other 
content</P>
<P align=left>types are "render".</P></FONT></SPAN></FONT><FONT size=2><FONT 
face="Century Gothic"></DIV></FONT></FONT></BODY></HTML>

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Subject: [Sip] Refer-To header and Replaces option
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Dear All,
 
Is the usage of Replaces parameter in a Refer-To header a standard usage. If
it it could you please point me to the RFC that specifies the usage.
 
Here is an example.
 
Refer-To: <sip:650916143674820@10.254.1.20>;transport=UDP?Require=replaces&Replaces=b7f4eb08d5674c5fa019ec2975173a4b4010.254.1.20

 
Thanks,
Nagesh Shekar

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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE id=3DridTitle>Blank</TITLE>

<STYLE>BODY {
	MARGIN-TOP: 25px; FONT-SIZE: 10pt; MARGIN-LEFT: 25px; COLOR: #000000; =
FONT-FAMILY: Arial, Helvetica
}
P.msoNormal {
	MARGIN-TOP: 0px; FONT-SIZE: 10pt; MARGIN-LEFT: 0px; COLOR: #ffffcc; =
FONT-FAMILY: Helvetica, "Times New Roman"
}
LI.msoNormal {
	MARGIN-TOP: 0px; FONT-SIZE: 10pt; MARGIN-LEFT: 0px; COLOR: #ffffcc; =
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<BODY id=3DridBody background=3Dcid:995245713@06012005-37a8>
<DIV><SPAN class=3D995245713-06012005>Dear All,</SPAN></DIV>
<DIV><SPAN class=3D995245713-06012005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D995245713-06012005>Is the usage of Replaces =
parameter in a=20
Refer-To header a standard usage. If</SPAN></DIV>
<DIV><SPAN class=3D995245713-06012005>it it could you please point me =
to the RFC=20
that specifies the usage.</SPAN></DIV>
<DIV><SPAN class=3D995245713-06012005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D995245713-06012005>Here is an example.</SPAN></DIV>
<DIV><SPAN class=3D995245713-06012005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D995245713-06012005>
<P>Refer-To:<SPAN class=3D995245713-06012005>=20
</SPAN>&lt;sip:650916143674820@10.254.1.20&gt;;transport=3DUDP?Require=3D=
replaces&amp;Replaces=3Db7f4eb08d5674c5fa019ec2975173a4b4010.254.1.20</P=
></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV align=3Dleft>
<DIV align=3Dleft><FONT face=3D"Century Gothic" =
color=3D#800080><STRONG><SPAN=20
class=3D995245713-06012005>Thanks,</SPAN></STRONG></FONT></DIV>
<DIV align=3Dleft><FONT face=3D"Century Gothic" =
color=3D#800080><STRONG><SPAN=20
class=3D995245713-06012005>Nagesh=20
Shekar</SPAN></STRONG></FONT></DIV></DIV></BODY></HTML>

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_______________________________________________
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From sip-bounces@ietf.org  Thu Jan  6 11:52:48 2005
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From: Eric Burger <eburger@brooktrout.com>
To: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
Subject: RE: [Sip] Interpreting Missing Content-Disposition
Date: Thu, 6 Jan 2005 11:21:15 -0500 
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No.  The language is clear:
"... if the Content-Disposition header field
is missing, the server SHOULD assume bodies of Content-Type
application/sdp are the disposition "session", while other content
types are "render"."

Answer: Yes, unless you have a good reason not to.

-----Original Message-----
From: Shekar, Nagesh Soma (Shekar)** CTR ** [mailto:shek@lucent.com]
Sent: Thursday, January 06, 2005 8:56 AM
To: 'sip-ietf'
Subject: [Sip] Interpreting Missing Content-Disposition


Could someone help me interpret RFC 3261 related to the the case when 
the Content-Disposition header is missing from an Invite. 

The verbage below clearly specifies that if the handling parameter is
missing 
from a Content disposition header, then it should be assumed that the
Disposition 
is required.  Extra-polating that, should a missing Content-Disposition
header be 
treated as a required Disposition. 

Here is the snippet from RFC 3261, section 20.11 about Content Disposition.

Several new "disposition-types" of the Content-Disposition header are
defined by SIP. The value "session" indicates that the body part
describes a session, for either calls or early (pre-call) media. The
value "render" indicates that the body part should be displayed or
otherwise rendered to the user. Note that the value "render" is used
rather than "inline" to avoid the connotation that the MIME body is
displayed as a part of the rendering of the entire message (since the
MIME bodies of SIP messages oftentimes are not displayed to users).
For backward-compatibility, if the Content-Disposition header field
is missing, the server SHOULD assume bodies of Content-Type
application/sdp are the disposition "session", while other content
types are "render".

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


From sip-bounces@ietf.org  Thu Jan  6 12:01:49 2005
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From: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
To: "'Eric Burger'" <eburger@brooktrout.com>
Subject: RE: [Sip] Interpreting Missing Content-Disposition
Date: Thu, 6 Jan 2005 11:47:47 -0500 
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Eric,

Thanks for your response. The text that you indicate below
talks about the default Disposition type based on the
Content-Type. What I am looking for is a default Content
Disposition Handling.

Say for example, If a UAS receives a INVITE message
with 
Content-Type: application/xxx

and missing content disposition header missing from the
INVITE. Should the Content handling be assumed as 
required or optional ?. In this case the Content Disposition
would be session.

Thanks,
Shekar

> -----Original Message-----
> From: Eric Burger [mailto:eburger@brooktrout.com]
> Sent: Thursday, January 06, 2005 11:21 AM
> To: Shekar, Nagesh Soma (Shekar)** CTR **
> Cc: 'sip-ietf'
> Subject: RE: [Sip] Interpreting Missing Content-Disposition
> 
> 
> No.  The language is clear:
> "... if the Content-Disposition header field
> is missing, the server SHOULD assume bodies of Content-Type
> application/sdp are the disposition "session", while other content
> types are "render"."
> 
> Answer: Yes, unless you have a good reason not to.
> 
> -----Original Message-----
> From: Shekar, Nagesh Soma (Shekar)** CTR ** [mailto:shek@lucent.com]
> Sent: Thursday, January 06, 2005 8:56 AM
> To: 'sip-ietf'
> Subject: [Sip] Interpreting Missing Content-Disposition
> 
> 
> Could someone help me interpret RFC 3261 related to the the case when 
> the Content-Disposition header is missing from an Invite. 
> 
> The verbage below clearly specifies that if the handling parameter is
> missing 
> from a Content disposition header, then it should be assumed that the
> Disposition 
> is required.  Extra-polating that, should a missing 
> Content-Disposition
> header be 
> treated as a required Disposition. 
> 
> Here is the snippet from RFC 3261, section 20.11 about 
> Content Disposition.
> 
> Several new "disposition-types" of the Content-Disposition header are
> defined by SIP. The value "session" indicates that the body part
> describes a session, for either calls or early (pre-call) media. The
> value "render" indicates that the body part should be displayed or
> otherwise rendered to the user. Note that the value "render" is used
> rather than "inline" to avoid the connotation that the MIME body is
> displayed as a part of the rendering of the entire message (since the
> MIME bodies of SIP messages oftentimes are not displayed to users).
> For backward-compatibility, if the Content-Disposition header field
> is missing, the server SHOULD assume bodies of Content-Type
> application/sdp are the disposition "session", while other content
> types are "render".
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
Subject: Re: [Sip] Interpreting Missing Content-Disposition
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Shekar, Nagesh Soma (Shekar)** CTR ** wrote:
> Eric,
> 
> Thanks for your response. The text that you indicate below
> talks about the default Disposition type based on the
> Content-Type. What I am looking for is a default Content
> Disposition Handling.
> 
> Say for example, If a UAS receives a INVITE message
> with 
> Content-Type: application/xxx
> 
> and missing content disposition header missing from the
> INVITE. Should the Content handling be assumed as 
> required or optional ?. In this case the Content Disposition
> would be session.

I disagree. The content disposition in this case would be "render" 
(because the content-type is not application/sdp.)

I would interpret the text that eric quotes ("the server SHOULD assume 
bodies of Content-Type application/sdp are the disposition "session", 
while other content types are "render".") to mean there is a default 
content disposition header of form

	Content-Disposition: render

This header then does not have a "handling" parameter, so the other rule 
says the default value for that is "required".

Bottom line: if you want a body part to be optional, then you must 
include a content-disposition header that has handling=optional.

	Paul

> Thanks,
> Shekar
> 
> 
>>-----Original Message-----
>>From: Eric Burger [mailto:eburger@brooktrout.com]
>>Sent: Thursday, January 06, 2005 11:21 AM
>>To: Shekar, Nagesh Soma (Shekar)** CTR **
>>Cc: 'sip-ietf'
>>Subject: RE: [Sip] Interpreting Missing Content-Disposition
>>
>>
>>No.  The language is clear:
>>"... if the Content-Disposition header field
>>is missing, the server SHOULD assume bodies of Content-Type
>>application/sdp are the disposition "session", while other content
>>types are "render"."
>>
>>Answer: Yes, unless you have a good reason not to.
>>
>>-----Original Message-----
>>From: Shekar, Nagesh Soma (Shekar)** CTR ** [mailto:shek@lucent.com]
>>Sent: Thursday, January 06, 2005 8:56 AM
>>To: 'sip-ietf'
>>Subject: [Sip] Interpreting Missing Content-Disposition
>>
>>
>>Could someone help me interpret RFC 3261 related to the the case when 
>>the Content-Disposition header is missing from an Invite. 
>>
>>The verbage below clearly specifies that if the handling parameter is
>>missing 
>>from a Content disposition header, then it should be assumed that the
>>Disposition 
>>is required.  Extra-polating that, should a missing 
>>Content-Disposition
>>header be 
>>treated as a required Disposition. 
>>
>>Here is the snippet from RFC 3261, section 20.11 about 
>>Content Disposition.
>>
>>Several new "disposition-types" of the Content-Disposition header are
>>defined by SIP. The value "session" indicates that the body part
>>describes a session, for either calls or early (pre-call) media. The
>>value "render" indicates that the body part should be displayed or
>>otherwise rendered to the user. Note that the value "render" is used
>>rather than "inline" to avoid the connotation that the MIME body is
>>displayed as a part of the rendering of the entire message (since the
>>MIME bodies of SIP messages oftentimes are not displayed to users).
>>For backward-compatibility, if the Content-Disposition header field
>>is missing, the server SHOULD assume bodies of Content-Type
>>application/sdp are the disposition "session", while other content
>>types are "render".
>>
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From: Vijay Gaur <vgaur_tech@yahoo.com>
Subject: Re: [Sip] Interpreting Missing Content-Disposition
To: "Shekar, Nagesh Soma \(Shekar\)** CTR **" <shek@lucent.com>,
        "'sip-ietf'" <sip@ietf.org>
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Hi Shekar,
  This is what I interpreted from RFC 3261 and RFC 3204.
 
   RFC 3261 says that for unknown Content-Disposition and Content-Types if handling parameter is missing then assume its value as required and reject the INVITE with 415. If there is no Content-Disposition header present and Content-Type is unknown then reject INVITE with 415 message. Because for unknown message bodies application doesn't know how to deal with the body, simply rendering unknown bodies don't help. 
 
Old SIP implementation will render the bodies instead of rejecting the INVIE.
 
Thanks,
Vijay Gaur

"Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com> wrote:
BODY {	MARGIN-TOP: 25px; FONT-SIZE: 10pt; MARGIN-LEFT: 25px; COLOR: #000000; FONT-FAMILY: Arial, Helvetica}P.msoNormal {	MARGIN-TOP: 0px; FONT-SIZE: 10pt; MARGIN-LEFT: 0px; COLOR: #ffffcc; FONT-FAMILY: Helvetica, "Times New Roman"}LI.msoNormal {	MARGIN-TOP: 0px; FONT-SIZE: 10pt; MARGIN-LEFT: 0px; COLOR: #ffffcc; FONT-FAMILY: Helvetica, "Times New Roman"}Could someone help me interpret RFC 3261 related to the the case when 
the Content-Disposition header is missing from an Invite. 
 
The verbage below clearly specifies that if the handling parameter is missing 
from a Content disposition header, then it should be assumed that the Disposition 
is required.  Extra-polating that, should a missing Content-Disposition header be 
treated as a required Disposition. 
 
Here is the snippet from RFC 3261, section 20.11 about Content Disposition.
 

Several new "disposition-types" of the Content-Disposition header are

defined by SIP. The value "session" indicates that the body part

describes a session, for either calls or early (pre-call) media. The

value "render" indicates that the body part should be displayed or

otherwise rendered to the user. Note that the value "render" is used

rather than "inline" to avoid the connotation that the MIME body is

displayed as a part of the rendering of the entire message (since the

MIME bodies of SIP messages oftentimes are not displayed to users).

For backward-compatibility, if the Content-Disposition header field

is missing, the server SHOULD assume bodies of Content-Type

application/sdp are the disposition "session", while other content

types are "render".



> ATTACHMENT part 2 image/gif name=Blank Bkgrd.gif
_______________________________________________
Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
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<DIV>Hi Shekar,</DIV>
<DIV>&nbsp; This is what I interpreted from RFC 3261 and RFC 3204.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;&nbsp; RFC 3261 says that for unknown Content-Disposition and Content-Types if handling parameter is missing then assume its value as required and reject the INVITE with 415. If there is no Content-Disposition header present and Content-Type is unknown then reject INVITE with 415 message. Because for unknown message bodies application doesn't know how to deal with the body, simply rendering unknown bodies don't help. </DIV>
<DIV>&nbsp;</DIV>
<DIV>Old SIP implementation will render the bodies instead of rejecting the&nbsp;INVIE.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Thanks,</DIV>
<DIV>Vijay Gaur</DIV>
<DIV><BR><B><I>"Shekar, Nagesh Soma (Shekar)** CTR **" &lt;shek@lucent.com&gt;</I></B> wrote:</DIV>
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<DIV><FONT color=#800080><SPAN class=459554513-06012005><FONT size=2><FONT face="Century Gothic"><FONT color=#000000><FONT face=Arial>Could someone help me interpret RFC 3261 related to the the case when </FONT></FONT></FONT></FONT></SPAN></FONT></DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005>the Content-Disposition header is missing from an Invite. </SPAN></FONT></DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005>The verbage below clearly specifies that if the handling parameter </SPAN></FONT><FONT size=+0><SPAN class=459554513-06012005>is missing </SPAN></FONT></DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005>from a Content disposition header, then it should be assumed that </SPAN></FONT><FONT size=+0><SPAN class=459554513-06012005>the Disposition </SPAN></FONT></DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005>is required.&nbsp; Extra-polating that, should a&nbsp;missing </SPAN></FONT><FONT size=+0><SPAN class=459554513-06012005>Content-Disposition header be </SPAN></FONT></DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005>treated as a required Disposition.<FONT size=+0><SPAN class=459554513-06012005>&nbsp;</SPAN></FONT></SPAN></FONT></DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005>Here is the snippet from RFC 3261, section 20.11 about Content Disposition.</SPAN></FONT></DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=+0><SPAN class=459554513-06012005><FONT face=Courier size=2>
<P align=left>Several new "disposition-types" of the Content-Disposition header are</P>
<P align=left>defined by SIP. The value "session" indicates that the body part</P>
<P align=left>describes a session, for either calls or early (pre-call) media. The</P>
<P align=left>value "render" indicates that the body part should be displayed or</P>
<P align=left>otherwise rendered to the user. Note that the value "render" is used</P>
<P align=left>rather than "inline" to avoid the connotation that the MIME body is</P>
<P align=left>displayed as a part of the rendering of the entire message (since the</P>
<P align=left>MIME bodies of SIP messages oftentimes are not displayed to users).</P>
<P align=left>For backward-compatibility, if the Content-Disposition header field</P>
<P align=left>is missing, the server SHOULD assume bodies of Content-Type</P>
<P align=left>application/sdp are the disposition "session", while other content</P>
<P align=left>types are "render".</P></FONT></SPAN></FONT><FONT size=2><FONT face="Century Gothic"></DIV></FONT></FONT><BR><BR>&gt; ATTACHMENT part 2 image/gif name=Blank Bkgrd.gif<BR>_______________________________________________<BR>Sip mailing list https://www1.ietf.org/mailman/listinfo/sip<BR>This list is for NEW development of the core SIP Protocol<BR>Use sip-implementors@cs.columbia.edu for questions on current sip<BR>Use sipping@ietf.org for new developments on the application of sip</BLOCKQUOTE><BR><BR>Vijay Gaur<br>It's how you deal with failure that determines how you achieve success.<br>Au revoir, Adios, Namaste<p>__________________________________________________<br>Do You Yahoo!?<br>Tired of spam?  Yahoo! Mail has the best spam protection around <br>http://mail.yahoo.com 
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From: Vijay Gaur <vgaur_tech@yahoo.com>
Subject: Re: [Sip] Refer-To header and Replaces option
To: "Shekar, Nagesh Soma \(Shekar\)** CTR **" <shek@lucent.com>,
        "'sip-ietf'" <sip@ietf.org>
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Hi Shekar,
  In different call transfer scenarios like Attended, call park Replaces parameter is used in Refer-To header. 
  You can refer draft for the usage of Replaces parameter:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-07.txt
 
Thanks,
Vijay

"Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com> wrote:
BODY {	MARGIN-TOP: 25px; FONT-SIZE: 10pt; MARGIN-LEFT: 25px; COLOR: #000000; FONT-FAMILY: Arial, Helvetica}P.msoNormal {	MARGIN-TOP: 0px; FONT-SIZE: 10pt; MARGIN-LEFT: 0px; COLOR: #ffffcc; FONT-FAMILY: Helvetica, "Times New Roman"}LI.msoNormal {	MARGIN-TOP: 0px; FONT-SIZE: 10pt; MARGIN-LEFT: 0px; COLOR: #ffffcc; FONT-FAMILY: Helvetica, "Times New Roman"}Dear All,
 
Is the usage of Replaces parameter in a Refer-To header a standard usage. If
it it could you please point me to the RFC that specifies the usage.
 
Here is an example.
 

Refer-To: <sip:650916143674820@10.254.1.20>;transport=UDP?Require=replaces&Replaces=b7f4eb08d5674c5fa019ec2975173a4b4010.254.1.20

 
Thanks,
Nagesh Shekar



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_______________________________________________
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Vijay Gaur
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<DIV>Hi Shekar,</DIV>
<DIV>&nbsp; In different call transfer scenarios like Attended, call park&nbsp;Replaces parameter is used in Refer-To header. </DIV>
<DIV>&nbsp; You can refer draft for the usage of Replaces parameter:</DIV>
<DIV><A href="http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-07.txt"><SPAN style="mso-ansi-font-size: 11.0pt"><FONT face="Times New Roman" size=3>http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-07.txt</FONT></SPAN></A></DIV>
<DIV>&nbsp;</DIV>
<DIV>Thanks,</DIV>
<DIV>Vijay<BR><BR><B><I>"Shekar, Nagesh Soma (Shekar)** CTR **" &lt;shek@lucent.com&gt;</I></B> wrote:</DIV>
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<DIV><SPAN class=995245713-06012005>Dear All,</SPAN></DIV>
<DIV><SPAN class=995245713-06012005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=995245713-06012005>Is the usage of Replaces parameter in a Refer-To header a standard usage. If</SPAN></DIV>
<DIV><SPAN class=995245713-06012005>it it could you please point me to the RFC that specifies the usage.</SPAN></DIV>
<DIV><SPAN class=995245713-06012005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=995245713-06012005>Here is an example.</SPAN></DIV>
<DIV><SPAN class=995245713-06012005></SPAN>&nbsp;</DIV>
<DIV><SPAN class=995245713-06012005>
<P>Refer-To:<SPAN class=995245713-06012005> </SPAN>&lt;sip:650916143674820@10.254.1.20&gt;;transport=UDP?Require=replaces&amp;Replaces=b7f4eb08d5674c5fa019ec2975173a4b4010.254.1.20</P></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV align=left>
<DIV align=left><FONT face="Century Gothic" color=#800080><STRONG><SPAN class=995245713-06012005>Thanks,</SPAN></STRONG></FONT></DIV>
<DIV align=left><FONT face="Century Gothic" color=#800080><STRONG><SPAN class=995245713-06012005>Nagesh Shekar</SPAN></STRONG></FONT></DIV></DIV><BR><BR>&gt; ATTACHMENT part 2 image/gif name=Blank Bkgrd.gif<BR>_______________________________________________<BR>Sip mailing list https://www1.ietf.org/mailman/listinfo/sip<BR>This list is for NEW development of the core SIP Protocol<BR>Use sip-implementors@cs.columbia.edu for questions on current sip<BR>Use sipping@ietf.org for new developments on the application of sip</BLOCKQUOTE><BR><BR>Vijay Gaur<br>It's how you deal with failure that determines how you achieve success.<br>Au revoir, Adios, Namaste<p>
		<hr size=1>Do you Yahoo!?<br> 
<a href="http://us.rd.yahoo.com/mail_us/taglines/virus/*http://promotions.yahoo.com/new_mail/static/protection.html">Yahoo! Mail</a> - Helps protect you from nasty viruses.
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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
--===============0634738004==--



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HI,
   To establish a Pre established session , a INVITE has to be sent to =
the UAS .=20
    What would be the TO and From header of the INVITE Message? Will =
they be the same ?



Girish Pai
Bharti Telesoft Intl. Pvt. Ltd.
Bangalore

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<DIV>HI,</DIV>
<DIV>&nbsp;&nbsp; To establish a Pre established session , a INVITE has =
to be=20
sent to the UAS . </DIV>
<DIV>&nbsp;&nbsp;&nbsp; What would be the TO and From header of the =
INVITE=20
Message? Will they be the same ?</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>Girish Pai<BR>Bharti Telesoft Intl. Pvt. Ltd.<BR>Bangalore</DIV>
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_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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--===============1969829674==--





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The second edition of the WiFi Voice Conference will stand in Paris on 10 to
13 May, 2005. 

 

- What about the standardization process and the regulatory issues? 
- What are the market drivers for VoWLAN? 
- How will VoWLAN systems prioritize and deliver data packets? 
- What are the planning and engineering issues? 
- What about security and privacy? 

Distinguished experts and key players in the field will address these
questions with detailed and technical explanations. 

Get details at:

 <http://www.upperside.fr/wifi05/wifivoice2005intro.htm>
http://www.upperside.fr/wifi05/wifivoice2005intro.htm

 


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>The second edition of the WiFi Voice =
Conference will
stand in <st1:place w:st=3D"on"><st1:City =
w:st=3D"on">Paris</st1:City></st1:place>
on <strong><b><font color=3D"#6495be" face=3DArial><span =
style=3D'font-family:Arial;
color:#6495BE'>10 to 13 May, 2005.</span></font></b></strong> =
<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>- What about the standardization process and =
the
regulatory issues? <br>
- What are the market drivers for VoWLAN? <br>
- How will VoWLAN systems prioritize and deliver data packets? <br>
- What are the planning and engineering issues?&nbsp;<br>
- What about security and privacy?&nbsp;<br>
<br>
Distinguished experts and key players in the field will address these =
questions
with detailed and technical explanations. <o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><span class=3Dkeynote><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'><a
href=3D"http://www.upperside.fr/wifi05/wifivoice2005intro.htm"
title=3D"http://www.upperside.fr/wifi05/wifivoice2005intro.htm"><span =
lang=3DEN-GB>http://www.upperside.fr/wifi05/wifivoice2005intro.htm</span>=
</a></span></font></span><span
class=3Dkeynote><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p></o:p></span></font></span></p>

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

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_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0683950329==--




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Subject: Re: [Sip] Invite Headrs
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WARNING: LONG

Normally, I wouldn't respond to something like this here, but would =20
refer it to the sip-implementors list. However, this does pose a =20
reasonable example for running down SIP details across the multiple =20
bodies developing specifications related to SIP. Consequently, I'll =20
give a long-winded, rambling, and potentially pointless discussion of =20=

how one might try to run this sort of question down. Perhaps somebody =20=

can suggest a more direct route . . .

On Jan 6, 2005, at 1:32 PM, girish wrote:

> HI,
> =A0=A0 To establish a Pre established session , a INVITE has to be =
sent to =20
> the UAS .
>  =A0=A0=A0 What would be the TO and =46rom header of the INVITE =
Message? Will =20
> they be the same ?
>
What difference does it make, and why?

"Pre Established Session", as I understand it, is an OMA POC =20
optimization technique wherein a POC client maintains a long-lived =20
dialog with a POC application server, and that POC application server =20=

acts a B2BUA to bridge the long-lived dialog together with short-lived =20=

dialogs created dynamically between the POC application server and =20
other network elements.

Since the To and =46rom of the INVITE that establishes the long-lived =20=

dialog is only seen by the POC application server, it may not matter a =20=

whole lot what goes in there.

However, assuming we use a "POC as conferencing" design model, which =20
then envisions the POC application server as a sort of specialized =20
conference bridge, it would seem reasonable to populate To: with the =20
identity of the POC application server or service provider, and From: =20=

with the AOR of the user sending that INVITE. This would facilitate =20
reuse of the From: field by the POC Application Server in the =20
establishment of further short-lived dialogs.

But assuming we're taking about "pre established sessions" in the OMA =20=

POC sense, we should observe that what goes into To: and From: isn't =20
really in the domain of the IETF. Rather, OMA specifications are likely =20=

to describe what they expect here, and unless they need some guidance =20=

on the matter from IETF (whether they know it or not) we generally =20
don't have a lot to say about their specifications.

If you're interested, the POC working group of OMA has some of their =20
working documents categorized as "public" and they can be retrieved =20
from the OMA web site at:

http://member.openmobilealliance.org/ftp/public_documents/POC/=20
Permanent_documents/

Note: The page at =20
http://www.openmobilealliance.org/tech/wg_committees/poc.html makes a =20=

lot more sense as a starting point.

The Architecture, Control plane, and User Plane documents may be of =20
interest:

see:
OMA-AD-PoC-V1_0-20041117-D.zip
OMA-CP-POC-V1_0-20041111-D.zip
OMA-UP-PoC-V1_0-20041005-D.zip

And in:

OMA-TS-PoC_ControlPlane-V1_0-20041117-D.zip

I found:

--------------

6.1.3.2.1 PoC Client initiates a Pre-established Session

Upon receiving a requestfrom a PoC User to initiate a Pre-established =20=

Session the PoC Client:

-=A0=A0=A0=A0=A0=A0=A01.=A0SHALL generate an initial SIP INVITE request =
as specified in =20
subclause 6.1.3.1 "General";

-=A0=A0=A0=A0=A0=A0=A02.=A0SHALL set the Request-URI of the SIP INVITE =
request to the =20
Conference-factory-URI for the PoC Service in the Home PoC Network of =20=

the PoC User;

-=A0=A0=A0=A0=A0=A0=A03.=A0SHALL include a MIME SDP body as a SDP offer =
according to =20
rules and procedures of [RFC3264] and [RFC2327] as follows;

a)=A0=A0=A0set the IP address and the port number for the RTP Session

b)=A0=A0set the supported codecs of the PoC Client

c)=A0=A0if another IP address or port number is used instead of the =
default =20
IP address and port number specified by [RFC3550], set the IP address =20=

and port number for the RTCP Session at the PoC Clientaccording to =20
rules and procedures of=A0[RFC3605];

d) set the offered Talk Burst Control Protocol and the port number(s) =20=

for the Talk Burst Control Protocol

e) if the PoC Client is not able/willing to receive media streams =20
immediately then the PoC Client SHALL set the media to "'inactive"' as =20=

specified in [RFC3108] when generating the SDP offer;

-=A0=A0=A0=A0=A0=A0=A05.=A0SHALL send the SIP INVITE request towards the =
PoC Server =20
according torules andprocedures of the SIP/IP core.

