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Subject: [Sip] Some question about draft:A Framework for Session Initiation
	Protocol (SIP) Session Policies 
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    I have some questions about the frame work proposed in the draft which defines a framework for SIP policy. See bellow:          1. What the advantage is in the proposed framework in which UA require its policy (which means it contacts policy server after contacts proxy) by itself? An alternative is, the proxy contacts the proxy server on behalf of the UA, and return or not return determined policies, which means UA need not involved in the procedure of contacting policy server. So why the one proposed in the draft is better?
   
      2. If it retrieves policies according to call flow this draft, how a proxy knows that the policies in the resending request for the UA is genuine, especially for the specific ongoing request?(which means there should be security mechanism to avoid replay attacks).Can anyone give me some ideas or solutions?
   
  Thanks.
  _______
  mjhorse



       
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<FORM name=frmAddAddrs action=http://cn.address.mail.yahoo.com/yab/cn?v=YM&amp;.rand=23523&amp;A=m&amp;simp=1 method=post>&nbsp;&nbsp;&nbsp; I have some questions about the frame work proposed in the draft which defines a framework for SIP policy. See bellow:</FORM>  <DIV class="msgbody clearfix" id=message71853755122710842052494275951080 style="VISIBILITY: visible">  <DIV id=yiv336752346>  <DIV>&nbsp;&nbsp;&nbsp; 1.&nbsp;What the advantage is in the proposed framework in which UA require its policy (which means it contacts policy server after contacts proxy) by itself? An alternative is, the proxy contacts the proxy server on behalf of the UA, and return or not return determined policies, which means UA need not involved in the procedure of contacting policy server. So why the one proposed in the draft is better?</DIV>  <DIV>&nbsp;</DIV>  <DIV>&nbsp;&nbsp;&nbsp; 2. If it retrieves policies according to&nbsp;call flow&nbsp;this draft, how a proxy knows that the policies in
 the resending request for the UA is genuine, especially for the specific ongoing request?(which means there should be security mechanism to avoid replay attacks).Can anyone give me some ideas or solutions?</DIV>  <DIV>&nbsp;</DIV>  <DIV>Thanks.</DIV>  <DIV>_______</DIV>  <DIV>mjhorse</DIV></DIV></DIV><p>&#32;


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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From: "Dave Burke" <daveburke@google.com>
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	draft-ietf-mediactrl-vxml-00
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Hi Paul,

Thanks a lot for the review. Mark and I have discussed each of these and
I've responded to the comments below.

Cheers,

Dave

[snip]

Section 2.1:
>
> The ABNF of the URI seems excessively rigid in the ordering of
> parameters - first the init-parameters, then the vxml-parameters, then
> then any other uri-parameters. It also doesn't note that
> 'uri-parameters' is defined in 3261.
>


DB> Good observation - ordering of the parameters was never intended. Your
proposal looks good. However, while looking over this again, we found
another issue which crept in late in the process related to the addition of
JSON support. Specifically, the problem is that it is impossible to
differentiate where vxml-params end and uri-parameters begin and hence which
parameter values to interpret as JSON values and which as plain strings. We
propose a change to limit JSON values to just the named VoiceXML session
variables 'ccxml' and 'aai' (these were the two variables for which JSON
support was originally sought).

  dialog-parameters = *(";" dialog-parameter)

  dialog-parameter  = init-param / url-parameter ; defined in [RFC 3261]

  init-param        = ";" (dialog-param /
                           maxage-param /
                           maxstale-param /
                           method-param /
                           postbody-param /
                           ccxml-param /
                           aai-param)

  dialog-param      = "voicexml=" vxml-url ; vxml-url follows the URI
                                           ; syntax defined in [RFC3986]
  maxage-param      = "maxage=" 1*DIGIT

  maxstale-param    = "maxstale=" 1*DIGIT

  method-param      = "method=" ("get" / "post")

  postbody-param    = "postbody=" token

  ccxml-param       = "ccxml=" json-value

  aai-param         = "aai=" json-value

  json-value        =  false /
                       null /
                       true /
                       object /
                       array /
                       number /
                       string ; see RFC4627

>
>
> I'm confused about postbody:
>
> [snip]

>
> The interaction between these two escaping rules seems potentially
> confusing. I *think* when this is all put together it means that the
> body must first be encoded according to the [HTML4] section above. At
> that point it will already be almost conformant to the 3261 syntax of a
> token, except for the use of '&'. Then it needs to be encoded again,
> which will take care of the ampersands, but which will re-encode the '%'
> characters of the first encoding.


[snip]

DB> Your interpretation of the intent is correct. The postbody value is an
application/x-www-form-urlencoded string which we subsequenty re-encode (the
latter step affecting each & and % character). We will clarify the text that
this is the case. Moreover, we'll be more specific and call out which
parameter values are subject to this encoding step , namely vxml-url, the
postbody token, and the json-value.

>
> Section 2.2:
>
>    The Application Server SHOULD insert its own URI in the Record-Route
>    header so that it remains in the signaling path for subsequent
>    signaling related to the session.  This is of particular importance
>    for call transfers so that upstream Application Servers or proxy
>    servers see signaling originating from the Application Server and not
>    the VoiceXML Media Server itself.
>
> I don't understand the purpose of the above. The SHOULD strength of this
> requirement suggests to me an assumption of a particular operating
> environment. In the general case, why should this be more than MAY
> strength?


DB> Actually, this paragraph is quite problematic. We shoudn't be mandating
how the AS behaves (and indeed a B2BUA doesn't Record-Route anyway). This
paragraph is also a hang over from before we recommended against using
VoiceXML transfer features with application servers. We propose to delete
it.

>
> Section 2.3:
>
> IMO the use of a media-less session is an entirely valid sip usage. The
> only concern I might potentially have is if it were to catch some UA
> unaware, because some UAs just aren't prepared to handle this case. But
> the recommended usage here always puts the choice of doing this in the
> hands of *other* UA, not the media server. So I see no problem.
>
> I do find it disconcerting that the initial invite and subsequent
> reinvites are handled in different ways. For the initial one you stall
> the VXML awaiting a media stream, but on subsequent ones you assume the
> absence of a stream is equivalent to having a stream that doesn't send
> anything. Why isn't the behavior consistent in these cases? If you need
> to support both behaviors, then it might be better to explicit and
> unique signaling for each. For instance, you might define that a media
> stream with a=inactive or a=sendonly (from the client's perspective -
> client putting media server "on hold") could be treated as the absence
> of input but the absence of the stream means you should wait for a
> stream to be negotiated.
>

DB> We're conscious of the inconsistency but we took a judgement call to
live with it. Preparing a session is hugely valuable for latency hiding.
However, to support this kind of pause/resume feature once the dialog is
executing is not practical since VoiceXML has no concept of such semantics
and it is difficult to implement especially when external speech recognition
and synthesis systems are used and operate on their own timers without
pausing capabilities.

[snip]

>       In addition, the array's toString() function returns the full SIP
>       Request-URI.  For example, assuming a Request-URI of sip:dialog@
>       example.com;voicexml=http://example.com;obj={<http://example.com;obj=%7B>"x":1,"y":true}
> then
>
> I don't believe the above URI is valid. The ',' '{' and '}' aren't
> syntactically correct in a URI pvalue. You would need to escape them.


DB> Good catch, will fix

>
> Section 2.6.2:
>
> IIUC, message (1) contains no offer, (5) contains an offer with media,
> and (6) accepts the call but rejects the media. If so, then (9) will
> most likely be invalid. To make it valid, the o-line from (8) needs to
> be replaced with one consistent to that used in (6), with the version
> number incremented. And if (5) has more m-lines than (8), then (9) needs
> to be padded with extra (rejected) m-lines.


DB> Good catch. The intent here is to that the similarly SDP messages share
the same m- and a- lines only. We propose to update the diagram and perhaps
using a prime to denote a derivative SDP message, e.g. [offer2'], and
clarifying in the text.

>
>
> Section 5.2:
>
>    On receipt of the REFER request, the VoiceXML Media Server MUST issue
>    a provisional response, 100 Trying.  The 202 Accepted response
>    indicates that the VoiceXML document has been fetched and parsed
>    correctly.  The VoiceXML Media Server proceeds to place the outbound
>    INVITE and will execute the application after the ACK is sent.
>
> The rules of RFC 4320 need to be followed here. REFER is a non-invite
> transaction and so the timing of the 100 must be as specified in 4320.
>

DB> Will fix

>
> In the call flow, the sending of the initial NOTIFY before the 202 for
> the REFER, and especially before determining if REFER is going to
> succeed or fail, is at best unusual and almost certainly incorrect.
> Sending the NOTIFY and then sending a failure response would certainly
> be incorrect.
>
> I think you have two choices:
>
> - wait until the get is complete before sending the NOTIFY, and probably
> send it after the 202.
>
> - send a 202 for the REFER before doing the GET. Inform of a GET failure
> via a NOTIFY.
>

DB> There is currently an ongoing thread on the list about whether we remove
this feature (and hence section) altogether. For the sake of argument, I'll
assume it stays, in which case we think option 2 is the most conventional
(especially since most systems responds to the REFER instantaneously). A GET
failure would be conveyed by a NOTIFY with a 500 Server Internal Failure
sipfrag message (same status used for the INVITE case).


>
> Section 6.3:
>
> In the call flow I think you probably need another NOTIFY between
> messages (6) and (7). Its potentially too long until (13).


DB> This is correct but we note in the text "(provisional responses and
NOTIFY
   messages corresponding to provisional responses have been omitted for
   clarity)"

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Hi Paul,<br><br>Thanks a lot for the review. Mark and I have discussed each of these and I&#39;ve responded to the comments below.<br><br>Cheers,<br><br>Dave<br><br>[snip]<br><br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

Section 
2.1:<br><br>The ABNF of the URI seems excessively rigid in the ordering of<br>parameters - first the init-parameters, then the vxml-parameters, then<br>then any other uri-parameters. It also doesn&#39;t note that<br>&#39;uri-parameters&#39; is defined in 3261.
<br></blockquote><div><br><br>DB&gt; Good observation - ordering of the parameters was never intended. Your proposal looks good. However, while looking over this again, we found another issue which crept in late in the process related to the addition of JSON support. Specifically, the problem is that it is impossible to differentiate where vxml-params end and uri-parameters begin and hence which parameter values to interpret as JSON values and which as plain strings. We propose a change to limit JSON values to just the named VoiceXML session variables &#39;ccxml&#39; and &#39;aai&#39; (these were the two variables for which JSON support was originally sought).
<br><br>&nbsp; dialog-parameters = *(&quot;;&quot; dialog-parameter)<br><br>&nbsp; dialog-parameter&nbsp; = init-param / url-parameter ; defined in [RFC 3261]<br><br>&nbsp; init-param&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = &quot;;&quot; (dialog-param /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; maxage-param /
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; maxstale-param /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; method-param /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; postbody-param /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ccxml-param /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; aai-param)
<br><br>&nbsp; dialog-param&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = &quot;voicexml=&quot; vxml-url ; vxml-url follows the URI<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ; syntax defined in [RFC3986]<br>&nbsp; maxage-param&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = &quot;maxage=&quot; 1*DIGIT<br>

<br>
&nbsp; maxstale-param&nbsp;&nbsp;&nbsp; = &quot;maxstale=&quot; 1*DIGIT<br><br>&nbsp; method-param&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = &quot;method=&quot; (&quot;get&quot; / &quot;post&quot;)<br><br>&nbsp; postbody-param&nbsp;&nbsp;&nbsp; = &quot;postbody=&quot; token<br><br>&nbsp; ccxml-param&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = &quot;ccxml=&quot; json-value
<br><br>&nbsp; aai-param&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; = &quot;aai=&quot; json-value<br><br>&nbsp; json-value&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =&nbsp; false /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; null /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; true /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; object /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; array /
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; number /<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; string ; see RFC4627<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">


<br><br>I&#39;m confused about postbody:<br><br></blockquote><div>[snip] <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>The interaction between these two escaping rules seems potentially
<br>confusing. I *think* when this is all put together it means that the<br>body must first be encoded according to the [HTML4] section above. At<br>

that point it will already be almost conformant to the 3261 syntax of a<br>token, except for the use of &#39;&amp;&#39;. Then it needs to be encoded again,<br>which will take care of the ampersands, but which will re-encode the &#39;%&#39;
<br>characters of the first encoding.</blockquote><div><br>[snip]<br><br>DB&gt; Your interpretation of the intent is correct. The postbody value is an application/x-www-form-urlencoded string which we subsequenty re-encode (the latter step affecting each &amp; and % character). We will clarify the text that this is the case. Moreover, we&#39;ll be more specific and call out which parameter values are subject to this encoding step , namely vxml-url, the postbody token, and the json-value.
<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Section 2.2:<br><br> &nbsp; &nbsp;The Application Server SHOULD insert its own URI in the Record-Route
<br> &nbsp; &nbsp;header so that it remains in the signaling path for subsequent<br> &nbsp; &nbsp;signaling related to the session. &nbsp;This is of particular importance
<br> &nbsp; &nbsp;for call transfers so that upstream Application Servers or proxy<br> &nbsp; &nbsp;servers see signaling originating from the Application Server and not<br> &nbsp; &nbsp;the VoiceXML Media Server itself.<br><br>I don&#39;t understand the purpose of the above. The SHOULD strength of this
<br>requirement suggests to me an assumption of a particular operating<br>environment. In the general case, why should this be more than MAY strength?</blockquote><div><br>DB&gt; Actually, this paragraph is quite problematic. We shoudn&#39;t be mandating how the AS behaves (and indeed a B2BUA doesn&#39;t Record-Route anyway). This paragraph is also a hang over from before we recommended against using VoiceXML transfer features with application servers. We propose to delete it.
<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Section 2.3:<br><br>IMO the use of a media-less session is an entirely valid sip usage. The
<br>only concern I might potentially have is if it were to catch some UA<br>unaware, because some UAs just aren&#39;t prepared to handle this case. But<br>the recommended usage here always puts the choice of doing this in the
<br>hands of *other* UA, not the media server. So I see no problem.<br><br>I do find it disconcerting that the initial invite and subsequent<br>reinvites are handled in different ways. For the initial one you stall<br>the VXML awaiting a media stream, but on subsequent ones you assume the
<br>absence of a stream is equivalent to having a stream that doesn&#39;t send<br>anything. Why isn&#39;t the behavior consistent in these cases? If you need<br>to support both behaviors, then it might be better to explicit and
<br>unique signaling for each. For instance, you might define that a media<br>stream with a=inactive or a=sendonly (from the client&#39;s perspective -<br>client putting media server &quot;on hold&quot;) could be treated as the absence
<br>of input but the absence of the stream means you should wait for a<br>stream to be negotiated.<br></blockquote><div><br>DB&gt; We&#39;re conscious of the inconsistency but we took a judgement call to live with it. Preparing a session is hugely valuable for latency hiding. However, to support this kind of pause/resume feature once the dialog is executing is not practical since VoiceXML has no concept of such semantics and it is difficult to implement especially when external speech recognition and synthesis systems are used and operate on their own timers without pausing capabilities.&nbsp;
<br></div><div><br>[snip] <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> &nbsp; &nbsp; &nbsp; In addition, the array&#39;s toString() function returns the full SIP
<br> &nbsp; &nbsp; &nbsp; Request-URI. &nbsp;For example, assuming a Request-URI of sip:dialog@<br> &nbsp; &nbsp; &nbsp; <a href="http://example.com" target="_blank">


example.com</a>;voicexml=<a href="http://example.com;obj=%7B" target="_blank">http://example.com;obj={</a>&quot;x&quot;:1,&quot;y&quot;:true} then<br><br>I don&#39;t believe the above URI is valid. The &#39;,&#39; &#39;{&#39; and &#39;}&#39; aren&#39;t
<br>syntactically correct in a URI pvalue. You would need to escape them.</blockquote><div><br>DB&gt; Good catch, will fix<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>Section 2.6.2:<br><br>IIUC, message (1) contains no offer, (5) contains an offer with media,<br>and (6) accepts the call but rejects the media. If so, then (9) will
<br>most likely be invalid. To make it valid, the o-line from (8) needs to<br>be replaced with one consistent to that used in (6), with the version<br>number incremented. And if (5) has more m-lines than (8), then (9) needs
<br>to be padded with extra (rejected) m-lines.</blockquote><div><br>DB&gt; Good catch. The intent here is to that the similarly SDP messages share the same m- and a- lines only. We propose to update the diagram and perhaps using a prime to denote a derivative SDP message, 
e.g. [offer2&#39;], and clarifying in the text.<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br><br>Section 5.2:<br><br> &nbsp; &nbsp;On receipt of the REFER request, the VoiceXML Media Server MUST issue
<br> &nbsp; &nbsp;a provisional response, 100 Trying. &nbsp;The 202 Accepted response<br>


 &nbsp; &nbsp;indicates that the VoiceXML document has been fetched and parsed<br> &nbsp; &nbsp;correctly. &nbsp;The VoiceXML Media Server proceeds to place the outbound<br> &nbsp; &nbsp;INVITE and will execute the application after the ACK is sent.<br><br>



The rules of RFC 4320 need to be followed here. REFER is a non-invite<br>transaction and so the timing of the 100 must be as specified in 4320.<br></blockquote><div><br>DB&gt; Will fix <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>In the call flow, the sending of the initial NOTIFY before the 202 for
<br>the REFER, and especially before determining if REFER is going to<br>succeed or fail, is at best unusual and almost certainly incorrect.<br>Sending the NOTIFY and then sending a failure response would certainly<br>be incorrect.
<br><br>I think you have two choices:<br><br>- wait until the get is complete before sending the NOTIFY, and probably<br>send it after the 202.<br><br>- send a 202 for the REFER before doing the GET. Inform of a GET failure
<br>via a NOTIFY.<br></blockquote><div><br>DB&gt; There is currently an ongoing thread on the list about whether we remove this feature (and hence section) altogether. For the sake of argument, I&#39;ll assume it stays, in which case we think option 2 is the most conventional (especially since most systems responds to the REFER instantaneously). A GET failure would be conveyed by a NOTIFY with a 500 Server Internal Failure sipfrag message (same status used for the INVITE case).
<br>&nbsp;</div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Section 6.3:<br><br>In the call flow I think you probably need another NOTIFY between
<br>messages (6) and (7). Its potentially too long until (13).</blockquote><div><br>DB&gt; This is correct but we note in the text &quot;(provisional responses and NOTIFY<br>&nbsp;&nbsp; messages corresponding to provisional responses have been omitted for
<br>&nbsp;&nbsp; clarity)&quot;<br></div></div><br>

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Thread-Topic: about offer/answer model
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From: "Zhang, Rocky Fan Wen \(Rocky\)" <fanwenz@alcatel-lucent.com>
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Hi, all,
=20
   In the scenario of session timer, AS acts as the refresher and send a
UPDATE to the endpoint without a SDP body,
   but the endpoint returns a 200 OK with a SDP. Does this action
allowed?
  =20
   Thanks in advance.
=20
=20
Best Regards,
=20
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Fanwen zhang
FS5000 CPE IOT, Alcatel-Lucent Qingdao, R&D
Tel: +86-532-88615493
Email:fanwenz@alcatel-lucent.com
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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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<DIV><SPAN class=3D773435101-29122007><FONT size=3D2>Hi, =
all,</FONT></SPAN></DIV>
<DIV><SPAN class=3D773435101-29122007><FONT =
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV>
<DIV><SPAN class=3D476064701-29122007><FONT size=3D2>&nbsp;&nbsp; In the =
scenario of=20
session timer, AS acts as the refresher and send a UPDATE to the =
endpoint=20
without a SDP body,</FONT></SPAN></DIV>
<DIV><SPAN class=3D476064701-29122007><FONT size=3D2>&nbsp;&nbsp; but =
the endpoint=20
returns a 200 OK&nbsp;with a SDP.&nbsp;Does this action=20
allowed?</FONT></SPAN></DIV>
<DIV><SPAN class=3D476064701-29122007><FONT size=3D2>&nbsp;&nbsp;=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D476064701-29122007><FONT size=3D2>&nbsp;&nbsp; Thanks =
in=20
advance.</FONT></SPAN></DIV></DIV>
<DIV align=3Dleft><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT size=3D2>Best Regards,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT =
size=3D2>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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></D=
IV>
<DIV><FONT size=3D2>Fanwen zhang<BR>FS5000 CPE IOT, Alcatel-Lucent =
Qingdao,=20
R&amp;D<BR>Tel:=20
+86-532-88615493<BR>Email:fanwenz@alcatel-lucent.com</FONT></DIV>
<DIV><FONT =
size=3D2>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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></D=
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Thread-Topic: INVITE after terminated dialog?
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Hi All,
=20
There is a weird situation with us:
(1) A Linksys phone sends an INVITE out with incorrect destination TN.
As a result, our application server will set up a call between the
Linksys phone and an IVR
(2) A user incorrectly hits the on-hold key, which is basically sending
a re-INVITE to the IVR
(3) Our server seems to have answered the re-INVITE with a BYE
(4) The Linksys phone responds to the BYE with a 200
(5) Afterwards, the Linksys phone still thinks that its original
re-INVITE has not been responded.  So, it will keep resend re-INVITE
=20
The consequence of these re-INVITEs is a registration migration, which
we don't like.  We have been trying to figure out which side is not
following RFC 3261, the phone, the SBCs are wrong, or the application
servers.
=20
I am wondering:
(A) Is it legal for a server to ignore the re-INVITE and respond with a
BYE and has nothing in between?
=20
(B) If a UA has responded to a BYE with 200, is it supposed to forget
about the re-INVITE request that it has sent out and not try to resend
the re-INVITE, assuming a dialog has already been terminated with the
BYE/200 transaction?  Maybe in another sentence, is it legal to send our
re-INVITEs outside an dialog, even if these re-INVITEs are the repeats
of an original in-dialog re-INVITE?
=20
Thanks!
=20
Jane
=20

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<DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>Hi=20
All,</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>There =
is a weird=20
situation with us:</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>(1) A =
Linksys phone=20
sends an INVITE out with incorrect destination&nbsp;TN.&nbsp; As a =
result,=20
our&nbsp;<SPAN class=3D184531405-18122007>application </SPAN>server will =
set up a=20
call between the Linksys phone an<SPAN =
class=3D184531405-18122007>d</SPAN> an=20
IVR</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial =
size=3D2>(2)&nbsp;<SPAN=20
class=3D184531405-18122007>A&nbsp;user</SPAN> incorrectly hit<SPAN=20
class=3D184531405-18122007>s</SPAN> the on-hold key, which is =
basically&nbsp;<SPAN=20
class=3D184531405-18122007>sending </SPAN>a re-INVITE to the=20
IVR</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>(3) =
Our=20
server&nbsp;<SPAN class=3D184531405-18122007>seems to have =
</SPAN>answer<SPAN=20
class=3D184531405-18122007>ed</SPAN> the re-INVITE with a =
BYE</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>(4) =
The Linksys=20
phone responds&nbsp;<SPAN class=3D184531405-18122007>to </SPAN>the BYE =
with a=20
200</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>(5) =
Afterwards, the=20
Linksys phone still thinks that its&nbsp;<SPAN =
class=3D184531405-18122007>original=20
</SPAN>re-INVITE has not been responded.&nbsp; So, it will keep=20
resend&nbsp;re-INVITE</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>The =
consequence of=20
these re-INVITEs is&nbsp;<SPAN class=3D184531405-18122007>a registration =

migration, which we don't like</SPAN>.&nbsp; We have been trying to=20
figure&nbsp;<SPAN class=3D184531405-18122007>out which side is not =
following RFC=20
3261, the phone,&nbsp;</SPAN>the SBCs are wrong, or&nbsp;<SPAN=20
class=3D184531405-18122007>the </SPAN>application =
servers.</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>I am=20
wondering:</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>(A) Is =
it legal for=20
a server to&nbsp;<SPAN class=3D184531405-18122007>ignore the re-INVITE =
and=20
respond&nbsp;</SPAN>with a BYE and has nothing in =
between?</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial size=3D2>(B) If =
a UA has=20
responded to a BYE with 200, is it supposed to forget about the =
re-INVITE=20
request that it has sent out and not try to resend the re-INVITE, =
assuming a=20
dialog has already been terminated with the BYE/200 transaction?&nbsp; =
Maybe in=20
another sentence, is it legal to send our re-INVITEs outside an dialog, =
even if=20
these re-INVITEs are the repeats of an original in-dialog=20
re-INVITE?</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2>Thanks!</FONT></SPAN></DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953294019-17122007><FONT face=3DArial=20
size=3D2>Jane</FONT></SPAN></DIV></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

------_=_NextPart_001_01C84135.F44D76F8--




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Date: Tue, 01 Jan 2008 21:33:28 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: "Zhang, Rocky Fan Wen \(Rocky\)" <fanwenz@alcatel-lucent.com>
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No, its not allowed.