Upon receiving a SIP 200 "OK" response to the SIP INVITE request the =20
PoC Client:

-=A0=A0=A0=A0=A0=A0=A01.=A0SHALL store the content of the received =
Contact header as =20
the PoC Session Identity

-=A0=A0=A0=A0=A0=A0=A02.=A0SHALL interact with the User Plane as =
specified in =20
[OMA-PoC-UP] "PoC Client procedure at Pre-established Session =20
initialization".

When the SIP/IP Core corresponds with 3GPP/3GPP2 IMS, the PoC Client =20
SHALL use 3GPP/3GPP2 IMS Session establishment mechanisms according to =20=

rules and procedures of=A0[TS24.229] / [3GPP2X.S0013.4]with the =20
clarifications given in this subclause.

--------------




Noting that this refers to 6.1.3.1, we look there and see:


--------------


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6.1.3.1=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0General

The PoC Client SHALL generate an initial SIP INVITE request according=20
to rules and procedures of=A0[RFC3261].=A0The PoC Client

-=A0=A0=A0=A0=A0=A0=A01. SHALL include an Accept-Contact header with the =
Poc=20
feature-tag '+g.poc.talkburst' along with =91require=92 and =91explicit=92=
=20
parameters according to the rules and procedures of=A0[RFC 3841]in all=20=

initial SIP INVITE requests.

-=A0=A0=A0=A0=A0=A0=A02. SHALL include User-Agent header to indicate the =
PoC release=20
version.

-=A0=A0=A0=A0=A0=A0=A03. SHALL include value ''id'' in a Privacy header =
if anonymity=20
is requested according to rules and procedures of [RFC3325].

NOTE 1:=A0=A0If anonymity is not allowed for the PoC Group indicated =
with=20
the Request-URI of the SIP INVITE based on the rules specified in the=20
[POC XDM Specification] the PoC Session will not be allowed by the PoC=20=

server hosting the PoC Group.

-=A0=A0=A0=A0=A0=A0=A04. SHALL include the feature tag =
'+g.poc.talkburst' in the=20
Contact header.

-=A0=A0=A0=A0=A0=A0=A05. SHALL include the option tag 'timer' in the =
Supported header.

-=A0=A0=A0=A0=A0=A0=A06. SHOULD include an Allow header with all =
supported SIP=20
methods.

-=A0=A0=A0=A0=A0=A0=A07. SHOULD include the Session-Expires header with =
the refresher=20
parameter set to ''uac''according to rules and procedures=20
of[draft-ietf-sip-session-timer].

NOTE:=A0=A0The use of the option tag 'precondition', as specified in=20
[RFC3312], and the option tag '100rel', specified in [RFC3262], is not=20=

defined for POC-1 reference point.

On receipt of a final response to the SIP INVITE request, the PoC=20
Client SHALLstore the list of SIP methods that the PoC Server=20
supportsif received in the Allow header.

On receipt of the SIP 200 "OK" response to the initial SIP INVITE=20
request the PoC Client SHALL start the SIP Session timer usingthe value=20=

received in the Session-Expires headeraccording to rules and procedures=20=

of[draft-ietf-sip-session-timer].

When the SIP/IP Core corresponds with 3GPP/3GPP2 IMS, the PoC Client=20
SHALL use 3GPP/3GPP2 IMS Session establishment mechanisms according to=20=

rules and procedures of=A0[TS24.229] / [3GPP2X.S0013.4]with the=20
clarifications given in this subclause.

--------------

So, in the final analysis, OMA doesn't seem to say anything direct=20
about populating the To: and From: fields of an INVITE intended to=20
establish a "pre-established session". Instead, they defer to the rules=20=

of the "SIP/IP Core", referencing 3GPP's TS 24.229, which one can see=20
at:

http://www.3gpp.org/ftp/Specs/html-info/24229.htm

Note that this reference is not normative unless one happens to be=20
running POC on IMS.

Looking at 24.229, we find in=20=

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5.1.2A=A0=A0=A0"Generic procedures applicable to all methods excluding =
the=20
REGISTER method":

-----------

Where privacy is required, in any initial request for a dialog or=20
request for a standalone transaction, the UE shall set the =46rom header=20=

to "Anonymous".

NOTE=A04:=A0=A0=A0=A0=A0=A0=A0=A0The contents of the =46rom header =
should not be relied=20
upon to be modified by the network based on any privacy specified by=20
the user either within the UE indication of privacy or by network=20
subscription or network policy. Therefore the user should include the=20
value "Anonymous" whenever privacy is explicitly required.As the user=20
may well have privacy requirements, terminal manufacturers should not=20
automatically derive and include values in this header from the public=20=

user identity or other values stored in or derived from the UICC. Where=20=

the user has not expressed a preference in the configuration of the=20
terminal implementation, the implementation should assume that privacy=20=

is required. Users that require to identify themselves, and are making=20=

calls to SIP destinations beyond the IM CN subsystem, where the=20
destination does not implement RFC=A03325=A0[34], will need to include a=20=

value in the =46rom header other than Anonymous.

------------

24.229, as far as I can tell, says little about the From: header field,=20=

and almost nothing about the From: header field needed for IMS. So, the=20=

reference from the OMA document seems a little spurious. Digging=20
deeper, we find that there are examples in 24.228. Unfortunately,=20
although they discuss the use of the user's requested public identity=20
in From: fields, these example do little to illuminate the use of the=20
To: field (generally using a Tel: URL). Now, since I vaguely recall=20
from years of working with 3GPP that 24.x specs are "stage 3" based on=20=

the 23.x stage 2 specs, I decided to fall back to stage 2 and see if=20
there is anything helpful.


In 23.228, we find something useful:


--------------

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4.3.3=A0=A0=A0Identification of users

There are various identities that may be associated with a user of IP=20
multimedia services. This section describes these identities and their=20=

use.


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4.3.3.1=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0Private user identities


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Every IM CN subsystem user shall have one or more private user=20
identities. The private identity is assigned by the home network=20
operator, and used, for example, for Registration, Authorisation,=20
Administration, and Accounting purposes. This identity shall take the=20
form of a Network Access Identifier (NAI) as defined in RFC 2486 [14].=20=

It is possible for a representation of the IMSI to be contained within=20=

the NAI for the private identity.

-=A0=A0The Private User Identity is not used for routing of SIP =
messages.

-=A0=A0The Private User Identity shall be contained in all Registration=20=

requests, (including Re-registration and De-registration requests)=20
passed from the UE to the home network.

-=A0=A0An ISIM application shall securely store one Private User =
Identity.=20
It shall not be possible for the UE to modify the Private User Identity=20=

information stored on the ISIM application.

-=A0=A0The Private User Identity is a unique global identity defined by =
the=20
Home Network Operator, which may be used within the home network to=20
uniquely identify the user from a network perspective.

-=A0=A0The Private User Identity shall be permanently allocated to a =
user=20
(it is not a dynamic identity), and is valid for the duration of the=20
user=92s subscription with the home network.

-=A0=A0The Private User Identity is used to identify the user=92s =
information=20
(for example authentication information) stored within the HSS (for use=20=

for example during Registration).

-=A0=A0The Private User Identity may be present in charging records =
based=20
on operator policies.

-=A0=A0The Private User Identity identifies the subscription (e.g. IM=20
service capability) not the user.

-=A0=A0The Private User Identity is authenticated only during =
registration=20
of the user, (including re-registration and de-registration).

-=A0=A0The HSS needs to store the Private User Identity.

-=A0=A0The S-CSCF needs to obtain and store the Private User Identity =
upon=20
registration and unregistered termination.


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4.3.3.2=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0Public user identities

Every IM CN subsystem user shall have one or more public user=20
identities [8]. The public user identity/identities are used by any=20
user for requesting communications to other users. For example, this=20
might be included on a business card.

-=A0=A0Both telecom numbering and Internet naming schemes can be used to=20=

address users depending on the Public User identities that the users=20
have.

-=A0=A0The public user identity/identities shall take the form of SIP =
URL=20
(as defined in RFC 3261 [12] and RFC2396 [13]) or the "tel:"-URL format=20=

[15].

-=A0=A0An ISIM application shall securely store at least one Public User=20=

Identity (it shall not be possible for the UE to modify the Public User=20=

Identity), but it is not required that all additional Public User=20
Identities be stored on the ISIM application.

-=A0=A0=A0=A0=A0=A0=A0A Public User Identity shall be registered either =
explicitly or=20
implicitly before the identity can be used to originate IMS sessions=20
and IMS session unrelated procedures.

-=A0=A0=A0=A0=A0=A0=A0A Public User Identity shall be registered either =
explicitly or=20
implicitly before terminating IMS sessions and terminating IMS session=20=

unrelated procedures can be delivered to the UE of the user that the=20
Public User Identity belongs to. Subscriber-specific services for=20
unregistered users may nevertheless be executed as described in chapter=20=

5.12.

-=A0=A0It shall be possible to register globally (i.e. through one =
single=20
UE request) a user that has more than one public identity via a=20
mechanism within the IP multimedia CN subsystem (e.g. by using an=20
Implicit Registration Set). This shall not preclude the user from=20
registering individually some of his/her public identities if needed.

-=A0=A0Public User Identities are not authenticated by the network =
during=20
registration.

-=A0=A0Public User Identities may be used to identify the user=92s=20
information within the HSS (for example during mobile terminated=20
session set-up).

--------------

So, from this we learn that the From: should be populated (if we aren't=20=

requesting anonymity) with a sip: or tel: URL and display name string=20
corresponding to a public identity of the requesting user. And if we=20
ARE requesting anonymity (which doesn't seem all that useful in this=20
context), we set it to "Anonymous".

I'm not at all sure why some people think figuring this stuff out is=20
easy. After all, I've only been working with this stuff for about six=20
years, and I was able to address this basic question with only an hour=20=

or so of research,

And I still don't know what goes in the To:. But I suspect it doesn't=20
really matter very much.


--
Dean


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--Apple-Mail-20-135677000--




From sip-bounces@ietf.org  Mon Jan 10 07:44:33 2005
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Date: Mon, 10 Jan 2005 18:12:17 +0530
From: Aby K Samuel <abysamuel@gmail.com>
To: sip@ietf.org
Subject: RE: [Sip] a question on tags when INVITE is challenged
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 Can someone tell me how having the same tag in the From fields of
INVITE(F1) and INVITE(F4), affect a call ??

> 
> Message: 5
> Date: Wed, 5 Jan 2005 07:20:20 -0800 (PST)
> From: Vijay Gaur <vgaur_tech@yahoo.com>
> Subject: RE: [Sip] a question on tags when INVITE is challenged
> To: Nataraju A B <bnataraju@kodiaknetworks.com>,        "'Wang, Jin-Gen'"
>        <Jin-Gen.Wang@Level3.com>, sip@ietf.org
> Message-ID: <20050105152020.76711.qmail@web61205.mail.yahoo.com>
> Content-Type: text/plain; charset="us-ascii"
> 
> Hi Wang,
>  Vendor1's SIP implementation should not expect same To-tag which is present in 401. There is no dialog established by 401 response. New INVITE with credentials is totaly different request. In fact few SIP implementations might send the same From tag of original INVITE.
> 
> You can see similar call flows in the draft below:
> http://www.iptel.org/info/players/ietf/callflows/draft-ietf-sipping-basic-call-flows-02.txt
> 
> Thanks,
> Vijay Gaur
> 
> Nataraju A B <bnataraju@kodiaknetworks.com> wrote:
> Comments inline...
> 
> Best Regards,
> Nataraju A.B.
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Wang, Jin-Gen
> Sent: Wednesday, January 05, 2005 10:45 AM
> To: sip@ietf.org
> Subject: [Sip] a question on tags when INVITE is challenged
> 
> We are running into an inter-op problem in SIP. It involves the
> interpretation of RFC 3261. Hopefully we can get some input from the
> list. Here is the call flow:
> 
> UAC UAS
> | ----- INVITE (F1) ------> |
> | <--- 401 (F2) ------------ |
> | ------ ACK (F3) ---------> |
> | ------ INVITE (F4) ------> |
> | ............................... |
> | <----- 200 (F5) ---------- |
> | ------- ACK (F6) --------> |
> 
> where INVITE (F1) has no credential and INVITE (F4) has credential.
> 
> The vendor 1 believes INVITE (F1) and INVITE (F4) should have same From
> tag and 401 (F2) and 200 (F5) should have same To tag. The following
> quote from section 22.2 in RFC 3261 indicates F4 is an updated request.
> 
> "When a UAC resubmits a request with its credentials after receiving a
> 401 (Unauthorized) or 407 (Proxy Authentication Required) response,
> it MUST increment the CSeq header field value as it would normally
> when sending an updated request."
> 
> It implicitly indicates that INVITE (F1) and INVITE (F4) are within the
> same dialog for a same call even though the dialog is created after
> INVITE (F4).
> 
> [ABN] this might be confusing text... may needs to be modified...
> 
> The vendor 2 believes INVITE (F1) and INVITE (F4) should have different
> >From tag and/or 401 (F2) and 200 (F5) should have different To tags
> because of the quote from section 19.3 in RFC 3261:
> 
> "When a tag is generated by a UA for insertion into a request or
> response, it MUST be globally unique and cryptographically random
> with at least 32 bits of randomness. A property of this selection
> requirement is that a UA will place a different tag into the From
> header of an INVITE than it would place into the To header of the
> response to the same INVITE. This is needed in order for a UA to
> invite itself to a session, a common case for "hairpinning" of calls
> in PSTN gateways. Similarly, two INVITEs for different calls will
> have different From tags, and two responses for different calls will
> have different To tags."
> 
> Which vendor interprets the tag requirements correctly?
> 
> [ABN] I feel the vendor 2 interprets correctly...
> Once the failure response received for an INVITE request and that
> transaction is closed the session will also be deleted since there are
> no other dialog exists as part of the same session. Hence the
> application (no the user) has to initiate a new session with new from
> tag to send the INVITE (F4) with challenge response...
> 
> Thanks.
> 
> Jin-Gen Wang
> Level3 Communications Inc.
> 1025 Eldorado Blvd.
> Broomfield, CO 80021
> 
> _______________________________________________
> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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
> 
> Vijay Gaur
> It's how you deal with failure that determines how you achieve success.
> Au revoir, Adios, Namaste
> 
> ---------------------------------
> Do you Yahoo!?
> Take Yahoo! Mail with you! Get it on your mobile phone.
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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
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> 
> End of Sip Digest, Vol 9, Issue 2
> *********************************
>

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From: "Nataraju A B" <bnataraju@kodiaknetworks.com>
To: <sip@ietf.org>
Subject: RE: [Sip] a question on tags when INVITE is challenged
Date: Mon, 10 Jan 2005 18:32:54 +0530
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Best Regards,
Nataraju A.B.
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Aby K Samuel
Sent: Monday, January 10, 2005 6:12 PM
To: sip@ietf.org
Subject: RE: [Sip] a question on tags when INVITE is challenged

 Can someone tell me how having the same tag in the From fields of
INVITE(F1) and INVITE(F4), affect a call ??

> 
> Message: 5
> Date: Wed, 5 Jan 2005 07:20:20 -0800 (PST)
> From: Vijay Gaur <vgaur_tech@yahoo.com>
> Subject: RE: [Sip] a question on tags when INVITE is challenged
> To: Nataraju A B <bnataraju@kodiaknetworks.com>,        "'Wang,
Jin-Gen'"
>        <Jin-Gen.Wang@Level3.com>, sip@ietf.org
> Message-ID: <20050105152020.76711.qmail@web61205.mail.yahoo.com>
> Content-Type: text/plain; charset="us-ascii"
> 
> Hi Wang,
>  Vendor1's SIP implementation should not expect same To-tag which is
present in 401. There is no dialog established by 401 response. New
INVITE with credentials is totaly different request. In fact few SIP
implementations might send the same From tag of original INVITE.
> 
> You can see similar call flows in the draft below:
>
http://www.iptel.org/info/players/ietf/callflows/draft-ietf-sipping-basi
c-call-flows-02.txt
> 
> Thanks,
> Vijay Gaur
> 
> Nataraju A B <bnataraju@kodiaknetworks.com> wrote:
> Comments inline...
> 
> Best Regards,
> Nataraju A.B.
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Wang, Jin-Gen
> Sent: Wednesday, January 05, 2005 10:45 AM
> To: sip@ietf.org
> Subject: [Sip] a question on tags when INVITE is challenged
> 
> We are running into an inter-op problem in SIP. It involves the
> interpretation of RFC 3261. Hopefully we can get some input from the
> list. Here is the call flow:
> 
> UAC UAS
> | ----- INVITE (F1) ------> |
> | <--- 401 (F2) ------------ |
> | ------ ACK (F3) ---------> |
> | ------ INVITE (F4) ------> |
> | ............................... |
> | <----- 200 (F5) ---------- |
> | ------- ACK (F6) --------> |
> 
> where INVITE (F1) has no credential and INVITE (F4) has credential.
> 
> The vendor 1 believes INVITE (F1) and INVITE (F4) should have same
From
> tag and 401 (F2) and 200 (F5) should have same To tag. The following
> quote from section 22.2 in RFC 3261 indicates F4 is an updated
request.
> 
> "When a UAC resubmits a request with its credentials after receiving a
> 401 (Unauthorized) or 407 (Proxy Authentication Required) response,
> it MUST increment the CSeq header field value as it would normally
> when sending an updated request."
> 
> It implicitly indicates that INVITE (F1) and INVITE (F4) are within
the
> same dialog for a same call even though the dialog is created after
> INVITE (F4).
> 
> [ABN] this might be confusing text... may needs to be modified...
> 
> The vendor 2 believes INVITE (F1) and INVITE (F4) should have
different
> >From tag and/or 401 (F2) and 200 (F5) should have different To tags
> because of the quote from section 19.3 in RFC 3261:
> 
> "When a tag is generated by a UA for insertion into a request or
> response, it MUST be globally unique and cryptographically random
> with at least 32 bits of randomness. A property of this selection
> requirement is that a UA will place a different tag into the From
> header of an INVITE than it would place into the To header of the
> response to the same INVITE. This is needed in order for a UA to
> invite itself to a session, a common case for "hairpinning" of calls
> in PSTN gateways. Similarly, two INVITEs for different calls will
> have different From tags, and two responses for different calls will
> have different To tags."
> 
> Which vendor interprets the tag requirements correctly?
> 
> [ABN] I feel the vendor 2 interprets correctly...
> Once the failure response received for an INVITE request and that
> transaction is closed the session will also be deleted since there are
> no other dialog exists as part of the same session. Hence the
> application (no the user) has to initiate a new session with new from
> tag to send the INVITE (F4) with challenge response...
> 
> Thanks.
> 
> Jin-Gen Wang
> Level3 Communications Inc.
> 1025 Eldorado Blvd.
> Broomfield, CO 80021
> 
> _______________________________________________
> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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
> 
> Vijay Gaur
> It's how you deal with failure that determines how you achieve
success.
> Au revoir, Adios, Namaste
> 
> ---------------------------------
> Do you Yahoo!?
> Take Yahoo! Mail with you! Get it on your mobile phone.
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> 
> ------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> End of Sip Digest, Vol 9, Issue 2
> *********************************
>

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From: Ke Ke Qi <qikeke@cn.ibm.com>
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I will be out of the office starting  2005-01-10 and will not return until
2005-01-12.

I will respond to your message when I return.
Any urgent thing, please contact with my manager Yan Da Lai
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<p>I will be out of the office starting  2005-01-10 and will not return until 2005-01-12.<br>
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sip] Draft defining the Accept-Disposition header field
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Folks,

I have submitted a draft that defines the Accept-Disposition header 
field. The usage of this header field would be similar to the usage of 
Accept, Accept-Encoding, or Accept-Language, but it would indicate 
support for disposition types.

http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept-disposition-00.txt

Your comments are welcome.

Thanks,

Gonzalo



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From sip-bounces@ietf.org  Wed Jan 12 09:32:12 2005
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Gonzalo,

I think work on this general subject area is long overdue. The whole 
notion of content disposition is significantly underspecified in sip. I 
would like to see it go further, in several respects:

- define the meaning when no Accept-Disposition is present.
   Are there any disposition types that are required to be
   supported by default? (E.g. is support for 'render' a default?)

- Define how to report the following situation:
   A request as been received containing a body part with
   content type T and content disposition D. Type T is
   supported with some dispositions, but not with disp D,
   and disposition D is supported, but not with type T.
   (E.g. if a MESSAGE was received containing body with
   c-t:application/sdp and c-d:render.)

- We really need better specification of specific content
   dispositions. E.g. I don't know how to decide if I support
   'render', because I don't know what that means. And what
   do I do if I can support 'render' for MESSAGE, but not for
   SUBSCRIBE?

- There is a real need for further specification of how
   content dispostion is used with multipart bodies. The
   contained body parts may have independent dispositions.
   Is the content type of the container related to that of
   the contents, or independent of it? If independent, then
   what should it be. (Most examples just don't include a
   c-d in multiparts, which means they default to 'render'.
   What does it mean to 'render' a multipart that contains
   parts with dispostions other than 'render'?)

- There is need for specification of what c-d to use with
   the distinguished parts of special multipart containers.
   E.g. the signature part of a multipart/signed.

- What should the disposition be for body parts that contain
   content-id and are then referenced from elsewhere in the
   message using a cid: uri? If such body parts have a dispostion
   and type that implies processing independent of the reference,
   are they to be processed both ways?
   (Strained example: message has an Alert-Info header with a
   cid: uri referencing a body part with c-d of 'render'. Should
   it be used to alert *and* also be rendered as part of the
   session?)

It seems likely that resolving all of the above would require more than 
one draft. But I think we should recognize that there is a program of 
required work, and adjust the charter accordingly, with the needed 
deliverables being scheduled. I would be willing to work on this.

	Paul

Gonzalo Camarillo wrote:
> Folks,
> 
> I have submitted a draft that defines the Accept-Disposition header 
> field. The usage of this header field would be similar to the usage of 
> Accept, Accept-Encoding, or Accept-Language, but it would indicate 
> support for disposition types.
> 
> http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept-disposition-00.txt 
> 
> 
> Your comments are welcome.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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Hello,
=20
As per section 10.3 step 6 the bindings must not be removed if there is =
an incoming REGISTER request with...
=20
CallID=3D=3DOne stored
CSeq=3D=3DSame as stored
Contact : *
Expires: 0
=20
But I found couple of Registrar implementations removing the bindings =
and sends 200Ok.It should have dropped the request.
=20
Any thoughts or inputs??
=20
Thanks
karthikeyan
=20
=20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML =
DIR=3Dltr><HEAD><META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1"></HEAD><BODY><DIV><FONT face=3D'Arial' =
color=3D#000000 size=3D2>Hello,</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>As per section 10.3 step 6 the bindings =
must not be =0A=
removed if there is an incoming REGISTER request with...</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>CallID=3D=3DOne stored</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>CSeq=3D=3DSame as stored</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Contact : *</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Expires: 0</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>But I found couple of Registrar =
implementations =0A=
removing the bindings and sends 200Ok.It should have dropped the =0A=
request.</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Any thoughts or inputs??</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>karthikeyan</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
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Hello,
=20
As per RFC 3261 the Registrars must ignore Record-Route field in any =
incoming REGISTER request and should not include the same in the =
response.
=20
But I find couple of Registrar implementations including the Record =
Route Field in the 200OK Response.
=20
Any advice or suggestion??
=20
Thanks
karthikeyan

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML =
DIR=3Dltr><HEAD><META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1"></HEAD><BODY><DIV><FONT face=3D'Arial' =
color=3D#000000 size=3D2>Hello,</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>As per RFC 3261 the Registrars must =
ignore =0A=
Record-Route field in any incoming REGISTER request and should not =
include the =0A=
same&nbsp;in the response.</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>But I find couple of Registrar =
implementations =0A=
including the Record Route Field in the 200OK Response.</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Any advice or suggestion??</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>karthikeyan</FONT></DIV></BODY></HTML>
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Hello everybody,
I'm trying to connect messenger 5 to my SIP proxy, but when I try to us=
e a
TLS connection, messenger gives me an error --> service non available.
I can log in with TCP connection but I would like to do it with TLS...
Any proposal?
Thank you
Mar=EDa S=E1nchez
=



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Cc: "sip@ietf.org" <sip@ietf.org>, Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: [Sip] Re: question on draft-ietf-sip-sctp-05.txt
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Cullen,

thanks for pointing this out. This problem was caused by the 
documentation process we followed. The original idea was to define 
everything in the SIP over SCTP draft, but then some parts were moved to 
RFC 3261 (e.g., the definition of the token for the via header and the 
value for the transport parameter) and to RFC 3263 (e.g., the definition 
of the SIP+D2S service). We obviously forgot to define (somewhere) both 
tokens and the new service for TLS over SCTP.

I will prepare a new revision of the draft including these definitions.

Thanks,

Gonzalo


Cullen Jennings wrote:

> What do the following two lines look like if I am using TLS over SCTP?
> 
>       sip:Bob.Johnson@example.com;transport=sctp
>       Via: SIP/2.0/SCTP ws1234.example.com:5060
> 
> And could you explicitly mention what the SRV lookups are like (particularly
> with TLS and SCTP)
> 
> Thanks, Cullen
> 

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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Hi,
   What do we mean by     PoC service indication  in a SIP Message. What =
does it imply ?


Girish Pai
Bharti Telesoft Intl Pvt. Ltd.
Airport Road
Bangalore

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<DIV><FONT face=3DArial>Hi,</FONT></DIV>
<DIV><FONT face=3DArial>&nbsp;&nbsp; What do we mean by <FONT =
size=3D2><FONT=20
face=3D"Times New Roman">&nbsp;&nbsp;&nbsp; </FONT></FONT><SPAN =
lang=3DEN-GB=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Times New Roman'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-GB; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA"><FONT=20
face=3DVerdana size=3D3>PoC service indication&nbsp; in a SIP Message. =
What does it=20
imply ?</FONT></SPAN></FONT></DIV>
<DIV><FONT face=3DVerdana size=3D3><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Times New Roman'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-GB; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA"></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DVerdana size=3D3><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Times New Roman'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-GB; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA"></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial>Girish Pai<BR>Bharti Telesoft Intl Pvt. =
Ltd.<BR>Airport=20
Road<BR>Bangalore<BR></FONT></DIV></BODY></HTML>

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From: "Brett Tate" <brett@broadsoft.com>
To: <innovation.interops@wipro.com>, <sip@ietf.org>
Subject: RE: [Sip] PROCESSING REGISTER REQUESTS - Section 10.3
Date: Thu, 13 Jan 2005 08:01:53 -0500
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> As per RFC 3261 the Registrars must 
> ignore Record-Route field in any 
> incoming REGISTER request and should 
> not include the same in the response.
>
> But I find couple of Registrar 
> implementations including the 
> Record Route Field in the 200OK Response.
>
> Any advice or suggestion??

The Registrar is compliant to RFC2543 concerning the Record-Route.  The
Registrar has not yet become compliant to that particular "MUST NOT" within
RFC3261.  There is likely no consequences.

Such questions should be posted to sip-implementors@cs.columbia.edu instead
of this list.



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From: "Brett Tate" <brett@broadsoft.com>
To: <innovation.interops@wipro.com>, <sip@ietf.org>
Subject: RE: [Sip] Processing REGISTER REQUESTS - STEP 6
Date: Thu, 13 Jan 2005 09:00:23 -0500
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> As per section 10.3 step 6 the bindings 
> must not be removed if there is an 
> incoming REGISTER request with...
>
> CallID==One stored
> CSeq==Same as stored
> Contact : *
> Expires: 0
>
> But I found couple of Registrar implementations 
> removing the bindings and sends 200Ok.
> It should have dropped the request.
>
> Any thoughts or inputs??

The Registrar is compliant to RFC2543 and has not yet become compliant to
that particular "MUST" within RFC3261.

The situation can occur if the REGISTER forked.  If a merged request was
observed, a 482 response might be returned.  If the merge was not detected,
the REGISTER might trigger a second deletion and return a 200 response.
During a race condition with a new REGISTER, this second removal might be
problematic.  I do not know if RFC3261 discusses this race condition or
provides a solution.



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From: Peter Macaulay <peter.macaulay@ZDSL.com>
Subject: Re: [Sip] PoC service indication
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One meaning of PoC is .. Push-To-Talk (PTT) over Cellular (PoC).

/PETER MACAULAY

At 07:48 AM 1/13/2005, girish wrote:
>Hi,
>    What do we mean by     PoC service indication  in a SIP Message. What 
> does it imply ?
>
>
>Girish Pai
>Bharti Telesoft Intl Pvt. Ltd.
>Airport Road
>Bangalore
>_______________________________________________
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Date: Thu, 13 Jan 2005 19:50:54 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Re: question on draft-ietf-sip-sctp-05.txt
References: <41E662B6.9030002@ericsson.com>
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Cullen,

please, have a look at Sections 4, 6, and 8 of the new revision of the 
draft (you can fetch it from the link below) and let me know if they 
address your concerns:

http://users.piuha.net/gonzalo/temp/draft-ietf-sip-sctp-06.txt

Thanks,

Gonzalo

Gonzalo Camarillo wrote:

> Cullen,
> 
> thanks for pointing this out. This problem was caused by the 
> documentation process we followed. The original idea was to define 
> everything in the SIP over SCTP draft, but then some parts were moved to
> 
> RFC 3261 (e.g., the definition of the token for the via header and the 
> value for the transport parameter) and to RFC 3263 (e.g., the definition
> 
> of the SIP+D2S service). We obviously forgot to define (somewhere) both 
> tokens and the new service for TLS over SCTP.
> 
> I will prepare a new revision of the draft including these definitions.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> Cullen Jennings wrote:
> 
> 
>>What do the following two lines look like if I am using TLS over SCTP?
>>
>>      sip:Bob.Johnson@example.com;transport=sctp
>>      Via: SIP/2.0/SCTP ws1234.example.com:5060
>>
>>And could you explicitly mention what the SRV lookups are like
> 
> (particularly
> 
>>with TLS and SCTP)
>>
>>Thanks, Cullen
>>
> 
> 

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


_______________________________________________
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Subject: Re: [Sip] Re: question on draft-ietf-sip-sctp-05.txt
From: Cullen Jennings <fluffy@cisco.com>
To: "gonzalo.camarillo@ericsson.com" <Gonzalo.Camarillo@ericsson.com>
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Yes - that dealt with everything I was worried about. Looks good. Thanks.

On 1/13/05 9:50 AM, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
wrote:

> Cullen,
> 
> please, have a look at Sections 4, 6, and 8 of the new revision of the
> draft (you can fetch it from the link below) and let me know if they
> address your concerns:
> 
> http://users.piuha.net/gonzalo/temp/draft-ietf-sip-sctp-06.txt
> 
> Thanks,
> 
> Gonzalo
> 
> Gonzalo Camarillo wrote:
> 
>> Cullen,
>> 
>> thanks for pointing this out. This problem was caused by the
>> documentation process we followed. The original idea was to define
>> everything in the SIP over SCTP draft, but then some parts were moved to
>> 
>> RFC 3261 (e.g., the definition of the token for the via header and the
>> value for the transport parameter) and to RFC 3263 (e.g., the definition
>> 
>> of the SIP+D2S service). We obviously forgot to define (somewhere) both
>> tokens and the new service for TLS over SCTP.
>> 
>> I will prepare a new revision of the draft including these definitions.
>> 
>> Thanks,
>> 
>> Gonzalo
>> 
>> 
>> Cullen Jennings wrote:
>> 
>> 
>>> What do the following two lines look like if I am using TLS over SCTP?
>>> 
>>>      sip:Bob.Johnson@example.com;transport=sctp
>>>      Via: SIP/2.0/SCTP ws1234.example.com:5060
>>> 
>>> And could you explicitly mention what the SRV lookups are like
>> 
>> (particularly
>> 
>>> with TLS and SCTP)
>>> 
>>> Thanks, Cullen
>>> 
>> 
>> 


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This list is for NEW development of the core SIP Protocol
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From: =?gb2312?B?1uzA1w==?= <lei.zhu@huawei.com>
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hello all
Can I ask some questions about RFC3261 for discuessing?

Zhu Lei
Huawei Technologies Co.,LTD.
Telephone: 01082882838
E-mail Addr: lei.zhu@huawei.com

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<DIV><FONT size=2>hello all</FONT></DIV>
<DIV><FONT size=2>Can I ask some questions about RFC3261 for 
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<DIV><FONT size=2>Zhu Lei<BR>Huawei Technologies Co.,LTD.<BR>Telephone: 
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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] ask SIP questions first time
Date: Fri, 14 Jan 2005 15:05:31 -0600
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On Jan 13, 2005, at 9:22 PM, $B<kMk(B wrote:

> hello all
> Can I ask some questions about RFC3261 for discuessing?
>  
> Zhu Lei
> Huawei Technologies Co.,LTD.
> Telephone: 01082882838
> E-mail Addr: lei.zhu@huawei.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

The general recommendation for discussing current(as opposed to "being 
developed") SIP specifications is to use the sip-implementors list at:

sip-implementors@cs.columbia.edu

The address for this list is conveniently attached to the footer of 
messages on the SIP mailing list by our mail server.

If you're suggesting requirements for changing or extending RFC3261, 
then probably the SIPPING list is right.

If you think you've found an error in RFC 3261 that needs to be 
reported, then this is the right list.

--
Dean Willis
SIP Co-Chair


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: Minor anomaly in RFC 3261 Registration example?
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Folks:

Not sure if this minor anomaly has been reported before and is already
logged in SIP bugzilla (which seems to be down right now). Apologies if
this is a repetition.


Section 10.3 of RFC 3261 states the following:

      8. The registrar returns a 200 (OK) response.  The response MUST
         contain Contact header field values enumerating all current
         bindings.  Each Contact value MUST feature an "expires"
         parameter indicating its expiration interval chosen by the
         registrar.  The response SHOULD include a Date header field.


Section 24.1 of RFC 3261 gives an example of 200 OK for a REGISTER as
the following:

        SIP/2.0 200 OK
        Via: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=3Dz9hG4bKnashds7
         ;received=3D192.0.2.4
        To: Bob <sip:bob@biloxi.com>;tag=3D2493k59kd
        From: Bob <sip:bob@biloxi.com>;tag=3D456248
        Call-ID: 843817637684230@998sdasdh09
        CSeq: 1826 REGISTER
        Contact: <sip:bob@192.0.2.4>
        Expires: 7200
        Content-Length: 0


The "expires" parameter in the 200 OK's Contact header is missing in
this example whereas it's a MUST in the text.

Thanks,
Rajnish

Email     mailto:rjain@excelswitching.com=20
SIP       sip:rjain@voip.excelswitching.com
PSTN      508-862-3215=20

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

<P><FONT SIZE=3D2 FACE=3D"Courier New">Not sure if this minor anomaly =
has been reported before and is already logged in SIP bugzilla (which =
seems to be down right now). Apologies if this is a =
repetition.</FONT></P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Section 10.3 of RFC 3261 states =
the following:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 8. =
The registrar returns a 200 (OK) response.&nbsp; The response =
MUST</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; contain Contact =
header field values enumerating all current</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; bindings.&nbsp; =
Each Contact value MUST feature an &quot;expires&quot;</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; parameter =
indicating its expiration interval chosen by the</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; registrar.&nbsp; =
The response SHOULD include a Date header field.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Section 24.1 of RFC 3261 gives an =
example of 200 OK for a REGISTER as the following:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SIP/2.0 200 OK</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP =
bobspc.biloxi.com:5060;branch=3Dz9hG4bKnashds7</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
;received=3D192.0.2.4</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; To: Bob =
&lt;sip:bob@biloxi.com&gt;;tag=3D2493k59kd</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; From: Bob =
&lt;sip:bob@biloxi.com&gt;;tag=3D456248</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Call-ID: =
843817637684230@998sdasdh09</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CSeq: 1826 =
REGISTER</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Contact: =
&lt;sip:bob@192.0.2.4&gt;</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Expires: 7200</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Content-Length: 0</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The &quot;expires&quot; parameter =
in the 200 OK's Contact header is missing in this example whereas it's a =
MUST in the text.</FONT>
</P>

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

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

<P><FONT SIZE=3D2 FACE=3D"Courier New">Email&nbsp;&nbsp;&nbsp;&nbsp; =
</FONT><A HREF=3D"mailto:rjain@excelswitching.com"><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Courier =
New">mailto:rjain@excelswitching.com</FONT></U></A><FONT SIZE=3D2 =
FACE=3D"Courier New"><U> </U></FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">SIP&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT><U> <FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Courier =
New">sip:rjain@voip.excelswitching.com</FONT></U>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">PSTN&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 508-862-3215</FONT>=20
</P>

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From: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Subject: RE: [Sip] PoC service indication
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Hi
    PoC service indication in SIP INVITE message provides the originating user an early indication of start to speak even before invited users accept the invitation.
 
 

Regards 
Ranjit 



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of girish
Sent: Thursday, January 13, 2005 6:18 PM
To: sip-implementors@cs.columbia.edu; sip@vovida.org; 'sip-ietf'
Subject: [Sip] PoC service indication


Hi,
   What do we mean by     PoC service indication  in a SIP Message. What does it imply ?
 
 
Girish Pai
Bharti Telesoft Intl Pvt. Ltd.
Airport Road
Bangalore


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To: "Avasarala Ranjit-A20990" <ranjit@motorola.com>,
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References: <653138C25D8AD6118292000347080A3714EA4E02@zin05exm02.corp.mot.com>
Subject: Re: [Sip] PoC service indication
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MessageHi Ranjit,
  Thanks for the explanation on the poc service indication . Could you =
please tell me how exactly is it indicated in the SIP Message .What is =
the keyword used to identify the poc service indication


Girish Pai
Bharti Telesoft Intl Pvt. Ltd.
Airport Road
Bangalore

  ----- Original Message -----=20
  From: Avasarala Ranjit-A20990=20
  To: girish ; sip-implementors@cs.columbia.edu ; sip@vovida.org ; =
'sip-ietf'=20
  Sent: Monday, January 17, 2005 12:13 PM
  Subject: RE: [Sip] PoC service indication


  Hi
      PoC service indication in SIP INVITE message provides the =
originating user an early indication of start to speak even before =
invited users accept the invitation.


  Regards=20
  Ranjit=20




  -----Original Message-----
  From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
girish
  Sent: Thursday, January 13, 2005 6:18 PM
  To: sip-implementors@cs.columbia.edu; sip@vovida.org; 'sip-ietf'
  Subject: [Sip] PoC service indication


  Hi,
     What do we mean by     PoC service indication  in a SIP Message. =
What does it imply ?


  Girish Pai
  Bharti Telesoft Intl Pvt. Ltd.
  Airport Road
  Bangalore

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1226" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial>Hi Ranjit,</FONT></DIV>
<DIV><FONT face=3DArial>&nbsp; Thanks for the explanation on the poc =
service=20
indication . Could you please tell me how exactly is it indicated in the =
SIP=20
Message .What is the keyword used to identify the poc service=20
indication</FONT></DIV>
<DIV><FONT face=3DArial></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>Girish Pai<BR>Bharti Telesoft Intl Pvt. Ltd.<BR>Airport=20
Road<BR>Bangalore<BR></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Dranjit@motorola.com =
href=3D"mailto:ranjit@motorola.com">Avasarala=20
  Ranjit-A20990</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
  title=3Dgirish.pai@bhartitelesoft.com=20
  href=3D"mailto:girish.pai@bhartitelesoft.com">girish</A> ; <A=20
  title=3Dsip-implementors@cs.columbia.edu=20
  =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
  ; <A title=3Dsip@vovida.org =
href=3D"mailto:sip@vovida.org">sip@vovida.org</A> ; <A=20
  title=3Dsip@ietf.org href=3D"mailto:sip@ietf.org">'sip-ietf'</A> =
</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Monday, January 17, 2005 =
12:13=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> RE: [Sip] PoC service=20
  indication</DIV>
  <DIV><BR></DIV>
  <DIV><SPAN class=3D450544006-17012005><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Hi</FONT></SPAN></DIV>
  <DIV><SPAN class=3D450544006-17012005><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>&nbsp;&nbsp;&nbsp; PoC service indication in SIP INVITE =
message=20
  provides the originating user an early indication of&nbsp;start =
to&nbsp;speak=20
  even before invited users accept the invitation.</FONT></SPAN></DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV><!-- Converted from text/rtf format -->
  <P><SPAN lang=3Den-us><I><FONT face=3DArial color=3D#008080=20
  size=3D2>Regards</FONT></I></SPAN> <BR><SPAN lang=3Den-us><I><FONT =
face=3DArial=20
  color=3D#008080 size=3D2>Ranjit</FONT></I></SPAN> </P><BR><BR>
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> <A=20
  href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of =
</B>girish<BR><B>Sent:</B>=20
  Thursday, January 13, 2005 6:18 PM<BR><B>To:</B> <A=20
  =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>;=20
  <A href=3D"mailto:sip@vovida.org">sip@vovida.org</A>;=20
  'sip-ietf'<BR><B>Subject:</B> [Sip] PoC service=20
indication<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial>Hi,</FONT></DIV>
  <DIV><FONT face=3DArial>&nbsp;&nbsp; What do we mean by <FONT =
size=3D2><FONT=20
  face=3D"Times New Roman">&nbsp;&nbsp;&nbsp; </FONT></FONT><SPAN =
lang=3DEN-GB=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Times New Roman'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-GB; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA"><FONT=20
  face=3DVerdana size=3D3>PoC service indication&nbsp; in a SIP Message. =
What does=20
  it imply ?</FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D3><SPAN lang=3DEN-GB=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Times New Roman'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-GB; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA"></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D3><SPAN lang=3DEN-GB=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Times New Roman'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-GB; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA"></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial>Girish Pai<BR>Bharti Telesoft Intl Pvt. =
Ltd.<BR>Airport=20
  Road<BR>Bangalore<BR></FONT></DIV></BLOCKQUOTE></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0055580869==--





From sip-bounces@ietf.org  Mon Jan 17 04:58:30 2005
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From: Avasarala Ranjit-A20990 <ranjit@motorola.com>
To: girish <girish.pai@bhartitelesoft.com>, sip-implementors@cs.columbia.edu,
        sip@vovida.org, "'sip-ietf'" <sip@ietf.org>
Subject: RE: [Sip] PoC service indication
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This is done using 'p' headers. Refer this draft for more details:  <http://ietfreport.isoc.org/idref/draft-allen-sipping-poc-p-headers/rfcmarkup?repository=/away/ietf&url=/away/ietf/all-ids/draft-allen-sipping-poc-p-headers-00.txt> draft-allen-sipping-poc-p-headers-00.txt

 
 

Regards 
Ranjit 



-----Original Message-----
From: girish [mailto:girish.pai@bhartitelesoft.com] 
Sent: Monday, January 17, 2005 2:21 PM
To: Avasarala Ranjit-A20990; sip-implementors@cs.columbia.edu; sip@vovida.org; 'sip-ietf'
Subject: Re: [Sip] PoC service indication


Hi Ranjit,
  Thanks for the explanation on the poc service indication . Could you please tell me how exactly is it indicated in the SIP Message .What is the keyword used to identify the poc service indication
 
 
Girish Pai
Bharti Telesoft Intl Pvt. Ltd.
Airport Road
Bangalore


----- Original Message ----- 
From: Avasarala  <mailto:ranjit@motorola.com> Ranjit-A20990 
To: girish <mailto:girish.pai@bhartitelesoft.com>  ; sip-implementors@cs.columbia.edu <mailto:sip-implementors@cs.columbia.edu>  ; sip@vovida.org <mailto:sip@vovida.org>  ; 'sip-ietf' <mailto:sip@ietf.org>  
Sent: Monday, January 17, 2005 12:13 PM
Subject: RE: [Sip] PoC service indication

Hi
    PoC service indication in SIP INVITE message provides the originating user an early indication of start to speak even before invited users accept the invitation.
 
 

Regards 
Ranjit 



-----Original Message-----
From: sip-bounces@ietf.org <mailto:sip-bounces@ietf.org>  [mailto:sip-bounces@ietf.org] On Behalf Of girish
Sent: Thursday, January 13, 2005 6:18 PM
To: sip-implementors@cs.columbia.edu <mailto:sip-implementors@cs.columbia.edu> ; sip@vovida.org <mailto:sip@vovida.org> ; 'sip-ietf'
Subject: [Sip] PoC service indication


Hi,
   What do we mean by     PoC service indication  in a SIP Message. What does it imply ?
 
 
Girish Pai
Bharti Telesoft Intl Pvt. Ltd.
Airport Road
Bangalore



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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0988500669==--



From sip-bounces@ietf.org  Tue Jan 18 00:24:09 2005
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Thread-Topic: Address-of-record not valid for the domain in the Request-URI -
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Hello.
=20
As per RFC 3261 section 10.3 Registration, REGISTRAR on receipt of a =
REGISTER request including a To header from which the extracted =
address-of-record is not valid for the domain in the Request-URI, sends =
a Request
Failure (404 Not Found) response.
=20
Is there an example available for this Scenario??
=20
Regds
karthikeyan.k

=20

=20
=20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML =
DIR=3Dltr><HEAD><META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1"></HEAD><BODY>=0A=
<DIV><FONT face=3D'Arial' color=3D#000000 size=3D2>Hello.</FONT></DIV>=0A=
<DIV><FONT face=3DArial color=3D#000000 size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial color=3D#000000 size=3D2>As per RFC 3261 section =
10.3 =0A=
Registration,&nbsp;REGISTRAR on receipt of a REGISTER request including =
a To =0A=
header from which the extracted address-of-record is not valid for the =
domain in =0A=
the Request-URI, sends a Request<BR>Failure (404 Not Found) =0A=
response.</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Is there an example available for this =0A=
Scenario??</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Regds</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>karthikeyan.k</FONT></DIV>=0A=
<DIV>=0A=
<P class=3DMsoNormal =0A=
style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt =
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1323251936==--



From sip-bounces@ietf.org  Tue Jan 18 10:52:09 2005
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Subject: RE: [Sip] FW: [Sipping]
	I-D	ACTION:draft-ietf-sipping-dialog-package-05.txt
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Robert,

I have added comments in-line below but first a comment about
SIPits and SIP software in general.  It seems that one of our
disconnects is that you, quite rightly, point out that partner
implementations shouldn't <insert protocol misbehavior of your
choice here>.  You then use this to argue that my software
must therefore act in one specific manner based on this
assumption that the partner will play nice.

Unfortunately, one of the things we hit often is partner
implementations that are far from perfect and do all manner
of bizarre things.  Whilst we always point these out to our
customers, we have to bow to their response that "we don't
care what implementation X does, why can't you just keep the
call up"?

Now proving that most implementations at SIPits "play nice"
does not mean that the real world is nice.  Many
implementations never seem to go to SIPits and even different
software levels of implementations that do, don't always get
things right.  This means that our implementation has to
be as resilient as possible against bad SIP.

I believe Jon Postel said "be conservative in what you send,
and liberal in what you receive".  I interpret this as
"obey the protocol when you send, but don't be surprised if
other don't".  I still don't agree that a blanket 481
means "the dialog has gone" because that is not what
the definitions in RFCs 3262 and 3265 say.  They have
mangled the definition of 481 to relate to transactions
and subscriptions respectively.

OK, so where does this leave us?  I think we should progress as
follows.

1. 481 should not be used to indicate "what was this errant
   PRACK for"?  A new status code should be defined meaning
   "the dialog was there but I have no outstanding reliable
   provisional response - your PRACK is in error".  RFC3262
   must be amended accordingly.  This new status code does
   not cause the dialog to end.

2. 481 should not be used to indicate "this NOTIFY was within
   a valid dialog but I know nothing about this subscription".
   A new status code should be defined for this.  RFC3265 must
   be amended accordingly.  This new status code does not
   cause the dialog to end.

3. 481 means one thing, and one thing only.  The dialog is not
   known and as far as everyone is concerned, it no longer
   exists.

For a period of time, 481s may be sent in error and may cause
the odd call to be lost but we will live with this.  Long term
we have a single, clear meaning of 481 and we can get back to
more interesting aspects of the protocol.

Does this sound like a reasonable way forward?  If so, I
believe the next step raise this through more specific
postings to the WG for further discussion with a view to
getting the final decision possible raised in BUGZILLA.

Do you agree?

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


-----Original Message-----
From: Robert Sparks [mailto:rjsparks@nostrum.com]
Sent: 18 January 2005 01:14
To: SIP Public Folder (E-mail)
Subject: Re: [Sip] FW: [Sipping] I-D
ACTION:draft-ietf-sipping-dialog-package-05.txt



On Jan 17, 2005, at 6:48 PM, Paul Kyzivat wrote:

>
>
> Robert Sparks wrote:
>> On Jan 17, 2005, at 7:53 AM, Paul D.Smith wrote:
>
>>> However, RFCs 3262 and 3265 have changed the use of 481 status =
codes.
>>> RFC3262 states:
>>>
>>>    If a PRACK request is received by the UA core that does not match =

>>> any
>>>    unacknowledged reliable provisional response, the UAS MUST=20
>>> respond to
>>>    the PRACK with a 481 response.
>>>
>>> RFC3265 states:
>>>
>>>                               Designers of such mechanisms are also
>>>    warned to make a distinction between sending a NOTIFY message to =
a
>>>    subscriber who is aware of the subscription, and sending a NOTIFY
>>>    message to an unsuspecting node.  The latter behavior is invalid, =

>>> and
>>>    MUST receive a "481 Subscription does not exist" response (unless
>>>    some other 400- or 500-class error code is more applicable), as
>>>    described in section 3.2.4.
>>>
>>> In both of these, the 481 is _NOT_ indicating that the dialog is=20
>>> flawed,
>>> rather it is a transaction or subscription specific error.
>> A subscription _is_ a usage in a dialog.
>
> One of potentially several usages.
>
>> In both of these cases, the dialog should be destroyed.
>
> Why? That might break another usage.

Because the _DIALOG STATE_ is no longer there on the
peer you touched. Things like CSeq are not there anymore.
Any other usage is already broken.

[PDS] What is wrong with CSeq?  The previous request failed but
so what?  My next reINVITE etc. simply increments the CSeq and
on the dialog goes.

>
> Each usage has its own way of timing out, etc. anyway.
> It makes much more sense to me to simply destroy the corresponding=20
> usage. If there are then no usages, the dialog can be destroyed.
CSeq.
Remote target
Local target

Do I need to go on?

>
> Of course I know you will bring up that multiple usages per dialog is=20
> evil, but there are still use cases for them.
>
>> I think you are thinking that if you are working along happily with
>> a dialog with identifier A and a message shows up with identifier B
>> then dialog A gets destroyed - this is clearly not the case.
>> What these requirement cover is when Element 1 thinks it is in
>> a dialog with Element 2, and Element 2 disagrees (for whatever=20
>> reason).
>> The 481 is the response that lets Element 1 know that things are in=20
>> trouble,
>> and they are in trouble at the dialog level.
>
> Yes, but it is rooted in the fact that 3261 created the concept of=20
> multiple usages per dialog, but mostly pretends that they aren't=20
> there.
>
> There is clearly a difference between:
> - I don't recognize this dialog id
> - I don't recognize this subscription on this dialog id
>
> But there is only one status for them both, and no contextual way to=20
> tell which is the intended meaning.
>
>>> Now consider my user, busily talking on their SIP phone.  The=20
>>> arrival of
>>> an errant PRACK or NOTIFY probably has no affect on their call.
>> A PRACK with dialog identifers B will never affect the dialog with=20
>> identifier A.
>> There is no "probably" here.
>
> I thought the intended case was a prack within the same dialog.
> In this case, PRACK is only meaningful in the context of an invite=20
> usage. So one might assume that getting an errant one implies=20
> something is wrong with your peer in the invite usage. So I can see=20
> killing the invite usage.
>
> But if there is another usage sharing the same dialog, then while they =

> are related (or else they wouldn't be sharing a dialog), they may not=20
> need to be fate shared in this case.
>
>>>   And if
>>> it does, then they are in a position to determine this and =
terminated
>>> the
>>> dialog.  But the combination of RFC3261, and your assertion that 481
>>> must
>>> mean "dialog is terminated" results in their call being dropped,=20
>>> despite
>>> there really being no reason why it should.
>> Have I seen where you misunderstood the intent? All these =
requirements
>> indicate that an element receiving a 481 with dialog identifier B=20
>> treats
>> dialog B as terminated. This says nothing about any other dialogs =
that
>> are in play at that endpoint.
>> If I haven't understood your concern yet, try again - preferably=20
>> showing an
>> example call-flow where you see a problem.
>
> OK, how about this:
>
> - A calls B (which perhaps is a conference focus)
>   this creates a dialog, and an invite usage.
>
> - within that dialog, A sends a REFER to B, to bring somebody
>   else into the call. This creates an implicit subscription
>   (usage) in the same dialog.
>
> - B starts sending NOTIFYs to A with refer status.
>
> - The refer completes and B sends A a final notify,
>   terminating the subscription.
>
> - A sends a 200 for the final notify. It also destroys
>   the usage. (But the dialog remains for the invite.)
A went non-standard on your second sentence. The
transaction will have to complete before it removes
the usage (the NOTIFY might get a 401 for example).

[PDS] I assume that this is because A should maintain=20
the transaction for a period which allows B to
retransmit NOTIFYs.  Surely though network interactions
might allow a late NOTIFY to arrive after A has given
up waiting?  Legitimately, A will now send a 481 meaning
"no such subcription", but the dialog is perfectly healthy.

>
> - The 200 is lost in the network.
>
> - Eventually, B times out the 200 and retransmits the
>   NOTIFY.
>
> - A gets the NOTIFY, but has no corresponding usage, so
>   responds with a 481.
A is broken.
>
> - When B gets the 481 it destroys the complete dialog,
>   taking down the call. WRONG!
No - B has no way to know that what it saw was because A
was broken, so it needs to behave as if A legitimately didn't
find that dialog state.

[PDS] As above, RFC3265 has changed the meaning of 481. It
states there is no subscription, not there is no dialog.

Can you create an example where one party hasn't
violated the protocol and terminating the dialog is
the wrong thing to do?

>
>>> I hope this has got across my concerns regarding "481 means=20
>>> terminated
>>> dialog".  I appreciate that at least some of this might be tidied up =

>>> in
>>> later versions of specifications, but SIP hardware is already using=20
>>> the
>>> current specifications, and will not adopt new behaviour for some=20
>>> time.
>>> Until that time, having all 481 responses cause dialogs to end will
>>> contribute to poor perception of SIP hardware as dialogs fail
>>> unnecessarily.
>> I disagree. This is something we've tested heavily at SIPIts. =
Existing
>> equipment is following the specifications, and the effect on the=20
>> applications
>> is correct.
>
> Well, maybe not every corner case.
>
> 	Paul


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Subject: [Sip] Sessions in B2BUA
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Dear All,

as I understood from RFC 3261 and from various posts of this list, a 
B2BUA is basically two SIP UAs with no clear definition of  how they 
interact or work together to manage sessions through the B2BUA.