	Paul

Zhang, Rocky Fan Wen (Rocky) wrote:
> Hi, all,
>  
>    In the scenario of session timer, AS acts as the refresher and send a 
> UPDATE to the endpoint without a SDP body,
>    but the endpoint returns a 200 OK with a SDP. Does this action allowed?
>   
>    Thanks in advance.
>  
>  
> Best Regards,
>  
> ===========================================
> Fanwen zhang
> FS5000 CPE IOT, Alcatel-Lucent Qingdao, R&D
> Tel: +86-532-88615493
> Email:fanwenz@alcatel-lucent.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 Wed Jan 02 13:55:24 2008
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I'll get these on the various sites shortly.

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

Draft Minutes, SIP at IETF 70
Edited by Dean Willis based on notes by Ben Campbell, Miguel Angel  
Garcia-Martin and Brian Rosen

Session 1
========

Topic: Agenda and Status
led by Chairs
Slides presented and included in proceedings

Issue: How to deal with extensions in experimental RFCS like "e2m"

Proposed that we change the process of RFC 3427 to allow experimental  
track RFCs to register SIP extension headers. Discussion concluded  
that using an "X-" prefix in the registry would be self-defeating.  
There are also concerns about setting the bar too low -- we want any  
registered extension headers fields to be both reasonable and well- 
documented. The group reached a consensus on making a change to RFC  
3427 allowing any RFC produced by the SIP working group to register  
extension headers.

Issue: Robert's fork-loop-fix draft

Not enough people appear to have read the draft yet for us to have a  
strong sense of whether the changes satisfy issues raised in WGLC.  
The consensus is that the chairs shall conduct another short WGLC.


Topic: Location Conveyance
led by James Polk
Slides presented and included in proceedings

Issue: Should we have an error message to describe what's bad about  
the civic address data contained in a prior message?

Several points were raised in discussion,  including whether we have  
enough of an understanding of location validation to be able to do  
this sort of detailed response coding at this time. The current draft  
allows for a human-readable string, and this may suffice for current  
needs. Whether this is sufficient depends in large part on whether we  
expect automata to be able to respond to a a validation error,  
correct the location data, and resubmit it. There may also be the  
possible that the automata could parse the error response and ask a  
human a specific question, such as "What floor are you on?" if needed.

No conclusion was reached in the meeting, and the author was asked to  
lead a discussion of the issue on the mailing list.

Issue: What are the semantics of the "recipient" parameter in the  
Geolocation header?

Debate centered on whether this parameter is a hint, a provider of  
context, or a binding constraint on authorization to access the  
location information. The discussion became somewhat heated.

The working group did indicate a strong consensus to develop a  
protocol mechanism for location-based routing.

Debate during the meeting session was inconclusive, and it was  
proposed that we have an ad-hoc breakout session on this topic.

The ad-hoc session resolved to move this semantic into the geopriv  
object itself, and to send the work item for this back to GEOPRIV to  
complete.



Topic: Info Harmful
led by led by Eric Burger
Slides presented and included in proceedings

Discussion was deferred until after the presentation of the INFO  
Events draft.


Topic: Info Events
led by Christer Holmberg
Slides presented and included in proceedings

Issue: If an endpoint can receive an info package (i.e., indicates it  
can receive such events), does that mean it will do something useful  
with an event if it receives it?

Discussion seems to indicate that each INFO event package will  
determine the requirements for that package, with the general trend  
being "only say you know it if you want it". So, event packages,  
where available, would be higher-priority than older techniques with  
less negotiation flexibility.

Issue: If the other end doesn't understand an INFO event you want to  
send, what do you do?

Discussion concluded that a node can only send events that the other  
side has indicated a willingness to receive, so if the other end  
doesn't understand something, don't send it. This may mean that event- 
dependent applications won't work, or that one might have to do  
something tricky like falling back to actually talking to the other  
party instead of sending events.

Issue: Backward Compatibility

Noted that in the bad-old-days of INFO, all backward compatibility  
was handled manually by configuration. That technique can still be  
used. However, for the ISUP and QSIG uses, we can provide for  
backward compatibility using Accept header fields as these payloads  
have MIME types that are not used in other contexts. Also noted that  
we need to consider these sort of backward compatibility issues in  
each event package.

Issue: Do we want to enable transmission of user-to-user information  
in an INVITE initiated dialog? This was one of the arguments against  
dialogs of MESSAGE methods.

Discussion indicated that we're probably OK with this for the class  
of things we seem to think will end up in INFO events. It was noted  
by a chair that there is little sense in trying to "protocol police"  
this sort of thing -- the requirement is just t make sure that the  
protocol allows the applications involved to understand what they're  
doing and not step on each other's extensions.

Issue: Use cases?

Whether DTMF is a convincing use case was discussed without conclusion.

Noted that other SDOs appear to be developing things using INFO. This  
MAY be because INFO is the right thing, or it may be simply because  
they don't have to negotiate with IETF to use it, and that if we  
adjust INFO to encourage them to work with IETF, they might just find  
some other loophole like X-headers.

Proposed that there are non-DTMF needs for end to end traffic, but  
that these needs may be better met by end-to-end sessions like MSRP.   
Of course, this raises the issue of negotiating the semantics of  
these end-to-end sessions, like "Why are you sending me these JPEG  
images in an MSRP session? What AM I supposed to do with them?", so  
there would still be a need for some sort of package definition.  
However, DTMF may be unique in being both signaling and media, and we  
might solve much of the problem by just finding a fix at the SIP  
level for DTMF and ignoring the remainder of the problem.

Discussion on whether use cases could motivate the work. At one  
point, a poll by the chairs indicated that few of the people opposed  
to INFO events would be persuaded even if confronted with a large  
number of valid use cases.

Issue: Registration policy. If we were to define INFO events, what  
would the IANA policy be on registering an INFO event package?

Chair Dean Willis proposed that the policy be "specification  
required", given that  our goal is not to police the architectures of  
everybody who wants to use SIP, but that we do need to make sure that  
the feature negotiation mechanism work well enough that conflicts are  
avoided. We really should NOT have to use SIP working group time to  
discuss whether or not some other SDO's use of their SIP-absed  
signaling channel is valid.

Issue: Where to go from here?

Noted that there ARE other SDOs actively engaged, and that we are not  
servcing the community well by giving no guidance.

Eric Burger noted that the two drafts are not incompatible, and that  
the INFO events draft solves all of the "harmful" aspects of current  
INFO use noted in his draft.

The meeting concluded with an agreement to further consider use  
cases, but no commitment to further action. (Note: subsequent  
discussion occurred on-list, shortly after the meeting session).


Session 2
========

Topic: Agenda and Status
led by Chairs
Slides presented and included in proceedings

Discussion of Essential Corrections will be added to the agenda if  
time allows.

The XCAP Diff event draft will be revised and submitted as draft-ietf- 
sip-xcapevent-00

Connection reuse was rescoped to allow some requests in the reverse  
direction and discourage connection reuse for virtual servers. Some  
further clarifications are needed.



Topic: Outbound
led by Rohan Mahy
Slides presented and included in proceedings

Issue:  TCP Keepalive

Current text removed. Noted that even if you use TCP keepalive, you  
still need to do the keepalives as specified in this draft.

Issue:  interaction with GRUU.

Group seemed to indicate consensus on the proposal by Rohan for the  
2nd option listed in the slides -- GRUU should check that an incoming  
REGISTER matches any Call-ID already registered by this device. This  
would appear to require a change in the GRUU document.

Issue: What error does the registrar send if it gets a register with  
require: outbound but the edge proxy doesn't support outbound?

Christer Holmberg suggested that most cases would need a 500-class.

Jonathan Rosenberg noted that this depends on what might happen if  
the requested were to be retried. If it is always going to fail, the  
response would be in the 400 class. If success on retry is possible,  
a 500 might be more reasonable.

Cullen Jennings noted that It is also important to consider what  
clients will do with the response.

Adam Roach observed that a 420 response would make it hard to debug  
the problem from a client perspective, and that a new error code  
would be better.

Jonathan suggested using a proxy require that the proxy would  
translate to require, but Rohan reminded us of prior discussion where  
we had decided this would not work.

Christer concurred with Adam, saying that this problem is more  
general, essentially an "invalid path".

The resolution is to develop a new 400 class response code indicating  
the problem with this path. Adam Roach volunteered to send text.

Issue:  How do subsequent in-dialog requests follow across an  
Outbound edge link?

Proposed by Rohan that the Outbound proxy MUST record route if the  
top Route header has an :ob" parameter.

Extended discussion followed, noting that this makes us dependent on  
proxies for mid-dialog requests, which has impacts on reliability  
designs.

Francois Audet noted that the current text is hard to understand.

Chairs conclude that the room seems to mostly support the resolution  
proposed by Rohan, but that the final text needs to be reviewed and  
agreed on-list.

Issue: The "keep" parameter.

Ted Hardie  argued for  keeping the text as-is.

Aki Niemi argued in favor of deprecating the "keep" parameter, based  
on its lack of usefulness and potential to confuse people.

Christer Holmberg suggested deprecating the parameter when used with  
an outbound proxy, but allowing it for non-outbound cases.

Poll by chairs indicated strong consensus for removing the "keep"  
parameter.


Issue: Replace timed-keepalive parameter with a new header

Proposed by Aki that weuse a new header, such as "Flow-Timer" to be  
returned in register response.

Noted by Rohan that this might fix the non-outbound case for keepalive.

Christer questions this, as it is still necessary to know if  
keepalive is supported.

Poll by chairs indicated strong consensus to define a new header for  
this purpose.


Issue: Detecting client support for keepalives without "outbound"

Proposed new option tag for client support, included by client in  
Supported.

Poll by chairs indicated preference to consider this approach outside  
of the Outbound draft. Christer Holmberg may pursue a draft on this  
topic.

Procedural point: Francois Audet has volunteered to act as the  
protocol shepherd for getting this draft through the process.

Derek reminded Rohan to add text relating to the route construction  
issue. Derek will try to send text on this to Rohan.


Topic: Resource Priority Header in Responses
led by Janet Gunn
Slides presented and included in proceedings

Discussion centered on the role of the RPH header in a response and  
the assumptions underlying a network in which it would be useful.   
The key issue is that since SIP cannot challenge responses that if  
the response is not signed (as in SIP Identity) then there is no way  
to tell upstream where the RPH header came from. There may be no  
coupling between the proxy layer and the access network for media  
priority enforcement.

Noted by EKR that given the security model, where the user is not  
trusted but all the proxy nodes are, that this requires integrity  
protection on the connections, with peer authentication. A proxy  
receiving a response with RPH has to know that it's getting this  
response from a trusted proxy and not somebody else.

Several parties suggested that this sort of mechanism is more  
appropriate for a P-header with an applicability statement that  
explains the transitive trust model.

Keith Drage, as chair, asked people to consider the stateless proxy  
use case from James' draft relative to the security assumptions, P- 
header concept, and authentication issues.


Topic: Media Identity
led by Dan Wing
Slides presented and included in proceedings

Issue: SBCs break 4474 because SDP signed by Identity is rewritten by  
SBC

Francois observed that this problem may only occur when it is an SBC  
outside the domain of the Identity server that is doing the rewriting.

Hannes suggested extending IDENTITY to be like DKIM, which can  
selectively sign pieces.

General discussion revealed that most of the material in the  
signaling is problematic from a signature perspective. IP Addresses  
are essentially meaningless, especially across domain boundaries.  
Phone numbers are little better, with end-nodes having no real way to  
test assertions about them (and perhaps a service provided signature  
is the best we can do here).  RFC 4474 already documents limitations  
with respect to telephone numbers.

There seems to be an emerging thread of consensus here that what we  
need is some way to bind media to signaling -- that is, to by  
inspection of the media and the signaling be able to say with some  
level of certainty that they are related -- and specifically, to say  
"this media is a result of that signaling".

Chairs noted that there is additional work required her to be able to  
frame this into the sort of requirement that could be added to a  
charter.


Topic: DTLS Framework
led by Jason Fischl
Slides presented and included in proceedings

Issue: 4474 for phone numbers

We may be lacking documentation on when RFC 4474 does and does not  
work, and what security we get when using it with phone numbers. Jon  
Peterson seems to think that 4474 already offers this guidance.   
Jonathan Rosenberg disagrees and believes the text in 4474 is  
inadequate and more is needed.

Consensus noted that even if we do need more here, it is outside of  
the scope of the DTLS framework. However, the framework should at  
least note the issue.

Issue; Anonymity

Agreed that text must be updated to say it does not break existing  
anonymity.

Issue: Middle boxes.

SBCs may block key exchange before 200 OK. This is not specific to  
DTLS-SRTP.

Proposed by Christer that we at least mention this issue in the draft.

RKR noted that much of this SBC difficulty should be documented in  
Brian's middlebox draft.

Issue: SRTP/TCP

Consensus: DTLS-SRTP over TCP rather than RTP over TLS.

Issue: Priority for advancing.

Chairs observed that an AVT document is dependent on this, making  
this a high priority to move along, However, the AVT chair reported  
that the AVT document is not as ready as we might have thought.


Topic: Essential Corrections
led by Keith Drage
Slides presented and included in proceedings

Issue: general intent of essential corrections

We seem to have some disagreement as to the intent and value of the  
essential corrections process. The chairs restated the goal as being  
to batch together the updates to RFC 3261 so that readers do not have  
to look in many places to find the current "corrected" specification.


Issue: record-route-fix

Is this an essential correction or an enhancement? It does identify  
specific failure cases that occur if RFC 3261 is used as-is. The  
chairs noted that drafts making normative changes to a standards  
track draft need to be standards track themselves, not BCPs. However,  
BCPs would be appropriate for documents that, for example, argue for  
using one option in an an existing standards-track document over an  
alternative in that document.

Robert argues that this fix is needed, although it would be  
acceptable to advance it individually rather than in the essential- 
corrections procss.

Conclusion: robert and Jonathan are to go discuss this and come back  
with a recommendation.

Issue: IPV6 ABNF error

Consensus that this is an essential correction we would consider  
anything that behaves as per 3261 to be erroneous.


Issue: 503 corrrection

JDR argues that it is known that 3261 doesn't perform well in  
overload conditions, and that we have ongoing work to improve this  
behavior.

Volker Hit concurs, noting that the overload team is still working to  
get better simulation results.

Conclusion: We'll continue to discuss this document but not move  
towards WGLC yet.

Issue: Mutual authentication

Noted that there does seem to be a use case for this extension  in  
IMS and cable networks, but that it probably doesn't meet the litmus  
test for an essential correction.

Conclusion: It shall be discussed as an extension.


Issue: invfix

Conclusion: this is an essential correction. Author Robert Sparks is  
to revise for WGLC.


Topic: INFO

Chairs proposal is for Hadriel and Christer to continue their draft,  
rolling in the supporting motivation from Eric's draft. We will also  
discuss use cases for the INFO package model. if we don't find three  
agreeable use cases by the next meeting, we'll go ahead and publish  
Eric's "INFO Considered Harmful" draft so that there will be some  
firm guidance in RFC format.




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Date: Wed, 2 Jan 2008 17:32:08 -0200
From: "Nadya Zanon" <nadya.zanon@gmail.com>
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Subject: [Sip] I'm a SIP beginner
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Hello All,

I'm a SIP beginner, probably my questions in the begin will be silly but I
believe that you will understand me.

I have two questions for now:

1) Are there some place where I can find a bunch of drafts for mapping of
PSTN features on SIP?
2) Are ther some gateway and standard that maps SMS messages on SIP? I mean,
are there some mapping from GSM for example to SIP?


Thanks in advance,

Nadya Zanon

------=_Part_26234_31273726.1199302328990
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<div>Hello All,</div>
<div>&nbsp;</div>
<div>I&#39;m a SIP beginner, probably my questions in the begin will be silly but I believe that you will understand me.</div>
<div>&nbsp;</div>
<div>I have two questions for now:</div>
<div>&nbsp;</div>
<div>1) Are there some place where I can find a bunch of drafts for mapping of PSTN features on SIP?</div>
<div>2) Are ther some gateway and standard that maps SMS messages on SIP? I mean, are there some mapping from GSM for example to SIP?</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>Thanks in advance,</div>
<div>&nbsp;</div>
<div>Nadya Zanon</div>

------=_Part_26234_31273726.1199302328990--




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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
--===============1204688262==--






From sip-bounces@ietf.org Wed Jan 02 22:17:40 2008
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From: Dale.Worley@comcast.net
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	(janej@hq.speakeasy.net)
Subject: Re: [Sip] INVITE after terminated dialog?
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   From: "Jane Jiang" <janej@hq.speakeasy.net>

   Hi All,

   There is a weird situation with us:
   (1) A Linksys phone sends an INVITE out with incorrect destination TN.
   As a result, our application server will set up a call between the
   Linksys phone and an IVR
   (2) A user incorrectly hits the on-hold key, which is basically sending
   a re-INVITE to the IVR
   (3) Our server seems to have answered the re-INVITE with a BYE
   (4) The Linksys phone responds to the BYE with a 200
   (5) Afterwards, the Linksys phone still thinks that its original
   re-INVITE has not been responded.  So, it will keep resend re-INVITE

   The consequence of these re-INVITEs is a registration migration, which
   we don't like.  We have been trying to figure out which side is not
   following RFC 3261, the phone, the SBCs are wrong, or the application
   servers.

   I am wondering:
   (A) Is it legal for a server to ignore the re-INVITE and respond with a
   BYE and has nothing in between?

The server can legally send a request before responding to a request
it has received.  Indeed, it must be allowed to do so, as from the
protocol point of view that action is indistinguishable from the
network delaying the re-INVITE.

However, the server remains obliged to send a response to the
re-INVITE, as each request is a separate transaction that demands a
response.  In this case, 481 Dialog Usage Does Not Exist seems most
appropriate.

   (B) If a UA has responded to a BYE with 200, is it supposed to forget
   about the re-INVITE request that it has sent out and not try to resend
   the re-INVITE, assuming a dialog has already been terminated with the
   BYE/200 transaction?

In principle, the re-INVITE is a separate transaction and the UA
should complete processing that transaction by re-sending to get a
response (until the time-out).  However, the server cannot enforce
that (as it is possible that the network has lost all sends of the
request), and if the UA decides that it no longer is concerned with
the response the re-INVITE might get, it can give up resending the
re-INVITE.

   Maybe in another sentence, is it legal to send our re-INVITEs
   outside an dialog, even if these re-INVITEs are the repeats of an
   original in-dialog re-INVITE?

A re-INVITE is by definition within a dialog, since a re-INVITE is (by
definition) an INVITE with a to-tag.  However, if the INVITE dialog
usage of the dialog has been terminated (by the server sending the
BYE), the re-INVITE will fail, as you can't create a new INVITE dialog
usage within a dialog that has had one which has been terminated.

Dale


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From sip-bounces@ietf.org Thu Jan 03 02:55:39 2008
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Nadya Zanon wrote:
>  
> 1) Are there some place where I can find a bunch of drafts for mapping 
> of PSTN features on SIP?
If, by 'PSTN features', you mean "SIP services" (e.g.: call fowarding, 
completion, transfers and so on) you MAY have a look at BLISS working 
group (http://ietf.org/html.charters/bliss-charter.html), and
on the "SIP services guide" individual draft tentative:
http://tools.ietf.org/html/draft-ligot-bliss-sip-services-guide

This is a work in progress, but it already contains many references to 
IETF and non-IETF specs (ETSI/TISPAN in particular).

Thomas



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From sip-bounces@ietf.org Thu Jan 03 03:39:20 2008
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
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Hi Nadya
=20
u can have a look at RFC 3398 for mapping of ISUP protocol to SIP.
=20
Regards
Ranjit

________________________________

From: Nadya Zanon [mailto:nadya.zanon@gmail.com]=20
Sent: Thursday, January 03, 2008 1:02 AM
To: sip@ietf.org
Subject: [Sip] I'm a SIP beginner


Hello All,
=20
I'm a SIP beginner, probably my questions in the begin will be silly but
I believe that you will understand me.
=20
I have two questions for now:
=20
1) Are there some place where I can find a bunch of drafts for mapping
of PSTN features on SIP?
2) Are ther some gateway and standard that maps SMS messages on SIP? I
mean, are there some mapping from GSM for example to SIP?
=20
=20
Thanks in advance,
=20
Nadya Zanon

------_=_NextPart_001_01C84DE4.0289E986
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	charset="us-ascii"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3199" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D177313608-03012008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi Nadya</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D177313608-03012008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D177313608-03012008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>u can have a look at RFC 3398 for mapping of =
ISUP protocol=20
to SIP.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D177313608-03012008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D177313608-03012008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D177313608-03012008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Ranjit</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Nadya Zanon =
[mailto:nadya.zanon@gmail.com]=20
<BR><B>Sent:</B> Thursday, January 03, 2008 1:02 AM<BR><B>To:</B>=20
sip@ietf.org<BR><B>Subject:</B> [Sip] I'm a SIP =
beginner<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>Hello All,</DIV>
<DIV>&nbsp;</DIV>
<DIV>I'm a SIP beginner, probably my questions in the begin will be =
silly but I=20
believe that you will understand me.</DIV>
<DIV>&nbsp;</DIV>
<DIV>I have two questions for now:</DIV>
<DIV>&nbsp;</DIV>
<DIV>1) Are there some place where I can find a bunch of drafts for =
mapping of=20
PSTN features on SIP?</DIV>
<DIV>2) Are ther some gateway and standard that maps SMS messages on =
SIP? I=20
mean, are there some mapping from GSM for example to SIP?</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>Thanks in advance,</DIV>
<DIV>&nbsp;</DIV>
<DIV>Nadya Zanon</DIV></BODY></HTML>

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Date: Thu, 3 Jan 2008 11:08:10 -0200
From: "Nadya Zanon" <nadya.zanon@gmail.com>
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Hello All,

Thanks for the all answers I received. They are very useful.

I have another question, what is intended as the future of SIP and
telephony?
I mean the development of gateways will still be developed and the analog
telephony will still be alive or the complete the replacement of the analog
telephony by SIP.

In the PBX world, if  SIP pbx are connected directly to "SIP trunks",  are
there differences between the end point SIP and the SIP from the carrier? I
have heard about SIP-T. Is there some relation with that?


Thanks.

Nadya Zanon

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<div>Hello All,</div>
<div>&nbsp;</div>
<div>Thanks for the all answers I received. They are very useful.</div>
<div>&nbsp;</div>
<div>I have another question, what is intended as the future&nbsp;of SIP and telephony?</div>
<div>I mean the development of gateways will still be developed and the analog telephony will still be alive or the complete&nbsp;the replacement of the analog telephony by SIP.</div>
<div>&nbsp;</div>
<div>In the PBX world,&nbsp;if&nbsp;&nbsp;SIP pbx&nbsp;are connected directly to &quot;SIP trunks&quot;,&nbsp; are there differences between the end point SIP and the SIP from the carrier? I have heard about SIP-T. Is there some relation with that?
</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>Thanks.</div>
<div>&nbsp;</div>
<div>Nadya Zanon</div>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
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Hello,

I have been looking for some time for an RFC or specification that=0D
details the mapping of SIP to Q.931. Does anyone know if such a=0D
document exists?

Thanks in advance,

Bruce Farnham
Time Warner Telecom


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<P><FONT COLOR=3D"#000000" FACE=3D"Comic Sans MS">I have been looking for=
 some time for an RFC or specification that </FONT>

<BR><FONT COLOR=3D"#000000" FACE=3D"Comic Sans MS">details the mapping of=
 SIP to Q.931. Does anyone know if such a </FONT>

<BR><FONT COLOR=3D"#000000" FACE=3D"Comic Sans MS">document exists?</FONT>
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<P><FONT COLOR=3D"#000000" FACE=3D"Comic Sans MS">Thanks in advance,</FONT>
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<P><FONT COLOR=3D"#000000" FACE=3D"Comic Sans MS">Bruce Farnham</FONT>

<BR><FONT COLOR=3D"#000000" FACE=3D"Comic Sans MS">Time Warner=
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From sip-bounces@ietf.org Thu Jan 03 12:27:56 2008
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James Polk pointed out I forgot to include the breakout session we  
had on Location Conveyance. I've added this to the minutes. Copies of  
these minutes are also posted in the iETF Proceedings and in the SIP  
WIKI on the Softarmor site.

Please point out any more issues.

Revised minutes follow:

Minutes, SIP at IETF 70

Edited by Dean Willis based on notes by Ben Campbell, Miguel Angel
Garcia-Martin and Brian Rosen

Session 1
========

Topic: Agenda and Status
led by Chairs
Slides presented and included in proceedings

Issue: How to deal with extensions in experimental RFCS like "e2m"

Proposed that we change the process of RFC 3427 to allow experimental
track RFCs to register SIP extension headers. Discussion concluded
that using an "X-" prefix in the registry would be
self-defeating. There are also concerns about setting the bar too low
-- we want any registered extension headers fields to be both
reasonable and well-documented. The group reached a consensus on
making a change to RFC 3427 allowing any RFC produced by the SIP
working group to register extension headers.

Issue: Robert's fork-loop-fix draft

Not enough people appear to have read the draft yet for us to have a
strong sense of whether the changes satisfy issues raised in WGLC. The
consensus is that the chairs shall conduct another short WGLC.


Topic: Location Conveyance
led by James Polk
Slides presented and included in proceedings

Issue: Should we have an error message to describe what's bad about
the civic address data contained in a prior message?

Several points were raised in discussion, including whether we have
enough of an understanding of location validation to be able to do
this sort of detailed response coding at this time. The current draft
allows for a human-readable string, and this may suffice for current
needs. Whether this is sufficient depends in large part on whether we
expect automata to be able to respond to a a validation error, correct
the location data, and resubmit it. There may also be the possible
that the automata could parse the error response and ask a human a
specific question, such as "What floor are you on?" if needed.

No conclusion was reached in the meeting, and the author was asked to
lead a discussion of the issue on the mailing list.

Issue: What are the semantics of the "recipient" parameter in the
Geolocation header?

Debate centered on whether this parameter is a hint, a provider of
context, or a binding constraint on authorization to access the
location information. The discussion became somewhat heated.

The working group did indicate a strong consensus to develop a
protocol mechanism for location-based routing.

Debate during the meeting session was inconclusive, and it was
proposed that we have an ad-hoc breakout session on this topic.

The ad-hoc session resolved to move this semantic into the geopriv
object itself, and to send the work item for this back to GEOPRIV to
complete.

Issue: Are the error codes proposed too complex or too granular?

Debate centered on whether the error codes helped users or
automata recover from correctable conditions or are intended
for post mortem debugging. Debate was extensive, with no
immediate consensus forthcoming, so this topic was also deferred
to the the breakout session.

The breakout session resulted in a consensus on restructuring the
error response codes. Minutes of this session are included
later in this document.


Topic: Info Harmful
led by led by Eric Burger
Slides presented and included in proceedings

Discussion was deferred until after the presentation of the INFO
Events draft.


Topic: Info Events
led by Christer Holmberg
Slides presented and included in proceedings

Issue: If an endpoint can receive an info package (i.e., indicates it
can receive such events), does that mean it will do something useful
with an event if it receives it?

Discussion seems to indicate that each INFO event package will
determine the requirements for that package, with the general trend
being "only say you know it if you want it". So, event packages, where
available, would be higher-priority than older techniques with less
negotiation flexibility.

Issue: If the other end doesn't understand an INFO event you want to
send, what do you do?

Discussion concluded that a node can only send events that the other
side has indicated a willingness to receive, so if the other end
doesn't understand something, don't send it. This may mean that
event-dependent applications won't work, or that one might have to do
something tricky like falling back to actually talking to the other
party instead of sending events.

Issue: Backward Compatibility

Noted that in the bad-old-days of INFO, all backward compatibility was
handled manually by configuration. That technique can still be
used. However, for the ISUP and QSIG uses, we can provide for backward
compatibility using Accept header fields as these payloads have MIME
types that are not used in other contexts. Also noted that we need to
consider these sort of backward compatibility issues in each event
package.

Issue: Do we want to enable transmission of user-to-user information
in an INVITE initiated dialog? This was one of the arguments against
dialogs of MESSAGE methods.

Discussion indicated that we're probably OK with this for the class of
things we seem to think will end up in INFO events. It was noted by a
chair that there is little sense in trying to "protocol police" this
sort of thing -- the requirement is just t make sure that the protocol
allows the applications involved to understand what they're doing and
not step on each other's extensions.

Issue: Use cases?

Whether DTMF is a convincing use case was discussed without conclusion.

Noted that other SDOs appear to be developing things using INFO. This
MAY be because INFO is the right thing, or it may be simply because
they don't have to negotiate with IETF to use it, and that if we
adjust INFO to encourage them to work with IETF, they might just find
some other loophole like X-headers.

Proposed that there are non-DTMF needs for end to end traffic, but
that these needs may be better met by end-to-end sessions like MSRP.
Of course, this raises the issue of negotiating the semantics of these
end-to-end sessions, like "Why are you sending me these JPEG images in
an MSRP session? What AM I supposed to do with them?", so there would
still be a need for some sort of package definition. However, DTMF may
be unique in being both signaling and media, and we might solve much
of the problem by just finding a fix at the SIP level for DTMF and
ignoring the remainder of the problem.

Discussion on whether use cases could motivate the work. At one point,
a poll by the chairs indicated that few of the people opposed to INFO
events would be persuaded even if confronted with a large number of
valid use cases.

Issue: Registration policy. If we were to define INFO events, what
would the IANA policy be on registering an INFO event package?

Chair Dean Willis proposed that the policy be "specification
required", given that our goal is not to police the architectures of
everybody who wants to use SIP, but that we do need to make sure that
the feature negotiation mechanism work well enough that conflicts are
avoided. We really should NOT have to use SIP working group time to
discuss whether or not some other SDO's use of their SIP-absed
signaling channel is valid.

Issue: Where to go from here?

Noted that there ARE other SDOs actively engaged, and that we are not
servicing the community well by giving no guidance.

Eric Burger noted that the two drafts are not incompatible, and that
the INFO events draft solves all of the "harmful" aspects of current
INFO use noted in his draft.

The meeting concluded with an agreement to further consider use cases,
but no commitment to further action. (Note: subsequent discussion
occurred on-list, shortly after the meeting session).


Session 2
========

Topic: Agenda and Status
led by Chairs
Slides presented and included in proceedings

Discussion of Essential Corrections will be added to the agenda if
time allows.

The XCAP Diff event draft will be revised and submitted as
draft-ietf-sip-xcapevent-00

Connection reuse was rescoped to allow some requests in the reverse
direction and discourage connection reuse for virtual servers. Some
further clarifications are needed.



Topic: Outbound
led by Rohan Mahy
Slides presented and included in proceedings

Issue:  TCP Keepalive

Current text removed. Noted that even if you use TCP keepalive, you
still need to do the keepalives as specified in this draft.

Issue:  interaction with GRUU.

Group seemed to indicate consensus on the proposal by Rohan for the
2nd option listed in the slides -- GRUU should check that an incoming
REGISTER matches any Call-ID already registered by this device. This
would appear to require a change in the GRUU document.

Issue: What error does the registrar send if it gets a register with
require: outbound but the edge proxy doesn't support outbound?

Christer Holmberg suggested that most cases would need a 500-class.

Jonathan Rosenberg noted that this depends on what might happen if the
requested were to be retried. If it is always going to fail, the
response would be in the 400 class. If success on retry is possible, a
500 might be more reasonable.

Cullen Jennings noted that It is also important to consider what
clients will do with the response.

Adam Roach observed that a 420 response would make it hard to debug
the problem from a client perspective, and that a new error code would
be better.

Jonathan suggested using a proxy require that the proxy would
translate to require, but Rohan reminded us of prior discussion where
we had decided this would not work.

Christer concurred with Adam, saying that this problem is more
general, essentially an "invalid path".

The resolution is to develop a new 400 class response code indicating
the problem with this path. Adam Roach volunteered to send text.

Issue: How do subsequent in-dialog requests follow across an Outbound
edge link?

Proposed by Rohan that the Outbound proxy MUST record route if the top
Route header has an :ob" parameter.

Extended discussion followed, noting that this makes us dependent on
proxies for mid-dialog requests, which has impacts on reliability
designs.

Francois Audet noted that the current text is hard to understand.

Chairs conclude that the room seems to mostly support the resolution
proposed by Rohan, but that the final text needs to be reviewed and
agreed on-list.

Issue: The "keep" parameter.

Ted Hardie  argued for  keeping the text as-is.

Aki Niemi argued in favor of deprecating the "keep" parameter, based
on its lack of usefulness and potential to confuse people.

Christer Holmberg suggested deprecating the parameter when used with
an outbound proxy, but allowing it for non-outbound cases.

Poll by chairs indicated strong consensus for removing the "keep"
parameter.


Issue: Replace timed-keepalive parameter with a new header

Proposed by Aki that weuse a new header, such as "Flow-Timer" to be
returned in register response.

Noted by Rohan that this might fix the non-outbound case for
keepalive.

Christer questions this, as it is still necessary to know if keepalive
is supported.

Poll by chairs indicated strong consensus to define a new header for
this purpose.


Issue: Detecting client support for keepalives without "outbound"

Proposed new option tag for client support, included by client in
Supported.

Poll by chairs indicated preference to consider this approach outside
of the Outbound draft. Christer Holmberg may pursue a draft on this
topic.

Procedural point: Francois Audet has volunteered to act as the
protocol shepherd for getting this draft through the process.

Derek reminded Rohan to add text relating to the route construction
issue. Derek will try to send text on this to Rohan.


Topic: Resource Priority Header in Responses
led by Janet Gunn
Slides presented and included in proceedings

Discussion centered on the role of the RPH header in a response and
the assumptions underlying a network in which it would be useful.  The
key issue is that since SIP cannot challenge responses that if the
response is not signed (as in SIP Identity) then there is no way to
tell upstream where the RPH header came from. There may be no coupling
between the proxy layer and the access network for media priority
enforcement.

Noted by EKR that given the security model, where the user is not
trusted but all the proxy nodes are, that this requires integrity
protection on the connections, with peer authentication. A proxy
receiving a response with RPH has to know that it's getting this
response from a trusted proxy and not somebody else.

Several parties suggested that this sort of mechanism is more
appropriate for a P-header with an applicability statement that
explains the transitive trust model.

Keith Drage, as chair, asked people to consider the stateless proxy
use case from James' draft relative to the security assumptions,
P-header concept, and authentication issues.


Topic: Media Identity
led by Dan Wing
Slides presented and included in proceedings

Issue: SBCs break 4474 because SDP signed by Identity is rewritten by
SBC

Francois observed that this problem may only occur when it is an SBC
outside the domain of the Identity server that is doing the rewriting.

Hannes suggested extending IDENTITY to be like DKIM, which can
selectively sign pieces.

General discussion revealed that most of the material in the signaling
is problematic from a signature perspective. IP Addresses are
essentially meaningless, especially across domain boundaries. Phone
numbers are little better, with end-nodes having no real way to test
assertions about them (and perhaps a service provided signature is the
best we can do here).  RFC 4474 already documents limitations with
respect to telephone numbers.

There seems to be an emerging thread of consensus here that what we
need is some way to bind media to signaling -- that is, to by
inspection of the media and the signaling be able to say with some
level of certainty that they are related -- and specifically, to say
"this media is a result of that signaling".

Chairs noted that there is additional work required her to be able to
frame this into the sort of requirement that could be added to a
charter.


Topic: DTLS Framework
led by Jason Fischl
Slides presented and included in proceedings

Issue: 4474 for phone numbers

We may be lacking documentation on when RFC 4474 does and does not
work, and what security we get when using it with phone numbers. Jon
Peterson seems to think that 4474 already offers this guidance.
Jonathan Rosenberg disagrees and believes the text in 4474 is
inadequate and more is needed.

Consensus noted that even if we do need more here, it is outside of
the scope of the DTLS framework. However, the framework should at
least note the issue.

Issue; Anonymity

Agreed that text must be updated to say it does not break existing
anonymity.

Issue: Middle boxes.

SBCs may block key exchange before 200 OK. This is not specific to
DTLS-SRTP.

Proposed by Christer that we at least mention this issue in the draft.

EKR noted that much of this SBC difficulty should be documented in
Brian's middlebox draft.

Issue: SRTP/TCP

Consensus: DTLS-SRTP over TCP rather than RTP over TLS.

Issue: Priority for advancing.

Chairs observed that an AVT document is dependent on this, making this
a high priority to move along, However, the AVT chair reported that
the AVT document is not as ready as we might have thought.


Topic: Essential Corrections
led by Keith Drage
Slides presented and included in proceedings

Issue: general intent of essential corrections

We seem to have some disagreement as to the intent and value of the
essential corrections process. The chairs restated the goal as being
to batch together the updates to RFC 3261 so that readers do not have
to look in many places to find the current "corrected" specification.


Issue: record-route-fix

Is this an essential correction or an enhancement? It does identify
specific failure cases that occur if RFC 3261 is used as-is. The
chairs noted that drafts making normative changes to a standards track
draft need to be standards track themselves, not BCPs. However, BCPs
would be appropriate for documents that, for example, argue for using
one option in an an existing standards-track document over an
alternative in that document.

Robert argues that this fix is needed, although it would be acceptable
to advance it individually rather than in the essential-corrections
procss.

Conclusion: robert and Jonathan are to go discuss this and come back
with a recommendation.

Issue: IPV6 ABNF error

Consensus that this is an essential correction we would consider
anything that behaves as per 3261 to be erroneous.


Issue: 503 corrrection

JDR argues that it is known that 3261 doesn't perform well in overload
conditions, and that we have ongoing work to improve this behavior.

Volker Hit concurs, noting that the overload team is still working to
get better simulation results.

Conclusion: We'll continue to discuss this document but not move
towards WGLC yet.

Issue: Mutual authentication

Noted that there does seem to be a use case for this extension in IMS
and cable networks, but that it probably doesn't meet the litmus test
for an essential correction.

Conclusion: It shall be discussed as an extension.


Issue: invfix

Conclusion: this is an essential correction. Author Robert Sparks is
to revise for WGLC.


Topic: INFO

Chairs proposal is for Hadriel and Christer to continue their draft,
rolling in the supporting motivation from Eric's draft. We will also
discuss use cases for the INFO package model. if we don't find three
agreeable use cases by the next meeting, we'll go ahead and publish
Eric's "INFO Considered Harmful" draft so that there will be some firm
guidance in RFC format.


Session: Location Ad-Hoc
========================

An additional break-out session was held to discuss issues related to
location conveyance and SIP, due to the failure to reach consensus
on key issues during the regular session.

Notes reported by Dean Willis
Session chaired by Keith Drage

Note-Well reviewed

Discussion led by James POlk
Slides presented


Issue: Error codes in Geolocation-Error header

Issues raised in SIP meeting include general sense that current error
codes are just "too much" and should be reduced.

Discussion of actionable vs. diagnostic errors ensued (started by
James Winterbottom). Are the text strings fixed so that they can be
processed by automata? Are we doing trend analysis or trying to make a
call succeed that was failing?

Comment by Henning Schulzrinne: Not really a question of whether 14 or
17 is the right number, but as to whether the civic address (CA) type
information is appropriately included in the responses. Just including
a CA type doesn't help solve the problem and may confuse the
situation. Robots cannot usefully "fix" defective CA information, but
require human interaction. I fear that the software will say "A1 and
A7 are wrong" and the user will have no idea what to do.

Response: Do you want to be able to say things like "The city field
was missing and we need it." We could put this in the current draft
for now.

Counteproposal: We should leave CA types out for now and add them back
in only if experience shows that they would have been useful. It's
better to drop it now.

We seem to have a consensus to delete the CA type information from  
error responses.

As for other error codes, EKR suggests that those that are repairable
by automata be included. Henning agrees, suggesting we also have one
error code for human-fixable errors that can be communicated in text.

Ted Hardie categorizes errors in 3 groups : 1) Same things that might
work later, 2) vague "something you gave me is wrong", 3)
bit-toggleable "permission" errors. Possibly a 4th class, "things an
automata can fix", like refreshing its location calc and resending.

Noted that SIP "Accept-Language" mechanism provides the
human-language negotiation mechanism.

Poll: does anybody have issues with Ted's plan?

Mostly agreed: James will revisit.


Issue: Recipient=

Henning discussed differences between URN routing and URI routing in
the presence of a location body. He has concerns that it might not be
possible to make a reasonable determination in an implementation and
this has lawsuit potential.

Ted counter-argues that this was concluded in GEOPRIV.

Massive debate ensued.

Proposed: Easy solution for SIP working group is to move the
functionality formally known as "recipient =" out of SIP and into
PIDF-LO as a routing-allowed flag. This will require deleting the
subject from the SIP draft and creating a new GEOPRIV draft. Jon
Peterson volunteered to write the GEOPRIV draft.

NOTE: The GEOPRIV draft will require careful definition of
"retransmission" so as to not interact with the fact that SIP proxies
operate by retransmitting SIP requests.







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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Thu Jan 03 13:08:24 2008
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Subject: RE: [Sip] That is the future
Date: Thu, 3 Jan 2008 19:07:30 +0100
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There is no relationship between IP-PBX interconnection and SIP-T unless =
you need to interwork with SS7 as part of this process.
=20
You can find work on interconnection of IP-PBX to public SIP networks in =
the SIP forum, and in ETSI TISPAN.