Does the B2BUA terminate sessions completely, so sessions are completely 
independent on the two session legs? If yes, it means that all session 
related descriptors have to be changed. How does the B2BUA then match 
the two legs? Is there some parameter that remains unaltered? If, e.g., 
Call-ID would remain the same then it were easy to correlate the 
signaling legs, but otherwise the B2BUA has to maintain some kind of 
mapping to identify that certain legs belong to the same session.

Is this behavior specified somewhere?

Regards
Peter 


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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] FW: [Sipping]
	I-D	ACTION:draft-ietf-sipping-dialog-package-05.txt
Date: Tue, 18 Jan 2005 12:21:48 -0600
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You know that I am a strong champion of Postel's Maxim -
its a major point we drive home carefully at every SIPIt and
is the cornerstone of the open source work I'm involved in.

Everyone must work hard to be liberal in what they receive.
However, you also have to be aware of the line past which
not following the protocol yourself will do more damage.
This is a place where it will.

Building new protocol, as you suggest below, might be
warranted (I respond more favorably for your point on
SUBSCRIBE than I do on PRACK because I can think
of corner cases (network race conditions) where things
following protocol can get into this condition. I'm not
yet convinced that ignoring a PRACK to a provisional
response you didn't send is a sane thing to do.

There is definitely need for clarification through multiple
drafts, and I will make the appropriate bugzilla entries.

Other comments inline.

On Jan 18, 2005, at 9:37 AM, Paul D.Smith wrote:

> Robert,
>
> I have added comments in-line below but first a comment about
> SIPits and SIP software in general.  It seems that one of our
> disconnects is that you, quite rightly, point out that partner
> implementations shouldn't <insert protocol misbehavior of your
> choice here>.  You then use this to argue that my software
> must therefore act in one specific manner based on this
> assumption that the partner will play nice.
>
> Unfortunately, one of the things we hit often is partner
> implementations that are far from perfect and do all manner
> of bizarre things.  Whilst we always point these out to our
> customers, we have to bow to their response that "we don't
> care what implementation X does, why can't you just keep the
> call up"?
>
> Now proving that most implementations at SIPits "play nice"
> does not mean that the real world is nice.  Many
> implementations never seem to go to SIPits and even different
> software levels of implementations that do, don't always get
> things right.  This means that our implementation has to
> be as resilient as possible against bad SIP.
>
> I believe Jon Postel said "be conservative in what you send,
> and liberal in what you receive".  I interpret this as
> "obey the protocol when you send, but don't be surprised if
> other don't".  I still don't agree that a blanket 481
> means "the dialog has gone" because that is not what
> the definitions in RFCs 3262 and 3265 say.  They have
> mangled the definition of 481 to relate to transactions
> and subscriptions respectively.

I completely disagree with your interpretation of 3262 and
3265. Looking at the path you are pointing at below, there
may be semantics you would _like_ to have that those don't
ask for.

>
> OK, so where does this leave us?  I think we should progress as
> follows.
>
> 1. 481 should not be used to indicate "what was this errant
>    PRACK for"?  A new status code should be defined meaning
>    "the dialog was there but I have no outstanding reliable
>    provisional response - your PRACK is in error".  RFC3262
>    must be amended accordingly.  This new status code does
>    not cause the dialog to end.
>
> 2. 481 should not be used to indicate "this NOTIFY was within
>    a valid dialog but I know nothing about this subscription".
>    A new status code should be defined for this.  RFC3265 must
>    be amended accordingly.  This new status code does not
>    cause the dialog to end.
>
> 3. 481 means one thing, and one thing only.  The dialog is not
>    known and as far as everyone is concerned, it no longer
>    exists.
Nuance -
The 481 means the recipient of the request doesn't know the dialog,
and the sender should stop talking to the recipient about it.
>
> For a period of time, 481s may be sent in error and may cause
> the odd call to be lost but we will live with this.  Long term
> we have a single, clear meaning of 481 and we can get back to
> more interesting aspects of the protocol.
>
> Does this sound like a reasonable way forward?  If so, I
> believe the next step raise this through more specific
> postings to the WG for further discussion with a view to
> getting the final decision possible raised in BUGZILLA.
>
> Do you agree?
>
> 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: Robert Sparks [mailto:rjsparks@nostrum.com]
> Sent: 18 January 2005 01:14
> To: SIP Public Folder (E-mail)
> Subject: Re: [Sip] FW: [Sipping] I-D
> ACTION:draft-ietf-sipping-dialog-package-05.txt
>
>
>
> On Jan 17, 2005, at 6:48 PM, Paul Kyzivat wrote:
>
>>
>>
>> Robert Sparks wrote:
>>> On Jan 17, 2005, at 7:53 AM, Paul D.Smith wrote:
>>
>>>> However, RFCs 3262 and 3265 have changed the use of 481 status 
>>>> codes.
>>>> RFC3262 states:
>>>>
>>>>    If a PRACK request is received by the UA core that does not match
>>>> any
>>>>    unacknowledged reliable provisional response, the UAS MUST
>>>> respond to
>>>>    the PRACK with a 481 response.
>>>>
>>>> RFC3265 states:
>>>>
>>>>                               Designers of such mechanisms are also
>>>>    warned to make a distinction between sending a NOTIFY message to 
>>>> a
>>>>    subscriber who is aware of the subscription, and sending a NOTIFY
>>>>    message to an unsuspecting node.  The latter behavior is invalid,
>>>> and
>>>>    MUST receive a "481 Subscription does not exist" response (unless
>>>>    some other 400- or 500-class error code is more applicable), as
>>>>    described in section 3.2.4.
>>>>
>>>> In both of these, the 481 is _NOT_ indicating that the dialog is
>>>> flawed,
>>>> rather it is a transaction or subscription specific error.
>>> A subscription _is_ a usage in a dialog.
>>
>> One of potentially several usages.
>>
>>> In both of these cases, the dialog should be destroyed.
>>
>> Why? That might break another usage.
>
> Because the _DIALOG STATE_ is no longer there on the
> peer you touched. Things like CSeq are not there anymore.
> Any other usage is already broken.
>
> [PDS] What is wrong with CSeq?  The previous request failed but
> so what?  My next reINVITE etc. simply increments the CSeq and
> on the dialog goes.
>
>>
>> Each usage has its own way of timing out, etc. anyway.
>> It makes much more sense to me to simply destroy the corresponding
>> usage. If there are then no usages, the dialog can be destroyed.
> CSeq.
> Remote target
> Local target
>
> Do I need to go on?
>
>>
>> Of course I know you will bring up that multiple usages per dialog is
>> evil, but there are still use cases for them.
>>
>>> I think you are thinking that if you are working along happily with
>>> a dialog with identifier A and a message shows up with identifier B
>>> then dialog A gets destroyed - this is clearly not the case.
>>> What these requirement cover is when Element 1 thinks it is in
>>> a dialog with Element 2, and Element 2 disagrees (for whatever
>>> reason).
>>> The 481 is the response that lets Element 1 know that things are in
>>> trouble,
>>> and they are in trouble at the dialog level.
>>
>> Yes, but it is rooted in the fact that 3261 created the concept of
>> multiple usages per dialog, but mostly pretends that they aren't
>> there.
>>
>> There is clearly a difference between:
>> - I don't recognize this dialog id
>> - I don't recognize this subscription on this dialog id
>>
>> But there is only one status for them both, and no contextual way to
>> tell which is the intended meaning.
>>
>>>> Now consider my user, busily talking on their SIP phone.  The
>>>> arrival of
>>>> an errant PRACK or NOTIFY probably has no affect on their call.
>>> A PRACK with dialog identifers B will never affect the dialog with
>>> identifier A.
>>> There is no "probably" here.
>>
>> I thought the intended case was a prack within the same dialog.
>> In this case, PRACK is only meaningful in the context of an invite
>> usage. So one might assume that getting an errant one implies
>> something is wrong with your peer in the invite usage. So I can see
>> killing the invite usage.
>>
>> But if there is another usage sharing the same dialog, then while they
>> are related (or else they wouldn't be sharing a dialog), they may not
>> need to be fate shared in this case.
>>
>>>>   And if
>>>> it does, then they are in a position to determine this and 
>>>> terminated
>>>> the
>>>> dialog.  But the combination of RFC3261, and your assertion that 481
>>>> must
>>>> mean "dialog is terminated" results in their call being dropped,
>>>> despite
>>>> there really being no reason why it should.
>>> Have I seen where you misunderstood the intent? All these 
>>> requirements
>>> indicate that an element receiving a 481 with dialog identifier B
>>> treats
>>> dialog B as terminated. This says nothing about any other dialogs 
>>> that
>>> are in play at that endpoint.
>>> If I haven't understood your concern yet, try again - preferably
>>> showing an
>>> example call-flow where you see a problem.
>>
>> OK, how about this:
>>
>> - A calls B (which perhaps is a conference focus)
>>   this creates a dialog, and an invite usage.
>>
>> - within that dialog, A sends a REFER to B, to bring somebody
>>   else into the call. This creates an implicit subscription
>>   (usage) in the same dialog.
>>
>> - B starts sending NOTIFYs to A with refer status.
>>
>> - The refer completes and B sends A a final notify,
>>   terminating the subscription.
>>
>> - A sends a 200 for the final notify. It also destroys
>>   the usage. (But the dialog remains for the invite.)
> A went non-standard on your second sentence. The
> transaction will have to complete before it removes
> the usage (the NOTIFY might get a 401 for example).
>
> [PDS] I assume that this is because A should maintain
> the transaction for a period which allows B to
> retransmit NOTIFYs.  Surely though network interactions
> might allow a late NOTIFY to arrive after A has given
> up waiting?  Legitimately, A will now send a 481 meaning
> "no such subcription", but the dialog is perfectly healthy.
>
>>
>> - The 200 is lost in the network.
>>
>> - Eventually, B times out the 200 and retransmits the
>>   NOTIFY.
>>
>> - A gets the NOTIFY, but has no corresponding usage, so
>>   responds with a 481.
> A is broken.
>>
>> - When B gets the 481 it destroys the complete dialog,
>>   taking down the call. WRONG!
> No - B has no way to know that what it saw was because A
> was broken, so it needs to behave as if A legitimately didn't
> find that dialog state.
>
> [PDS] As above, RFC3265 has changed the meaning of 481. It
> states there is no subscription, not there is no dialog.
>
> Can you create an example where one party hasn't
> violated the protocol and terminating the dialog is
> the wrong thing to do?
>
>>
>>>> I hope this has got across my concerns regarding "481 means
>>>> terminated
>>>> dialog".  I appreciate that at least some of this might be tidied up
>>>> in
>>>> later versions of specifications, but SIP hardware is already using
>>>> the
>>>> current specifications, and will not adopt new behaviour for some
>>>> time.
>>>> Until that time, having all 481 responses cause dialogs to end will
>>>> contribute to poor perception of SIP hardware as dialogs fail
>>>> unnecessarily.
>>> I disagree. This is something we've tested heavily at SIPIts. 
>>> Existing
>>> equipment is following the specifications, and the effect on the
>>> applications
>>> is correct.
>>
>> Well, maybe not every corner case.
>>
>> 	Paul
>


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From sip-bounces@ietf.org  Tue Jan 18 15:03:00 2005
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Subject: Re: [Sip] Sessions in B2BUA
Date: Tue, 18 Jan 2005 13:53:11 -0600
To: =?ISO-8859-1?Q?P=E9ter_Barta?= <peter.barta@ericsson.com>
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On Jan 18, 2005, at 11:56 AM, P=E9ter Barta wrote:

> Dear All,
>
> as I understood from RFC 3261 and from various posts of this list, a=20=

> B2BUA is basically two SIP UAs with no clear definition of  how they=20=

> interact or work together to manage sessions through the B2BUA.
>
That's it precisely.

> Does the B2BUA terminate sessions completely, so sessions are=20
> completely independent on the two session legs? If yes, it means that=20=

> all session related descriptors have to be changed. How does the B2BUA=20=

> then match the two legs? Is there some parameter that remains=20
> unaltered? If, e.g., Call-ID would remain the same then it were easy=20=

> to correlate the signaling legs, but otherwise the B2BUA has to=20
> maintain some kind of mapping to identify that certain legs belong to=20=

> the same session.
>

That's up to the B2BUA. Some "hide" the two sessions from each other,=20
perhaps even masking or altering the identities expressed (an=20
"anonymizer") function. Others try to preserve higher-order=20
transparency functions, acting as much like a proxy as they can while=20
still breaking the rules of being a proxy.

> Is this behavior specified somewhere?
>

No.

Individual B2BUA implementations may specify their own behavior. This=20
behavior will be contingent on the purpose and mission of the specific=20=

B2BUA, as well as the operational requirements, design philosophy, and=20=

toolset used for implementation, and quite possibly on the intoxication=20=

level of the developer who writes the code.

This question keeps coming up. It's no more possible to specify what=20
all B2BUAs do than it is to specify what all possible "C" programs do.=20=

It's an unbounded set.

It MIGHT be possible to learn from the world of object oriented=20
programming and talk about "B2BUA patterns", meaning archetypal=20
behavior sets that might be used to solve certain classes of problems.=20=

This would probably be an interesting doctoral problem for someone who=20=

is also interested in software engineering and systems modeling. But=20
I'm not aware of any serious work ongoing on the topic.


--
Dean


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From sip-bounces@ietf.org  Tue Jan 18 15:34:01 2005
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From: Nils Ohlmeier <nils.ohlmeier@snom.de>
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Subject: [Sip] Running ESC and proxy together
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Hello,

section 7. of RFC3903 states that the ESC should include PUBLISH in the Allow 
header of an OPTIONS reply. But according to RFC3261 a proxy SHOULD omit the 
Allow header because it usually is able to route any request method. So how 
should a UAS signal that it accepts PUBLISH as ESC but is also able to route 
such requests?
Or is it simply not desirable to run an ESC and a proxy together and thus we 
always have to include an "ESC target option" in every UA?
Root of the problem: I do not see any other option to check for RFC3903 
support then using OPTIONS.

Regards
  Nils Ohlmeier

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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		: An Extension to the Session Initiation Protocol 
			  for Request History Information
	Author(s)	: M. Barnes
	Filename	: draft-ietf-sip-history-info-06.txt
	Pages		: 48
	Date		: 2005-1-18
	
This document defines a standard mechanism for capturing the history 
   information associated with a SIP request.  This capability enables 
   many enhanced services by providing the information as to how and why 
   a call arrives at a specific application or user.  This document 
   defines a new optional SIP header, History-Info, for capturing the 
   history information in requests.

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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Subject: FW: [Sip] I-D ACTION:draft-ietf-sip-history-info-06.txt
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Hi all,

Just to let you know the context of this new version;  I received a set of
comments just after last call ended, which were not incorporated in the -05
version.  They were primarily in the nit category of tightening the
specification in preparation for IETF LC/IESG review.  The changes are
summarized in detail at the back of the document, with the largest change
being removing the paragraphs in the security section describing the
alternatives such as adding message bodies, AIB and securing inserted
information documents.  There were no normative changes. The text that was
removed had been previously in section 3 (in the -03 version) and had been
moved in the -03 to -04 updates to section 6, my thoughts at the time being
that it was still useful for informational purposes.  However, since it
doesn't impact or add any normative value to the History-Info header at this
time, it's appropriate and sensible to remove and thus also remove some
informational references that are likely to be outdated by the time the
document is published as an RFC.

Of course, if you do review and see any errors that I may have inadvertently
introduced in this last round of edits, please let me know.

Thanks,
Mary.


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Internet-Drafts@ietf.org
Sent: Tuesday, January 18, 2005 2:38 PM
To: i-d-announce@ietf.org
Cc: sip@ietf.org
Subject: [Sip] I-D ACTION:draft-ietf-sip-history-info-06.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		: An Extension to the Session Initiation Protocol 
			  for Request History Information
	Author(s)	: M. Barnes
	Filename	: draft-ietf-sip-history-info-06.txt
	Pages		: 48
	Date		: 2005-1-18
	
This document defines a standard mechanism for capturing the history 
   information associated with a SIP request.  This capability enables 
   many enhanced services by providing the information as to how and why 
   a call arrives at a specific application or user.  This document 
   defines a new optional SIP header, History-Info, for capturing the 
   history information in requests.

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

To remove yourself from the I-D Announcement list, send a message to 
i-d-announce-request@ietf.org with the word unsubscribe in the body of the
message.  
You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce 
to change your subscription settings.


Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sip-history-info-06.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:
	"FILE /internet-drafts/draft-ietf-sip-history-info-06.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.


------_=_NextPart_001_01C4FDA3.AE4AFD2F
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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charset=3Dus-ascii">
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<BODY>

<P><FONT SIZE=3D2>Hi all,</FONT>
</P>

<P><FONT SIZE=3D2>Just to let you know the context of this new =
version;&nbsp; I received a set of comments just after last call ended, =
which were not incorporated in the -05 version.&nbsp; They were =
primarily in the nit category of tightening the specification in =
preparation for IETF LC/IESG review.&nbsp; The changes are summarized =
in detail at the back of the document, with the largest change being =
removing the paragraphs in the security section describing the =
alternatives such as adding message bodies, AIB and securing inserted =
information documents.&nbsp; There were no normative changes. The text =
that was removed had been previously in section 3 (in the -03 version) =
and had been moved in the -03 to -04 updates to section 6, my thoughts =
at the time being that it was still useful for informational =
purposes.&nbsp; However, since it doesn't impact or add any normative =
value to the History-Info header at this time, it's appropriate and =
sensible to remove and thus also remove some informational references =
that are likely to be outdated by the time the document is published as =
an RFC.</FONT></P>

<P><FONT SIZE=3D2>Of course, if you do review and see any errors that I =
may have inadvertently introduced in this last round of edits, please =
let me know.</FONT></P>

<P><FONT SIZE=3D2>Thanks,</FONT>
<BR><FONT SIZE=3D2>Mary.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On Behalf Of Internet-Drafts@ietf.org</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, January 18, 2005 2:38 PM</FONT>
<BR><FONT SIZE=3D2>To: i-d-announce@ietf.org</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [Sip] I-D =
ACTION:draft-ietf-sip-history-info-06.txt</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>A New Internet-Draft is available from the on-line =
Internet-Drafts directories.</FONT>
<BR><FONT SIZE=3D2>This draft is a work item of the Session Initiation =
Protocol Working Group of the IETF.</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
An Extension to the Session Initiation Protocol </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&nbsp; for =
Request History Information</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : M. =
Barnes</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-ietf-sip-history-info-06.txt</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
48</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2005-1-18</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>This document defines a standard mechanism for =
capturing the history </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; information associated with a SIP =
request.&nbsp; This capability enables </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; many enhanced services by providing the =
information as to how and why </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; a call arrives at a specific =
application or user.&nbsp; This document </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; defines a new optional SIP header, =
History-Info, for capturing the </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; history information in requests.</FONT>
</P>

<P><FONT SIZE=3D2>A URL for this Internet-Draft is:</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-=
06.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sip-his=
tory-info-06.txt</A></FONT>
</P>

<P><FONT SIZE=3D2>To remove yourself from the I-D Announcement list, =
send a message to </FONT>
<BR><FONT SIZE=3D2>i-d-announce-request@ietf.org with the word =
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</P>
<BR>

<P><FONT SIZE=3D2>Internet-Drafts are also available by anonymous FTP. =
Login with the username</FONT>
<BR><FONT SIZE=3D2>&quot;anonymous&quot; and a password of your e-mail =
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<BR><FONT SIZE=3D2>type &quot;cd internet-drafts&quot; and then</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&quot;get =
draft-ietf-sip-history-info-06.txt&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>A list of Internet-Drafts directories can be found =
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<BR>

<P><FONT SIZE=3D2>Internet-Drafts can also be obtained by =
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</P>

<P><FONT SIZE=3D2>Send a message to:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>mailserv@ietf.org.</FONT>
<BR><FONT SIZE=3D2>In the body type:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;FILE =
/internet-drafts/draft-ietf-sip-history-info-06.txt&quot;.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>NOTE:&nbsp;&nbsp; The mail server at ietf.org can =
return the document in</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>MIME-encoded form by using the &quot;mpack&quot; =
utility.&nbsp; To use this</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>feature, =
insert the command &quot;ENCODING mime&quot; before the =
&quot;FILE&quot;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>command.&nbsp; To decode the response(s), you will need =
&quot;munpack&quot; or</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>a =
MIME-compliant mail reader.&nbsp; Different MIME-compliant mail =
readers</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>exhibit =
different behavior, especially when dealing with</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;multipart&quot; MIME messages (i.e. documents which have =
been split</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>up into =
multiple messages), so check your local documentation on</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>how to =
manipulate these messages.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>Below is the data which will enable a MIME compliant =
mail reader</FONT>
<BR><FONT SIZE=3D2>implementation to automatically retrieve the ASCII =
version of the</FONT>
<BR><FONT SIZE=3D2>Internet-Draft.</FONT>
</P>

<P><FONT FACE=3D"Arial" SIZE=3D2 COLOR=3D"#000000"></FONT><FONT =
FACE=3D"Arial" SIZE=3D2 COLOR=3D"#000000"></FONT><FONT FACE=3D"Arial" =
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org  Tue Jan 18 18:53:24 2005
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Date: Tue, 18 Jan 2005 13:24:27 -1000
Subject: Re: [Sip] Draft defining the Accept-Disposition header field
From: Cullen Jennings <fluffy@cisco.com>
To: "gonzalo.camarillo@ericsson.com" <Gonzalo.Camarillo@ericsson.com>,
        "sip@ietf.org" <sip@ietf.org>
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I'm not really sure what the problem is that this solves.

Ignoring that, it seems that the dispositions need to be mime specific. For
example, I may be able to have a session with SDP but I doubt that just
because a UA support render means I can render SDP.

Do we have any MIME types where the UA could support that MIME type and more
than one disposition would be useful for that MIME type?


Cullen


On 1/12/05 4:36 AM, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
wrote:

> Folks,
> 
> I have submitted a draft that defines the Accept-Disposition header
> field. The usage of this header field would be similar to the usage of
> Accept, Accept-Encoding, or Accept-Language, but it would indicate
> support for disposition types.
> 
> http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept-disposition-00.
> txt
> 
> Your comments are welcome.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


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Cullen Jennings wrote:
> 
> I'm not really sure what the problem is that this solves.
> 
> Ignoring that, it seems that the dispositions need to be mime specific. For
> example, I may be able to have a session with SDP but I doubt that just
> because a UA support render means I can render SDP.
> 
> Do we have any MIME types where the UA could support that MIME type and more
> than one disposition would be useful for that MIME type?

application/sdp: session & early-session

application/sipfrag: render? (for refer notifies), ??? (for smime)

text/plain: render (for MESSAGE), alert, ...

image/*: render (for MESSAGE), icon

multipart/*: what disposition are these supposed to get???

	Paul

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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] FW: [Sipping]
	I-D	ACTION:draft-ietf-sipping-dialog-package-05.txt
Date: Wed, 19 Jan 2005 15:37:51 +0100
To: Robert Sparks <rjsparks@nostrum.com>
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On Jan 18, 2005, at 7:21 PM, Robert Sparks wrote:
>
>>
>> OK, so where does this leave us?  I think we should progress as
>> follows.
>>
>> 1. 481 should not be used to indicate "what was this errant
>>    PRACK for"?  A new status code should be defined meaning
>>    "the dialog was there but I have no outstanding reliable
>>    provisional response - your PRACK is in error".  RFC3262
>>    must be amended accordingly.  This new status code does
>>    not cause the dialog to end.
>>
>> 2. 481 should not be used to indicate "this NOTIFY was within
>>    a valid dialog but I know nothing about this subscription".
>>    A new status code should be defined for this.  RFC3265 must
>>    be amended accordingly.  This new status code does not
>>    cause the dialog to end.
>>
>> 3. 481 means one thing, and one thing only.  The dialog is not
>>    known and as far as everyone is concerned, it no longer
>>    exists.
> Nuance -
> The 481 means the recipient of the request doesn't know the dialog,
> and the sender should stop talking to the recipient about it.
>
Robert,

I agree with you here, but you need to consider the dialog stateful 
intermediaries in the case that an endpoint no longer knows about a 
dialog and sends a 481. Should the dialog stateful intermediaries 
terminate that dialog abnormally or should they wait for the BYE coming 
from the originator. I suggest the former. Any statistics on what 
current dialog stateful proxy implementations do?

Regards,
Hisham


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From sip-bounces@ietf.org  Wed Jan 19 11:38:06 2005
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From: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        "gonzalo.camarillo@ericsson.com"
	<Gonzalo.Camarillo@ericsson.com>,
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Subject: RE: [Sip] Draft defining the Accept-Disposition header field
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A formal specification of the Acceptable Disposition types is
useful. I do not necessarily agree that the proposed way is the 
best way to do it.

The Accept-Disposition needs to be associated with the corresponding 
Acceped Content Types. A Accept-Disposition header by itself will not 
be effective unless there is an associated Content Type specififed. 

I think that an effective way to specify the  Disposition is to 
Change the Accept Header to also specify a required implicit/explicit
Disposition Type.

Example: 

Accept:application/sdp;session

Thanks,
Shekar

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, January 18, 2005 6:24 PM
> To: gonzalo.camarillo@ericsson.com; sip@ietf.org
> Subject: Re: [Sip] Draft defining the Accept-Disposition header field
> 
> 
> 
> 
> I'm not really sure what the problem is that this solves.
> 
> Ignoring that, it seems that the dispositions need to be mime 
> specific. For
> example, I may be able to have a session with SDP but I doubt 
> that just
> because a UA support render means I can render SDP.
> 
> Do we have any MIME types where the UA could support that 
> MIME type and more
> than one disposition would be useful for that MIME type?
> 
> 
> Cullen
> 
> 
> On 1/12/05 4:36 AM, "Gonzalo Camarillo" 
> <Gonzalo.Camarillo@ericsson.com>
> wrote:
> 
> > Folks,
> > 
> > I have submitted a draft that defines the Accept-Disposition header
> > field. The usage of this header field would be similar to 
> the usage of
> > Accept, Accept-Encoding, or Accept-Language, but it would indicate
> > support for disposition types.
> > 
> > 
> http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept
-disposition-00.
> txt
> 
> Your comments are welcome.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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

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Subject: Re: [Sip] Draft defining the Accept-Disposition header field
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Shekar, Nagesh Soma (Shekar)** CTR ** wrote:
> A formal specification of the Acceptable Disposition types is
> useful. I do not necessarily agree that the proposed way is the 
> best way to do it.
> 
> The Accept-Disposition needs to be associated with the corresponding 
> Acceped Content Types. A Accept-Disposition header by itself will not 
> be effective unless there is an associated Content Type specififed. 
> 
> I think that an effective way to specify the  Disposition is to 
> Change the Accept Header to also specify a required implicit/explicit
> Disposition Type.
> 
> Example: 
> 
> Accept:application/sdp;session

Seems somewhat appealing. Because parameters are already accepted, I 
don't think the above way of using parameters is ideal - its confusing 
about the role of the parameter, and runs risk of conflicting with other 
parameter usages, and would need to be updated every time a new 
disposition was invented. Instead, perhaps:

	Accept:application/sdp;disposition="session,early-session"
         Accept:multipart/mixed;disposition="*"

It might also be necessary to allow:

	Accept:text/*;disposition="render",
	       text/plain;disposition="alert"

For backward compatibility, probably need to say that default for 
disposition parameter is "*".

	Paul

>>-----Original Message-----
>>From: Cullen Jennings [mailto:fluffy@cisco.com]
>>Sent: Tuesday, January 18, 2005 6:24 PM
>>To: gonzalo.camarillo@ericsson.com; sip@ietf.org
>>Subject: Re: [Sip] Draft defining the Accept-Disposition header field
>>
>>
>>
>>
>>I'm not really sure what the problem is that this solves.
>>
>>Ignoring that, it seems that the dispositions need to be mime 
>>specific. For
>>example, I may be able to have a session with SDP but I doubt 
>>that just
>>because a UA support render means I can render SDP.
>>
>>Do we have any MIME types where the UA could support that 
>>MIME type and more
>>than one disposition would be useful for that MIME type?
>>
>>
>>Cullen
>>
>>
>>On 1/12/05 4:36 AM, "Gonzalo Camarillo" 
>><Gonzalo.Camarillo@ericsson.com>
>>wrote:
>>
>>
>>>Folks,
>>>
>>>I have submitted a draft that defines the Accept-Disposition header
>>>field. The usage of this header field would be similar to 
>>
>>the usage of
>>
>>>Accept, Accept-Encoding, or Accept-Language, but it would indicate
>>>support for disposition types.
>>>
>>>
>>
>>http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept
> 
> -disposition-00.
> 
>>txt
>>
>>Your comments are welcome.
>>
>>Thanks,
>>
>>Gonzalo
>>
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP 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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Date: Wed, 19 Jan 2005 23:16:33 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [Sip] Draft defining the Accept-Disposition header field
References: <41EEBF8A.8000708@cisco.com>
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Hi,

yes, there is a trade-off here. On the one hand, we can just list the 
disposition types supported by the client, as defined in the draft. This 
has the advantage of efficiency (each disposition type is listed only 
once). On the other hand, we can associate each disposition type with 
the content type it applies to. This is not so efficient (disposition 
types that apply to more than one content type will be listed more than 
one), but it provides more accurate information.

If folks feel that having better accuracy is important, we should 
definitively go for the second option.

Regards,

Gonzalo



Paul Kyzivat wrote:

> 
> Shekar, Nagesh Soma (Shekar)** CTR ** wrote:
> 
>>A formal specification of the Acceptable Disposition types is
>>useful. I do not necessarily agree that the proposed way is the 
>>best way to do it.
>>
>>The Accept-Disposition needs to be associated with the corresponding 
>>Acceped Content Types. A Accept-Disposition header by itself will not 
>>be effective unless there is an associated Content Type specififed. 
>>
>>I think that an effective way to specify the  Disposition is to 
>>Change the Accept Header to also specify a required implicit/explicit
>>Disposition Type.
>>
>>Example: 
>>
>>Accept:application/sdp;session
> 
> 
> Seems somewhat appealing. Because parameters are already accepted, I 
> don't think the above way of using parameters is ideal - its confusing 
> about the role of the parameter, and runs risk of conflicting with other
> 
> parameter usages, and would need to be updated every time a new 
> disposition was invented. Instead, perhaps:
> 
> 	Accept:application/sdp;disposition="session,early-session"
>          Accept:multipart/mixed;disposition="*"
> 
> It might also be necessary to allow:
> 
> 	Accept:text/*;disposition="render",
> 	       text/plain;disposition="alert"
> 
> For backward compatibility, probably need to say that default for 
> disposition parameter is "*".
> 
> 	Paul
> 
> 
>>>-----Original Message-----
>>>From: Cullen Jennings [mailto:fluffy@cisco.com]
>>>Sent: Tuesday, January 18, 2005 6:24 PM
>>>To: gonzalo.camarillo@ericsson.com; sip@ietf.org
>>>Subject: Re: [Sip] Draft defining the Accept-Disposition header field
>>>
>>>
>>>
>>>
>>>I'm not really sure what the problem is that this solves.
>>>
>>>Ignoring that, it seems that the dispositions need to be mime 
>>>specific. For
>>>example, I may be able to have a session with SDP but I doubt 
>>>that just
>>>because a UA support render means I can render SDP.
>>>
>>>Do we have any MIME types where the UA could support that 
>>>MIME type and more
>>>than one disposition would be useful for that MIME type?
>>>
>>>
>>>Cullen
>>>
>>>
>>>On 1/12/05 4:36 AM, "Gonzalo Camarillo" 
>>><Gonzalo.Camarillo@ericsson.com>
>>>wrote:
>>>
>>>
>>>
>>>>Folks,
>>>>
>>>>I have submitted a draft that defines the Accept-Disposition header
>>>>field. The usage of this header field would be similar to 
>>>
>>>the usage of
>>>
>>>
>>>>Accept, Accept-Encoding, or Accept-Language, but it would indicate
>>>>support for disposition types.
>>>>
>>>>
>>>
>>>http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept
>>
>>-disposition-00.
>>
>>
>>>txt
>>>
>>>Your comments are welcome.
>>>
>>>Thanks,
>>>
>>>Gonzalo
>>>
>>>
>>>
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>This list is for NEW development of the core SIP 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
>>

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


_______________________________________________
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Subject: Re: [Sip] Draft defining the Accept-Disposition header field
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Gonzalo Camarillo wrote:
> Hi,
> 
> yes, there is a trade-off here. On the one hand, we can just list the 
> disposition types supported by the client, as defined in the draft. This 
> has the advantage of efficiency (each disposition type is listed only 
> once). On the other hand, we can associate each disposition type with 
> the content type it applies to. This is not so efficient (disposition 
> types that apply to more than one content type will be listed more than 
> one), but it provides more accurate information.
> 
> If folks feel that having better accuracy is important, we should 
> definitively go for the second option.

For some purposes the less accurate approach might be sufficient to give 
a hint of what is possible and what isn't.

But I am concerned about one of the key uses of these headers - 
reporting what is accepted as part of a response saying that something 
isn't accepted. It is not good form to reject a message and yet say that 
you support the content type and disposition it contained.

Getting both accuracy and efficiency seems difficult. I think the key 
may be in defining the scope over which these declarations apply, and 
what is meant when something has not been specified. Then perhaps it 
would only be necessary to be precise in the error response to an 
unsupported usage.

BTW, now that I think about it more, the notation problem is more 
complex - really need to specify the supported triples of 
type,disposition,message. And even that leaves out responses. Its harder 
to cram all of that into a parameter syntax.

Part of this could be dealt with by a new error code that means the 
particular combination of types and dispositions in the request was not 
supported. However that only works for requests. It doesn't give a way 
to indicate what combinations can/can't be used in *responses*.

	Paul

> Regards,
> 
> Gonzalo
> 
> 
> 
> Paul Kyzivat wrote:
> 
>>
>> Shekar, Nagesh Soma (Shekar)** CTR ** wrote:
>>
>>> A formal specification of the Acceptable Disposition types is
>>> useful. I do not necessarily agree that the proposed way is the best 
>>> way to do it.
>>>
>>> The Accept-Disposition needs to be associated with the corresponding 
>>> Acceped Content Types. A Accept-Disposition header by itself will not 
>>> be effective unless there is an associated Content Type specififed.
>>> I think that an effective way to specify the  Disposition is to 
>>> Change the Accept Header to also specify a required implicit/explicit
>>> Disposition Type.
>>>
>>> Example:
>>> Accept:application/sdp;session
>>
>>
>>
>> Seems somewhat appealing. Because parameters are already accepted, I 
>> don't think the above way of using parameters is ideal - its confusing 
>> about the role of the parameter, and runs risk of conflicting with other
>>
>> parameter usages, and would need to be updated every time a new 
>> disposition was invented. Instead, perhaps:
>>
>>     Accept:application/sdp;disposition="session,early-session"
>>          Accept:multipart/mixed;disposition="*"
>>
>> It might also be necessary to allow:
>>
>>     Accept:text/*;disposition="render",
>>            text/plain;disposition="alert"
>>
>> For backward compatibility, probably need to say that default for 
>> disposition parameter is "*".
>>
>>     Paul
>>
>>
>>>> -----Original Message-----
>>>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>>>> Sent: Tuesday, January 18, 2005 6:24 PM
>>>> To: gonzalo.camarillo@ericsson.com; sip@ietf.org
>>>> Subject: Re: [Sip] Draft defining the Accept-Disposition header field
>>>>
>>>>
>>>>
>>>>
>>>> I'm not really sure what the problem is that this solves.
>>>>
>>>> Ignoring that, it seems that the dispositions need to be mime 
>>>> specific. For
>>>> example, I may be able to have a session with SDP but I doubt that just
>>>> because a UA support render means I can render SDP.
>>>>
>>>> Do we have any MIME types where the UA could support that MIME type 
>>>> and more
>>>> than one disposition would be useful for that MIME type?
>>>>
>>>>
>>>> Cullen
>>>>
>>>>
>>>> On 1/12/05 4:36 AM, "Gonzalo Camarillo" 
>>>> <Gonzalo.Camarillo@ericsson.com>
>>>> wrote:
>>>>
>>>>
>>>>
>>>>> Folks,
>>>>>
>>>>> I have submitted a draft that defines the Accept-Disposition header
>>>>> field. The usage of this header field would be similar to 
>>>>
>>>>
>>>> the usage of
>>>>
>>>>
>>>>> Accept, Accept-Encoding, or Accept-Language, but it would indicate
>>>>> support for disposition types.
>>>>>
>>>>>
>>>>
>>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-accept
>>>
>>>
>>> -disposition-00.
>>>
>>>
>>>> txt
>>>>
>>>> Your comments are welcome.
>>>>
>>>> Thanks,
>>>>
>>>> Gonzalo
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP 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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Date: Wed, 19 Jan 2005 17:32:28 -1000
Subject: Re: [Sip] Draft defining the Accept-Disposition header field
From: Cullen Jennings <fluffy@cisco.com>
To: Cullen Jennings <fluffy@cisco.com>,
        "gonzalo.camarillo@ericsson.com" <Gonzalo.Camarillo@ericsson.com>,
        "sip@ietf.org" <sip@ietf.org>
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On 1/18/05 1:24 PM, "Cullen Jennings" <fluffy@cisco.com> wrote:

> I'm not really sure what the problem is that this solves.

I like how the discussion is going on *how* to do this but I'm still missing
the *why* to do it. I'm failing to imagine when this problem comes up.



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To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Draft defining the Accept-Disposition header field
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Cullen Jennings wrote:
> On 1/18/05 1:24 PM, "Cullen Jennings" <fluffy@cisco.com> wrote:
> 
> 
>>I'm not really sure what the problem is that this solves.
> 
> 
> I like how the discussion is going on *how* to do this but I'm still missing
> the *why* to do it. I'm failing to imagine when this problem comes up.

Well, I think most of this ultimately is just debugging info. In 
principle a UA could change its behavior based on what it learns from 
these kinds of headers. But how much of that really happens.

While I think this kind of info is useful to fill in the logical holes 
in the protocol, I am more interested in carefully defining all the 
currently underdefined behaviors wrt disposition.

	Paul

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi,

> I'm not really sure what the problem is that this solves.

it lets a UA know whether the remote UA supports a particular 
disposition type. Right now, UAs guess this information based on the 
support for MIME types. For example, if the remote UA supports 
application/sdp, my UA assumes that the remote UA supports the 'session' 
disposition type... but what about 'early-session'?

The same thing with uri-list bodies. A conference server may support 
uri-list bodies for some reason, but may not support the 
'recipient-list' disposition type. The disposition type is what tells me 
that the conference server supports (or does not support) the feature I 
want to use. (So far there are two disposition types associated with 
uri-list bodies and may be more in the future.)

Regarding debugging, the UAC can know what went wrong when it received a 
415 response from a UAS that supports all the MIME types needed for the 
service. In this sense, the draft completes the procedures defined in 
RFC 3204 regarding rejection of body parts whose handling is required.


The thing is that, as it has been pointed out in this thread, 
disposition types are usually supported in relation with a particular 
MIME type and even a particular method (e.g., a server is a message 
URI-list service but does not support the 'recipient-list' disposition 
type for INVITEs).

The solution proposed in the draft follows the definition of the 
existing Allow and Accept header fields in the sense that they do not 
provide cross header information: Allow tells you which methods are 
supported and Accept which MIME types, but there is no info about which 
MIME type is acceptable for which method. We need to decide if this is 
good enough or if we want to introduce some level of cross header 
information in the solution.

Regards,

Gonzalo


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From: "Vishnu Yadav" <vishnuyadav@deccanetworld.com>
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Subject: [Sip] 183 progress & 190 ringing
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Hi All,

I have a doubt on usage & handling of 183 & 180.
1. What is difference between 180 & 183? Why 183 needed ?
2. What is the order in which we get both of these responses? I mean to say,
we get 183 then we get 180 or in some other ways.
3. Is 183 progress sent by proxy or UAS? Can B2BUA send 183 to UAC?

Thanks in advance.
Vishnu Yadav



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From sip-bounces@ietf.org  Thu Jan 20 02:55:52 2005
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From: Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
Subject: RE: [Sip] 183 progress & 190 ringing
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One use of 183 is while dealing with SIP to PSTN Interworking.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
Vishnu Yadav
Sent: Thursday, January 20, 2005 12:25 PM
To: sip@ietf.org
Subject: [Sip] 183 progress & 190 ringing


Hi All,

I have a doubt on usage & handling of 183 & 180.
1. What is difference between 180 & 183? Why 183 needed ?
2. What is the order in which we get both of these responses? I mean to say,
we get 183 then we get 180 or in some other ways.
3. Is 183 progress sent by proxy or UAS? Can B2BUA send 183 to UAC?

Thanks in advance.
Vishnu Yadav



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


_______________________________________________
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From sip-bounces@ietf.org  Thu Jan 20 05:00:14 2005
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Subject: [Sip] CSEQ Number in ACK to 2xx Response for INVITE???
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This is a multi-part message in MIME format.

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------_=_NextPart_001_01C4FED3.7894EB58
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Hello all,

=20

In section 13.2.2.4 2xx Responses,it is mentioned that while generating =
an ACK for a 2xx response should have same CSeq number as in INVITE but =
the method as ACK.

=20

But is mentioned in Definition section that ACK for a 2xx response to an =
INVITE request is a separate transaction.Does it mean that CSEQ number =
to be incremented?

=20

Regds

karthikeyan.k


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New'"></SPAN>&nbsp;</P>=0A=
<P class=3DMsoNormal =0A=
style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt =
229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt =
641.2pt 687.0pt 732.8pt"><SPAN =0A=
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'">In =
section =0A=
13.2.2.4 2xx Responses,it is mentioned that w</SPAN><SPAN =0A=
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'">hile =0A=
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as in =0A=
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style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'"></SPAN></P>=0A=
<P class=3DMsoNormal =0A=
style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt =
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641.2pt 687.0pt 732.8pt"><SPAN =0A=
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'"></SPAN>&nbsp;</P>=0A=
<P class=3DMsoNormal =0A=
style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt =
229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt =
641.2pt 687.0pt 732.8pt"><SPAN =0A=
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'">But =
is =0A=
mentioned in Definition section that ACK<SPAN =0A=
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'">&nbsp;for a =0A=
2xx response to</SPAN><SPAN =0A=
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA"><SPAN =0A=
style=3D"mso-spacerun: yes">&nbsp;</SPAN>an INVITE request is a separate =0A=
transaction.Does it mean that CSEQ number to be =
incremented?</SPAN></SPAN></P>=0A=
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style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt =
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641.2pt 687.0pt 732.8pt"><SPAN =0A=
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New'"><SPAN =0A=
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA"></SPAN></SPAN>&nbsp;</P>=0A=
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style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt =
229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt =
641.2pt 687.0pt 732.8pt"><SPAN =0A=
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Date: Thu, 20 Jan 2005 16:16:55 +0530
From: Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
Subject: RE: [Sip] CSEQ Number in ACK to 2xx Response for INVITE???
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Hi,
   No transaction for ACK of 2xx response for INVITE.

Rajeew

  -----Original Message-----
  From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
innovation.interops@wipro.com
  Sent: Thursday, January 20, 2005 3:06 PM
  To: sip@ietf.org
  Subject: [Sip] CSEQ Number in ACK to 2xx Response for INVITE???


  Hello all,



  In section 13.2.2.4 2xx Responses,it is mentioned that while generating an
ACK for a 2xx response should have same CSeq number as in INVITE but the
method as ACK.



  But is mentioned in Definition section that ACK for a 2xx response to an
INVITE request is a separate transaction.Does it mean that CSEQ number to be
incremented?



  Regds

  karthikeyan.k


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<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=608043910-20012005>Hi,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=608043910-20012005>&nbsp; 
&nbsp;No&nbsp;transaction&nbsp;for ACK of&nbsp;2xx response for 
INVITE.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=608043910-20012005></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=608043910-20012005>Rajeew</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> sip-bounces@ietf.org 
  [mailto:sip-bounces@ietf.org]<B>On Behalf Of 
  </B>innovation.interops@wipro.com<BR><B>Sent:</B> Thursday, January 20, 2005 
  3:06 PM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] CSEQ Number in ACK 
  to 2xx Response for INVITE???<BR><BR></DIV></FONT>
  <DIV><FONT color=#000000 face=Arial size=2>
  <P class=MsoNormal 
  style="MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt 641.2pt 687.0pt 732.8pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">Hello 
  all,</SPAN></P>
  <P class=MsoNormal 
  style="MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt 641.2pt 687.0pt 732.8pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"></SPAN>&nbsp;</P>
  <P class=MsoNormal 
  style="MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt 641.2pt 687.0pt 732.8pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">In section 
  13.2.2.4 2xx Responses,it is mentioned that w</SPAN><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">hile 
  generating an ACK for a 2xx response&nbsp;should have same CSeq number as in 
  INVITE but the method as ACK.</SPAN><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"></SPAN></P>
  <P class=MsoNormal 
  style="MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt 641.2pt 687.0pt 732.8pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"></SPAN>&nbsp;</P>
  <P class=MsoNormal 
  style="MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt 641.2pt 687.0pt 732.8pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">But is 
  mentioned in Definition section that ACK<SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt">&nbsp;for a 
  2xx response to</SPAN><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt; mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; mso-fareast-language: EN-US; mso-bidi-language: AR-SA"><SPAN 
  style="mso-spacerun: yes">&nbsp;</SPAN>an INVITE request is a separate 
  transaction.Does it mean that CSEQ number to be incremented?</SPAN></SPAN></P>
  <P class=MsoNormal 
  style="MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt 641.2pt 687.0pt 732.8pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt; mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; mso-fareast-language: EN-US; mso-bidi-language: AR-SA"></SPAN></SPAN>&nbsp;</P>
  <P class=MsoNormal 
  style="MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt 183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt 595.4pt 641.2pt 687.0pt 732.8pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt"><SPAN 
  style="COLOR: black; FONT-FAMILY: 'Courier New'; FONT-SIZE: 10pt; mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; mso-fareast-language: EN-US; mso-bidi-language: AR-SA">Regds</SPAN></SPAN></P>
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0243137181==--



From sip-bounces@ietf.org  Thu Jan 20 09:01:11 2005
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To: Vishnu Yadav <vishnuyadav@deccanetworld.com>
Subject: Re: [Sip] 183 progress & 190 ringing
References: <NIBBJFCFKLEFDAHDNMBGMEDPCEAA.vishnuyadav@deccanetworld.com>
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The 183 says something is happening, but is not so precise as 180.  It is a 
response from the UAS (which may be the final destination or the originator's side 
of a B2BUA).  The 183 is assigned specific roles in Q.1912.5 and in 3GPP call flows.

Vishnu Yadav wrote:
> Hi All,
> 
> I have a doubt on usage & handling of 183 & 180.
> 1. What is difference between 180 & 183? Why 183 needed ?
> 2. What is the order in which we get both of these responses? I mean to say,
> we get 183 then we get 180 or in some other ways.
> 3. Is 183 progress sent by proxy or UAS? Can B2BUA send 183 to UAC?
> 
> Thanks in advance.
> Vishnu Yadav
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel
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To: "'innovation.interops@wipro.com'" <innovation.interops@wipro.com>,
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Subject: RE: [Sip] CSEQ Number in ACK to 2xx Response for INVITE???
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It means that the ACK is different transaction. For a 6361 implementation, this means that
the branch ID should be different from the branch ID that was used for the INVITE transaction,
 
Shekar

-----Original Message-----
From: innovation.interops@wipro.com [mailto:innovation.interops@wipro.com]
Sent: Thursday, January 20, 2005 4:36 AM
To: sip@ietf.org
Subject: [Sip] CSEQ Number in ACK to 2xx Response for INVITE???


Hello all,

 

In section 13.2.2.4 2xx Responses,it is mentioned that while generating an ACK for a 2xx response should have same CSeq number as in INVITE but the method as ACK.

 

But is mentioned in Definition section that ACK for a 2xx response to an INVITE request is a separate transaction.Does it mean that CSEQ number to be incremented?

 

Regds

karthikeyan.k


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<DIV><SPAN class=3D646303414-20012005><FONT face=3DArial =
color=3D#0000ff size=3D2>It=20
means that the ACK is different transaction. For a 6361 implementation, =
this=20
means that</FONT></SPAN></DIV>
<DIV><SPAN class=3D646303414-20012005><FONT face=3DArial =
color=3D#0000ff size=3D2>the=20
branch ID should be different from the branch ID that was used for the =
INVITE=20
transaction,</FONT></SPAN></DIV>
<DIV><SPAN class=3D646303414-20012005><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D646303414-20012005><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Shekar</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B>=20
  innovation.interops@wipro.com=20
  [mailto:innovation.interops@wipro.com]<BR><B>Sent:</B> Thursday, =
January 20,=20
  2005 4:36 AM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] CSEQ =
Number=20
  in ACK to 2xx Response for INVITE???<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#000000 size=3D2>
  <P class=3DMsoNormal=20
  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
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  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
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  all,</SPAN></P>
  <P class=3DMsoNormal=20
  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
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595.4pt 641.2pt 687.0pt 732.8pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'"></SPAN>&nbsp;</P>
  <P class=3DMsoNormal=20
  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt =
595.4pt 641.2pt 687.0pt 732.8pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'">In section=20
  13.2.2.4 2xx Responses,it is mentioned that w</SPAN><SPAN=20
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New'">hile=20
  generating an ACK for a 2xx response&nbsp;should have same CSeq =
number as in=20
  INVITE but the method as ACK.</SPAN><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'"></SPAN></P>
  <P class=3DMsoNormal=20
  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt =
595.4pt 641.2pt 687.0pt 732.8pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'"></SPAN>&nbsp;</P>
  <P class=3DMsoNormal=20
  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt =
595.4pt 641.2pt 687.0pt 732.8pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'">But is=20
  mentioned in Definition section that ACK<SPAN=20
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New'">&nbsp;for a=20
  2xx response to</SPAN><SPAN=20
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mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA"><SPAN=20
  style=3D"mso-spacerun: yes">&nbsp;</SPAN>an INVITE request is a =
separate=20
  transaction.Does it mean that CSEQ number to be =
incremented?</SPAN></SPAN></P>
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  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
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595.4pt 641.2pt 687.0pt 732.8pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA"></SPAN></SPAN>&nbsp;</P>
  <P class=3DMsoNormal=20
  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
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New'"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
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AR-SA">Regds</SPAN></SPAN></P>
  <P class=3DMsoNormal=20
  style=3D"MARGIN: 0in 0in 0pt; tab-stops: 45.8pt 91.6pt 137.4pt =
183.2pt 229.0pt 274.8pt 320.6pt 366.4pt 412.2pt 458.0pt 503.8pt 549.6pt =
595.4pt 641.2pt 687.0pt 732.8pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier =
New'"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: 'Courier New'; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: EN-US; =
mso-fareast-language: EN-US; mso-bidi-language: =
AR-SA">karthikeyan.k</SPAN></SPAN></P></FONT></DIV></BLOCKQUOTE></BODY><=
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From sip-bounces@ietf.org  Thu Jan 20 10:52:02 2005
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Draft defining the Accept-Disposition header field
Date: Thu, 20 Jan 2005 14:39:15 +0100
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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This sounds like some presence event package related info or some other 
event package, not something that is carried in a SIP request or 
response.

Regards,
Hisham

On Jan 20, 2005, at 7:43 AM, Gonzalo Camarillo wrote:

> Hi,
>
>> I'm not really sure what the problem is that this solves.
>
> it lets a UA know whether the remote UA supports a particular 
> disposition type. Right now, UAs guess this information based on the 
> support for MIME types. For example, if the remote UA supports 
> application/sdp, my UA assumes that the remote UA supports the 
> 'session' disposition type... but what about 'early-session'?
>
> The same thing with uri-list bodies. A conference server may support 
> uri-list bodies for some reason, but may not support the 
> 'recipient-list' disposition type. The disposition type is what tells 
> me that the conference server supports (or does not support) the 
> feature I want to use. (So far there are two disposition types 
> associated with uri-list bodies and may be more in the future.)
>
> Regarding debugging, the UAC can know what went wrong when it received 
> a 415 response from a UAS that supports all the MIME types needed for 
> the service. In this sense, the draft completes the procedures defined 
> in RFC 3204 regarding rejection of body parts whose handling is 
> required.
>
>
> The thing is that, as it has been pointed out in this thread, 
> disposition types are usually supported in relation with a particular 
> MIME type and even a particular method (e.g., a server is a message 
> URI-list service but does not support the 'recipient-list' disposition 
> type for INVITEs).
>
> The solution proposed in the draft follows the definition of the 
> existing Allow and Accept header fields in the sense that they do not 
> provide cross header information: Allow tells you which methods are 
> supported and Accept which MIME types, but there is no info about 
> which MIME type is acceptable for which method. We need to decide if 
> this is good enough or if we want to introduce some level of cross 
> header information in the solution.
>
> Regards,
>
> Gonzalo
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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] Draft defining the Accept-Disposition header field
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This is very much required to accept a Content carried by a SIP
request or a Response. This is a real problem, in that sense that
the nodes trying to accept the Content needs to look at the Content
Type and Content Disposition associated with the Content.

Having incorrect or unsupported comibnations will result in call
failures, which obviously is not desirable.

By specifying the acceptable combinations of Content Type and Content 
Disposition in some form of Accept Header, we can suppose that the 
chance of a call succeeding would increase.

Nagesh Shekar.

> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no]
> Sent: Thursday, January 20, 2005 8:39 AM
> To: Gonzalo Camarillo
> Cc: Cullen Jennings; sip@ietf.org; Paul Kyzivat
> Subject: Re: [Sip] Draft defining the Accept-Disposition header field
> 
> 
> This sounds like some presence event package related info or 
> some other 
> event package, not something that is carried in a SIP request or 
> response.
> 
> Regards,
> Hisham
> 
> On Jan 20, 2005, at 7:43 AM, Gonzalo Camarillo wrote:
> 
> > Hi,
> >
> >> I'm not really sure what the problem is that this solves.
> >
> > it lets a UA know whether the remote UA supports a particular 
> > disposition type. Right now, UAs guess this information 
> based on the 
> > support for MIME types. For example, if the remote UA supports 
> > application/sdp, my UA assumes that the remote UA supports the 
> > 'session' disposition type... but what about 'early-session'?
> >
> > The same thing with uri-list bodies. A conference server 
> may support 
> > uri-list bodies for some reason, but may not support the 
> > 'recipient-list' disposition type. The disposition type is 
> what tells 
> > me that the conference server supports (or does not support) the 
> > feature I want to use. (So far there are two disposition types 
> > associated with uri-list bodies and may be more in the future.)
> >
> > Regarding debugging, the UAC can know what went wrong when 
> it received 
> > a 415 response from a UAS that supports all the MIME types 
> needed for 
> > the service. In this sense, the draft completes the 
> procedures defined 
> > in RFC 3204 regarding rejection of body parts whose handling is 
> > required.
> >
> >
> > The thing is that, as it has been pointed out in this thread, 
> > disposition types are usually supported in relation with a 
> particular 
> > MIME type and even a particular method (e.g., a server is a message 
> > URI-list service but does not support the 'recipient-list' 
> disposition 
> > type for INVITEs).
> >
> > The solution proposed in the draft follows the definition of the 
> > existing Allow and Accept header fields in the sense that 
> they do not 
> > provide cross header information: Allow tells you which methods are 
> > supported and Accept which MIME types, but there is no info about 
> > which MIME type is acceptable for which method. We need to 
> decide if 
> > this is good enough or if we want to introduce some level of cross 
> > header information in the solution.
> >
> > Regards,
> >
> > Gonzalo
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From sip-bounces@ietf.org  Thu Jan 20 17:17:03 2005
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From: "Karnam, Swamy" <SKarnam@verisign.com>
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Hi,

 

   This is my first post here and am a little bit unfamiliar with the
process.