The SIP Forum work is at:

http://www.sipforum.org/content/view/290/290/
=20
The ETSI TISPAN work is still in progress (except first document below =
which is complete), but is publically available at:
=20
http://portal.etsi.org/docbox/TISPAN/Open/NGN_LATEST_DRAFTS/RELEASE2/
=20
and take whatever is the latest version of:
=20
01047
02040
02041
02042
=20
=20
Regards

Keith=20
=20


________________________________

	From: Nadya Zanon [mailto:nadya.zanon@gmail.com]=20
	Sent: Thursday, January 03, 2008 1:08 PM
	To: sip@ietf.org
	Subject: [Sip] That is the future
=09
=09
	Hello All,
	=20
	Thanks for the all answers I received. They are very useful.
	=20
	I have another question, what is intended as the future of SIP and =
telephony?
	I mean the development of gateways will still be developed and the =
analog telephony will still be alive or the complete the replacement of =
the analog telephony by SIP.
	=20
	In the PBX world, if  SIP pbx are connected directly to "SIP trunks",  =
are there differences between the end point SIP and the SIP from the =
carrier? I have heard about SIP-T. Is there some relation with that?=20
	=20
	=20
	Thanks.
	=20
	Nadya Zanon



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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Thu Jan 03 13:58:25 2008
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Date: Fri, 4 Jan 2008 01:57:46 +0700
From: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>
To: sip@ietf.org
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--===============0161668270==
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Hi,,

I'm new to this mailing list. Right now, i'm studying about interworking
between megaco and sip.
I simulated a call from SIP phone to Analog phone which was busy. You can
download the captured data using ethereal in here:
http://www.4shared.com/file/33781721/3b2ac786/4577020ke4580220busy.html, and
you can view the callflow by writing "*(megaco and ip.addr==10.14.32.186) or
(sip and ip.addr==10.14.32.185) or rtp*" in filter textbox.

I have a few questions about that captured data :
1. Why did SIP Phone send a CANCEL twice?
2. What did line 2115 mean?

I hope my questions are all clear to be understood,,,because i'm not too
good in english.
Hopefully, you guys can help me.

Regards,

Jati

------=_Part_12857_17200001.1199386666802
Content-Type: text/html; charset=ISO-8859-1
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Hi,,<br>
<br>
I&#39;m new to this mailing list. Right now, i&#39;m studying about interworking between megaco and sip.<br>
I simulated a call from SIP phone to Analog phone which was busy. You can download the captured data using ethereal in here: <a href="http://www.4shared.com/file/33781721/3b2ac786/4577020ke4580220busy.html">http://www.4shared.com/file/33781721/3b2ac786/4577020ke4580220busy.html
</a>, and you can view the callflow by
writing &quot;<i>(megaco and ip.addr==10.14.32.186) or (sip and ip.addr==10.14.32.185) or rtp</i>&quot; in filter textbox.<br>
<br>
I have a few questions about that captured data :<br>
1. Why did SIP Phone send a CANCEL twice?<br>2. What did line 2115 mean?<br><br>
I hope my questions are all clear to be understood,,,because i&#39;m not too good in english.<br>
Hopefully, you guys can help me.<br>
<br>
Regards,<br><font color="#888888">
<br>
Jati</font>

------=_Part_12857_17200001.1199386666802--



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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
--===============0161668270==--





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Subject: [Sip] SIP2Megaco Case:Callee onhook first
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Hi,,

Now, i have questions too in a call from SIP phone to analog phone which the
case was the callee onhooked first, which is the analog phone. You can
download captured data in this url:
http://www.4shared.com/file/33781592/42539b99/4577020ke4580220ttupdtlpn.html.
And you can view the data using this filter : *(megaco and
ip.addr==10.14.32.186) or (sip and ip.addr==10.14.32.185) or rtp.

*The questions are:
1. When the callee onhooked, why did the MGC send INVITE to the SIP Phone
(line 3845)?
2. Is that true, that the ring back tone, busy tone, dial tone can be sent
using RTP? I mean, why did after 180 Ringing (line 1621), RTP was sent
between 10.14.32.185 and 10.14.32.186? Did the RTP contain ring tone?If yes,
is there a standard explain this?

That's all that i want to ask,,,I hope you guys can help me.
Thx in advance before.

Regards,

Jati

------=_Part_12866_9118058.1199386783128
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Hi,,<br><br>Now, i have questions too in a call from SIP phone to
analog phone which the case was the callee onhooked first, which is the
analog phone. You can download captured data in this url: <a href="http://www.4shared.com/file/33781592/42539b99/4577020ke4580220ttupdtlpn.html">http://www.4shared.com/file/33781592/42539b99/4577020ke4580220ttupdtlpn.html
</a>.&nbsp; And you can view the data
using this filter : <i>(megaco and ip.addr==10.14.32.186) or (sip and ip.addr==10.14.32.185) or rtp.<br><br></i>The questions are:<br>1. When the callee onhooked, why did the MGC send INVITE to the SIP Phone (line 3845)?
<br>2.
Is that true, that the ring back tone, busy tone, dial tone can be sent
using RTP? I mean, why did after 180 Ringing (line 1621), RTP was sent
between <a href="http://10.14.32.185/" target="_blank">10.14.32.185</a> and <a href="http://10.14.32.186/" target="_blank">10.14.32.186</a>? Did the RTP contain ring tone?If yes, is there a standard explain this?<br><br>
That&#39;s all that i want to ask,,,I hope you guys can help me.
<br>Thx in advance before.<br><br>Regards,<br><font color="#888888"><br>Jati</font>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1809516649==--





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Date: Fri, 4 Jan 2008 02:14:17 +0700
From: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>
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This is the right url:
http://www.4shared.com/account/file/33781721/3b2ac786/4577020ke4580220busy.html?sId=x8tjOAyXOEYhJmc5
.

Thx b4...

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This is the right url: <a href="http://www.4shared.com/account/file/33781721/3b2ac786/4577020ke4580220busy.html?sId=x8tjOAyXOEYhJmc5" target="_blank">http://www.4shared.com/account/file/33781721/3b2ac786/4577020ke4580220busy.html?sId=x8tjOAyXOEYhJmc5
</a>.<br><br>Thx b4...

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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
--===============1443313885==--





From sip-bounces@ietf.org Thu Jan 03 14:16:08 2008
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From: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>
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Subject: [Sip] SIP2Megaco Case:Callee onhook first
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This is the right url:
<http://www.4shared.com/account/file/33781721/3b2ac786/4577020ke4580220busy.html?sId=x8tjOAyXOEYhJmc5>
http://www.4shared.com/account/file/33781592/42539b99/4577020ke4580220ttupdtlpn.html?sId=x8tjOAyXOEYhJmc5

Thx b4...

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This is the right url: <a href="http://www.4shared.com/account/file/33781721/3b2ac786/4577020ke4580220busy.html?sId=x8tjOAyXOEYhJmc5" target="_blank"></a><a href="http://www.4shared.com/account/file/33781592/42539b99/4577020ke4580220ttupdtlpn.html?sId=x8tjOAyXOEYhJmc5">
http://www.4shared.com/account/file/33781592/42539b99/4577020ke4580220ttupdtlpn.html?sId=x8tjOAyXOEYhJmc5</a><br><br>Thx b4...

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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
--===============1629385480==--





From sip-bounces@ietf.org Fri Jan 04 09:43:36 2008
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Thread-Topic: [Megaco] SIP2Megaco Case:Callee onhook first
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Subject: [Sip] RE: [Megaco] SIP2Megaco Case:Callee onhook first
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Tones can be played over the RTP stream.  Typically a tone is defined as
proceeding from the Termination towards the exterior of the context so
if a tone is requested to be played on a Phy termination it is
played/generated locally while if it is played to the EPH termination it
is played on the RTP stream.

=20

See megaco standards document section 7.1.11 ( V2 document )

=20

John

=20

=20

________________________________

From: Jati Kalingga Praja [mailto:king.of.kalingga@gmail.com]=20
Sent: Thursday, January 03, 2008 12:25 PM
To: megaco@ietf.org; sip@ietf.org
Subject: [Megaco] SIP2Megaco Case:Callee onhook first

=20

Hi,,

Now, i have questions too in a call from SIP phone to analog phone which
the case was the callee onhooked first, which is the analog phone. Here,
i attached captured data too. You can view the data using this filter :
(megaco and ip.addr=3D=3D10.14.32.186) or (sip and =
ip.addr=3D=3D10.14.32.185) or
rtp.

The questions are:
1. When the callee onhooked, why did the MGC send INVITE to the SIP
Phone (line 3845)?
2. Is that true, that the ring back tone, busy tone, dial tone can be
sent using RTP? I mean, why did after 180 Ringing (line 1621), RTP was
sent between 10.14.32.185 and 10.14.32.186? Did the RTP contain ring
tone?If yes, is there a standard explain this?

That's all that i want to ask,,,I hope you guys can help me.=20
Thx in advance before.

Regards,

Jati


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xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns=3D"http://www.w3.org/TR/REC-html40">

<head>
<meta http-equiv=3DContent-Type content=3D"text/html; =
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Tones can be played over the RTP
stream.&nbsp; Typically a tone is defined as proceeding from the =
Termination
towards the exterior of the context so if a tone is requested to be =
played on a
Phy termination it is played/generated locally while if it is played to =
the EPH
termination it is played on the RTP stream.<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'>See megaco standards document =
section
7.1.11 ( V2 document )<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'>John<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>

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

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

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

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Jati =
Kalingga
Praja [mailto:king.of.kalingga@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, January =
03, 2008
12:25 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> megaco@ietf.org; =
sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Megaco] =
SIP2Megaco
Case:Callee onhook first</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=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Hi,,<br>
<br>
Now, i have questions too in a call from SIP phone to analog phone which =
the
case was the callee onhooked first, which is the analog phone. Here, i =
attached
captured data too. You can view the data using this filter : <i><span
style=3D'font-style:italic'>(megaco and ip.addr=3D=3D10.14.32.186) or =
(sip and
ip.addr=3D=3D10.14.32.185) or rtp.<br>
<br>
</span></i>The questions are:<br>
1. When the callee onhooked, why did the MGC send INVITE to the SIP =
Phone (line
3845)?<br>
2. Is that true, that the ring back tone, busy tone, dial tone can be =
sent
using RTP? I mean, why did after 180 Ringing (line 1621), RTP was sent =
between <a
href=3D"http://10.14.32.185">10.14.32.185</a> and <a =
href=3D"http://10.14.32.186">10.14.32.186</a>?
Did the RTP contain ring tone?If yes, is there a standard explain =
this?<br>
<br>
That's all that i want to ask,,,I hope you guys can help me. <br>
Thx in advance before.<br>
<br>
Regards,<br>
<br>
Jati<o:p></o:p></span></font></p>

</div>

</body>

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Date: Fri, 4 Jan 2008 21:08:41 +0530
From: "Hulbut hulbut" <hulbut@gmail.com>
To: rohan@ekabal.com
Subject: [Sip] Few Doubts/comments in draft-ietf-sip-outbound-11
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Hi Rohan,

I have following doubts in draft:

1) There is a typo (I think) in section 5.3

   If the flow in the flow token in the topmost Route header field value
   matches the source of the request, the request *in* an "outgoing"
   request.  For an "outgoing" request, the edge proxy just removes the
   Route header and continues processing the request.  Otherwise, this
   is an "incoming" request.

 Instead of "in" there should be "is"

2) Section 5.3(Forwarding Non-Register requests), evaluates inbound/outbound
on the basis of Flow token.

   If the flow in the flow token in the topmost Route header field value
   matches the source of the request, the request in an "outgoing"
   request.  For an "outgoing" request, the edge proxy just removes the
   Route header and continues processing the request.  Otherwise, this
   is an "incoming" request.

However in the mentioned decode algorithm, above mismatch will lead to
sending 403 forbidden.

   Example Algorithm:  To decode the flow token, take the flow
      identifier in the user portion of the URI and base64 decode it,
      then verify the HMAC is correct by recomputing the HMAC and
      checking that it matches.  If the HMAC is not correct, the proxy
      SHOULD send a 403 (Forbidden) response.  If the HMAC is correct
      then the proxy SHOULD forward the request on the flow that was
      specified by the information in the flow identifier.  If this flow
      no longer exists, the proxy SHOULD send a 430 (Flow Failed)
      response to the request.


I think inbound/outbound determination can be left to the proxy's own
implementation.
Regarding sending 403 response when HMAC mismatch happens, I am not clear
how will any request be sent to UEs
Considering an Edge proxy receives request from some other server
(authoritative server), the flow token in route header would be
pointing to UEs source address(or NAT mapped address) but the value which
Edge proxy will compute will be based on the source address of other server.
Thus, in case of inbound all requests will be rejected by 403 .


Please correct me !

Thanks
Hulbut

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<p style="font-family: courier new,monospace;" class="MsoNormal"><font size="2"><span style="font-size: 10pt;"></span>Hi Rohan,<br><br>
I have following doubts in draft: <br><br>
1) There is a typo (I think) in section 5.3</font>

</p>

<pre style="font-family: courier new,monospace;"><font size="2"><span style="">&nbsp;&nbsp; </span>If the flow in the flow token in the topmost Route header field value<br><span style="">&nbsp;&nbsp; </span>matches the source of the request, the request 
<b><i>in</i></b> an &quot;outgoing&quot; <br><span style="">&nbsp;&nbsp; </span>request.<span style="">&nbsp; </span>For an &quot;outgoing&quot; request, the edge proxy just removes the<br><span style="">&nbsp; </span>&nbsp;Route header and continues processing the request.
<span style="">&nbsp; </span>Otherwise, this<br><span style="">&nbsp;&nbsp; </span>is an &quot;incoming&quot; request. </font></pre><pre style="font-family: courier new,monospace;"><font size="2">&nbsp;Instead of &quot;in&quot; there should be &quot;is&quot;
<br></font></pre>

<p style="font-family: courier new,monospace;" class="MsoNormal"><font size="2">2) Section 5.3(Forwarding Non-Register requests), evaluates inbound/outbound on
the basis of Flow token.</font>
</p>



<pre style="font-family: courier new,monospace;"><font size="2"><span style="">&nbsp;&nbsp; </span>If the flow in the flow token in the topmost Route header field value<br><span style="">&nbsp;&nbsp; </span>matches the source of the request, the request in an &quot;outgoing&quot;
<br><span style="">&nbsp;&nbsp; </span>request.<span style="">&nbsp; </span>For an &quot;outgoing&quot; request, the edge proxy just removes the<br><span style="">&nbsp;&nbsp; </span>Route header and continues processing the request.<span style="">
&nbsp; </span>Otherwise, this<br><span style="">&nbsp;&nbsp; </span>is an &quot;incoming&quot; request. <br><br><span class="NormalcouriernewChar"><span style="font-size: 12pt;">However in the mentioned decode algorithm, above mismatch will lead to sending 403 forbidden.
</span></span><span style="font-size: 12pt;"></span></font></pre>

<p style="font-family: courier new,monospace;" class="Normalcouriernew"><font size="2"><span style="">&nbsp;&nbsp; </span><span class="HTMLPreformattedChar"><span style="font-size: 10pt;">Example Algorithm:<span style="">&nbsp;
</span>To decode the flow token, take the flow</span></span><br><span class="HTMLPreformattedChar"><span style="font-size: 10pt;"><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>identifier in the user portion of the URI
and base64 decode it,<br>
<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>then verify the HMAC is correct by
recomputing the HMAC and<br>
<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>checking that it matches.<span style="">&nbsp; </span>If the HMAC is not correct, the proxy<br>
<span style="">&nbsp;&nbsp;&nbsp; </span><span style="">&nbsp;&nbsp;</span>SHOULD send a 403 (Forbidden) response.<span style="">&nbsp; </span>If the HMAC is correct<br>
<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>then the proxy SHOULD forward the
request on the flow that was<br>
<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>specified by the information in the
flow identifier.<span style="">&nbsp; </span>If this flow<br>
<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>no longer exists, the proxy SHOULD
send a 430 (Flow Failed)<br>
<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>response to the request.</span></span><br><br><br>
I think inbound/outbound determination can be left to the proxy&#39;s own
implementation.<br>
Regarding sending 403 response when HMAC mismatch happens, I am not clear how
will any request be sent to UEs<br>
Considering an Edge proxy receives request from some other server
(authoritative server), the flow token in route header would be<br>
pointing to UEs source address(or NAT mapped address) but the value which Edge
proxy will compute will be based on the source address of other server.<br>
Thus, in case of inbound all requests will be rejected by 403 .</font>




</p>

<p style="font-family: courier new,monospace;" class="Normalcouriernew"><font size="2"><br>
Please correct me !<br><br>
Thanks<br>
Hulbut</font>
</p>

<p style="font-family: courier new,monospace;" class="MsoNormal"><font size="2"><span style="font-size: 10pt;">&nbsp;</span></font></p><br>

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From: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>
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Hi,,,

Right now i'm studying interworking between megaco and sip. Is there any
exact definition about delay translation?And is there any specs about delay
translation in this interworking? And how about the message mapping?
I'm confused in counting the delay translation value. For example, if the
caller is using sip, MG will send only INVITE to MGC,,it is different if the
caller is analog phone, MG will send a lot of Megaco message to MGC (Notify
Offhook, Modify, Notify digits, Add),,how can i count the delay translation
here?

Hopefully you guys can help me,,

Regards,

Jati

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Hi,,,<br><br>Right now i&#39;m studying interworking between megaco and sip. Is there any exact definition about delay translation?And is there any specs about delay translation in this interworking? And how about the message mapping?
<br>I&#39;m confused in counting the delay translation value. For example, if the caller is using sip, MG will send only INVITE to MGC,,it is different if the caller is analog phone, MG will send a lot of Megaco message to MGC (Notify Offhook, Modify, Notify digits, Add),,how can i count the delay translation here?
<br><br>Hopefully you guys can help me,,<br><br>Regards,<br><br>Jati<br>

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Content-Transfer-Encoding: quoted-printable

=20
About your second question, you can find help from RFC 3959 and RFC 3960
which tells early media.
Yes, after 180 there can follows RTP packets to let the endpoint to play
remote media.
=20
=20
=20
Best Regards,
=20
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Fanwen zhang
FS5000 CPE IOT, Alcatel-Lucent Qingdao, R&D
Tel: +86-532-88615493
Email:fanwenz@alcatel-lucent.com
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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

________________________________

From: Jati Kalingga Praja [mailto:king.of.kalingga@gmail.com]=20
Sent: Friday, January 04, 2008 3:00 AM
To: megaco@ietf.org; sip@ietf.org
Subject: [Sip] SIP2Megaco Case:Callee onhook first


Hi,,

Now, i have questions too in a call from SIP phone to analog phone which
the case was the callee onhooked first, which is the analog phone. You
can download captured data in this url:
http://www.4shared.com/file/33781592/42539b99/4577020ke4580220ttupdtlpn.
html .  And you can view the data using this filter : (megaco and
ip.addr=3D=3D10.14.32.186) or (sip and ip.addr=3D=3D10.14.32.185) or =
rtp.

The questions are:
1. When the callee onhooked, why did the MGC send INVITE to the SIP
Phone (line 3845)?=20
2. Is that true, that the ring back tone, busy tone, dial tone can be
sent using RTP? I mean, why did after 180 Ringing (line 1621), RTP was
sent between 10.14.32.185 <http://10.14.32.185/>  and 10.14.32.186
<http://10.14.32.186/> ? Did the RTP contain ring tone?If yes, is there
a standard explain this?

That's all that i want to ask,,,I hope you guys can help me.=20
Thx in advance before.

Regards,

Jati=20

------_=_NextPart_001_01C850CF.C6ED2033
Content-Type: text/html;
	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1595" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3D&#23435;&#20307; =
color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D140374801-07012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>About your second question, you can find help =
from RFC 3959=20
and RFC 3960 which tells early media.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D140374801-07012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>Yes, after 180 there can follows RTP packets to =
let the=20
endpoint to play remote media.</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV align=3Dleft><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT size=3D2>Best Regards,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT =
size=3D2>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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></D=
IV>
<DIV><FONT size=3D2>Fanwen zhang<BR>FS5000 CPE IOT, Alcatel-Lucent =
Qingdao,=20
R&amp;D<BR>Tel:=20
+86-532-88615493<BR>Email:fanwenz@alcatel-lucent.com</FONT></DIV>
<DIV><FONT =
size=3D2>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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></D=
IV>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Jati Kalingga Praja=20
[mailto:king.of.kalingga@gmail.com] <BR><B>Sent:</B> Friday, January 04, =
2008=20
3:00 AM<BR><B>To:</B> megaco@ietf.org; sip@ietf.org<BR><B>Subject:</B> =
[Sip]=20
SIP2Megaco Case:Callee onhook first<BR></FONT><BR></DIV>
<DIV></DIV>Hi,,<BR><BR>Now, i have questions too in a call from SIP =
phone to=20
analog phone which the case was the callee onhooked first, which is the =
analog=20
phone. You can download captured data in this url: <A=20
href=3D"http://www.4shared.com/file/33781592/42539b99/4577020ke4580220ttu=
pdtlpn.html">http://www.4shared.com/file/33781592/42539b99/4577020ke45802=
20ttupdtlpn.html=20
</A>.&nbsp; And you can view the data using this filter : <I>(megaco and =

ip.addr=3D=3D10.14.32.186) or (sip and ip.addr=3D=3D10.14.32.185) or =
rtp.<BR><BR></I>The=20
questions are:<BR>1. When the callee onhooked, why did the MGC send =
INVITE to=20
the SIP Phone (line 3845)? <BR>2. Is that true, that the ring back tone, =
busy=20
tone, dial tone can be sent using RTP? I mean, why did after 180 Ringing =
(line=20
1621), RTP was sent between <A href=3D"http://10.14.32.185/"=20
target=3D_blank>10.14.32.185</A> and <A href=3D"http://10.14.32.186/"=20
target=3D_blank>10.14.32.186</A>? Did the RTP contain ring tone?If yes, =
is there a=20
standard explain this?<BR><BR>That's all that i want to ask,,,I hope you =
guys=20
can help me. <BR>Thx in advance before.<BR><BR>Regards,<BR><FONT=20
color=3D#888888><BR>Jati</FONT> </BODY></HTML>

------_=_NextPart_001_01C850CF.C6ED2033--



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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
--===============1464347566==--





From sip-bounces@ietf.org Sun Jan 06 22:20:09 2008
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Subject: RE: [Sip] INVITE after terminated dialog?
Date: Mon, 7 Jan 2008 11:18:23 +0800
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From: "Zhang, Rocky Fan Wen \(Rocky\)" <fanwenz@alcatel-lucent.com>
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=20

It seems you have used TISPAN mode.
I think the reasonable process should be:
- When flashes at endpoint it will send out a INVITE but it is not a
re-INVITE.
- When AS receive this INVITE it should return 200 OK
- The endpoint return a ACK, and the second INVITE is ignored and the
announcement goes on.
=20

=20
Best Regards,
=20
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Fanwen zhang
FS5000 CPE IOT, Alcatel-Lucent Qingdao, R&D
Tel: +86-532-88615493
Email:fanwenz@alcatel-lucent.com
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

-----Original Message-----
From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]=20
Sent: Thursday, January 03, 2008 11:17 AM
To: sip@ietf.org
Subject: Re: [Sip] INVITE after terminated dialog?

   From: "Jane Jiang" <janej@hq.speakeasy.net>

   Hi All,

   There is a weird situation with us:
   (1) A Linksys phone sends an INVITE out with incorrect destination
TN.
   As a result, our application server will set up a call between the
   Linksys phone and an IVR
   (2) A user incorrectly hits the on-hold key, which is basically
sending
   a re-INVITE to the IVR
   (3) Our server seems to have answered the re-INVITE with a BYE
   (4) The Linksys phone responds to the BYE with a 200
   (5) Afterwards, the Linksys phone still thinks that its original
   re-INVITE has not been responded.  So, it will keep resend re-INVITE

   The consequence of these re-INVITEs is a registration migration,
which
   we don't like.  We have been trying to figure out which side is not
   following RFC 3261, the phone, the SBCs are wrong, or the application
   servers.

   I am wondering:
   (A) Is it legal for a server to ignore the re-INVITE and respond with
a
   BYE and has nothing in between?

The server can legally send a request before responding to a request it
has received.  Indeed, it must be allowed to do so, as from the protocol
point of view that action is indistinguishable from the network delaying
the re-INVITE.

However, the server remains obliged to send a response to the re-INVITE,
as each request is a separate transaction that demands a response.  In
this case, 481 Dialog Usage Does Not Exist seems most appropriate.

   (B) If a UA has responded to a BYE with 200, is it supposed to forget
   about the re-INVITE request that it has sent out and not try to
resend
   the re-INVITE, assuming a dialog has already been terminated with the
   BYE/200 transaction?

In principle, the re-INVITE is a separate transaction and the UA should
complete processing that transaction by re-sending to get a response
(until the time-out).  However, the server cannot enforce that (as it is
possible that the network has lost all sends of the request), and if the
UA decides that it no longer is concerned with the response the
re-INVITE might get, it can give up resending the re-INVITE.

   Maybe in another sentence, is it legal to send our re-INVITEs
   outside an dialog, even if these re-INVITEs are the repeats of an
   original in-dialog re-INVITE?

A re-INVITE is by definition within a dialog, since a re-INVITE is (by
definition) an INVITE with a to-tag.  However, if the INVITE dialog
usage of the dialog has been terminated (by the server sending the BYE),
the re-INVITE will fail, as you can't create a new INVITE dialog usage
within a dialog that has had one which has been terminated.

Dale


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


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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--===============1258268591==
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	boundary="=_alternative 00422567C22573C9_="

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Wish to get a clarification regarding what will happen when we subscribe 
to the 
resources below and the first entry does not exist while the other two do 
exist. 
Assume that we get a pending notification, when and how will it be 
possible to 
know which resource does exist and which one does not.

What will be the content of the pending notification and how it will be 
possible 
to know which one exist?


   <?xml version="1.0" encoding="UTF-8"?>
   <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists">
    <list>
     <entry uri="resource-lists/users/sip:joe@example.com/index"/>
     <entry uri="rls-services/users/sip:joe@example.com/index/
   ~~/*/service%5b@uri='sip:marketing@example.com'%5d"/>
     <entry uri="pidf-manipulation/"/>
    </list>
   </resource-lists>
--Avshalom


--=_alternative 00422567C22573C9_=
Content-Type: text/html; charset="US-ASCII"


<br><font size=2 face="Courier New">Wish to get a clarification regarding
what will happen when we subscribe to the </font>
<br><font size=2 face="Courier New">resources below and the first entry
does not exist while the other two do exist. </font>
<br><font size=2 face="Courier New">Assume that we get a pending notification,
when and how will it be possible to </font>
<br><font size=2 face="Courier New">know which resource does exist and
which one does not.</font>
<br>
<br><font size=2 face="Courier New">What will be the content of the pending
notification and how it will be possible </font>
<br><font size=2 face="Courier New">to know which one exist?</font>
<br>
<br>
<br><font size=2 face="Courier New">&nbsp; &nbsp;&lt;?xml version=&quot;1.0&quot;
encoding=&quot;UTF-8&quot;?&gt;</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;&lt;resource-lists xmlns=&quot;urn:ietf:params:xml:ns:resource-lists&quot;&gt;</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &lt;list&gt;</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp;&lt;entry uri=&quot;resource-lists/users/sip:joe@example.com/index&quot;/&gt;</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp;&lt;entry uri=&quot;rls-services/users/sip:joe@example.com/index/</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;~~/*/service%5b@uri='sip:marketing@example.com'%5d&quot;/&gt;</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp;&lt;entry uri=&quot;pidf-manipulation/&quot;/&gt;</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &lt;/list&gt;</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;&lt;/resource-lists&gt;<br>
--Avshalom<br>
<br>
</font>
--=_alternative 00422567C22573C9_=--



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Use sipping@ietf.org for new developments on the application of sip
--===============1258268591==--





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From: "Karl Heinz Wolf" <khwolf1@gmail.com>
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Subject: [Sip] Re: SIP Location Conveyance revved (-09)
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by James M. Polk  Nov 21, 2007; 09:56pm
< - made "inserted-by=" a mandatory header parameter.  This is because
< there is a comparable header for errors that needs this parameter's
< value. I'll mention that in a bit.


Hi James,

I just had a look at the draft because of the "inserted-by" parameter
and noticed the following inconsistency:

Section 3 says:

   The Geolocation header can have parameters that are associated
   with a URI in the header field.  The "inserted-by" parameter has
   values of "endpoint" or "server", indicating which entry added
   location to the message. This header parameter MAY be added every
   time a new location is added into a message.

But below it is a "must" and the parameter contains the host name.

-karl heinz


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Subject: RE: [Sip] doubt regarding early session in sip RFC 3959/3960
Date: Mon, 7 Jan 2008 18:41:54 -0500
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Thread-Topic: [Sip] doubt regarding early session in sip RFC 3959/3960
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From: "Brett Tate" <brett@broadsoft.com>
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> UAS is expecting " early-answer " in PRACK from=20
> the UAC, what will happen if UAC does not=20
> support "100rel" (RFC 3262).=20

RFC 3959 section 4 indicates that early-session must follow the
offer/answer model.  Thus an early-session cannot be setup by indicating
an offer within 18x when 100rel is not being used.


> And RFC 3959 is not mandating the "100rel" extension=20
> at UA for the same. Can anyone give an example of =20
> early session flow in case of  UAC is not=20
> supporting "100rel". (UPDATE..?)=20
=09
It is generically discussed within rfc3959 section 4. =20

UPDATE and rfc3311 are not specifically discussed with rfc3959.  I'm not
sure if this was an oversight or because maybe UPDATE currently defined
to "modify" a session instead of "establish" a session.


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Date: Tue, 08 Jan 2008 09:29:51 -0600
From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Subject: [Sip] Review of draft-ietf-speechsc-mrcpv2-15
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Hi:

The following is a review of draft-ietf-speechsc-mrcpv2-15 with
emphasis on S1-S5, S12 and S14.  There are a few general comments
first; the rest of the comments follow in linear fashion with respect
to the sections enumerated above.

General comment 1:
------------------
A general comment I have is what is the relationship of MRCPv2 with
RTSP?  Both of these seem to provide some similar functionality,
such as a recording resource.  Also, RFC4463, the precursor to MRCPv2
appears to discuss RTSP more than this document does.  In this
document, SIP essentially replaces RTSP as the preferred protocol to
establish a control session.  This is fine, but it would seem to me
that the average reader familiar with RTSP may benefit from some
thoughts about the relationship of MRCPv2 to RTSP.

General comment 2:
------------------
Another general comment I have concerns the SIP message flow in S4.2.
There are a number of tactical omissions and implicit assumptions in
that call flow that are repeated in the call flows of other sections
as well.  I will list these out for S4.2 and trust that the authors
will retrofit other sections based on this discussion.

  1. On page 15, when the C->S, the R-URI of the request is shown as
   sip:mresources@server.example.com.  While this is not inaccurate,
   what will typically happen in SIP is that the initial request is
   targeted to a domain (i.e., sip:mresources@example.com) instead
   of a specific server within that domain
   (i.e., sip:mresources@server.example.com).  Once the SIP dialog
   has been established, subsequent requests will contain a R-URI
   that is reflective of the specific server within that domain.

   Thus, on the very first request of a dialog-forming method like
   INVITE, you may want to consider changing the R-URI to reflect
   the domain URI of "sip:mresources@example.com".  When that server
   replies with a 200 OK, the Contact URI of that response will
   contain a FQDN (i.e., "sip:mresources@server.example.com") -- and
   this is indeed the case in your call flow (see 200 OK on Page 16.)

   Note that subsequent requests in the dialog will contain a R-URI
   that corresponds to the FQDN of the server.  Thus, in S4.2, at the
   end of page 16, the R-URI of the server is accurately shown as
   being "sip:mresources@server.example.com".

  2. On page 15, the Contact URI of the initial INVITE request should
   contain a specific client within that domain.  That is, the Contact
   URI must be "sip:sarvi@client.example.com" and not
   "sip:sarvi@example.com".

  3. In SIP, a LWS is used to denote continuation.  Thus, instead of
   writing:

     Via:SIP/2.0/TCP client.atlanta.example.com:5060;
     branch=z9hG4bK74bf9

   you must instead write:

     Via:SIP/2.0/TCP client.atlanta.example.com:5060;
       branch=z9hG4bK74bf9

   Note the LWS on the second line above.

  4. In SIP, when a server responds with an affirmative response,
   it always adds a tag parameter to the From header.  In the call
   flow on page 16, the From header of the 200 OK does not contain
   a tag parameter, it must.  Interestingly enough, the subsequent
   ACK request contains the tag in the From parameter (as indeed it
   should.)  Please add a tag parameter to the From header in the
   200 OK, and ensure that it matches the one in the ACK request.
   The tags are maintained for the duration of the lifetime of that
   dialog.

   Thus, in the re-INV that starts at the bottom of page 16, the
   From tag must be maintained.

   Aside: the second m= line in the 200 OK body should have one 0
   (i.e., s/00/0).

  5. In SIP, the CSeq number of the ACK is kept the same as the
   preceding INVITE.  Thus, on page 16 the CSeq number of the ACK
   request must be 314161.

General comment 3:
------------------
In S6.1.1, page 29, there is an example message:

    MRCP/2.0 124 SET-PARAMS 543256
    Channel-Identifier:32AECB23433802@speechsynth
    Voice-gender:female
    Voice-variant:3

The second token (124) refers to the message length of the
message, including the message body, if any.  In the above
example, if I assume that each line is terminated by a '\n'
and a '\r\n' terminates the message itself, then I come up
with a message length of 115 octets, which does not match
the value shown above of 124.  I would advise that, for the sake of
completeness, the message-length field matches the actual value
of the number of octets in the message for this example as well
as the rest of the examples in the draft, including the examples
in S14.

As another example, take a look at the S->C message at the end
of page 42 in S7.  I calculate the payload length to be 291
octets, which does not match the Content-Length field value of
269 in the example.  I would advise that, for the sake of
completeness, the Content-Length field value matches the actual
value of the number of octets in the SIP message body for this
example as well as the rest of the examples in the draft.

General comment 4:
------------------
The state machines in the draft are depicted as a time-flow
diagram -- this is interesting.  While I have not seen a state machine
depicted as such -- all of the ones I am used to employ the
canonical vertex and edges between them to signify transitions --
if a time-flow diagram suffices for the population reading the draft,
then I am perfectly fine with leaving it in as is.  I will depend on
your discretion to strike out this comment if the depiction of the
state machine is consistent with reader expectations.

Rest of my comments follow:

- Abstract: It is stated that MRCPv2 "relies on a session management
   protocol such as the Session Initiation Protocol (SIP) to establish
   the MRCPv2 control session between the client and the server, and
   for rendezvous and capability discovery."  True, SIP is a rendezvous
   protocol, but in and of itself, it is not a (media) capability
   discovery protocol; SDP is probably closer to the mark for (media)
   capability discovery and description (incidentally, SDP is mentioned
   in the immediate next sentence.)  I would suggest a modest re-write,
   something along these lines:

   OLD:
   - it relies on a session management protocol such as the Session
   Initiation Protocol (SIP) to establish the MRCPv2 control session
   between the client and the server, and for rendezvous and capability
   discovery. It also depends on SIP and SDP ...

   NEW:
   - it relies on other protocols, such as Session Initiation Protocol
   (SIP) to rendezvous MRCPv2 client and servers and manage sessions
   between them, and the Session Description Protocol (SDP) to describe,
   discover and exchange capabilities.  It also depends on SIP and SDP
   ...

- S1, second paragraph: It is stated that "In addition, the client
   can use a SIP re-INVITE method (an INVITE dialog sent within an
   existing SIP Session) to change the characteristics of these
   media and control session while maintaining the SIP dialog between
   the client and server."  It is true that you can use re-INV to do
   what is mentioned.  However, I am curious: did you consider the
   UPDATE method?  It also allows a client to update the parameters
   of a session, but is lightweight than the re-INV (for instance,
   it does not require an ACK, and does not require the server to
   keep retransmission state of the response until it gets an ACK.)

   Now, using a re-INV for offer/answer is sometimes advantageous
   in situations where, for instance, the sender wants to send a re-
   INV but does not specify any offer in it.  The receiver sends an
   offer in the 2xx, and the sender answers that offer in the ACK.
   But I do not see MRCPv2 using such an exchange in the call flows
   present in the document.  These call flows are of the normal pattern
   of "offer in request, answer in response".  I realize that call
   flows do not normatively reflect the text, but if MRCPv2 can benefit
   from some reduced complexity by using a UPDATE/200 to change the
   session characteristics, then why not think about it?  I will
   leave the final decision to you.

- S1, third paragraph, first line: consider taking out the phrase
   "client/server" from the sentence.  In SIP, once a dialog is set
   up, either end can act as a client.

- S2.1: There appears to be some formatting problems.  The second
   column invariably starts at a line below the first column.

   S3.1: Same comment as above when describing the media processing
   resources.

- S2.2: Given that the reader of this document has no idea (at this
   point in the document) what PENDING, IN-PROGRESS, or COMPLETE
   are, you may wish to consider moving this paragraph before the
   first state diagram, by which time these states would have been
   explained.  Or alternatively, you may want to consider putting a
   forward reference to S5.3, where these states are defined.

- S3, third paragraph: s/server using this/server on this/

- S3, third paragraph: You may want to put a forward reference to the
   framing issue when doing I/O over a stream transport.  When I was
   reading this paragraph, it naturally occurred to me whether MRCPv2
   provides framing (it does, and this is discussed in S5.)

   Same comment as above for the opening paragraph of S4.

- S3.1, in the definition of a Recorder:
   s/A resource capable of recording audio and saving it to a URI./A
    resource capable of recording audio and saving it as resource
    accessible by a URI./

- S3.1, in the defintion of a Recorder, second line:
   s/end pointing//

- S3.2: the opening paragraph can best be re-written as follows:

      The MRCPv2 server is a generic SIP server, and is thus addressed
      by a SIP URI, for example: sip:mrcpv2@example.net.

   This is more concise, and I believe sufficient.  How this address is
   resolved to reach the SIP server, and how that SIP server registers
   with its registrar are all handled by rfc3261.  I do not see any
   advantage of discussing SIP registrars and Contact URIs here.

- S4.1, third line: s/SIP transactions/SIP requests/

- S4.3, page 21: s/and fork it/and give it/

- S4.3, last paragraph: I am always a proponent of better error
   diagnostics.  Thus, instead of putting the stock reason-phrase of
   "Not Implemented" on the 501 status-code, why not be more
   descriptive and put a reason-phrase of "Cannot mix/multimix or
   fork media"?  That allows the automata to consume the 501
   status-code as intended, but also allows for a descriptive
   reason-phrase to be displayed to the user.

- S5.1, page 22: In the ABNF for generic-message, the generic-
   header production rule is not expanded until S6.2.  I would
   suggest that it be expanded in S5.1, along with the resource-
   header production rule, or barring that, at least a forward
   reference to S6.2 be maintained to let the reader quickly
   find where generic-header is expanded.

- S5.1: It may be best to move the discussion of the message-length
   field on page 23 to S5.2.  It fits better there since the token
   is described in S5.2.  Otherwise, you straddle critical information
   in between two sections and the reader has to go back and forth
   to make sense (BTW, FWIW, I am glad you have this field; framing is
   made much easier and code can be optimized by having it.)

   Same comment for mrcp-version production rule; describe it in S5.2
   as well.

- S5.2: I think it may be best if you describe the fields in the
   order that they appear in the request-line production rule.  So,
   describe mrcp-version production rule first, followed by
   message-length, and so on.  And consider consolidating the
   description of mrcp-version and message-length from S5.1 to this
   section (see previous comment.)

   As an added incentive to do this, note that in S5.3, the order
   of description matches the order in which the tokens appear in
   the response-line production rule.

- S5.4: I note that you have some sort of an asynchronous
   notification system of responses (i.e., a PENDING or IN-PROGRESS
   indicates further Event messages will be delivered.)  Why not have
   an specific status-code to that effect?  In SIP, we have the
   notion of a "202 Accepted" response, which indicates that a request
   has been accepted but yet to be fully processed.

   Would a 202 response code be valuable to MRCPv2?  I cannot say for
   sure, but did want to point this out to you so you can make a
   judgment on whether this is indeed the case.  If you would like
   to read up more on the semantics of the 202 response code in SIP,
   it is defined in rfc3265.

- S5.5, first paragraph: s/The status value is COMPLETE/The request-
   state value is COMPLETE/

   Also, same comment as in S5.2 about making sure that the description
   of the tokens in the event-line production rule match the order in
   which the tokens appear in the production rule.

- S7: missing terminating period on the last sentence of the section
   (end of page 42.)

- The state machines in the draft are depicted as a time-flow
   diagram -- this is interesting.  While I have not seen a state machine
   depicted as such -- all of the ones I am used to employ the
   canonical vertex and edges between them to signify transitions --
   if a time-flow diagram suffices for the population reading the draft,
   then I am perfectly fine with leaving it in as is.  I will depend on
   your discretion to strike out this comment if the depiction of the
   state machine is consistent with reader expectations.

- S12.3 - Does anything need to be said about how to key the SRTP
   stream?

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
WWW:   http://www.alcatel-lucent.com/bell-labs


_______________________________________________
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Vijay,

A very thorough review. I have one comment on it, below.

	Thanks,
	Paul

Vijay K. Gurbani wrote:
> Hi:
> 
> The following is a review of draft-ietf-speechsc-mrcpv2-15 with
> emphasis on S1-S5, S12 and S14.  There are a few general comments
> first; the rest of the comments follow in linear fashion with respect
> to the sections enumerated above.

[snip]

>  4. In SIP, when a server responds with an affirmative response,
>   it always adds a tag parameter to the From header. 

Don't you mean To: header?

The initial INVITE should have a From-tag and the response should 
contain that and a To-tag. All subsequent requests and responses in the 
dialog should have both.

>   In the call
>   flow on page 16, the From header of the 200 OK does not contain
>   a tag parameter, it must.  Interestingly enough, the subsequent
>   ACK request contains the tag in the From parameter (as indeed it
>   should.)  Please add a tag parameter to the From header in the
>   200 OK, and ensure that it matches the one in the ACK request.
>   The tags are maintained for the duration of the lifetime of that
>   dialog.
> 
>   Thus, in the re-INV that starts at the bottom of page 16, the
>   From tag must be maintained.


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Paul Kyzivat wrote:
> Vijay,
> 
> A very thorough review. I have one comment on it, below.
> 
> 
> Vijay K. Gurbani wrote:
> 
>>  4. In SIP, when a server responds with an affirmative response,
>>   it always adds a tag parameter to the From header. 
> 
> Don't you mean To: header?

Paul: Yes, my bad.  Thanks for catching this.  Indeed, I meant the
To header.

> The initial INVITE should have a From-tag and the response should 
> contain that and a To-tag. All subsequent requests and responses in the 
> dialog should have both.

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
WWW:   http://www.alcatel-lucent.com/bell-labs


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From: "Elwell, John" <john.elwell@siemens.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, IETF SIP List <sip@ietf.org>
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Hadriel,

Some possible further use cases:

1. Along the same lines as 6.2/6.3/6.4 (vcard, user-icon, vcalendar), we
could also consider pictures, movie clips, audio clips, etc.. Of course,
we don't want to encourage sending anything too big along the signalling
path, so we would need to consider where to draw the line.

2. We already have the conference event package for obtaining conference
membership details. In many situations the conference focus is also the
source of such notifications, so the subscribe-initiated dialog
parallels the INVITE-initiate dialog, which is rather wasteful. It would
be great to subscribe as part of the INVITE transaction and receive
notifications along the INVITE-initiated dialog.

3. Concerning mobility, I like the suggestion in 6.8 (geo-location
information). Also, there may sometimes be intermediaries on the
signalling path (B2BUAs) that are involved in mobility in some way, and
these might be interested in movement-related notifications (e.g., a
dual mode device moving in or out of WLAN range).