   Hopefully, I am not asking something that has been answered.

 

   My comment is on RFC 3261.

  

   In a 302 Response message, there is a contact field that represents a
direct route to the destination.

  

   What if I want the message to be redirected to this contact via a
sequence of hops ?  Basically I want the

   Contact field to contact a Via string.

 

   Thanks,

   Swamy


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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp; This is my first post here and am a little bit
unfamiliar with the process.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp; Hopefully, I am not asking something that has
been answered.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp; My comment is on RFC 3261.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp; </span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp; In a 302 Response message, there is a contact
field that represents a direct route to the destination.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp; </span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp; &nbsp;What if I want the message to be redirected to
this contact via a sequence of hops ?&nbsp; Basically I want the</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp; Contact field to contact a Via string.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp; Thanks,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp; Swamy</span></font></p>

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--===============1186338853==--



From sip-bounces@ietf.org  Thu Jan 20 17:38:37 2005
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From: Volker Hilt <volkerh@bell-labs.com>
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Cc: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Subject: [Sip] draft-hilt-sip-ext-neg-00
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Folks,

we have submitted a draft, which defines a new Accept header field 
parameter that can be used to indicate support for MIME type extensions. 
In particular, this parameter can be used to indicate support for 
additional XML namespaces, which extend an XML-based media type.

http://www.ietf.org/internet-drafts/draft-hilt-sip-ext-neg-00.txt

Comments are very welcome.

-- Volker



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From sip-bounces@ietf.org  Thu Jan 20 17:51:58 2005
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Hey Swamy,

=20

If you want the response to travel back along the same path it took,
then the proxies along that path will need to insert a Record-route
header into the request. Have a read through section 16 of RFC 3261.

=20

HTH,

=20

B

=20

=20

Ben Gatewood

Lead Architect - IP Telephony

Streamdoor Ltd.

+44 20 7422 0439

+44 7835 136 187

bgatewood@streamdoor.net

INVITE. TRYING. RINGING. OK. ACK.

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Karnam, Swamy
Sent: 20 January 2005 21:21
To: 'sip@ietf.org'
Subject: [Sip] Via

=20

Hi,

=20

   This is my first post here and am a little bit unfamiliar with the
process.

   Hopefully, I am not asking something that has been answered.

=20

   My comment is on RFC 3261.

 =20

   In a 302 Response message, there is a contact field that represents a
direct route to the destination.

 =20

   What if I want the message to be redirected to this contact via a
sequence of hops ?  Basically I want the

   Contact field to contact a Via string.

=20

   Thanks,

   Swamy

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<div class=3DSection1>

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

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>If you want the response to travel b=
ack
along the same path it took, then the proxies along that path will need t=
o
insert a Record-route header into the request. Have a read through sectio=
n 16
of RFC 3261.<o:p></o:p></span></font></p>

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

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

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

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

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

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

<div>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Lead Architect - IP Telephony</span>=
</font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>+44 20 7422 0439</span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>+44 7835 136 187</span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><a href=3D"mailto:bgatewood@streamdo=
or.net">bgatewood@streamdoor.net</a></span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>INVITE. TRYING. RINGING. OK. ACK.</s=
pan></font><o:p></o:p></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font s=
ize=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-=
size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D=
2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style=3D'font=
-weight:
bold'>On Behalf Of </span></b>Karnam, Swamy<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 20 January 2005 21:2=
1<br>
<b><span style=3D'font-weight:bold'>To:</span></b> 'sip@ietf.org'<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Via</span><=
/font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

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

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size=
:10.0pt;
font-family:Arial'>&nbsp;&nbsp; This is my first post here and am a littl=
e bit
unfamiliar with the process.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size=
:10.0pt;
font-family:Arial'>&nbsp;&nbsp; Hopefully, I am not asking something that=
 has
been answered.</span></font><o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size=
:10.0pt;
font-family:Arial'>&nbsp;&nbsp; My comment is on RFC 3261.</span></font><=
o:p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size=
:10.0pt;
font-family:Arial'>&nbsp;&nbsp; In a 302 Response message, there is a con=
tact
field that represents a direct route to the destination.</span></font><o:=
p></o:p></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size=
:10.0pt;
font-family:Arial'>&nbsp; &nbsp;What if I want the message to be redirect=
ed to
this contact via a sequence of hops ?&nbsp; Basically I want the</span></=
font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size=
:10.0pt;
font-family:Arial'>&nbsp;&nbsp; Contact field to contact a Via string.</s=
pan></font><o:p></o:p></p>

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

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

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

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>*****************************************************************=
*********************<br>
Disclaimer:<br>
<br>
This electronic message may contain privileged or confidential informatio=
n. If
you are not the intended recipient, be advised that any disclosure, copyi=
ng,
distribution or use of this information is strictly prohibited. If you ar=
e not
the intended recipient, please notify the sender and delete this message.=
 Views
expressed in this message are those of the individual sender, and are not=

necessarily the views of the Streamdoor Limited, unless otherwise stated.=
<br>
<br>
Although Streamdoor Limited has taken reasonable precautions to ensure no=

viruses are present in this email, the company cannot accept responsibili=
ty for
any loss or damage arising from the use of this email or attachement<br>
<br>
Employees of Streamdoor Limited and its affiliates are expressly required=
 not
to make defamatory statements and not to infringe or authorise any infrin=
gements
of copyrights or any other legal right by email communications. Any such
communication is contrary to company policy and outside the scope of the
employment of the individual concerned.<br>
<br>
For further assistance on email policy, or if you have received this emai=
l in
error, please contact Streamdoor Group IT&amp;C Helpdesk by email at
it&amp;c_helpdesk@streamdoor.net or write to Streamdoor Ltd , Head of IT&=
amp;C
Department , 114 Power Road , Chiswick, London , W4 5PY.<br>
<br>
www.streamdoor.com<br>
*************************************************************************=
*************<o:p></o:p></span></font></p>

</div>

*************************************************************************=
*************<br>Disclaimer:<br><br>This electronic message may contain p=
rivileged or confidential information. If you are not the intended recipi=
ent, be advised that any disclosure, copying, distribution or use of this=
 information is strictly prohibited. If you are not the intended recipien=
t, please notify the sender and delete this message. Views expressed in t=
his message are those of the individual sender, and are not necessarily t=
he views of the Streamdoor Limited, unless otherwise stated.<br><br>Altho=
ugh Streamdoor Limited has taken reasonable precautions to ensure no viru=
ses are present in this email, the company cannot accept responsibility f=
or any loss or damage arising from the use of this email or attachement<b=
r><br>Employees of Streamdoor Limited and its affiliates are expressly re=
quired not to make defamatory statements and not to infringe or authorise=
 any infringements of copyrights or any other legal right by email commun=
ications. Any such communication is contrary to company policy and outsid=
e the scope of the employment of the individual concerned.<br><br>For fur=
ther assistance on email policy, or if you have received this email in er=
ror, please contact Streamdoor Group IT&C Helpdesk by email at it&c_helpd=
esk@streamdoor.net or write to Streamdoor Ltd , Head of IT&C Department ,=
 114 Power Road , Chiswick, London , W4 5PY.<br><br>www.streamdoor.com<br=
>************************************************************************=
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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From sip-bounces@ietf.org  Thu Jan 20 18:05:42 2005
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From: "Karnam, Swamy" <SKarnam@verisign.com>
To: "'Ben Gatewood'" <bgatewood@streamdoor.net>, sip@ietf.org
Subject: RE: [Sip] Via
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Ben,

   I don't want the response to travel back the same path,

   I want it to be redirected to the contact using a least cost path I want
to specify.

 

   Thanks,

   Swamy

 

 

-----Original Message-----
From: Ben Gatewood [mailto:bgatewood@streamdoor.net] 
Sent: Thursday, January 20, 2005 5:31 PM
To: Karnam, Swamy; sip@ietf.org
Subject: RE: [Sip] Via

 

Hey Swamy,

 

If you want the response to travel back along the same path it took, then
the proxies along that path will need to insert a Record-route header into
the request. Have a read through section 16 of RFC 3261.

 

HTH,

 

B

 

 

Ben Gatewood

Lead Architect - IP Telephony

Streamdoor Ltd.

+44 20 7422 0439

+44 7835 136 187

bgatewood@streamdoor.net <mailto:bgatewood@streamdoor.net> 

INVITE. TRYING. RINGING. OK. ACK.

  _____  

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Karnam, Swamy
Sent: 20 January 2005 21:21
To: 'sip@ietf.org'
Subject: [Sip] Via

 

Hi,

 

   This is my first post here and am a little bit unfamiliar with the
process.

   Hopefully, I am not asking something that has been answered.

 

   My comment is on RFC 3261.

  

   In a 302 Response message, there is a contact field that represents a
direct route to the destination.

  

   What if I want the message to be redirected to this contact via a
sequence of hops ?  Basically I want the

   Contact field to contact a Via string.

 

   Thanks,

   Swamy

****************************************************************************
**********
Disclaimer:

This electronic message may contain privileged or confidential information.
If you are not the intended recipient, be advised that any disclosure,
copying, distribution or use of this information is strictly prohibited. If
you are not the intended recipient, please notify the sender and delete this
message. Views expressed in this message are those of the individual sender,
and are not necessarily the views of the Streamdoor Limited, unless
otherwise stated.

Although Streamdoor Limited has taken reasonable precautions to ensure no
viruses are present in this email, the company cannot accept responsibility
for any loss or damage arising from the use of this email or attachement

Employees of Streamdoor Limited and its affiliates are expressly required
not to make defamatory statements and not to infringe or authorise any
infringements of copyrights or any other legal right by email
communications. Any such communication is contrary to company policy and
outside the scope of the employment of the individual concerned.

For further assistance on email policy, or if you have received this email
in error, please contact Streamdoor Group IT&C Helpdesk by email at
it&c_helpdesk@streamdoor.net or write to Streamdoor Ltd , Head of IT&C
Department , 114 Power Road , Chiswick, London , W4 5PY.

www.streamdoor.com
****************************************************************************
**********

****************************************************************************
**********
Disclaimer:

This electronic message may contain privileged or confidential information.
If you are not the intended recipient, be advised that any disclosure,
copying, distribution or use of this information is strictly prohibited. If
you are not the intended recipient, please notify the sender and delete this
message. Views expressed in this message are those of the individual sender,
and are not necessarily the views of the Streamdoor Limited, unless
otherwise stated.

Although Streamdoor Limited has taken reasonable precautions to ensure no
viruses are present in this email, the company cannot accept responsibility
for any loss or damage arising from the use of this email or attachement

Employees of Streamdoor Limited and its affiliates are expressly required
not to make defamatory statements and not to infringe or authorise any
infringements of copyrights or any other legal right by email
communications. Any such communication is contrary to company policy and
outside the scope of the employment of the individual concerned.

For further assistance on email policy, or if you have received this email
in error, please contact Streamdoor Group IT&C Helpdesk by email at
it&c_helpdesk@streamdoor.net or write to Streamdoor Ltd , Head of IT&C
Department , 114 Power Road , Chiswick, London , W4 5PY.

www.streamdoor.com
****************************************************************************
**********


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<div class=Section1>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Ben,</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp; I don&#8217;t want the response to
travel back the same path,</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp; I want it to be redirected to the
contact using a least cost path I want to specify.</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp; Thanks,</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp; Swamy</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> Ben Gatewood
[mailto:bgatewood@streamdoor.net] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, January 20, 2005
5:31 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Karnam, Swamy; sip@ietf.org<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Sip] Via</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>Hey Swamy,</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>If you want the response
to travel back along the same path it took, then the proxies along that path
will need to insert a Record-route header into the request. Have a read through
section 16 of RFC 3261.</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>HTH,</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>B</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<div>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>Ben Gatewood</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>Lead Architect - IP
Telephony</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>Streamdoor Ltd.</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>+44 20 7422 0439</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>+44 7835 136 187</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'><a
href="mailto:bgatewood@streamdoor.net">bgatewood@streamdoor.net</a></span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:navy'>INVITE. TRYING. RINGING.
OK. ACK.</span></font></p>

</div>

<div>

<div class=MsoNormal align=center style='margin-left:.5in;text-align:center'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>

<hr size=2 width="100%" align=center>

</span></font></div>

<p class=MsoNormal style='margin-left:.5in'><b><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Karnam, Swamy<br>
<b><span style='font-weight:bold'>Sent:</span></b> 20 January 2005 21:21<br>
<b><span style='font-weight:bold'>To:</span></b> 'sip@ietf.org'<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Sip] Via</span></font></p>

</div>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Hi,</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp; This is my first post
here and am a little bit unfamiliar with the process.</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp; Hopefully, I am not
asking something that has been answered.</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp; My comment is on RFC
3261.</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp; </span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp; In a 302 Response
message, there is a contact field that represents a direct route to the
destination.</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp; </span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp; &nbsp;What if I want the
message to be redirected to this contact via a sequence of hops ?&nbsp;
Basically I want the</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp; Contact field to
contact a Via string.</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp; Thanks,</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp; Swamy</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>**************************************************************************************<br>
Disclaimer:<br>
<br>
This electronic message may contain privileged or confidential information. If
you are not the intended recipient, be advised that any disclosure, copying,
distribution or use of this information is strictly prohibited. If you are not
the intended recipient, please notify the sender and delete this message. Views
expressed in this message are those of the individual sender, and are not
necessarily the views of the Streamdoor Limited, unless otherwise stated.<br>
<br>
Although Streamdoor Limited has taken reasonable precautions to ensure no
viruses are present in this email, the company cannot accept responsibility for
any loss or damage arising from the use of this email or attachement<br>
<br>
Employees of Streamdoor Limited and its affiliates are expressly required not
to make defamatory statements and not to infringe or authorise any
infringements of copyrights or any other legal right by email communications.
Any such communication is contrary to company policy and outside the scope of
the employment of the individual concerned.<br>
<br>
For further assistance on email policy, or if you have received this email in
error, please contact Streamdoor Group IT&amp;C Helpdesk by email at
it&amp;c_helpdesk@streamdoor.net or write to Streamdoor Ltd , Head of IT&amp;C
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<br>
www.streamdoor.com<br>
**************************************************************************************</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>**************************************************************************************<br>
Disclaimer:<br>
<br>
This electronic message may contain privileged or confidential information. If
you are not the intended recipient, be advised that any disclosure, copying,
distribution or use of this information is strictly prohibited. If you are not
the intended recipient, please notify the sender and delete this message. Views
expressed in this message are those of the individual sender, and are not
necessarily the views of the Streamdoor Limited, unless otherwise stated.<br>
<br>
Although Streamdoor Limited has taken reasonable precautions to ensure no
viruses are present in this email, the company cannot accept responsibility for
any loss or damage arising from the use of this email or attachement<br>
<br>
Employees of Streamdoor Limited and its affiliates are expressly required not
to make defamatory statements and not to infringe or authorise any
infringements of copyrights or any other legal right by email communications.
Any such communication is contrary to company policy and outside the scope of
the employment of the individual concerned.<br>
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For further assistance on email policy, or if you have received this email in
error, please contact Streamdoor Group IT&amp;C Helpdesk by email at
it&amp;c_helpdesk@streamdoor.net or write to Streamdoor Ltd , Head of IT&amp;C
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<br>
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**************************************************************************************</span></font></p>

</div>

</body>

</html>

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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
--===============1712254785==--



From sip-bounces@ietf.org  Fri Jan 21 05:10:07 2005
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From: "Samuel Osorio Calvo" <samuel.osorio@nl.thalesgroup.com>
To: <sip@ietf.org>, <bgatewood@streamdoor.net>, <SKarnam@verisign.com>
Subject: RE: [Sip] Via
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Hi,

Maybe you should take a look at the Path [RFC 3327] and the
Service-Route [RFC 3608] headers.

Path: is added in the REGISTER message way by all those servers that
want to stay in the path for further messages going to the Contact
address included in the registration process. Thus, after a user has
registered, when his/her AoR is checked in the Location Service, it
returns not only the current contact but also  a list of path headers
that represents proxies to be in the path of the request.
Summarising, its behaviour similar to the Record-Route in INVITES but
inserted in the REGISTER requests and obtained when the Location Service
is queried for the Contact.

The other option, Service-Route, is a header inserted by the Registrar
itself and it's meant to represent a list of proxies that must be
traversed when the Contact is queried. Its behaviour is much similar to
the Path header but Service-Route is inserted --via local policy-- by
the registrar and not by the proxies forwarding the REGISTER.

I hope I was clear and not wrong and that this is what you were looking
for.

Samuel. 


Unclassified.
>>> "Karnam, Swamy" <SKarnam@verisign.com> 01/20/05 11:48PM >>>
Ben,

   I don't want the response to travel back the same path,

   I want it to be redirected to the contact using a least cost path I
want
to specify.

 

   Thanks,

   Swamy

 

 

-----Original Message-----
From: Ben Gatewood [mailto:bgatewood@streamdoor.net] 
Sent: Thursday, January 20, 2005 5:31 PM
To: Karnam, Swamy; sip@ietf.org 
Subject: RE: [Sip] Via

 

Hey Swamy,

 

If you want the response to travel back along the same path it took,
then
the proxies along that path will need to insert a Record-route header
into
the request. Have a read through section 16 of RFC 3261.

 

HTH,

 

B

 

 

Ben Gatewood

Lead Architect - IP Telephony

Streamdoor Ltd.

+44 20 7422 0439

+44 7835 136 187

bgatewood@streamdoor.net <mailto:bgatewood@streamdoor.net> 

INVITE. TRYING. RINGING. OK. ACK.

  _____  

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Karnam, Swamy
Sent: 20 January 2005 21:21
To: 'sip@ietf.org' 
Subject: [Sip] Via

 

Hi,

 

   This is my first post here and am a little bit unfamiliar with the
process.

   Hopefully, I am not asking something that has been answered.

 

   My comment is on RFC 3261.

  

   In a 302 Response message, there is a contact field that represents
a
direct route to the destination.

  

   What if I want the message to be redirected to this contact via a
sequence of hops ?  Basically I want the

   Contact field to contact a Via string.

 

   Thanks,

   Swamy

****************************************************************************
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this
message. Views expressed in this message are those of the individual
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Although Streamdoor Limited has taken reasonable precautions to ensure
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and
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in error, please contact Streamdoor Group IT&C Helpdesk by email at
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www.streamdoor.com 
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and
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www.streamdoor.com 
****************************************************************************
**********


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

is anyone working with Avaya SIP Proxy (CCS), telephones (4602) and/or =
SIP
softphones?
I would be pleased if we could share our opinions and problems...

Thank you,
Mar=EDa S=E1nchez=



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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Fri Jan 21 11:10:22 2005
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Minor anomaly in RFC 3261 Registration example?
Date: Fri, 21 Jan 2005 09:50:53 -0600
To: "Jain, Rajnish" <rjain@excelswitching.com>
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The bugzilla database is available at bugs.sipit.net.

RjS

On Jan 14, 2005, at 4:06 PM, Jain, Rajnish wrote:

> Folks:
>
> Not sure if this minor anomaly has been reported before and is already=20=

> logged in SIP bugzilla (which seems to be down right now). Apologies=20=

> if this is a repetition.
>
>
>
> Section 10.3 of RFC 3261 states the following:
>
> =A0=A0=A0=A0=A0 8. The registrar returns a 200 (OK) response.=A0 The =
response MUST
>  =A0=A0=A0=A0=A0=A0=A0=A0 contain Contact header field values =
enumerating all current
>  =A0=A0=A0=A0=A0=A0=A0=A0 bindings.=A0 Each Contact value MUST feature =
an "expires"
>  =A0=A0=A0=A0=A0=A0=A0=A0 parameter indicating its expiration interval =
chosen by the
>  =A0=A0=A0=A0=A0=A0=A0=A0 registrar.=A0 The response SHOULD include a =
Date header field.
>
>
>
> Section 24.1 of RFC 3261 gives an example of 200 OK for a REGISTER as=20=

> the following:
>
> =A0=A0=A0=A0=A0=A0=A0 SIP/2.0 200 OK
>  =A0=A0=A0=A0=A0=A0=A0 Via: SIP/2.0/UDP =
bobspc.biloxi.com:5060;branch=3Dz9hG4bKnashds7
> =A0=A0=A0=A0=A0=A0=A0=A0 ;received=3D192.0.2.4
> =A0=A0=A0=A0=A0=A0=A0 To: Bob <sip:bob@biloxi.com>;tag=3D2493k59kd
> =A0=A0=A0=A0=A0=A0=A0 From: Bob <sip:bob@biloxi.com>;tag=3D456248
> =A0=A0=A0=A0=A0=A0=A0 Call-ID: 843817637684230@998sdasdh09
> =A0=A0=A0=A0=A0=A0=A0 CSeq: 1826 REGISTER
>  =A0=A0=A0=A0=A0=A0=A0 Contact: <sip:bob@192.0.2.4>
> =A0=A0=A0=A0=A0=A0=A0 Expires: 7200
>  =A0=A0=A0=A0=A0=A0=A0 Content-Length: 0
>
>
>
> The "expires" parameter in the 200 OK's Contact header is missing in=20=

> this example whereas it's a MUST in the text.
>
> Thanks,
>  Rajnish
>
> Email=A0=A0=A0=A0 mailto:rjain@excelswitching.com
> SIP=A0=A0=A0=A0=A0=A0 sip:rjain@voip.excelswitching.com
> PSTN=A0=A0=A0=A0=A0 508-862-3215
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip=


_______________________________________________
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From sip-bounces@ietf.org  Fri Jan 21 11:30:13 2005
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From: "Karnam, Swamy" <SKarnam@verisign.com>
To: "'Samuel Osorio Calvo'" <samuel.osorio@nl.thalesgroup.com>, sip@ietf.org,
        bgatewood@streamdoor.net
Subject: RE: [Sip] Via
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Thanks Samuel,

   The 2 RFC's were quite helpful.
   Swamy

-----Original Message-----
From: Samuel Osorio Calvo [mailto:samuel.osorio@nl.thalesgroup.com] 
Sent: Friday, January 21, 2005 4:57 AM
To: sip@ietf.org; bgatewood@streamdoor.net; Karnam, Swamy
Subject: RE: [Sip] Via

Hi,

Maybe you should take a look at the Path [RFC 3327] and the
Service-Route [RFC 3608] headers.

Path: is added in the REGISTER message way by all those servers that
want to stay in the path for further messages going to the Contact
address included in the registration process. Thus, after a user has
registered, when his/her AoR is checked in the Location Service, it
returns not only the current contact but also  a list of path headers
that represents proxies to be in the path of the request.
Summarising, its behaviour similar to the Record-Route in INVITES but
inserted in the REGISTER requests and obtained when the Location Service
is queried for the Contact.

The other option, Service-Route, is a header inserted by the Registrar
itself and it's meant to represent a list of proxies that must be
traversed when the Contact is queried. Its behaviour is much similar to
the Path header but Service-Route is inserted --via local policy-- by
the registrar and not by the proxies forwarding the REGISTER.

I hope I was clear and not wrong and that this is what you were looking
for.

Samuel. 


Unclassified.
>>> "Karnam, Swamy" <SKarnam@verisign.com> 01/20/05 11:48PM >>>
Ben,

   I don't want the response to travel back the same path,

   I want it to be redirected to the contact using a least cost path I
want
to specify.

 

   Thanks,

   Swamy

 

 

-----Original Message-----
From: Ben Gatewood [mailto:bgatewood@streamdoor.net] 
Sent: Thursday, January 20, 2005 5:31 PM
To: Karnam, Swamy; sip@ietf.org 
Subject: RE: [Sip] Via

 

Hey Swamy,

 

If you want the response to travel back along the same path it took,
then
the proxies along that path will need to insert a Record-route header
into
the request. Have a read through section 16 of RFC 3261.

 

HTH,

 

B

 

 

Ben Gatewood

Lead Architect - IP Telephony

Streamdoor Ltd.

+44 20 7422 0439

+44 7835 136 187

bgatewood@streamdoor.net <mailto:bgatewood@streamdoor.net> 

INVITE. TRYING. RINGING. OK. ACK.

  _____  

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Karnam, Swamy
Sent: 20 January 2005 21:21
To: 'sip@ietf.org' 
Subject: [Sip] Via

 

Hi,

 

   This is my first post here and am a little bit unfamiliar with the
process.

   Hopefully, I am not asking something that has been answered.

 

   My comment is on RFC 3261.

  

   In a 302 Response message, there is a contact field that represents
a
direct route to the destination.

  

   What if I want the message to be redirected to this contact via a
sequence of hops ?  Basically I want the

   Contact field to contact a Via string.

 

   Thanks,

   Swamy

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From sip-bounces@ietf.org  Fri Jan 21 15:19:04 2005
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From: Dean Willis <dean.willis@softarmor.com>
Date: Fri, 21 Jan 2005 14:08:32 -0600
To: SIPPING WG <sipping@ietf.org>, "'sip-ietf'" <sip@ietf.org>
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Subject: [Sip] Move Location Conveyance to SIP?
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The location conveyance draft:

	draft-ietf-sipping-location-requirements-02.txt

has grown well beyond requirements and into implementation. The group 
consensus a while back was to keep the requirements and implementation 
together for this one, as it helps with the review/publication cycle 
slightly and seems to fit together well. We seem to have a reasonable 
consensus around the technical approach in the draft. We expect the 
next rev of the draft to register a couple of additional SIP 400 class 
response codes making it a SIP extension, albeit a minor one. Since the 
process says this level of implementation belongs in SIP, we probably 
need to migrate this draft over to SIP.

James Polk is working on a revision of the draft. I propose that he 
submit this, when ready, as:

	draft-ietf-sip-location-conveyance-00


Anybody have major issues with this?

--
Dean Willis
SIP, SIPPING co-chair


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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Via
Date: Fri, 21 Jan 2005 14:16:34 -0600
To: "Karnam, Swamy" <SKarnam@verisign.com>
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Followups to sip-implementors please.

On Jan 20, 2005, at 3:21 PM, Karnam, Swamy wrote:

>  =A0=A0 In a 302 Response message, there is a contact field that=20
> represents a direct route to the destination.
>  =A0 =A0What if I want the message to be redirected to this contact =
via a=20
> sequence of hops ?=A0 Basically I want the
> =A0=A0 Contact field to contact a Via string.
>

Not QUITE sure what your use case here. It sounds like what you want to=20=

do is something like the service-route operation and return a string=20
that would be used as a default route on requests to the target=20
returned as a result of redirection.

This presumes that the party issuing the redirect understands the=20
routing topology as viewed by the originating UA well enough to build a=20=

route, which is not likely to be the general case. So this is pretty=20
likely to explode in some sort of useless way.

But I suspect you could return a Contact URI with an attached Route=20
header-field to get something like this beahvior.

--
Dean


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From sip-bounces@ietf.org  Fri Jan 21 16:46:02 2005
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Subject: Re: [Sip] Draft defining the Accept-Disposition header field
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On 1/19/05 8:43 PM, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
wrote:

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

Seems like Supported header is a better way to solve this particular issue

> The same thing with uri-list bodies. A conference server may support
> uri-list bodies for some reason, but may not support the
> 'recipient-list' disposition type. The disposition type is what tells me
> that the conference server supports (or does not support) the feature I
> want to use. (So far there are two disposition types associated with
> uri-list bodies and may be more in the future.)
>

I guess I am wondering if using the MIME dispositions is really the right
way to basically tell the UA what you want it to do with a given SIP command
- what does recipient-list mean in the context of email or other users of
MIME?
 