4. The use of offer-answer to achieve hold-retrieve fails to give an
explicit indication of what is happening. An explicit indication of
things like hold, retrieve, microphone mute, speaker mute, camera
on/off, recorder on/off, etc. might be quite helpful. Of course, there
will normally be an offer-answer exchange at the same time, so inclusion
of the notification in a re-INVITE or UPDATE request might be more
useful than using INFO.

5. Concerning 6.7 (soft-key labels), a problem here is that the
notifying UA might not have knowledge of soft key availability at the
receiving UA. In certain closed environments (e.g., involving a UA and
its local voicemail server) it may work, but I am not sure it would work
well in general. Having said that, where it does work, perhaps other
parameters at the device can be controlled in this way, e.g., microphone
and speaker volume (especially if this control is based on say a web
interface, so the user is really controlling his/her own settings from
the web page rather than directly on the device).

6. When a B2BUA performs 3PCC there is plenty of scope for the UAs to
lose track of what is happening. For example, when a B2BUA holds,
retrieves, transfers calls, joins calls, pre-empts calls, etc., there is
currently no explicit information - only perhaps a re-INVITE request.
Thus it can be difficult to know how to interpret things like identity.
Say a UA receives a re-INVITE request with a new PAI, it might deduce
that something like call transfer has occurred. However, if it receives
a re-INVITE request without a PAI, does it mean the previous PAI is
still valid, or does it mean, following transfer, there is no longer any
identity available? Of course, INFO might not always be the most
appropriate method to use for such notification if an offer-answer
exchange is needed at the same time.

7. There might be some usages to do with interworking with TDM, e.g.,
advice of charge during a call, user-to-user information during a call.

John


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From sip-bounces@ietf.org Wed Jan 09 05:25:04 2008
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Hi,


Chapter 9 of RFC3840 says:

"There are additional constraints on the usage of feature-param that
cannot be represented in a BNF. There MUST only be one instance of any
feature tag in feature-param."


Chapter 10 of RFC3840 says:

"Furthermore, there can only be one instance of any feature tag in
feature-param."



So, this means that the following is NOT allowed:

Accept-Contact: *;myTag,myTag


However, since an Accept-Contact header can contain multiple ac-values,
each with its own sets of ac-parameters (including feature-params), the
following should be allowed:

Accept-Contact: *;myTag, *;myTag

...or:

Accept-Contact: *;myTag
Accept-Contact: *;myTag


Correct?


Regards,

Christer





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To: "Sudhir Kumar Reddy" <t.sudhirkumar@gmail.com>
Subject: Re: [Sip] 200 OK in calling state for INVITE client transaction
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Hi Sudhir,

ACK must contain the same To- tag as recevied in 200 OK for invite, 
because to-tag is used to identify dialog for which this ACK is received.

Regards,
Vinay Sharma
Tata Consultancy Services
Mailto: vin.sharma@tcs.com
Website: http://www.tcs.com
____________________________________________
Experience certainty.   IT Services
                        Business Solutions
                        Outsourcing
____________________________________________



"Sudhir Kumar Reddy" <t.sudhirkumar@gmail.com> 
12/21/2007 09:50 AM

To
sip@ietf.org
cc

Subject
[Sip] 200 OK in calling state for INVITE client transaction






Hi All,

when an INVITE client transaction is in the Calling state, on receipt of 
Success (200 OK) response. As per RFC 17.1.1.2(Figure 5: INVITE client 
transaction), the 2xx response MUST cause the client transaction to enter 
the "Terminated" state. But, I'm not sure what would be To_Tag in 
generating ACK for that 200 OK. 

Any response is highly appericated.

Thanks,
Sudhir_______________________________________________
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<br><font size=2 face="sans-serif">Hi Sudhir,</font>
<br>
<br><font size=2 face="sans-serif">ACK must contain the same To- tag as
recevied in 200 OK for invite, because to-tag is used to identify dialog
for which this ACK is received.</font>
<br>
<br><font size=2 face="sans-serif">Regards,<br>
Vinay Sharma<br>
Tata Consultancy Services<br>
Mailto: vin.sharma@tcs.com<br>
Website: http://www.tcs.com<br>
____________________________________________<br>
Experience certainty. &nbsp; &nbsp; &nbsp; &nbsp;IT Services<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;Business Solutions<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;Outsourcing<br>
____________________________________________</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td width=40%><font size=1 face="sans-serif"><b>&quot;Sudhir Kumar Reddy&quot;
&lt;t.sudhirkumar@gmail.com&gt;</b> </font>
<p><font size=1 face="sans-serif">12/21/2007 09:50 AM</font>
<td width=59%>
<table width=100%>
<tr valign=top>
<td>
<div align=right><font size=1 face="sans-serif">To</font></div>
<td><font size=1 face="sans-serif">sip@ietf.org</font>
<tr valign=top>
<td>
<div align=right><font size=1 face="sans-serif">cc</font></div>
<td>
<tr valign=top>
<td>
<div align=right><font size=1 face="sans-serif">Subject</font></div>
<td><font size=1 face="sans-serif">[Sip] 200 OK in calling state for INVITE
client transaction</font></table>
<br>
<table>
<tr valign=top>
<td>
<td></table>
<br></table>
<br>
<br>
<br><font size=3>Hi All,<br>
<br>
when an INVITE client transaction is in the Calling state, on receipt of
Success (200 OK) response. As per RFC 17.1.1.2(Figure 5: INVITE client
transaction), the 2xx response MUST cause the client transaction to enter
the &quot;Terminated&quot; state. But, I'm not sure what would be To_Tag
in generating ACK for that 200 OK. <br>
<br>
Any response is highly appericated.<br>
<br>
Thanks,<br>
Sudhir</font><font size=2><tt>_______________________________________________<br>
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From sip-bounces@ietf.org Wed Jan 09 08:28:18 2008
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Date: Wed, 09 Jan 2008 19:00:36 +0530
From: Mahesh Anjanappa <mahesha@samsung.com>
Subject: Re: [Sip] Question on RFC3840/3841
To: Christer Holmberg <christer.holmberg@ericsson.com>, sip@ietf.org
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hi
That is my understanding as well. But if that is the case then it should 
apply to Contact header as well
when describing Capabilities using Feature parameters in Contact Header.
But i'm not sure if it applies the same way. I had posted this question long 
ago but didnt get any comments on it.

regards
Mahesh

----- Original Message ----- 
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: <sip@ietf.org>
Cc: "Paul Kyzivat" <pkyzivat@cisco.com>
Sent: Wednesday, January 09, 2008 3:54 PM
Subject: [Sip] Question on RFC3840/3841



Hi,


Chapter 9 of RFC3840 says:

"There are additional constraints on the usage of feature-param that
cannot be represented in a BNF. There MUST only be one instance of any
feature tag in feature-param."


Chapter 10 of RFC3840 says:

"Furthermore, there can only be one instance of any feature tag in
feature-param."



So, this means that the following is NOT allowed:

Accept-Contact: *;myTag,myTag


However, since an Accept-Contact header can contain multiple ac-values,
each with its own sets of ac-parameters (including feature-params), the
following should be allowed:

Accept-Contact: *;myTag, *;myTag

...or:

Accept-Contact: *;myTag
Accept-Contact: *;myTag


Correct?


Regards,

Christer





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Christer Holmberg wrote:
> Hi,
> 
> 
> Chapter 9 of RFC3840 says:
> 
> "There are additional constraints on the usage of feature-param that
> cannot be represented in a BNF. There MUST only be one instance of any
> feature tag in feature-param."
> 
> 
> Chapter 10 of RFC3840 says:
> 
> "Furthermore, there can only be one instance of any feature tag in
> feature-param."
> 
> 
> 
> So, this means that the following is NOT allowed:
> 
> Accept-Contact: *;myTag,myTag

The above isn't syntactically correct. I presume you meant to write:

Accept-Contact: *;myTag;myTag

And yes, that isn't allowed.

> However, since an Accept-Contact header can contain multiple ac-values,
> each with its own sets of ac-parameters (including feature-params), the
> following should be allowed:
> 
> Accept-Contact: *;myTag, *;myTag
> 
> ...or:
> 
> Accept-Contact: *;myTag
> Accept-Contact: *;myTag
> 
> 
> Correct?

Yes.

	Thanks,
	Paul


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Mahesh Anjanappa wrote:
> hi
> That is my understanding as well. But if that is the case then it should 
> apply to Contact header as well
> when describing Capabilities using Feature parameters in Contact Header.
> But i'm not sure if it applies the same way. I had posted this question 
> long ago but didnt get any comments on it.

The rule forbidding multiple instances of a parameter applies to all 
headers, so it applies to Contact.  E.g. the following is illegal:

	Contact:sip:alice@atlanta.com;methods="INVITE";methods="CANCEL"

Paul


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Thread-Topic: Delivering request-URI and parameters to UAS via proxy
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A reminder of the deadline on the 11th January for submitting
alternative proposals on the way forward.

Regards

Keith=20

> -----Original Message-----
> From: DRAGE, Keith (Keith)=20
> Sent: Tuesday, December 04, 2007 3:27 PM
> To: sip@ietf.org
> Subject: Delivering request-URI and parameters to UAS via proxy
>=20
> (As WG chair)
>=20
> We have a couple of milestones that we generated as a result=20
> of discussion of=20
>=20
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> se-route-01.txt
>=20
> Dec 2007    Delivering request-URI and parameters to UAS via=20
> proxy to WGLC =20
> Feb 2008    Delivering request-URI and parameters to UAS via=20
> proxy to IESG (PS) =20
>=20
> This work is currently stalled and the editor needs input.
>=20
> The document contains various example scenarios where a=20
> solution is required, for which there appears to be no=20
> dispute that a solution is needed.
>=20
> The document proposes one solution to resolve these example=20
> scenarios, but this solution is not gaining consensus. At=20
> least one other solution has been talked about, but there is=20
> no documentation on the table.
>=20
> This mail is to identify a deadline for other solutions to=20
> the example scenarios to be documented as internet drafts,=20
> showing how the solution works for those scenarios. This deadline is:
>=20
> 	January 11th 2008
>=20
> It is appropriate fo these documents to identify any other=20
> scenarios where such a solution is appropriate. Any other=20
> input is also welcome in identifying other scenarios.
>=20
> If we have no such internet-drafts by this deadline, we will=20
> proceed with completing the solution we have.
>=20
>=20
> Regards
>=20
> Keith
>=20
>=20


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Date: Wed, 09 Jan 2008 22:48:26 +0530
From: Mahesh Anjanappa <mahesha@samsung.com>
Subject: Re: [Sip] Question on RFC3840/3841
To: Paul Kyzivat <pkyzivat@cisco.com>
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Thanks Paul.
So would it be right if i have a Contact Headers describing Duplex Audio & 
one-way Video Capabilities as follows?

    Contact:sip:alice@atlanta.com;
    Contact:sip:audio;duplex=full;
    Contact:sip:video;duplex=recieve-only;

NOTE: The Predicate(for above) for which i'm attempting is
& ( & (sip.audio) (sip.duplex=full) )
    (  & (sip.video) (sip.duplex=receive-only) )

regards
Mahesh

----- Original Message ----- 
From: "Paul Kyzivat" <pkyzivat@cisco.com>
To: "Mahesh Anjanappa" <mahesha@samsung.com>
Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>; <sip@ietf.org>
Sent: Wednesday, January 09, 2008 7:55 PM
Subject: Re: [Sip] Question on RFC3840/3841


>
>
> Mahesh Anjanappa wrote:
>> hi
>> That is my understanding as well. But if that is the case then it should 
>> apply to Contact header as well
>> when describing Capabilities using Feature parameters in Contact Header.
>> But i'm not sure if it applies the same way. I had posted this question 
>> long ago but didnt get any comments on it.
>
> The rule forbidding multiple instances of a parameter applies to all 
> headers, so it applies to Contact.  E.g. the following is illegal:
>
> Contact:sip:alice@atlanta.com;methods="INVITE";methods="CANCEL"
>
> Paul 



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From sip-bounces@ietf.org Wed Jan 09 13:36:47 2008
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Hi,

>>Chapter 9 of RFC3840 says:
>>
>>"There are additional constraints on the usage of feature-param that
>>cannot be represented in a BNF. There MUST only be one instance of any
>>feature tag in feature-param."
>>
>>
>>Chapter 10 of RFC3840 says:
>>
>>"Furthermore, there can only be one instance of any feature tag in =
feature-param."
>>
>>
>>
>>So, this means that the following is NOT allowed:
>>
>>Accept-Contact: *;myTag,myTag
>
>The above isn't syntactically correct. I presume you meant to write:
>
>Accept-Contact: *;myTag;myTag

Yes.

Regards,

Christer



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From sip-bounces@ietf.org Wed Jan 09 14:01:40 2008
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Date: Thu, 10 Jan 2008 02:00:29 +0700
From: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>
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Subject: [Sip] Translation Megaco 2 SIP
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Hi,,

Right now, i'm studying about interworking between megaco and sip. Here i
attached the call flow diagram which i have concluded from my study. But i
found some questions here :
1. Look at the attached file, if i want to count delay translation from
megaco to sip "INVITE" message, which megaco message that i must use as the
time reference to count the delay?Is it notify digits or notify digits
response or add response?
2. Loot at the part of before the RTP, there is
3. Loot at the end part of the attached file, theoritically, when the analog
phone onhooked first, MGC must send the BYE message to the SIP phone, don't
it?,,,then, why did the SIP phone sent the BYE message?

Thx before in advance,,,

Best Regards,


Jati

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Hi,,<br><br>Right now, i&#39;m studying about interworking between megaco and sip.  Here i attached the call flow diagram which i have concluded from my study. But i found some questions here :<br>1. Look at the attached file, if i want to count delay translation from megaco to sip &quot;INVITE&quot; message, which megaco message that i must use as the time reference to count the delay?Is it notify digits or notify digits response or add response?
<br>2. Loot at the part of before the RTP, there is <br>3. Loot at the end part of the attached file, theoritically, when the analog phone onhooked first, MGC must send the BYE message to the SIP phone, don&#39;t it?,,,then, why did the SIP phone sent the BYE message?
<br><br>Thx before in advance,,,<br><br>Best Regards,<br><br><br>Jati<br>

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------=_Part_10989_26389566.1199905229342
Content-Type: text/plain; charset="us-ascii"
MIME-Version: 1.0
Content-Transfer-Encoding: 7bit
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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
------=_Part_10989_26389566.1199905229342--





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Date: Thu, 10 Jan 2008 02:15:46 +0700
From: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>
To: sip@ietf.org, megaco@ietf.org, sip-implementors@lists.cs.columbia.edu
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Subject: [Sip] Translation SIP 2 Megaco
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Hi,

I have another translation questions again,,
I attached the file too. Or you can download it in :
http://www.4shared.com/file/34360429/6e0d5471/sip2megaco_interworking.html.

If i want to count delay translation from Megaco message to "180 Ringing",
which megaco message do i use?Is *modify (signal:{cg/bt})*?or modify
response?or add response?

Thx before in advance,

Best Regards,

Jati

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Hi,<br><br>I have another translation questions again,,<br>I attached the file too. Or you can download it in : <a href="http://www.4shared.com/file/34360429/6e0d5471/sip2megaco_interworking.html">http://www.4shared.com/file/34360429/6e0d5471/sip2megaco_interworking.html
</a>.<br><br>If i want to count delay translation from Megaco message to &quot;180 Ringing&quot;, which megaco message do i use?Is <i>modify (signal:{cg/bt})</i>?or modify response?or add response?<br><br>Thx before in advance,
<br><br>Best Regards,<br><br>Jati<br>

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------=_Part_11016_23566565.1199906146345
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From sip-bounces@ietf.org Wed Jan 09 14:28:17 2008
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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] draft-kaplan-sip-info-use-cases-00 - more use cases?
Date: Wed, 9 Jan 2008 13:26:22 -0600
To: "Elwell, John" <john.elwell@siemens.com>
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On Jan 9, 2008, at 3:58 AM, Elwell, John wrote:

> Hadriel,
>
> Some possible further use cases:
>
> 1. Along the same lines as 6.2/6.3/6.4 (vcard, user-icon,  
> vcalendar), we
> could also consider pictures, movie clips, audio clips, etc.. Of  
> course,
> we don't want to encourage sending anything too big along the  
> signalling
> path, so we would need to consider where to draw the line.

So, how is this different from MESSAGE with the "gruu" as the target?  
Does it gain anything from being in-dialog? Does it gain from the  
specific content-type and usage negotiation? If so, does this mean we  
should fix this problem for MESSAGE too?

>
> 2. We already have the conference event package for obtaining  
> conference
> membership details. In many situations the conference focus is also  
> the
> source of such notifications, so the subscribe-initiated dialog
> parallels the INVITE-initiate dialog, which is rather wasteful. It  
> would
> be great to subscribe as part of the INVITE transaction and receive
> notifications along the INVITE-initiated dialog.

I like this one a lot. It could rip a LOT of the overhead out of  
conference setup, especially for conferences with restricted  
functionality.

A variant of this is synchronized browsing within the conference --  
INFO messages that convey the current URL that should be rendered.

>
> 3. Concerning mobility, I like the suggestion in 6.8 (geo-location
> information). Also, there may sometimes be intermediaries on the
> signalling path (B2BUAs) that are involved in mobility in some way,  
> and
> these might be interested in movement-related notifications (e.g., a
> dual mode device moving in or out of WLAN range).
>
> 4. The use of offer-answer to achieve hold-retrieve fails to give an
> explicit indication of what is happening. An explicit indication of
> things like hold, retrieve, microphone mute, speaker mute, camera
> on/off, recorder on/off, etc. might be quite helpful. Of course, there
> will normally be an offer-answer exchange at the same time, so  
> inclusion
> of the notification in a re-INVITE or UPDATE request might be more
> useful than using INFO.

I suspect adding something to the reINVITE to convey explicit  
semantics would be much better than an INFO here.

>
> 5. Concerning 6.7 (soft-key labels), a problem here is that the
> notifying UA might not have knowledge of soft key availability at the
> receiving UA. In certain closed environments (e.g., involving a UA and
> its local voicemail server) it may work, but I am not sure it would  
> work
> well in general. Having said that, where it does work, perhaps other
> parameters at the device can be controlled in this way, e.g.,  
> microphone
> and speaker volume (especially if this control is based on say a web
> interface, so the user is really controlling his/her own settings from
> the web page rather than directly on the device).

A soft-key INFO package would need to describe availability. This is  
a fundamental problem from KPML, and to the extent it is solvable by  
KPML in a SUBSCRIBE dialog, it should be solvable in an INVITE dialog.

>
> 6. When a B2BUA performs 3PCC there is plenty of scope for the UAs to
> lose track of what is happening. For example, when a B2BUA holds,
> retrieves, transfers calls, joins calls, pre-empts calls, etc.,  
> there is
> currently no explicit information - only perhaps a re-INVITE request.
> Thus it can be difficult to know how to interpret things like  
> identity.
> Say a UA receives a re-INVITE request with a new PAI, it might deduce
> that something like call transfer has occurred. However, if it  
> receives
> a re-INVITE request without a PAI, does it mean the previous PAI is
> still valid, or does it mean, following transfer, there is no  
> longer any
> identity available? Of course, INFO might not always be the most
> appropriate method to use for such notification if an offer-answer
> exchange is needed at the same time.

I'm thinking this is totally NOT an INFO problem. If it needs to be  
solved, it needs to happen in the 3PCC reINVITE sequence.

>
> 7. There might be some usages to do with interworking with TDM, e.g.,
> advice of charge during a call, user-to-user information during a  
> call.
>

Clearly, all the old TDM interworking mid-call events work better  
with the new INFO. Advice-of-charge would appear to be easily solved  
with INFO, except that some use cases for it require delivering the  
final AOC after the call has completed. There may some flexibility on  
this, of course, but a dialog-correlated MESSAGE to the GRUU may be  
cleaner.

--
Dean




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From sip-bounces@ietf.org Wed Jan 09 14:35:13 2008
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Thread-Topic: Question on Accept ABNF
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	<sip@ietf.org>
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Subject: [Sip] Question on Accept ABNF
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Hi,
=20
I appologize if the following issue has already been dealt with.
=20
In RFC3261 the ABNF for the Accept header is the following:
=20
Accept              =3D  "Accept" HCOLON[ accept-range *(COMMA =
accept-range) ]
accept-range    =3D  media-range *(SEMI accept-param)
media-range     =3D  ( "*/*" / ( m-type SLASH "*" )  / ( m-type SLASH =
m-subtype )) *( SEMI m-parameter )
accept-param   =3D  ("q" EQUAL qvalue) / generic-param
qvalue               =3D  ( "0" [ "." 0*3DIGIT ] ) / ( "1" [ "." =
0*3("0") ] )
generic-param  =3D  token [ EQUAL gen-value ]
gen-value         =3D  token / host / quoted-string
...
m-parameter     =3D  m-attribute EQUAL m-value
m-attribute       =3D  token
m-value            =3D  token / quoted-string

Now, when I receive: Accept: xxxx/yyyy;myParam=3DisThis

...how does my parser know whether "myParam" is a m-parameter, or an =
accept-param?

=20
In RFC2616 (HTTP) it is solved by specifying that the first accept-param =
MUST be a "q" value, so unless you also have a "q" m-parameter you know =
that anything that comes before "q" is m-parameter, and the rest =
(including "q" is accept-param):
=20
accept-params  =3D ";" "q" "=3D" qvalue *( accept-extension )

RFC2616 also says:
=20
"Each media-range MAY be followed by one or more accept-params, =
beginning with the "q" parameter for indicating a relative quality
factor. The first "q" parameter (if any) separates the media-range =
parameter(s) from the accept-params."
=20
...and:
=20
"Note: Use of the "q" parameter name to separate media type
 parameters from Accept extension parameters is due to historical
 practice. Although this prevents any media type parameter named
 "q" from being used with a media range, such an event is believed
 to be unlikely given the lack of any "q" parameters in the IANA
 media type registry and the rare usage of any media type
 parameters in Accept. Future media types are discouraged from
 registering any parameter named "q"."
=20
=20
Now, RFC3261 DOES say that Accept follows the syntax defined in RFC2616, =
but then it is confusing that the 3261 Accept ABNF is not identical.
=20
=20
Regards,
=20
Christer





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From sip-bounces@ietf.org Wed Jan 09 15:44:46 2008
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Thread-Topic: draft-rosenberg-sip-ua-loose-route-01: lr impacts
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From: "Brett Tate" <brett@broadsoft.com>
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	<jdrosen@cisco.com>
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Subject: [Sip] draft-rosenberg-sip-ua-loose-route-01: lr impacts
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Hi Jonathan,

Draft-rosenberg-sip-ua-loose-route-01 proposes the use of 'lr' parameter
within registered and redirected contact URIs.  The 'lr' is currently
not listed as optional within RFC3261.  Thus the additional impact upon
registrars not supporting this draft should maybe be reflected with the
Backwards Compatibility section.

Concerning the use of 'lr' within registered contact, what contact
(contact-param) should be placed within the Route?  More specifically,
should it be the one received within REGISTER or the one sent within the
REGISTER response?  I ask because there are Contact header parameters
(contact-params) which can be added or modified by the registrar.

Concerning the use of 'lr' within registered contact, the draft should
mention what should occur for contact-params applicable to registration
but not relevant (or would appear strange) within Route.  The 'q' and
'expires' parameters are some which fall into this category.  Section 10
currently shows none of the contact-params being copied to the rr-param
portion of the route-param.  Was this intentional or an error within the
example?

There are also sip-uri parameters which are valid within a Contact but
not currently optional to include within Route.  And some of these like
'ttl' might also be valid within a request-uri.  I thought that I'd
mention it incase you think it warrants discussion within the draft.

Thanks,
Brett


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Date: Wed, 09 Jan 2008 20:50:45 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] draft-kaplan-sip-info-use-cases-00 - more use cases?
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Dean,

Thanks for your comments. More below.

John=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 09 January 2008 19:26
> To: Elwell, John
> Cc: Hadriel Kaplan; IETF SIP List
> Subject: Re: [Sip] draft-kaplan-sip-info-use-cases-00 - more=20
> use cases?
>=20
>=20
> On Jan 9, 2008, at 3:58 AM, Elwell, John wrote:
>=20
> > Hadriel,
> >
> > Some possible further use cases:
> >
> > 1. Along the same lines as 6.2/6.3/6.4 (vcard, user-icon, =20
> > vcalendar), we
> > could also consider pictures, movie clips, audio clips, etc.. Of =20
> > course,
> > we don't want to encourage sending anything too big along the =20
> > signalling
> > path, so we would need to consider where to draw the line.
>=20
> So, how is this different from MESSAGE with the "gruu" as the=20
> target? =20
> Does it gain anything from being in-dialog? Does it gain from the =20
> specific content-type and usage negotiation? If so, does this=20
> mean we =20
> should fix this problem for MESSAGE too?
[JRE] Not very different, except that it is correlated to the dialog
without having to do extra work such as using Target-Dialog. Whether
dependence on GRUU is seen as a significant problem I am not sure.

>=20
> >
> > 2. We already have the conference event package for obtaining =20
> > conference
> > membership details. In many situations the conference focus=20
> is also =20
> > the
> > source of such notifications, so the subscribe-initiated dialog
> > parallels the INVITE-initiate dialog, which is rather wasteful. It =20
> > would
> > be great to subscribe as part of the INVITE transaction and receive
> > notifications along the INVITE-initiated dialog.
>=20
> I like this one a lot. It could rip a LOT of the overhead out of =20
> conference setup, especially for conferences with restricted =20
> functionality.
>=20
> A variant of this is synchronized browsing within the conference -- =20
> INFO messages that convey the current URL that should be rendered.
[JRE] Another good one.

>=20
> >
> > 3. Concerning mobility, I like the suggestion in 6.8 (geo-location
> > information). Also, there may sometimes be intermediaries on the
> > signalling path (B2BUAs) that are involved in mobility in=20
> some way, =20
> > and
> > these might be interested in movement-related notifications (e.g., a
> > dual mode device moving in or out of WLAN range).
> >
> > 4. The use of offer-answer to achieve hold-retrieve fails to give an
> > explicit indication of what is happening. An explicit indication of
> > things like hold, retrieve, microphone mute, speaker mute, camera
> > on/off, recorder on/off, etc. might be quite helpful. Of=20
> course, there
> > will normally be an offer-answer exchange at the same time, so =20
> > inclusion
> > of the notification in a re-INVITE or UPDATE request might be more
> > useful than using INFO.
>=20
> I suspect adding something to the reINVITE to convey explicit =20
> semantics would be much better than an INFO here.
[JRE] Yes. I guess what I am driving at is that if we had a
INVITE-initiated dialog notification mechanism, it should not be
restricted to INFO, but should be applicable to other mid-dialog methods
where appropriate, e.g., re-INVITE, UPDATE, BYE.

>=20
> >
> > 5. Concerning 6.7 (soft-key labels), a problem here is that the
> > notifying UA might not have knowledge of soft key=20
> availability at the
> > receiving UA. In certain closed environments (e.g.,=20
> involving a UA and
> > its local voicemail server) it may work, but I am not sure=20
> it would =20
> > work
> > well in general. Having said that, where it does work, perhaps other
> > parameters at the device can be controlled in this way, e.g., =20
> > microphone
> > and speaker volume (especially if this control is based on say a web
> > interface, so the user is really controlling his/her own=20
> settings from
> > the web page rather than directly on the device).
>=20
> A soft-key INFO package would need to describe availability. This is =20
> a fundamental problem from KPML, and to the extent it is solvable by =20
> KPML in a SUBSCRIBE dialog, it should be solvable in an INVITE dialog.
>=20
> >
> > 6. When a B2BUA performs 3PCC there is plenty of scope for=20
> the UAs to
> > lose track of what is happening. For example, when a B2BUA holds,
> > retrieves, transfers calls, joins calls, pre-empts calls, etc., =20
> > there is
> > currently no explicit information - only perhaps a=20
> re-INVITE request.
> > Thus it can be difficult to know how to interpret things like =20
> > identity.
> > Say a UA receives a re-INVITE request with a new PAI, it=20
> might deduce
> > that something like call transfer has occurred. However, if it =20
> > receives
> > a re-INVITE request without a PAI, does it mean the previous PAI is
> > still valid, or does it mean, following transfer, there is no =20
> > longer any
> > identity available? Of course, INFO might not always be the most
> > appropriate method to use for such notification if an offer-answer
> > exchange is needed at the same time.
>=20
> I'm thinking this is totally NOT an INFO problem. If it needs to be =20
> solved, it needs to happen in the 3PCC reINVITE sequence.
[JRE] In general, yes. I am not sure whether there are any exceptions
where INFO might still fit.

>=20
> >
> > 7. There might be some usages to do with interworking with=20
> TDM, e.g.,
> > advice of charge during a call, user-to-user information during a =20
> > call.
> >
>=20
> Clearly, all the old TDM interworking mid-call events work better =20
> with the new INFO. Advice-of-charge would appear to be easily solved =20
> with INFO, except that some use cases for it require delivering the =20
> final AOC after the call has completed. There may some=20
> flexibility on =20
> this, of course, but a dialog-correlated MESSAGE to the GRUU may be =20
> cleaner.
>=20
> --
> Dean
>=20
>=20
>=20


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Date: Thu, 10 Jan 2008 09:46:17 +0530
From: Mahesh Anjanappa <mahesha@samsung.com>
Subject: Re: [Sip] Question on RFC3840/3841
To: Paul Kyzivat <pkyzivat@cisco.com>
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hi Paul,
Sorry there is a mistake in Contact header i described.Following is what i 
wanted to:
Contact:sip:alice@atlanta.com;
Contact:audio;duplex=full;
Contact:video;duplex=recieve-only;

Thanks
Mahesh
----- Original Message ----- 
From: "Mahesh Anjanappa" <mahesha@samsung.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: <sip@ietf.org>; "Christer Holmberg" <christer.holmberg@ericsson.com>
Sent: Wednesday, January 09, 2008 10:48 PM
Subject: Re: [Sip] Question on RFC3840/3841


> Thanks Paul.
> So would it be right if i have a Contact Headers describing Duplex Audio & 
> one-way Video Capabilities as follows?
>
>    Contact:sip:alice@atlanta.com;
>    Contact:sip:audio;duplex=full;
>    Contact:sip:video;duplex=recieve-only;
>
> NOTE: The Predicate(for above) for which i'm attempting is
> & ( & (sip.audio) (sip.duplex=full) )
>    (  & (sip.video) (sip.duplex=receive-only) )
>
> regards
> Mahesh
>
> ----- Original Message ----- 
> From: "Paul Kyzivat" <pkyzivat@cisco.com>
> To: "Mahesh Anjanappa" <mahesha@samsung.com>
> Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>; <sip@ietf.org>
> Sent: Wednesday, January 09, 2008 7:55 PM
> Subject: Re: [Sip] Question on RFC3840/3841
>
>
>>
>>
>> Mahesh Anjanappa wrote:
>>> hi
>>> That is my understanding as well. But if that is the case then it should 
>>> apply to Contact header as well
>>> when describing Capabilities using Feature parameters in Contact Header.
>>> But i'm not sure if it applies the same way. I had posted this question 
>>> long ago but didnt get any comments on it.
>>
>> The rule forbidding multiple instances of a parameter applies to all 
>> headers, so it applies to Contact.  E.g. the following is illegal:
>>
>> Contact:sip:alice@atlanta.com;methods="INVITE";methods="CANCEL"
>>
>> Paul
>
>
>
> _______________________________________________
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Subject: RE: [Sip] Question on RFC3840/3841
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Mahesh Anjanappa" <mahesha@samsung.com>,
	"Paul Kyzivat" <pkyzivat@cisco.com>
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Shouldn't each Contact also contain "alice@atlanta.com"?

Also, why do you have ";" at the end of each Contact?

Regards,

Christer=20



> -----Original Message-----
> From: Mahesh Anjanappa [mailto:mahesha@samsung.com]=20
> Sent: 10. tammikuuta 2008 6:16
> To: Paul Kyzivat
> Cc: sip@ietf.org; Christer Holmberg
> Subject: Re: [Sip] Question on RFC3840/3841
>=20
> hi Paul,
> Sorry there is a mistake in Contact header i=20
> described.Following is what i wanted to:
> Contact:sip:alice@atlanta.com;
> Contact:audio;duplex=3Dfull;
> Contact:video;duplex=3Drecieve-only;
>=20
> Thanks
> Mahesh
> ----- Original Message -----
> From: "Mahesh Anjanappa" <mahesha@samsung.com>
> To: "Paul Kyzivat" <pkyzivat@cisco.com>
> Cc: <sip@ietf.org>; "Christer Holmberg"=20
> <christer.holmberg@ericsson.com>
> Sent: Wednesday, January 09, 2008 10:48 PM
> Subject: Re: [Sip] Question on RFC3840/3841
>=20
>=20
> > Thanks Paul.
> > So would it be right if i have a Contact Headers describing=20
> Duplex Audio &=20
> > one-way Video Capabilities as follows?
> >
> >    Contact:sip:alice@atlanta.com;
> >    Contact:sip:audio;duplex=3Dfull;
> >    Contact:sip:video;duplex=3Drecieve-only;
> >
> > NOTE: The Predicate(for above) for which i'm attempting is
> > & ( & (sip.audio) (sip.duplex=3Dfull) )
> >    (  & (sip.video) (sip.duplex=3Dreceive-only) )
> >
> > regards
> > Mahesh
> >
> > ----- Original Message -----=20
> > From: "Paul Kyzivat" <pkyzivat@cisco.com>
> > To: "Mahesh Anjanappa" <mahesha@samsung.com>
> > Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>;=20
> <sip@ietf.org>
> > Sent: Wednesday, January 09, 2008 7:55 PM
> > Subject: Re: [Sip] Question on RFC3840/3841
> >
> >
> >>
> >>
> >> Mahesh Anjanappa wrote:
> >>> hi
> >>> That is my understanding as well. But if that is the case=20
> then it should=20
> >>> apply to Contact header as well
> >>> when describing Capabilities using Feature parameters in=20
> Contact Header.
> >>> But i'm not sure if it applies the same way. I had posted=20
> this question=20
> >>> long ago but didnt get any comments on it.
> >>
> >> The rule forbidding multiple instances of a parameter=20
> applies to all=20
> >> headers, so it applies to Contact.  E.g. the following is illegal:
> >>
> >> Contact:sip:alice@atlanta.com;methods=3D"INVITE";methods=3D"CANCEL"
> >>
> >> Paul
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip=20
>=20
>=20


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



From sip-bounces@ietf.org Thu Jan 10 07:41:35 2008
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Date: Thu, 10 Jan 2008 18:13:06 +0530
From: Mahesh Anjanappa <mahesha@samsung.com>
Subject: Re: [Sip] Question on RFC3840/3841
To: Christer Holmberg <christer.holmberg@ericsson.com>,
	Paul Kyzivat <pkyzivat@cisco.com>
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Yes the ';' is a mistake. But i'm not sure if  each Contact should contain 
"alice@atlanta.com"
because the predicate that i'm trying to represent is as follows meanig to 
say
alice supports a communication which whose capability is Duplex Audio + One 
Way Video.
If each of the COntact contains alice@atlanta.com wouldnt it be interpreted 
as " alice
supports duplex audio and oneway video separatley and not both at the same 
time".
Actually i'm not quite sure how to represesnt the predicate in the Contact 
header correctly, hence
my Queries. It all stems from the statement which you too noted " There MUST 
only be one instance of any
feature tag in feature-param." in RFC 3840


Predicate:
------------
 & ( & (sip.audio) (sip.duplex=full) )
    (  & (sip.video) (sip.duplex=receive-only) )

regards
Mahesh
----- Original Message ----- 
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Mahesh Anjanappa" <mahesha@samsung.com>; "Paul Kyzivat" 
<pkyzivat@cisco.com>
Cc: <sip@ietf.org>
Sent: Thursday, January 10, 2008 11:30 AM
Subject: RE: [Sip] Question on RFC3840/3841



Shouldn't each Contact also contain "alice@atlanta.com"?

Also, why do you have ";" at the end of each Contact?

Regards,

Christer



> -----Original Message-----
> From: Mahesh Anjanappa [mailto:mahesha@samsung.com]
> Sent: 10. tammikuuta 2008 6:16
> To: Paul Kyzivat
> Cc: sip@ietf.org; Christer Holmberg
> Subject: Re: [Sip] Question on RFC3840/3841
>
> hi Paul,
> Sorry there is a mistake in Contact header i
> described.Following is what i wanted to:
> Contact:sip:alice@atlanta.com;
> Contact:audio;duplex=full;
> Contact:video;duplex=recieve-only;
>
> Thanks
> Mahesh
> ----- Original Message -----
> From: "Mahesh Anjanappa" <mahesha@samsung.com>
> To: "Paul Kyzivat" <pkyzivat@cisco.com>
> Cc: <sip@ietf.org>; "Christer Holmberg"
> <christer.holmberg@ericsson.com>
> Sent: Wednesday, January 09, 2008 10:48 PM
> Subject: Re: [Sip] Question on RFC3840/3841
>
>
> > Thanks Paul.
> > So would it be right if i have a Contact Headers describing
> Duplex Audio &
> > one-way Video Capabilities as follows?
> >
> >    Contact:sip:alice@atlanta.com;
> >    Contact:sip:audio;duplex=full;
> >    Contact:sip:video;duplex=recieve-only;
> >
> > NOTE: The Predicate(for above) for which i'm attempting is
> > & ( & (sip.audio) (sip.duplex=full) )
> >    (  & (sip.video) (sip.duplex=receive-only) )
> >
> > regards
> > Mahesh
> >
> > ----- Original Message ----- 
> > From: "Paul Kyzivat" <pkyzivat@cisco.com>
> > To: "Mahesh Anjanappa" <mahesha@samsung.com>
> > Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>;
> <sip@ietf.org>
> > Sent: Wednesday, January 09, 2008 7:55 PM
> > Subject: Re: [Sip] Question on RFC3840/3841
> >
> >
> >>
> >>
> >> Mahesh Anjanappa wrote:
> >>> hi
> >>> That is my understanding as well. But if that is the case
> then it should
> >>> apply to Contact header as well
> >>> when describing Capabilities using Feature parameters in
> Contact Header.
> >>> But i'm not sure if it applies the same way. I had posted
> this question
> >>> long ago but didnt get any comments on it.
> >>
> >> The rule forbidding multiple instances of a parameter
> applies to all
> >> headers, so it applies to Contact.  E.g. the following is illegal:
> >>
> >> Contact:sip:alice@atlanta.com;methods="INVITE";methods="CANCEL"
> >>
> >> Paul
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP 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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Mahesh Anjanappa wrote:
> Yes the ';' is a mistake. But i'm not sure if  each Contact should 
> contain "alice@atlanta.com"

Well, it has to have a URI to be syntactically correct.

Your problem is that you are trying to express something that cannot be 
expressed with the callee caps syntax. It only maps to a subset of the 
syntax supported by RFC 2533.

> because the predicate that i'm trying to represent is as follows meanig 
> to say
> alice supports a communication which whose capability is Duplex Audio + 
> One Way Video.

You just can't say that. Sorry.

A little history may help you to understand, though it won't solve your 
problem:

The initial draft for callerprefs came out a very long time ago. (The 
earliest draft I have is -03 which came out in november of 2000.) It set 
the basic syntax and the initial set of capabilities.

It was however very weak on the semantics of matching. I eventually got 
involved because I was interested in it and identified many of the 
semantic issues. Jonathan and I worked on it for a long time. 
Eventually, because of some review comments I think, Jonathan tried to 
define the semantics using RFC 2533 because it was a standard that 
seemed to be at least partially suitable to the task. (In retrospect 
this may not have been such a good idea, since it adds an additional 
layer of complexity.)

RFC 2533 syntax is much richer. But it was not a goal to provide that 
richness. (Mapping that all onto SIP header syntax would be awful.) It 
was only used as an intermediate form to define the semantics of the SIP 
headers.

Its understood that there are lots of things one might want to express 
that you can't express.

If you aren't already aware of it, you might want to look at RFC 4596 
(Guidelines for Usage of the Session Initiation Protocol (SIP) Caller 
Preferences Extension).

	Paul

> If each of the COntact contains alice@atlanta.com wouldnt it be 
> interpreted as " alice
> supports duplex audio and oneway video separatley and not both at the 
> same time".
> Actually i'm not quite sure how to represesnt the predicate in the 
> Contact header correctly, hence
> my Queries. It all stems from the statement which you too noted " There 
> MUST only be one instance of any
> feature tag in feature-param." in RFC 3840
> 
> 
> Predicate:
> ------------
> & ( & (sip.audio) (sip.duplex=full) )
>    (  & (sip.video) (sip.duplex=receive-only) )
> 
> regards
> Mahesh
> ----- Original Message ----- From: "Christer Holmberg" 
> <christer.holmberg@ericsson.com>
> To: "Mahesh Anjanappa" <mahesha@samsung.com>; "Paul Kyzivat" 
> <pkyzivat@cisco.com>
> Cc: <sip@ietf.org>
> Sent: Thursday, January 10, 2008 11:30 AM
> Subject: RE: [Sip] Question on RFC3840/3841
> 
> 
> 
> Shouldn't each Contact also contain "alice@atlanta.com"?
> 
> Also, why do you have ";" at the end of each Contact?
> 
> Regards,
> 
> Christer
> 
> 
> 
>> -----Original Message-----
>> From: Mahesh Anjanappa [mailto:mahesha@samsung.com]
>> Sent: 10. tammikuuta 2008 6:16
>> To: Paul Kyzivat
>> Cc: sip@ietf.org; Christer Holmberg
>> Subject: Re: [Sip] Question on RFC3840/3841
>>
>> hi Paul,
>> Sorry there is a mistake in Contact header i
>> described.Following is what i wanted to:
>> Contact:sip:alice@atlanta.com;
>> Contact:audio;duplex=full;
>> Contact:video;duplex=recieve-only;
>>
>> Thanks
>> Mahesh
>> ----- Original Message -----
>> From: "Mahesh Anjanappa" <mahesha@samsung.com>
>> To: "Paul Kyzivat" <pkyzivat@cisco.com>
>> Cc: <sip@ietf.org>; "Christer Holmberg"
>> <christer.holmberg@ericsson.com>
>> Sent: Wednesday, January 09, 2008 10:48 PM
>> Subject: Re: [Sip] Question on RFC3840/3841
>>
>>
>> > Thanks Paul.
>> > So would it be right if i have a Contact Headers describing
>> Duplex Audio &
>> > one-way Video Capabilities as follows?
>> >
>> >    Contact:sip:alice@atlanta.com;
>> >    Contact:sip:audio;duplex=full;
>> >    Contact:sip:video;duplex=recieve-only;
>> >
>> > NOTE: The Predicate(for above) for which i'm attempting is
>> > & ( & (sip.audio) (sip.duplex=full) )
>> >    (  & (sip.video) (sip.duplex=receive-only) )
>> >
>> > regards
>> > Mahesh
>> >
>> > ----- Original Message ----- > From: "Paul Kyzivat" 
>> <pkyzivat@cisco.com>
>> > To: "Mahesh Anjanappa" <mahesha@samsung.com>
>> > Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>;
>> <sip@ietf.org>
>> > Sent: Wednesday, January 09, 2008 7:55 PM
>> > Subject: Re: [Sip] Question on RFC3840/3841
>> >
>> >
>> >>
>> >>
>> >> Mahesh Anjanappa wrote:
>> >>> hi
>> >>> That is my understanding as well. But if that is the case
>> then it should
>> >>> apply to Contact header as well
>> >>> when describing Capabilities using Feature parameters in
>> Contact Header.
>> >>> But i'm not sure if it applies the same way. I had posted
>> this question
>> >>> long ago but didnt get any comments on it.
>> >>
>> >> The rule forbidding multiple instances of a parameter
>> applies to all
>> >> headers, so it applies to Contact.  E.g. the following is illegal:
>> >>
>> >> Contact:sip:alice@atlanta.com;methods="INVITE";methods="CANCEL"
>> >>
>> >> Paul
>> >
>> >
>> >
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP 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 10 09:42:52 2008
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	"Mahesh Anjanappa" <mahesha@samsung.com>
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Paul,

Regarding the matching, are you saying that reading RFC3840 is enough to
understand how the matching is done in a "SIP environment"? Or, do I
also need to read RFC2533?