> Regarding debugging, the UAC can know what went wrong when it received a
> 415 response from a UAS that supports all the MIME types needed for the
> service. In this sense, the draft completes the procedures defined in
> RFC 3204 regarding rejection of body parts whose handling is required.
> 
> 
> The thing is that, as it has been pointed out in this thread,
> disposition types are usually supported in relation with a particular
> MIME type and even a particular method (e.g., a server is a message
> URI-list service but does not support the 'recipient-list' disposition
> type for INVITEs).

I agree with this observation

> 
> The solution proposed in the draft follows the definition of the
> existing Allow and Accept header fields in the sense that they do not
> provide cross header information: Allow tells you which methods are
> supported and Accept which MIME types, but there is no info about which
> MIME type is acceptable for which method. We need to decide if this is
> good enough or if we want to introduce some level of cross header
> information in the solution.

I think we should consider if we need this at all or if we are trying to do
something with disposition that just should not be done in the MIME level
but should be done in the SIP level

> 
> Regards, 
> 
> Gonzalo 
> 


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From sip-bounces@ietf.org  Mon Jan 24 13:24:32 2005
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Hi Henning,

I've taken a look at the 05 draft and noticed most of the changes you acknowledged below (for draft-04) don't seem to have made it into this draft.

In addition I noticed that there is a hanging reference from section 3.4 to section to 4.2.2 (which is not present in draft-05).

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

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

In the definition of strict mode - it doesn't seem to be quite clear whether you mean that all SIP messages in every dialogue MUST contain the RP header or every message in the domain? Is it assumed that user agents are all authorised in a 'strict' mode environment? (or just don't need to include an authorisation header or trait-based auth as with semi-strict mode)

Also in the 'semi-strict' mode a similar question arises - when it is stated that "it is not required that every SIP message have a Resource-Priority header" - is this per dialogue or per domain?

It's not quite clear whether the strict and semi-strict 'modes' apply to the configuration of a user agent, proxy or a whole domain? And how each entity behaves in each mode. In the previous draft it seemed strict mode was just 'activated' by the presence of the Require header. 

Thanks,

Piers O'Hanlon
--------------
Computer Science 
University College London


> > 
> > General comments: - The fact that it isn't mandatory to mark all
> > responses with RP headers means that one should at least make it a
> > SHOULD(MUST?) to provide "resource prioritization" based on a dialog
> > or session initiated with the RP header (possibly requiring Stateful
> > proxy?). Otherwise it seems that being able to know if a node
> > "supports Resource priority" (using OPTIONS or a Require hdr) may not
> > be of use if you can't be assured the whole dialog/session will be
> > assured of the "resource prioritization"? If it's not specified in
> > this document then I guess it will need to be specified in some
> > external policy document...(out of scope)
> 
> There are two types of resource prioritization, as alluded to in the 
> document (Introduction), namely to prioritize the signaling handling 
> itself and, more importantly, the resources such signaling controls, 
> e.g., trunk circuits.
> 
> Resources such as trunk circuits are inherently stateful, as they are at 
> UAs, and thus don't require the response to carry the R-P indication.
> 
> The only time the response would need to carry the R-P header is for 
> prioritizing the handling of the SIP request itself. Since a response is 
> part of a transaction, transaction-stateful proxies (i.e., most of them) 
> would already know that the transaction has high priority and can handle 
> this without additional indication. Thus, only the rare case of a 
> stateless proxy is relevant. However, since that proxy can't readily 
> ascertain that the response is authorized (ignoring S/MIME for now), 
> this isn't likely to be useful.
> 
> I'll add a sentence that summarizes this.
> 
> 
> > 
> > 
> > Section 3.3 - minor comment: It might be useful to label table
> > something like 'Table 1'. 
> 
> Done.
> 
> - In the second the paragraph the 420
> > response is incorrectly defined as "(Not Supported)" - when it should
> > be "(Bad Extension)". 
> 
> Fixed.
> 
> - In the same paragraph the explanation
> > following the handling of the 420 response seems at odds with the
> > description in section 4.2.2 and that of RFC3261. Firstly this is
> > because it indicates that a 420 is generated when it encounters a
> > unknown RP value (on an RP capable node) -  shouldn't that be covered
> > in section 4.2.2 
> 
> That part of the paragraph below the table is an orphaned remnant of 
> some earlier behavioral description. The error behavior is now described 
> in Section 4.2. I ditched the confusing set of sentences.
> 
> - Shouldn't a 420 be generated only by non RP
> > capable node in response to Require? Secondly it seems to reccomend
> > adding an 'Accept-Resource-Priority' header when RFC3261 specifies
> > adding an "Unsupported" header for a 420 response? Or maybe it was
> > intended to talk about the 417 handling here?
> 
> See above.
> 
> 
> > 
> > Example 7.2 - In this example - according to section 4.2.2 a 417
> > response should only generated when in strict mode, in which case the
> > shouldn't the INVITE contain the Require header?
> 
> Indeed.
> 
> > 
> > 
> > Piers O'Hanlon --------------_____________ Computer Science
> > Department University College London
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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Subject: [Sip] Abbreviated WGLC for
	draft-ietf-sip-refer-with-norefersub-00.txt
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Hi Folks,

I'd like to begin an abbreviated WGLC for:

	http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-with- 
norefersub-00.txt

Since this document is extremely short and has been discussed  
extensively, WGLC will end on February 1st.

thanks,
-rohan


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This is a multi-part message in MIME format.

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=20

Hi,list,


Is there any standard in  pri and sip mapping for the following three =
items:=20

1)user to user infomation

2)bear capacity( 64K data),UDI.

3) call transfer service indication mapping.

Thank you for your response  !

Best Regards=20
Wang Julin(Jerry Wang)=20
Product Management
Alcatel Shanghai  Bell=20
Phone:+86-21-58541240-9531
Fax:+86-21-58342941
Mobile: 13817861240
E-mail:Julin.Wang@alcatel-sbell.com.cn



=20

=20

=20

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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dgb2312">


<META content=3D"MSHTML 6.00.2800.1479" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<P align=3Dleft><FONT face=3DArial><FONT size=3D2>Hi,<SPAN=20
class=3D442494208-21012005>list,</SPAN></FONT></FONT></P>
<P>
<P align=3Dleft><FONT face=3DArial><FONT size=3D2><SPAN=20
class=3D448062207-21012005></SPAN></FONT></FONT></P>
<P align=3Dleft><FONT size=3D2><SPAN class=3D448062207-21012005><FONT =
face=3DArial>Is=20
there any standard in&nbsp;<FONT size=3D3> pri and sip mapping for the=20
following&nbsp;<SPAN class=3D398093501-25012005>three=20
</SPAN>items:</FONT></FONT></SPAN></FONT> </P>
<P></P>
<P align=3Dleft><FONT face=3DArial><FONT size=3D2><SPAN=20
class=3D448062207-21012005>1)</SPAN></FONT></FONT><FONT =
face=3DArial><FONT=20
size=3D2><SPAN class=3D448062207-21012005>user to user=20
infomation</SPAN></FONT></FONT></P>
<P align=3Dleft><FONT face=3DArial><FONT size=3D2><SPAN=20
class=3D448062207-21012005>2)bear capacity( <FONT size=3D3>64K=20
data</FONT>),UDI.</SPAN></FONT></FONT></P>
<P align=3Dleft><FONT><FONT face=3DArial size=3D2><SPAN =
class=3D448062207-21012005><SPAN=20
class=3D398093501-25012005>3) call transfer service indication=20
mapping.</SPAN></SPAN></FONT></FONT></P>
<P align=3Dleft><FONT face=3DArial size=3D2><SPAN =
class=3D448062207-21012005>Thank=20
you<FONT face=3D=CB=CE=CC=E5><SPAN class=3D442494208-21012005> for=20
your&nbsp;response&nbsp;</SPAN><SPAN=20
class=3D442494208-21012005>&nbsp;</SPAN></FONT>!</SPAN></FONT></P>
<DIV><FONT size=3D2>B<FONT face=3DArial>est</FONT></FONT> <FONT =
face=3D=CB=CE=CC=E5=20
size=3D2>R</FONT><FONT face=3DArial size=3D2>egards</FONT> <BR><FONT =
face=3DArial=20
size=3D2><FONT face=3D=CB=CE=CC=E5>Wang Julin(Jerry Wang) =
</FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Product Management</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Alcatel Shanghai&nbsp; Bell =
</FONT></DIV><FONT=20
face=3DArial size=3D2>
<DIV>Phone:+86-21-58541240-9531<BR>Fax:+86-21-58342941<BR>Mobile:=20
13817861240</DIV>
<DIV>E-mail:Julin.Wang@alcatel-sbell.com.cn</FONT><BR></DIV>
<P><FONT size=3D2></FONT><BR>&nbsp;</P>
<P><FONT size=3D2></FONT>&nbsp;</P>
<DIV>&nbsp;</DIV></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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From: "Tai-Hsing Yu" <fhubert@ms37.hinet.net>
To: <sip@ietf.org>
Date: Tue, 25 Jan 2005 15:47:12 +0800
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Subject: [Sip] Questions about request message larger than 1300 bytes
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Hello,

        I have some questions about handling SIP msgs that are larger =
than 1300 bytes. RFC 3261, chapter 18 as listed, mentions that
if request message is larger than 1300 bytes, it MUST be sent using TCP =
because of the meesage size constraints. But I don=A1=A6t know
how to handle the following scenarios, could anyone help me?=20

1. The UA1 registers to Proxy/Registrar with =A1=A7transport=3Dudp=A1=A8 =
or without =A1=A7transport=A1=A8 parameter. According to RFC3261, the =
proxy
should try to send the request message to UA1 using default transport, =
which means UDP. Now suppose that a request comes and is
larger than 1500, what should the proxy do? Still try UDP first or =
always use TCP to send the request to UA1? If the proxy selects
the first stretagy, and, unfortunately, the message is splitted into =
more than one pieces and only the first one is received by UA1.
The received message is incorrect and might cause UA1 to do incorrect =
response. Can the proxy select the second stretagy, i.e, using
TCP, even the client only registers with =A1=A7transport=3Dudp=A1=A8? =
Which stretagy is correct or better?

=20

2. The UA2 sends the request to proxy using UDP with request message =
size smaller than 1300 bytes. The Via header in the request is
UDP. But the response backs to the proxy is larger than 1500 bytes. What =
should the proxy do? Using TCP or UDP to transmit the
response to UA2?

=20

Any acknowledgement will be deeply appreciated.

Sincerely,

Fish

=20

=A1=A7If a request is within 200 bytes of the path MTU, or if it is =
larger than 1300 bytes and the path MTU is unknown, the request MUST
be sent using an RFC 2914 [43] congestion controlled transport protocol, =
such as TCP. If this causes a change in the transport
protocol from the one indicated in the top Via, the value in the top Via =
MUST be changed. This prevents fragmentation of messages
over UDP and provides congestion control for larger messages. However, =
implementations MUST be able to handle messages up to the
maximum datagram packet size. For UDP, this size is 65,535 bytes, =
including IP and UDP headers.

The 200 byte "buffer" between the message size and the MTU accommodates =
the fact that the response in SIP can be larger than the
request. This happens due to the addition of Record-Route header field =
values to the responses to INVITE, for example. With the
extra buffer, the response can be about 170 bytes larger than the =
request, and still not be fragmented on IPv4 (about 30 bytes is
consumed by IP/UDP, assuming no IPSec). 1300 is chosen when path MTU is =
not known, based on the assumption of a 1500 byte Ethernet
MTU.

If an element sends a request over TCP because of these message size =
constraints, and that request would have otherwise been sent
over UDP, if the attempt to establish the connection generates either an =
ICMP Protocol Not Supported, or results in a TCP reset, the
element SHOULD retry the request, using UDP. This is only to provide =
backwards compatibility with RFC 2543 compliant implementations
that do not support TCP. It is anticipated that this behavior will be =
deprecated in a future revision of this specification.=A1=A8

=20

=20

=20


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<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>Hello,<o:p><=
/o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1
face=3D"Times New Roman"><span lang=3DEN-US =
style=3D'font-size:9.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</spa=
n></font><font
size=3D1 face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'> I
have some questions about handling SIP msgs that are larger than 1300 =
bytes.
RFC 3261, chapter 18 as listed, mentions that if request message is =
larger than
1300 bytes, it MUST be sent using TCP because of the meesage size =
constraints.
But I don</span></font><font size=3D1><span lang=3DEN-US =
style=3D'font-size:9.0pt'>=A1=A6</span></font><font
size=3D1 face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>t
know how to handle the following scenarios, could anyone help me? =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>1. The UA1 =
registers to
Proxy/Registrar with </span></font><font size=3D1><span lang=3DEN-US
style=3D'font-size:9.0pt'>=A1=A7</span></font><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>transport=3D=
udp</span></font><font
size=3D1><span lang=3DEN-US =
style=3D'font-size:9.0pt'>=A1=A8</span></font><font size=3D1
face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'> or without
</span></font><font size=3D1><span lang=3DEN-US =
style=3D'font-size:9.0pt'>=A1=A7</span></font><font
size=3D1 face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>transport</s=
pan></font><font
size=3D1><span lang=3DEN-US =
style=3D'font-size:9.0pt'>=A1=A8</span></font><font size=3D1
face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'> parameter.
According to RFC3261, the proxy should try to send the request message =
to UA1
using default transport, which means UDP. Now suppose that a request =
comes and
is larger than 1500, what should the proxy do? Still try UDP first or =
always
use TCP to send the request to UA1? If the proxy selects the first =
stretagy,
and, unfortunately, the message is splitted into more than one pieces =
and only
the first one is received by UA1. The received message is incorrect and =
might
cause UA1 to do incorrect response. Can the proxy select the second =
stretagy,
i.e, using TCP, even the client only registers with </span></font><font =
size=3D1><span
lang=3DEN-US style=3D'font-size:9.0pt'>=A1=A7</span></font><font =
size=3D1 face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>transport=3D=
udp</span></font><font
size=3D1><span lang=3DEN-US =
style=3D'font-size:9.0pt'>=A1=A8</span></font><font size=3D1
face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>? Which
stretagy is correct or better?<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>2. The UA2 =
sends the
request to proxy using UDP with request message size smaller than 1300 =
bytes.
The Via header in the request is UDP. But the response backs to the =
proxy is
larger than 1500 bytes. What should the proxy do? Using TCP or UDP to =
transmit
the response to UA2?<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>Any =
acknowledgement will be
deeply appreciated.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>Sincerely,<o=
:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>Fish<o:p></o=
:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1
face=3D"Times New Roman"><span lang=3DEN-US =
style=3D'font-size:9.0pt'>=A1=A7</span></font><font
size=3D1 face=3D=B7s=B2=D3=A9=FA=C5=E9><span lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>If a
request is within 200 bytes of the path MTU, or if it is larger than =
1300 bytes
and the path MTU is unknown, the request MUST be sent using an RFC 2914 =
[43]
congestion controlled transport protocol, such as TCP. If this causes a =
change
in the transport protocol from the one indicated in the top Via, the =
value in
the top Via MUST be changed. This prevents fragmentation of messages =
over UDP
and provides congestion control for larger messages. However, =
implementations
MUST be able to handle messages up to the maximum datagram packet size. =
For
UDP, this size is 65,535 bytes, including IP and UDP =
headers.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>The 200 =
byte
&quot;buffer&quot; between the message size and the MTU accommodates the =
fact
that the response in SIP can be larger than the request. This happens =
due to
the addition of Record-Route header field values to the responses to =
INVITE,
for example. With the extra buffer, the response can be about 170 bytes =
larger
than the request, and still not be fragmented on IPv4 (about 30 bytes is
consumed by IP/UDP, assuming no IPSec). 1300 is chosen when path MTU is =
not
known, based on the assumption of a 1500 byte Ethernet =
MTU.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'>If an =
element sends a
request over TCP because of these message size constraints, and that =
request
would have otherwise been sent over UDP, if the attempt to establish the
connection generates either an ICMP Protocol Not Supported, or results =
in a TCP
reset, the element SHOULD retry the request, using UDP. This is only to =
provide
backwards compatibility with RFC 2543 compliant implementations that do =
not
support TCP. It is anticipated that this behavior will be deprecated in =
a
future revision of this specification.</span></font><font size=3D1><span
lang=3DEN-US style=3D'font-size:9.0pt'>=A1=A8</span></font><font =
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lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'><o:p></o:p><=
/span></font></p>

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lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D1 =
face=3D=B7s=B2=D3=A9=FA=C5=E9><span
lang=3DEN-US =
style=3D'font-size:9.0pt;font-family:=B7s=B2=D3=A9=FA=C5=E9'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-US =
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--===============1616295453==--




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From: Avasarala Ranjit-A20990 <ranjit@motorola.com>
To: FCG Wang JuLin <JuLin.Wang@alcatel-sbell.com.cn>, sip@ietf.org,
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Subject: RE: [Sip] pri and sip mapping issue
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Hi
    Check RFC 3398. It provides a mapping between ISDN and SIP signaling. 
 
 

Regards 
Ranjit 



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of FCG Wang JuLin
Sent: Tuesday, January 25, 2005 7:09 AM
To: sip@ietf.org; sipforum-discussion@lists.su.se
Subject: [Sip] pri and sip mapping issue


 

Hi,list,


Is there any standard in  pri and sip mapping for the following three items: 

1)user to user infomation

2)bear capacity( 64K data),UDI.

3) call transfer service indication mapping.

Thank you for your response  !

Best Regards 
Wang Julin(Jerry Wang) 
Product Management
Alcatel Shanghai  Bell 
Phone:+86-21-58541240-9531
Fax:+86-21-58342941
Mobile: 13817861240
E-mail:Julin.Wang@alcatel-sbell.com.cn



 

 

 

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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--===============0003179538==--



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Cc: merisanchez@yahoo.es
Subject: [Sip] Calls emulator
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Hi!
Do you know any phone calls emulator software?
I need it to emulate calls against the SIP proxy.
Thank you all
Mar=EDa=



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Subject: RE: [Sip] Calls emulator
Date: Tue, 25 Jan 2005 09:54:26 +0100
Message-ID: <B7192C0D8D60754DADA9E22294C573698112B7@mailserver.hotsip.com>
Thread-Topic: [Sip] Calls emulator
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From: "Fevzi Konduk" <fevzi.konduk@hotsip.com>
To: <mslabrador@sztele.com>, <sip@ietf.org>
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Hi,=20
You could try:=20
http://sipp.sourceforge.net/

/Fevzi

> -----Original Message-----
> From: mslabrador@sztele.com [mailto:mslabrador@sztele.com]
> Sent: den 25 januari 2005 09:32
> To: sip@ietf.org
> Cc: merisanchez@yahoo.es
> Subject: [Sip] Calls emulator [bcc][faked-from]
>=20
> Hi!
> Do you know any phone calls emulator software?
> I need it to emulate calls against the SIP proxy.
> Thank you all
> Mar=EDa
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


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From: Tom Taylor <taylor@nortelnetworks.com>
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To: FCG Wang JuLin <JuLin.Wang@alcatel-sbell.com.cn>
Subject: Re: [Sip] pri and sip mapping issue
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Hi.  My responses below marked with [PTT].

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

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

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

[PTT] Leaving aside 3xx redirection responses, you may want to look at the
Request_History header field
(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
> 
> Thank you for your response  !
> 
> Best Regards
> Wang Julin(Jerry Wang)
> Product Management
> Alcatel Shanghai  Bell
> Phone:+86-21-58541240-9531
> Fax:+86-21-58342941
> Mobile: 13817861240
> E-mail:Julin.Wang@alcatel-sbell.com.cn
> 
> 
>  
> 
>  
> 
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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: Tom Taylor <taylor@nortelnetworks.com>
Subject: Re: [Sip] pri and sip mapping issue
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Hi,
I have sent a request for information to the author about the ClearMode
Draft and have not received any responce. Are we sure that this draft is
alive?

Thanks
Ramesh
Tom Taylor wrote:

>Hi.  My responses below marked with [PTT].
>
>FCG Wang JuLin wrote:
>  
>
>> 
>>
>>Hi,list,
>>
>>Is there any standard in  pri and sip mapping for the following three 
>>items:
>>
>>1)user to user infomation
>>    
>>
>
>[PTT]  To carry the information itself, the obvious method is to add a body of
>type application/octet-stream.  The issue comes in how to signal the service --
>especially, how to indicate what that body is for.
>  
>
>>2)bear capacity( 64K data),UDI.
>>    
>>
>
>[PTT] You can signal and use CLEARMODE
>(http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmode-05.txt).  This
>has been approved by the IESG but is stuck in the RFC Editor's queue because it
>has a dependency on draft-ietf-mmusic-sdp-new-23.txt, which still has to pass Last
>Call.
>  
>
>>3) call transfer service indication mapping.
>>    
>>
>
>[PTT] Leaving aside 3xx redirection responses, you may want to look at the
>Request_History header field
>(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
>  
>
>>Thank you for your response  !
>>
>>Best Regards
>>Wang Julin(Jerry Wang)
>>Product Management
>>Alcatel Shanghai  Bell
>>Phone:+86-21-58541240-9531
>>Fax:+86-21-58342941
>>Mobile: 13817861240
>>E-mail:Julin.Wang@alcatel-sbell.com.cn
>>
>>
>> 
>>
>> 
>>
>> 
>>
>>
>>------------------------------------------------------------------------
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>    
>>
>
>  
>

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


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

Thanks
Ramesh
Tom Taylor wrote:

>Hi.  My responses below marked with [PTT].
>
>FCG Wang JuLin wrote:
>  
>
>> 
>>
>>Hi,list,
>>
>>Is there any standard in  pri and sip mapping for the following three 
>>items:
>>
>>1)user to user infomation
>>    
>>
>
>[PTT]  To carry the information itself, the obvious method is to add a body of
>type application/octet-stream.  The issue comes in how to signal the service --
>especially, how to indicate what that body is for.
>  
>
>>2)bear capacity( 64K data),UDI.
>>    
>>
>
>[PTT] You can signal and use CLEARMODE
>(http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmode-05.txt).  This
>has been approved by the IESG but is stuck in the RFC Editor's queue because it
>has a dependency on draft-ietf-mmusic-sdp-new-23.txt, which still has to pass Last
>Call.
>  
>
>>3) call transfer service indication mapping.
>>    
>>
>
>[PTT] Leaving aside 3xx redirection responses, you may want to look at the
>Request_History header field
>(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
>  
>
>>Thank you for your response  !
>>
>>Best Regards
>>Wang Julin(Jerry Wang)
>>Product Management
>>Alcatel Shanghai  Bell
>>Phone:+86-21-58541240-9531
>>Fax:+86-21-58342941
>>Mobile: 13817861240
>>E-mail:Julin.Wang@alcatel-sbell.com.cn
>>
>>
>> 
>>
>> 
>>
>> 
>>
>>
>>------------------------------------------------------------------------
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>    
>>
>
>  
>
------
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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.  My responses below marked with [PTT].

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

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

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

[PTT] Leaving aside 3xx redirection responses, you may want to look at the
Request_History header field
(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
> 
> Thank you for your response  !
> 
> Best Regards
> Wang Julin(Jerry Wang)
> Product Management
> Alcatel Shanghai  Bell
> Phone:+86-21-58541240-9531
> Fax:+86-21-58342941
> Mobile: 13817861240
> E-mail:Julin.Wang@alcatel-sbell.com.cn
> 
> 
>  
> 
>  
> 
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com
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_______________________________________________
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I have deleted the SIPForum cross-posting -- I think Justin Millner has
(inadvertently, I'm sure) set up some sort of mail loop through that list.

Now that the IESG has approved the document (some months ago) the availability of
the author is irrelevant.

Ramesh Ramamurthy wrote:
> Hi,
> I have sent a request for information to the author about the ClearMode
> Draft and have not received any responce. Are we sure that this draft is
> alive?
> 
> Thanks
> Ramesh
> Tom Taylor wrote:
> 
> 
>>Hi.  My responses below marked with [PTT].
>>
>>FCG Wang JuLin wrote:
>> 
>>
>>
>>>
>>>Hi,list,
>>>
>>>Is there any standard in  pri and sip mapping for the following three 
>>>items:
>>>
>>>1)user to user infomation
>>>   
>>>
>>
>>[PTT]  To carry the information itself, the obvious method is to add a body of
>>type application/octet-stream.  The issue comes in how to signal the service --
>>especially, how to indicate what that body is for.
>> 
>>
>>
>>>2)bear capacity( 64K data),UDI.
>>>   
>>>
>>
>>[PTT] You can signal and use CLEARMODE
>>(http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmode-05.txt).  This
>>has been approved by the IESG but is stuck in the RFC Editor's queue because it
>>has a dependency on draft-ietf-mmusic-sdp-new-23.txt, which still has to pass Last
>>Call.
>> 
>>
>>
>>>3) call transfer service indication mapping.
>>>   
>>>
>>
>>[PTT] Leaving aside 3xx redirection responses, you may want to look at the
>>Request_History header field
>>(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-06.txt).
>> 
>>
>>
>>>Thank you for your response  !
>>>
>>>Best Regards
>>>Wang Julin(Jerry Wang)
>>>Product Management
>>>Alcatel Shanghai  Bell
>>>Phone:+86-21-58541240-9531
>>>Fax:+86-21-58342941
>>>Mobile: 13817861240
>>>E-mail:Julin.Wang@alcatel-sbell.com.cn
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>------------------------------------------------------------------------
>>>
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>This list is for NEW development of the core SIP Protocol
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sipping@ietf.org for new developments on the application of sip
>>>   
>>>
>>
>> 
>>
> 
> 
> 

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

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


From sip-bounces@ietf.org  Wed Jan 26 17:28:23 2005
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Abbreviated WGLC for
	draft-ietf-sip-refer-with-norefersub-00.txt
Date: Wed, 26 Jan 2005 16:22:00 -0600
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Initial LC Comments:

First paragraph of Section 3 calls out 481ing an initial NOTIFY. The  
-dialog-usage-
draft shows that is a bad idea, especially for in-dialog REFERs.  
Suggest trimming
the sentence to end after (Expires 0).

Section 4 fourth paragraph: It would help implementers to call out  
explicitly that the
200 response to the REFER will not contain a Supported or Require  
header containing
the norefersub token.

Section 4 fifth paragraph: This needs more definition. How is it  
rejected? 603, 420, or
something else? Implementors will need some guidance here on when to  
use which
error response. I think (but am not dead certain at the moment) that  
420 is almost always
what you want to use.

Section 4 sixth paragraph: Is Supported: in the response really the  
right thing? Eventlist
uses Require: in the response to a SUBSCRIBE. The intent of the  
normative behavior in
this paragraph is to let a Refer-issuer who only included Supported:  
know whether or
not a subscription was actually created. Require: (I'm using this  
extension) in the response
fits that better than Supported: (I could use this extension if you  
want to).

Section 4 sixth paragraph: Suggest replacing "In this case, no dialog  
is created" with
"In this case, no implicit subscription is created. As a consequence,  
no new dialog
is created if this REFER was issued outside any existing dialog."

Section 5: Probably worth also calling out using Max-Forwards: 0 to  
prevent forking
for those (rare now, but hopefully not rare forever) cases where  
signaling can get to
the refer target without going through a proxy.

Section 6: Must replace "tradewind" with "example" throughout.

Section 8: "threads" should be "threats", It would be good to pull any  
discussion
that led to this conclusion together for when the IESG asks if we  
_really_ thought
about this. Have we talked about what would happen if a SIP  
intermediary (or an
unauthorized MitM) just injected a Require: norefersub into a REFER  
request?

RjS


On Jan 24, 2005, at 5:07 PM, Rohan Mahy wrote:

> Hi Folks,
>
> I'd like to begin an abbreviated WGLC for:
>
> 	http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-with- 
> norefersub-00.txt
>
> Since this document is extremely short and has been discussed  
> extensively, WGLC will end on February 1st.
>
> thanks,
> -rohan
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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


From sip-bounces@ietf.org  Wed Jan 26 19:04:52 2005
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Hi All,

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

Can an  Implementation ( Gateway ) send Media before sending out the 183
Session in Progress over UDP Transport in response to a INVITE   ? Is
there any RFC/draft on the details of 183 Session in progress and Media
handling ?=20

Thanks in advance .

Best Regards,  =20

=20

Dipankar Bhattacharyya=20
Systems Engineer=20
Communication Infrastructure Group

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

E M P I R I X=20
www.empirix.com
<file:///C:\Documents%20and%20Settings\rknaub\Application%20Data\Microso
ft\Signatures\www.empirix.com>=20

=20

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Hi All,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp; This is query regarding Early =
Media ( 183
Session in Progress ) .<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Can an &nbsp;Implementation ( Gateway ) send Media =
before
sending out the 183 Session in Progress over UDP Transport in response =
to a
INVITE &nbsp;&nbsp;? Is there any RFC/draft on the details of 183 =
Session in
progress and Media handling ? <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks in advance .<o:p></o:p></span></font></p>

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

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

<p><b><font size=3D2 color=3Dnavy face=3D"Times New Roman"><span =
style=3D'font-size:
10.0pt;color:navy;font-weight:bold'>Dipankar Bhattacharyya <br>
</span></font></b><font size=3D2 color=3Dnavy><span =
style=3D'font-size:10.0pt;
color:navy'>Systems Engineer <br>
Communication Infrastructure Group</span></font><o:p></o:p></p>

<p><b><font size=3D2 color=3Dred face=3D"Times New Roman"><span =
style=3D'font-size:
10.0pt;color:red;font-weight:bold'>p |&nbsp; </span></font></b><b><font =
size=3D2
color=3Dnavy><span =
style=3D'font-size:10.0pt;color:navy;font-weight:bold'>972.401.4112</span=
></font></b><b><font
size=3D2 color=3Dred><span =
style=3D'font-size:10.0pt;color:red;font-weight:bold'>&nbsp;f
|</span></font></b><b><font size=3D2 color=3Dnavy><span =
style=3D'font-size:10.0pt;
color:navy;font-weight:bold'> 972.596.4909 </span></font></b><b><font =
size=3D2
color=3Dred><span =
style=3D'font-size:10.0pt;color:red;font-weight:bold'>m =
|</span></font></b><b><font
size=3D2 color=3Dnavy><span =
style=3D'font-size:10.0pt;color:navy;font-weight:bold'>
214.766.1479</span></font></b><font size=3D2 color=3Dnavy><span =
style=3D'font-size:
10.0pt;color:navy'><br>
</span></font><b><font size=3D2 color=3Dred><span =
style=3D'font-size:10.0pt;
color:red;font-weight:bold'>email |</span></font></b><font size=3D2 =
color=3Dnavy><span
style=3D'font-size:10.0pt;color:navy'> <a =
href=3D"mailto:dbhattacharyya@empirix.com"
title=3D"mailto:dbhattacharyya@empirix.com">dbhattacharyya@empirix.com</a=
></span></font><font
color=3Dnavy><span style=3D'color:navy'><br>
</span></font><st1:Street w:st=3D"on"><st1:address w:st=3D"on"><font =
size=3D2
  color=3Dnavy><span style=3D'font-size:10.0pt;color:navy'>6100 Ohio =
Drive , Suite
  2011</span></font></st1:address></st1:Street><font size=3D2 =
color=3Dnavy><span
style=3D'font-size:10.0pt;color:navy'> </span></font><b><font size=3D2 =
color=3Dred><span
style=3D'font-size:10.0pt;color:red;font-weight:bold'>|</span></font></b>=
<font
size=3D2 color=3Dnavy><span =
style=3D'font-size:10.0pt;color:navy'>&nbsp;&nbsp;&nbsp;<st1:place
w:st=3D"on"><st1:City w:st=3D"on">Plano</st1:City>,&nbsp;<st1:State =
w:st=3D"on">TX</st1:State></st1:place>
&nbsp;</span></font><b><font size=3D2 color=3Dred><span =
style=3D'font-size:10.0pt;
color:red;font-weight:bold'>|</span></font></b><font size=3D2 =
color=3Dred><span
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size=3D2
color=3Dnavy><span style=3D'font-size:10.0pt;color:navy'>75024 =
.</span></font><o:p></o:p></p>

<p><b><font size=3D2 color=3Dnavy face=3D"Times New Roman"><span =
style=3D'font-size:
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</span></font></b><a
href=3D"file:///C:\Documents%20and%20Settings\rknaub\Application%20Data\M=
icrosoft\Signatures\www.empirix.com"
title=3D"file:///C:/Documents%20and%20Settings/rknaub/Application%20Data/=
Microsoft/Signatures/www.empirix.com"><font
size=3D2
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Microsoft/Signatures/www.empirix.com"><span
title=3D"file:///C:/Documents%20and%20Settings/rknaub/Application%20Data/=
Microsoft/Signatures/www.empirix.com"><span
style=3D'font-size:10.0pt'>www.empirix.com</span></span></font></a><o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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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
--===============1611550584==--



From sip-bounces@ietf.org  Thu Jan 27 08:53:50 2005
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Tom Taylor'" <taylor@nortelnetworks.com>,
        Ramesh Ramamurthy
	<ramesh@peakss.com>
Subject: RE: [Sip] pri and sip mapping issue
Date: Thu, 27 Jan 2005 13:45:36 -0000
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Not completely irrelevant, as you still need the author participation to get it out of the Auth48 at the end of the publication process.

I believe it is currently stuck in the queue because it is waiting for draft-ietf-mmusic-sdp-new to get published - you know that document that never quite seems to get there because despite all the polishing, it never shines brightly enough!!

regards

Keith

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On 
> Behalf Of Tom Taylor
> Sent: 26 January 2005 15:46
> To: Ramesh Ramamurthy
> Cc: sip@ietf.org; Tom-PT Taylor; FCG Wang JuLin
> Subject: Re: [Sip] pri and sip mapping issue
> 
> 
> I have deleted the SIPForum cross-posting -- I think Justin 
> Millner has
> (inadvertently, I'm sure) set up some sort of mail loop 
> through that list.
> 
> Now that the IESG has approved the document (some months ago) 
> the availability of
> the author is irrelevant.
> 
> Ramesh Ramamurthy wrote:
> > Hi,
> > I have sent a request for information to the author about 
> the ClearMode
> > Draft and have not received any responce. Are we sure that 
> this draft is
> > alive?
> > 
> > Thanks
> > Ramesh
> > Tom Taylor wrote:
> > 
> > 
> >>Hi.  My responses below marked with [PTT].
> >>
> >>FCG Wang JuLin wrote:
> >> 
> >>
> >>
> >>>
> >>>Hi,list,
> >>>
> >>>Is there any standard in  pri and sip mapping for the 
> following three 
> >>>items:
> >>>
> >>>1)user to user infomation
> >>>   
> >>>
> >>
> >>[PTT]  To carry the information itself, the obvious method 
> is to add a body of
> >>type application/octet-stream.  The issue comes in how to 
> signal the service --
> >>especially, how to indicate what that body is for.
> >> 
> >>
> >>
> >>>2)bear capacity( 64K data),UDI.
> >>>   
> >>>
> >>
> >>[PTT] You can signal and use CLEARMODE
> >>(http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clea
> rmode-05.txt).  This
> >>has been approved by the IESG but is stuck in the RFC 
> Editor's queue because it
> >>has a dependency on draft-ietf-mmusic-sdp-new-23.txt, which 
> still has to pass Last
> >>Call.
> >> 
> >>
> >>
> >>>3) call transfer service indication mapping.
> >>>   
> >>>
> >>
> >>[PTT] Leaving aside 3xx redirection responses, you may want 
> to look at the
> >>Request_History header field
> >>(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-
info-06.txt).
>> 
>>
>>
>>>Thank you for your response  !
>>>
>>>Best Regards
>>>Wang Julin(Jerry Wang)
>>>Product Management
>>>Alcatel Shanghai  Bell
>>>Phone:+86-21-58541240-9531
>>>Fax:+86-21-58342941
>>>Mobile: 13817861240
>>>E-mail:Julin.Wang@alcatel-sbell.com.cn
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>------------------------------------------------------------------------
>>>
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>This list is for NEW development of the core SIP Protocol
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sipping@ietf.org for new developments on the application of sip
>>>   
>>>
>>
>> 
>>
> 
> 
> 

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

_______________________________________________
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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

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


From sip-bounces@ietf.org  Thu Jan 27 09:23:14 2005
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To: "Bhattacharyya, Dipankar" <DBhattacharyya@empirix.com>
Subject: Re: [Sip] Query on Early Media ( 183 Session in Progress )
References: <2A9AA9C52024F94A96634E39B9858FF1013473EE@empwilex1.empirix.com>
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You can send media as soon as the offer/answer exchange has provided you 
with a place to send it.

So, if the initial invite contains an offer, the UAS may begin sending 
media as soon as it receives that offer, even before sending any 
response. In this case, what the answer in the 183 does is permit the 
UAC to send media to the UAS, even prior to call completion.

Whether you *want* to send media that early, and whether you want to 
render it if you receive it, is another matter.

	Paul

Bhattacharyya, Dipankar wrote:
> Hi All,
> 
>     This is query regarding Early Media ( 183 Session in Progress ) .
> 
> Can an  Implementation ( Gateway ) send Media before sending out the 183 
> Session in Progress over UDP Transport in response to a INVITE   ? Is 
> there any RFC/draft on the details of 183 Session in progress and Media 
> handling ?
> 
> Thanks in advance .
> 
> Best Regards,   
> 
>  
> 
> *Dipankar Bhattacharyya
> *Systems Engineer
> Communication Infrastructure Group
> 
> *p |  **972.401.4112** f |** 972.596.4909 **m |** 214.766.1479*
> *email |* dbhattacharyya@empirix.com <mailto:dbhattacharyya@empirix.com>
> 6100 Ohio Drive , Suite 2011 *|*   Plano, TX  *|*  75024 .
> 
> *E M P I R I X
> *www.empirix.com 
> <file:///C:\Documents%20and%20Settings\rknaub\Application%20Data\Microsoft\Signatures\www.empirix.com>
> 
>  
> 
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From sip-bounces@ietf.org  Sat Jan 29 11:10:52 2005
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Date: Fri, 28 Jan 2005 17:31:13 -0800
Subject: Re: [Sip] draft-hilt-sip-ext-neg-00
From: Cullen Jennings <fluffy@cisco.com>
To: Volker Hilt <volkerh@bell-labs.com>, "sip@ietf.org" <sip@ietf.org>
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I don't know what to think of this. My gut reaction is that the Accept
mechanism is on the verge of breaking. If we do this, pretty soon every
single SIP messages is going to have 5k bytes of accept headers.

Perhaps we need to rethink about *when* certain things that could go in the
accept header are put in so not every messages has all of them.

Adam pointed out that the number of MIME types the average PC client can
accept is truly staggering and most clients just don't bother to provide
most of the but still accept them. This resulted in MSRP defining * to just
say, ok yah and give me whatever else you want.

If someone receives a geopriv XML extension inside some PIDF, they still
work because the XML allows them to skip what they don't understand. So I
guess I am asking what is a use case where we need this level of detail.

Cullen

PS - nice draft

On 1/20/05 12:03 PM, "Volker Hilt" <volkerh@bell-labs.com> wrote:

> Folks,
> 
> we have submitted a draft, which defines a new Accept header field
> parameter that can be used to indicate support for MIME type extensions.
> In particular, this parameter can be used to indicate support for
> additional XML namespaces, which extend an XML-based media type.
> 
> http://www.ietf.org/internet-drafts/draft-hilt-sip-ext-neg-00.txt
> 
> Comments are very welcome.
> 
> -- Volker
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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


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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] draft-hilt-sip-ext-neg-00
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Cc: "sip@ietf.org" <sip@ietf.org>, Volker Hilt <volkerh@bell-labs.com>,
        "gonzalo.camarillo@ericsson.com" <Gonzalo.Camarillo@ericsson.com>
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I think generally Accept-* makes sense if the receiver of that header 
has true choices in sending alternate content. This makes some, very 
limited, sense for content, where sometimes content is available in 
multiple formats, although this also seems to be rather rare in 
practice. In other cases, it doesn't really help much except to 
accelerate failure discovery and, maybe, avoid transmission of optional 
data that the receiver has to toss due to lack of understanding.

It is not clear that either case really applies to the labeling of XML 
schema.

Cullen Jennings wrote:
> I don't know what to think of this. My gut reaction is that the Accept
> mechanism is on the verge of breaking. If we do this, pretty soon every
> single SIP messages is going to have 5k bytes of accept headers.
> 
> Perhaps we need to rethink about *when* certain things that could go in the
> accept header are put in so not every messages has all of them.
> 
> Adam pointed out that the number of MIME types the average PC client can
> accept is truly staggering and most clients just don't bother to provide
> most of the but still accept them. This resulted in MSRP defining * to just
> say, ok yah and give me whatever else you want.
> 
> If someone receives a geopriv XML extension inside some PIDF, they still
> work because the XML allows them to skip what they don't understand. So I
> guess I am asking what is a use case where we need this level of detail.
> 
> Cullen
> 
> PS - nice draft
> 
> On 1/20/05 12:03 PM, "Volker Hilt" <volkerh@bell-labs.com> wrote:
> 
> 
>>Folks,
>>
>>we have submitted a draft, which defines a new Accept header field
>>parameter that can be used to indicate support for MIME type extensions.
>>In particular, this parameter can be used to indicate support for
>>additional XML namespaces, which extend an XML-based media type.
>>
>>http://www.ietf.org/internet-drafts/draft-hilt-sip-ext-neg-00.txt
>>
>>Comments are very welcome.
>>
>>-- Volker
>>
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP 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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Henning Schulzrinne wrote:
> I think generally Accept-* makes sense if the receiver of that header 
> has true choices in sending alternate content. This makes some, very 
> limited, sense for content, where sometimes content is available in 
> multiple formats, although this also seems to be rather rare in 
> practice. In other cases, it doesn't really help much except to 
> accelerate failure discovery and, maybe, avoid transmission of optional 
> data that the receiver has to toss due to lack of understanding.
> 
I fully agree. The intension of this parameter is not that the sender of 
announces every single XML namespace it supports. The idea is that there 
are cases where it is useful for the receiver of an Accept header 
receiver to know which extensions the sender supports because it can 
provide content in different formats or wants to avoid costs for 
gathering and/or transmitting information that is tossed away.

An example where this is useful is PIDF-LO where the sender of an Accept 
header can announce the location object formats it supports (e.g. 
civilLoc). Another example are session policies, where the sender can 
announce supported policy extensions. E.g. if a firewall traversal 
extension is supported, the policy server can open pinholes and return 
the external addresses (which it shouldn't do if the client uses a 
different firewall traversal method). A third example is a wireless 
client that wants to use a minimal schema to minimize the size of a 
received document.

I agree with Cullens point that the use of this parameter needs to be 
limited to these (probably rare) cases where it is useful.

Thanks,

Volker



> It is not clear that either case really applies to the labeling of XML 
> schema.
> 
> Cullen Jennings wrote:
>> I don't know what to think of this. My gut reaction is that the Accept
>> mechanism is on the verge of breaking. If we do this, pretty soon every
>> single SIP messages is going to have 5k bytes of accept headers.
>>
>> Perhaps we need to rethink about *when* certain things that could go 
>> in the
>> accept header are put in so not every messages has all of them.
>>
>> Adam pointed out that the number of MIME types the average PC client can
>> accept is truly staggering and most clients just don't bother to provide
>> most of the but still accept them. This resulted in MSRP defining * to 
>> just
>> say, ok yah and give me whatever else you want.
>>
>> If someone receives a geopriv XML extension inside some PIDF, they still
>> work because the XML allows them to skip what they don't understand. So I
>> guess I am asking what is a use case where we need this level of detail.
>>
>> Cullen
>>
>> PS - nice draft
>>
>> On 1/20/05 12:03 PM, "Volker Hilt" <volkerh@bell-labs.com> wrote:
>>
>>
>>> Folks,
>>>
>>> we have submitted a draft, which defines a new Accept header field
>>> parameter that can be used to indicate support for MIME type extensions.
>>> In particular, this parameter can be used to indicate support for
>>> additional XML namespaces, which extend an XML-based media type.
>>>
>>> http://www.ietf.org/internet-drafts/draft-hilt-sip-ext-neg-00.txt
>>>
>>> Comments are very welcome.
>>>
>>> -- Volker
>>>
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP 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


-- 
Volker Hilt                      Bell Labs / Lucent Technologies
phone: +1 732 332 6432           101 Crawfords Corner Rd
email: volkerh@bell-labs.com     Holmdel, NJ 07733


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

Today, when a request gets forked, the forking proxy will "collect" final responses received from the different dialogs.

The calling UAC, however, does now know when specific early dialogs are
terminated, since the forking proxy will not forward the final answer.

That means that the UAC may have to keep resources etc that it reserved when the early dialog was created (by a 18x provisional response)
available until the proxy forwards a final response.

Would it somehow be possible to define a "dialog has been terminated" response code that the proxy could
send to the UAC directly each time it receives an error response from any of the forked dialogs. A provisional response code would probably be good, because it would be backward compatible (nodes that don't support it simply keeps the early dialog alive until the final response comes from the forking proxy, as today).

There may have been some work on this issue, or similar, before, but I can't remember the details.

Regards,

Christer Holmberg
Ericsson Finland

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Subject: Re: [Sip] Provisional response to terminate early dialog
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Christer Holmberg (JO/LMF) wrote:

> Hi,
> 
> Today, when a request gets forked, the forking proxy will "collect" final responses received from the different dialogs.
> 
> The calling UAC, however, does now know when specific early dialogs are
> terminated, since the forking proxy will not forward the final answer.
> 
> That means that the UAC may have to keep resources etc that it reserved when the early dialog was created (by a 18x provisional response)
> available until the proxy forwards a final response.

This is a relatively small amount of time - 32 seconds.

> 
> Would it somehow be possible to define a "dialog has been terminated" response code that the proxy could
> send to the UAC directly each time it receives an error response from any of the forked dialogs. A provisional response code would probably be good, because it would be backward compatible (nodes that don't support it simply keeps the early dialog alive until the final response comes from the forking proxy, as today).

I don't really see what the problem is. Is this 32 seconds or so that 
critical? Besides shaving a few seconds off the amount of time that a 
UAC holds transaction state, is there any other benefit?

-Jonathan R.

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

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From sip-bounces@ietf.org  Mon Jan 31 02:24:36 2005
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>
Subject: RE: [Sip] Provisional response to terminate early dialog
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Hi Jonathan,

It's not so much about the time, and in paralell forking scenarios I think that time could be even longer, couldn't it?

I'll try to explain the problem by giving a use case:

A UAC sends an INVITE. A proxy forwards the INVITE to UAS_1, which establish an early dialog and starts sending some kind of early media to the UAC (there may also be additional offer/answer transactions before the media starts). After that UAS_2 sends an error response to the proxy, which then forwards the INVITE to UAS_2.

Now, UAC has established, and may have reserved media/security/etc resources, for UAS_1, so it may choose not to accept the early dialog (or, at least possible early media) from UAS_2. Eg in 3GPP it is said that an UE may choose to reject additional early dialogs if they - for whatever reason - see a need to do so. 

However, if the UAC would know that the early dialog with UAS_1 has been terminated it could free all resources and possibly reserve newones for UAS_2. Also, it wouldn't have to think about things like media mixing etc - issues which have been very much discussed on this list.

Regards,

Christer Holmberg
Ericsson Finland




> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: 30. tammikuuta 2005 16:39
> To: Christer Holmberg (JO/LMF)
> Cc: 'sip@ietf.org'
> Subject: Re: [Sip] Provisional response to terminate early dialog
> 
> 
> 
> 
> Christer Holmberg (JO/LMF) wrote:
> 
> > Hi,
> > 
> > Today, when a request gets forked, the forking proxy will 
> "collect" final responses received from the different dialogs.
> > 
> > The calling UAC, however, does now know when specific early 
> dialogs are
> > terminated, since the forking proxy will not forward the 
> final answer.
> > 
> > That means that the UAC may have to keep resources etc that 
> it reserved when the early dialog was created (by a 18x 
> provisional response)
> > available until the proxy forwards a final response.
> 
> This is a relatively small amount of time - 32 seconds.
> 
> > 
> > Would it somehow be possible to define a "dialog has been 
> terminated" response code that the proxy could
> > send to the UAC directly each time it receives an error 
> response from any of the forked dialogs. A provisional 
> response code would probably be good, because it would be 
> backward compatible (nodes that don't support it simply keeps 
> the early dialog alive until the final response comes from 
> the forking proxy, as today).
> 
> I don't really see what the problem is. Is this 32 seconds or so that 
> critical? Besides shaving a few seconds off the amount of time that a 
> UAC holds transaction state, is there any other benefit?
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org  Mon Jan 31 12:49:35 2005
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Provisional response to terminate early dialog
Date: Mon, 31 Jan 2005 11:30:55 -0600
To: Jonathan Rosenberg <jdrosen@cisco.com>
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How is this bound to 64T1?

It's an INVITE transaction - it can pend potentially indefinitely 
(though timer C will likely kick in to muck up the works).

I can see the kind of cases Christer is considering:

   1) You parallel fork to several places. All initiate early media. All 
but one
        return an error code that doesn't stimulate immediate forwarding 
in the
        proxy response context (say, 401just to see the situation, as 
odd as that
        would be to mix w/ early media). One leg just stays pending. The 
initiating
        UA will hold onto the resources for early media for _all_ of 
those legs until
        the pending leg completes.

2) If you have a serial forking proxy, and each leg initiates early 
media, at the
     end of the fork, the initiating UA is holding early media resources 
for each
     previous leg - it has no way to let go before it gets a final 
response.

Personally, I see this as just one of the consequences of the
terrible thing we did when we allowed early media in the first place.
I am reticent to further complicate the protocol by introducing 
provisionals
along the lines Christer is proposing - that step takes us most, if not 
all, of the way to having
transactions within transactions. Taken to an extreme, this could solve 
HERFP
by tunneling all the final responses on the other side of a proxy back 
to the UA
so it might be able to do something about it (say they were 421 
Extension
Required responses: A UA could deal with those immediately by 
initiating a
new branch, parallel forking at the UA (yuck)). This could even lead to 
pressure
to change the merging rules so that a UA can deal with 401s as part of 
early dialogs
by reissuing the request on a new branch (double-yuck).

In shorter words, I think this is a problem we should admit we have, 
but not
try to do anything about. If we _do_ choose to do something about it, 
then we
better carefully bound the scope.

RjS

On Jan 30, 2005, at 8:39 AM, Jonathan Rosenberg wrote:

>
>
> Christer Holmberg (JO/LMF) wrote:
>
>> Hi,
>> Today, when a request gets forked, the forking proxy will "collect" 
>> final responses received from the different dialogs.
>> The calling UAC, however, does now know when specific early dialogs 
>> are
>> terminated, since the forking proxy will not forward the final answer.
>> That means that the UAC may have to keep resources etc that it 
>> reserved when the early dialog was created (by a 18x provisional 
>> response)
>> available until the proxy forwards a final response.
>
> This is a relatively small amount of time - 32 seconds.
>
>> Would it somehow be possible to define a "dialog has been terminated" 
>> response code that the proxy could
>> send to the UAC directly each time it receives an error response from 
>> any of the forked dialogs. A provisional response code would probably 
>> be good, because it would be backward compatible (nodes that don't 
>> support it simply keeps the early dialog alive until the final 
>> response comes from the forking proxy, as today).
>
> I don't really see what the problem is. Is this 32 seconds or so that 
> critical? Besides shaving a few seconds off the amount of time that a 
> UAC holds transaction state, is there any other benefit?
>
> -Jonathan R.
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Abbreviated WGLC for
	draft-ietf-sip-refer-with-norefersub-00.txt
Date: Mon, 31 Jan 2005 11:34:28 -0600
To: Robert Sparks <rjsparks@nostrum.com>
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Cc: SIP WG <sip@ietf.org>, Rohan Mahy <rohan@ekabal.com>,
        Orit Levin <oritl@microsoft.com>,
        Dean Willis <dean.willis@softarmor.com>
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The LC the chairs initiated ends today.

Please read this draft and comment. It's very short - please do this  
now.

We've talked about this idea a lot - let make sure the talk is captured  
in the draft.

RjS

On Jan 26, 2005, at 4:22 PM, Robert Sparks wrote:

> Initial LC Comments:
>
> First paragraph of Section 3 calls out 481ing an initial NOTIFY. The  
> -dialog-usage-
> draft shows that is a bad idea, especially for in-dialog REFERs.  
> Suggest trimming
> the sentence to end after (Expires 0).
>
> Section 4 fourth paragraph: It would help implementers to call out  
> explicitly that the
> 200 response to the REFER will not contain a Supported or Require  
> header containing
> the norefersub token.
>
> Section 4 fifth paragraph: This needs more definition. How is it  
> rejected? 603, 420, or
> something else? Implementors will need some guidance here on when to  
> use which
> error response. I think (but am not dead certain at the moment) that  
> 420 is almost always
> what you want to use.
>
> Section 4 sixth paragraph: Is Supported: in the response really the  
> right thing? Eventlist
> uses Require: in the response to a SUBSCRIBE. The intent of the  
> normative behavior in
> this paragraph is to let a Refer-issuer who only included Supported:  
> know whether or
> not a subscription was actually created. Require: (I'm using this  
> extension) in the response
> fits that better than Supported: (I could use this extension if you  
> want to).
>
> Section 4 sixth paragraph: Suggest replacing "In this case, no dialog  
> is created" with
> "In this case, no implicit subscription is created. As a consequence,  
> no new dialog
> is created if this REFER was issued outside any existing dialog."
>
> Section 5: Probably worth also calling out using Max-Forwards: 0 to  
> prevent forking
> for those (rare now, but hopefully not rare forever) cases where  
> signaling can get to
> the refer target without going through a proxy.
>
> Section 6: Must replace "tradewind" with "example" throughout.
>
> Section 8: "threads" should be "threats", It would be good to pull any  
> discussion
> that led to this conclusion together for when the IESG asks if we  
> _really_ thought
> about this. Have we talked about what would happen if a SIP  
> intermediary (or an
> unauthorized MitM) just injected a Require: norefersub into a REFER  
> request?
>
> RjS
>
>
> On Jan 24, 2005, at 5:07 PM, Rohan Mahy wrote:
>
>> Hi Folks,
>>
>> I'd like to begin an abbreviated WGLC for:
>>
>> 	http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-with- 
>> norefersub-00.txt
>>
>> Since this document is extremely short and has been discussed  
>> extensively, WGLC will end on February 1st.
>>
>> thanks,
>> -rohan
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
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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		: Management Information Base for Session Initiation Protocol (SIP)
	Author(s)	: K. Lingle, et al.
	Filename	: draft-ietf-sip-mib-09.txt
	Pages		: 93
	Date		: 2005-1-31
	
This memo defines a portion of the Management Information Base (MIB)
   for use with network management protocols in the Internet community.
   In particular, it describes a set of managed objects that are used to
   manage Session Initiation Protocol (SIP) entities, which include User
   Agents, Proxy, Redirect and Registrar servers.

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