I have some questions on the matching, but I will come back to them
later. I still have some reading to do.

Regards,

Christer

=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: 10. tammikuuta 2008 16:21
> To: Mahesh Anjanappa
> Cc: Christer Holmberg; sip@ietf.org
> Subject: Re: [Sip] Question on RFC3840/3841
>=20
>=20
>=20
> Mahesh Anjanappa wrote:
> > Yes the ';' is a mistake. But i'm not sure if  each Contact should=20
> > contain "alice@atlanta.com"
>=20
> Well, it has to have a URI to be syntactically correct.
>=20
> Your problem is that you are trying to express something that=20
> cannot be expressed with the callee caps syntax. It only maps=20
> to a subset of the syntax supported by RFC 2533.
>=20
> > because the predicate that i'm trying to represent is as follows=20
> > meanig to say alice supports a communication which whose=20
> capability is=20
> > Duplex Audio + One Way Video.
>=20
> You just can't say that. Sorry.
>=20
> A little history may help you to understand, though it won't=20
> solve your
> problem:
>=20
> The initial draft for callerprefs came out a very long time=20
> ago. (The earliest draft I have is -03 which came out in=20
> november of 2000.) It set the basic syntax and the initial=20
> set of capabilities.
>=20
> It was however very weak on the semantics of matching. I=20
> eventually got involved because I was interested in it and=20
> identified many of the semantic issues. Jonathan and I worked=20
> on it for a long time.=20
> Eventually, because of some review comments I think, Jonathan=20
> tried to define the semantics using RFC 2533 because it was a=20
> standard that seemed to be at least partially suitable to the=20
> task. (In retrospect this may not have been such a good idea,=20
> since it adds an additional layer of complexity.)
>=20
> RFC 2533 syntax is much richer. But it was not a goal to=20
> provide that richness. (Mapping that all onto SIP header=20
> syntax would be awful.) It was only used as an intermediate=20
> form to define the semantics of the SIP headers.
>=20
> Its understood that there are lots of things one might want=20
> to express that you can't express.
>=20
> If you aren't already aware of it, you might want to look at=20
> RFC 4596 (Guidelines for Usage of the Session Initiation=20
> Protocol (SIP) Caller Preferences Extension).
>=20
> 	Paul
>=20
> > If each of the COntact contains alice@atlanta.com wouldnt it be=20
> > interpreted as " alice supports duplex audio and oneway video=20
> > separatley and not both at the same time".
> > Actually i'm not quite sure how to represesnt the predicate in the=20
> > Contact header correctly, hence my Queries. It all stems from the=20
> > statement which you too noted " There MUST only be one=20
> instance of any=20
> > feature tag in feature-param." in RFC 3840
> >=20
> >=20
> > Predicate:
> > ------------
> > & ( & (sip.audio) (sip.duplex=3Dfull) )
> >    (  & (sip.video) (sip.duplex=3Dreceive-only) )
> >=20
> > regards
> > Mahesh
> > ----- Original Message ----- From: "Christer Holmberg"=20
> > <christer.holmberg@ericsson.com>
> > To: "Mahesh Anjanappa" <mahesha@samsung.com>; "Paul Kyzivat"=20
> > <pkyzivat@cisco.com>
> > Cc: <sip@ietf.org>
> > Sent: Thursday, January 10, 2008 11:30 AM
> > Subject: RE: [Sip] Question on RFC3840/3841
> >=20
> >=20
> >=20
> > Shouldn't each Contact also contain "alice@atlanta.com"?
> >=20
> > Also, why do you have ";" at the end of each Contact?
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >> -----Original Message-----
> >> From: Mahesh Anjanappa [mailto:mahesha@samsung.com]
> >> Sent: 10. tammikuuta 2008 6:16
> >> To: Paul Kyzivat
> >> Cc: sip@ietf.org; Christer Holmberg
> >> Subject: Re: [Sip] Question on RFC3840/3841
> >>
> >> hi Paul,
> >> Sorry there is a mistake in Contact header i
> >> described.Following is what i wanted to:
> >> Contact:sip:alice@atlanta.com;
> >> Contact:audio;duplex=3Dfull;
> >> Contact:video;duplex=3Drecieve-only;
> >>
> >> Thanks
> >> Mahesh
> >> ----- Original Message -----
> >> From: "Mahesh Anjanappa" <mahesha@samsung.com>
> >> To: "Paul Kyzivat" <pkyzivat@cisco.com>
> >> Cc: <sip@ietf.org>; "Christer Holmberg"
> >> <christer.holmberg@ericsson.com>
> >> Sent: Wednesday, January 09, 2008 10:48 PM
> >> Subject: Re: [Sip] Question on RFC3840/3841
> >>
> >>
> >> > Thanks Paul.
> >> > So would it be right if i have a Contact Headers describing
> >> Duplex Audio &
> >> > one-way Video Capabilities as follows?
> >> >
> >> >    Contact:sip:alice@atlanta.com;
> >> >    Contact:sip:audio;duplex=3Dfull;
> >> >    Contact:sip:video;duplex=3Drecieve-only;
> >> >
> >> > NOTE: The Predicate(for above) for which i'm attempting is
> >> > & ( & (sip.audio) (sip.duplex=3Dfull) )
> >> >    (  & (sip.video) (sip.duplex=3Dreceive-only) )
> >> >
> >> > regards
> >> > Mahesh
> >> >
> >> > ----- Original Message ----- > From: "Paul Kyzivat"=20
> >> <pkyzivat@cisco.com>
> >> > To: "Mahesh Anjanappa" <mahesha@samsung.com>
> >> > Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>;
> >> <sip@ietf.org>
> >> > Sent: Wednesday, January 09, 2008 7:55 PM
> >> > Subject: Re: [Sip] Question on RFC3840/3841
> >> >
> >> >
> >> >>
> >> >>
> >> >> Mahesh Anjanappa wrote:
> >> >>> hi
> >> >>> That is my understanding as well. But if that is the case
> >> then it should
> >> >>> apply to Contact header as well
> >> >>> when describing Capabilities using Feature parameters in
> >> Contact Header.
> >> >>> But i'm not sure if it applies the same way. I had posted
> >> this question
> >> >>> long ago but didnt get any comments on it.
> >> >>
> >> >> The rule forbidding multiple instances of a parameter
> >> applies to all
> >> >> headers, so it applies to Contact.  E.g. the following=20
> is illegal:
> >> >>
> >> >> =
Contact:sip:alice@atlanta.com;methods=3D"INVITE";methods=3D"CANCEL"
> >> >>
> >> >> Paul
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> > This list is for NEW development of the core SIP Protocol
> >> > Use sip-implementors@cs.columbia.edu for questions on current sip
> >> > Use sipping@ietf.org for new developments on the=20
> application of sip
> >>
> >>
> >=20
> >=20
>=20


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



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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Question on RFC3840/3841
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Christer,

RFC3840 by itself, without 2533, is *not* sufficient to understand how 
to do the matching.

I put an appendix in RFC 4596 that gives an algorithm without reference 
to 2533. I *think* it is right, but of course it is not normative.

Note that IMO none of this is intuitive or straightforward. Every time I 
get a detailed question on it I have to do a bunch of review before I 
can answer it. (But maybe that's just because I don't have enough 
neurons left.)

	Paul

Christer Holmberg wrote:
> Paul,
> 
> Regarding the matching, are you saying that reading RFC3840 is enough to
> understand how the matching is done in a "SIP environment"? Or, do I
> also need to read RFC2533?
> 
> I have some questions on the matching, but I will come back to them
> later. I still have some reading to do.
> 
> Regards,
> 
> Christer
> 
>  
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: 10. tammikuuta 2008 16:21
>> To: Mahesh Anjanappa
>> Cc: Christer Holmberg; sip@ietf.org
>> Subject: Re: [Sip] Question on RFC3840/3841
>>
>>
>>
>> Mahesh Anjanappa wrote:
>>> Yes the ';' is a mistake. But i'm not sure if  each Contact should 
>>> contain "alice@atlanta.com"
>> Well, it has to have a URI to be syntactically correct.
>>
>> Your problem is that you are trying to express something that 
>> cannot be expressed with the callee caps syntax. It only maps 
>> to a subset of the syntax supported by RFC 2533.
>>
>>> because the predicate that i'm trying to represent is as follows 
>>> meanig to say alice supports a communication which whose 
>> capability is 
>>> Duplex Audio + One Way Video.
>> You just can't say that. Sorry.
>>
>> A little history may help you to understand, though it won't 
>> solve your
>> problem:
>>
>> The initial draft for callerprefs came out a very long time 
>> ago. (The earliest draft I have is -03 which came out in 
>> november of 2000.) It set the basic syntax and the initial 
>> set of capabilities.
>>
>> It was however very weak on the semantics of matching. I 
>> eventually got involved because I was interested in it and 
>> identified many of the semantic issues. Jonathan and I worked 
>> on it for a long time. 
>> Eventually, because of some review comments I think, Jonathan 
>> tried to define the semantics using RFC 2533 because it was a 
>> standard that seemed to be at least partially suitable to the 
>> task. (In retrospect this may not have been such a good idea, 
>> since it adds an additional layer of complexity.)
>>
>> RFC 2533 syntax is much richer. But it was not a goal to 
>> provide that richness. (Mapping that all onto SIP header 
>> syntax would be awful.) It was only used as an intermediate 
>> form to define the semantics of the SIP headers.
>>
>> Its understood that there are lots of things one might want 
>> to express that you can't express.
>>
>> If you aren't already aware of it, you might want to look at 
>> RFC 4596 (Guidelines for Usage of the Session Initiation 
>> Protocol (SIP) Caller Preferences Extension).
>>
>> 	Paul
>>
>>> If each of the COntact contains alice@atlanta.com wouldnt it be 
>>> interpreted as " alice supports duplex audio and oneway video 
>>> separatley and not both at the same time".
>>> Actually i'm not quite sure how to represesnt the predicate in the 
>>> Contact header correctly, hence my Queries. It all stems from the 
>>> statement which you too noted " There MUST only be one 
>> instance of any 
>>> feature tag in feature-param." in RFC 3840
>>>
>>>
>>> Predicate:
>>> ------------
>>> & ( & (sip.audio) (sip.duplex=full) )
>>>    (  & (sip.video) (sip.duplex=receive-only) )
>>>
>>> regards
>>> Mahesh
>>> ----- Original Message ----- From: "Christer Holmberg" 
>>> <christer.holmberg@ericsson.com>
>>> To: "Mahesh Anjanappa" <mahesha@samsung.com>; "Paul Kyzivat" 
>>> <pkyzivat@cisco.com>
>>> Cc: <sip@ietf.org>
>>> Sent: Thursday, January 10, 2008 11:30 AM
>>> Subject: RE: [Sip] Question on RFC3840/3841
>>>
>>>
>>>
>>> Shouldn't each Contact also contain "alice@atlanta.com"?
>>>
>>> Also, why do you have ";" at the end of each Contact?
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: Mahesh Anjanappa [mailto:mahesha@samsung.com]
>>>> Sent: 10. tammikuuta 2008 6:16
>>>> To: Paul Kyzivat
>>>> Cc: sip@ietf.org; Christer Holmberg
>>>> Subject: Re: [Sip] Question on RFC3840/3841
>>>>
>>>> hi Paul,
>>>> Sorry there is a mistake in Contact header i
>>>> described.Following is what i wanted to:
>>>> Contact:sip:alice@atlanta.com;
>>>> Contact:audio;duplex=full;
>>>> Contact:video;duplex=recieve-only;
>>>>
>>>> Thanks
>>>> Mahesh
>>>> ----- Original Message -----
>>>> From: "Mahesh Anjanappa" <mahesha@samsung.com>
>>>> To: "Paul Kyzivat" <pkyzivat@cisco.com>
>>>> Cc: <sip@ietf.org>; "Christer Holmberg"
>>>> <christer.holmberg@ericsson.com>
>>>> Sent: Wednesday, January 09, 2008 10:48 PM
>>>> Subject: Re: [Sip] Question on RFC3840/3841
>>>>
>>>>
>>>>> Thanks Paul.
>>>>> So would it be right if i have a Contact Headers describing
>>>> Duplex Audio &
>>>>> one-way Video Capabilities as follows?
>>>>>
>>>>>    Contact:sip:alice@atlanta.com;
>>>>>    Contact:sip:audio;duplex=full;
>>>>>    Contact:sip:video;duplex=recieve-only;
>>>>>
>>>>> NOTE: The Predicate(for above) for which i'm attempting is
>>>>> & ( & (sip.audio) (sip.duplex=full) )
>>>>>    (  & (sip.video) (sip.duplex=receive-only) )
>>>>>
>>>>> regards
>>>>> Mahesh
>>>>>
>>>>> ----- Original Message ----- > From: "Paul Kyzivat" 
>>>> <pkyzivat@cisco.com>
>>>>> To: "Mahesh Anjanappa" <mahesha@samsung.com>
>>>>> Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>;
>>>> <sip@ietf.org>
>>>>> Sent: Wednesday, January 09, 2008 7:55 PM
>>>>> Subject: Re: [Sip] Question on RFC3840/3841
>>>>>
>>>>>
>>>>>>
>>>>>> Mahesh Anjanappa wrote:
>>>>>>> hi
>>>>>>> That is my understanding as well. But if that is the case
>>>> then it should
>>>>>>> apply to Contact header as well
>>>>>>> when describing Capabilities using Feature parameters in
>>>> Contact Header.
>>>>>>> But i'm not sure if it applies the same way. I had posted
>>>> this question
>>>>>>> long ago but didnt get any comments on it.
>>>>>> The rule forbidding multiple instances of a parameter
>>>> applies to all
>>>>>> headers, so it applies to Contact.  E.g. the following 
>> is illegal:
>>>>>> Contact:sip:alice@atlanta.com;methods="INVITE";methods="CANCEL"
>>>>>>
>>>>>> Paul
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sipping@ietf.org for new developments on the 
>> application of sip
>>>>
>>>
> 


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



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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>,
	<sip@ietf.org>
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Hi,

We have submitted a draft with an alternative proposal.

It can also be found at:

http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-target-uri-del
ivery-00.txt

Regards,

Christer

=20

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Sent: 9. tammikuuta 2008 18:32
> To: sip@ietf.org
> Subject: [Sip] RE: Delivering request-URI and parameters to=20
> UAS via proxy
>=20
> A reminder of the deadline on the 11th January for submitting=20
> alternative proposals on the way forward.
>=20
> Regards
>=20
> Keith=20
>=20
> > -----Original Message-----
> > From: DRAGE, Keith (Keith)
> > Sent: Tuesday, December 04, 2007 3:27 PM
> > To: sip@ietf.org
> > Subject: Delivering request-URI and parameters to UAS via proxy
> >=20
> > (As WG chair)
> >=20
> > We have a couple of milestones that we generated as a result of=20
> > discussion of
> >=20
> > http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> > se-route-01.txt
> >=20
> > Dec 2007    Delivering request-URI and parameters to UAS via=20
> > proxy to WGLC =20
> > Feb 2008    Delivering request-URI and parameters to UAS via=20
> > proxy to IESG (PS)
> >=20
> > This work is currently stalled and the editor needs input.
> >=20
> > The document contains various example scenarios where a solution is=20
> > required, for which there appears to be no dispute that a=20
> solution is=20
> > needed.
> >=20
> > The document proposes one solution to resolve these example=20
> scenarios,=20
> > but this solution is not gaining consensus. At least one other=20
> > solution has been talked about, but there is no=20
> documentation on the=20
> > table.
> >=20
> > This mail is to identify a deadline for other solutions to=20
> the example=20
> > scenarios to be documented as internet drafts, showing how the=20
> > solution works for those scenarios. This deadline is:
> >=20
> > 	January 11th 2008
> >=20
> > It is appropriate fo these documents to identify any other=20
> scenarios=20
> > where such a solution is appropriate. Any other input is=20
> also welcome=20
> > in identifying other scenarios.
> >=20
> > If we have no such internet-drafts by this deadline, we=20
> will proceed=20
> > with completing the solution we have.
> >=20
> >=20
> > Regards
> >=20
> > Keith
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Date: Fri, 11 Jan 2008 10:28:30 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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	<CA9998CD4A020D418654FCDEF4E707DF040266B1@esealmw113.eemea.ericsson.se>
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Christer,

I have some questions and comments:

- I don't understand your examples in section 3. They are a bit sketchy 
about the assumptions they are making, and in notation. I get lost about 
which referenced component has which address, etc. I am far from 
convinced that these are problems with appropriate use of the mechanism.

- It seems from your analysis of use cases that it is P-Called-Party 
that solves many of them, not Target. So both headers seem to be part of 
the solution. Its not entirely clear to me at the moment whether the 
R-URI in the loose-route approach aligns with Target or P-Called-Party. 
Since they are different, it can't align with both. So there must be 
some features it doesn't cover. I haven't fully grokked that yet.

	Paul


Christer Holmberg wrote:
> Hi,
> 
> We have submitted a draft with an alternative proposal.
> 
> It can also be found at:
> 
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-target-uri-del
> ivery-00.txt
> 
> Regards,
> 
> Christer
> 
>  
> 
>> -----Original Message-----
>> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com] 
>> Sent: 9. tammikuuta 2008 18:32
>> To: sip@ietf.org
>> Subject: [Sip] RE: Delivering request-URI and parameters to 
>> UAS via proxy
>>
>> A reminder of the deadline on the 11th January for submitting 
>> alternative proposals on the way forward.
>>
>> Regards
>>
>> Keith 
>>
>>> -----Original Message-----
>>> From: DRAGE, Keith (Keith)
>>> Sent: Tuesday, December 04, 2007 3:27 PM
>>> To: sip@ietf.org
>>> Subject: Delivering request-URI and parameters to UAS via proxy
>>>
>>> (As WG chair)
>>>
>>> We have a couple of milestones that we generated as a result of 
>>> discussion of
>>>
>>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
>>> se-route-01.txt
>>>
>>> Dec 2007    Delivering request-URI and parameters to UAS via 
>>> proxy to WGLC  
>>> Feb 2008    Delivering request-URI and parameters to UAS via 
>>> proxy to IESG (PS)
>>>
>>> This work is currently stalled and the editor needs input.
>>>
>>> The document contains various example scenarios where a solution is 
>>> required, for which there appears to be no dispute that a 
>> solution is 
>>> needed.
>>>
>>> The document proposes one solution to resolve these example 
>> scenarios, 
>>> but this solution is not gaining consensus. At least one other 
>>> solution has been talked about, but there is no 
>> documentation on the 
>>> table.
>>>
>>> This mail is to identify a deadline for other solutions to 
>> the example 
>>> scenarios to be documented as internet drafts, showing how the 
>>> solution works for those scenarios. This deadline is:
>>>
>>> 	January 11th 2008
>>>
>>> It is appropriate fo these documents to identify any other 
>> scenarios 
>>> where such a solution is appropriate. Any other input is 
>> also welcome 
>>> in identifying other scenarios.
>>>
>>> If we have no such internet-drafts by this deadline, we 
>> will proceed 
>>> with completing the solution we have.
>>>
>>>
>>> Regards
>>>
>>> Keith
>>>
>>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP 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: Fri, 11 Jan 2008 17:01:21 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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Christer,

In addition to Paul's concerns, I am having difficulty understanding why
the To header field is insufficient for the cases where the Target
header field is proposed. The initial source of this confusion was the
statement "the Target header field represents
   the initial target identity that was used to initiate a session to
   the target"

So we now have 4 fields (ignoring Route):
- To
- Target
- P-Called-ID
- Request-URI
I think it would benefit from a clearer exposition of the semantics and
relationship between these.

John

=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: 11 January 2008 15:29
> To: Christer Holmberg
> Cc: sip@ietf.org; DRAGE, Keith (Keith)
> Subject: Re: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy
>=20
> Christer,
>=20
> I have some questions and comments:
>=20
> - I don't understand your examples in section 3. They are a=20
> bit sketchy=20
> about the assumptions they are making, and in notation. I get=20
> lost about=20
> which referenced component has which address, etc. I am far from=20
> convinced that these are problems with appropriate use of the=20
> mechanism.
>=20
> - It seems from your analysis of use cases that it is P-Called-Party=20
> that solves many of them, not Target. So both headers seem to=20
> be part of=20
> the solution. Its not entirely clear to me at the moment whether the=20
> R-URI in the loose-route approach aligns with Target or=20
> P-Called-Party.=20
> Since they are different, it can't align with both. So there must be=20
> some features it doesn't cover. I haven't fully grokked that yet.
>=20
> 	Paul
>=20
>=20
> Christer Holmberg wrote:
> > Hi,
> >=20
> > We have submitted a draft with an alternative proposal.
> >=20
> > It can also be found at:
> >=20
> >=20
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> et-uri-del
> > ivery-00.txt
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> > =20
> >=20
> >> -----Original Message-----
> >> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> >> Sent: 9. tammikuuta 2008 18:32
> >> To: sip@ietf.org
> >> Subject: [Sip] RE: Delivering request-URI and parameters to=20
> >> UAS via proxy
> >>
> >> A reminder of the deadline on the 11th January for submitting=20
> >> alternative proposals on the way forward.
> >>
> >> Regards
> >>
> >> Keith=20
> >>
> >>> -----Original Message-----
> >>> From: DRAGE, Keith (Keith)
> >>> Sent: Tuesday, December 04, 2007 3:27 PM
> >>> To: sip@ietf.org
> >>> Subject: Delivering request-URI and parameters to UAS via proxy
> >>>
> >>> (As WG chair)
> >>>
> >>> We have a couple of milestones that we generated as a result of=20
> >>> discussion of
> >>>
> >>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> >>> se-route-01.txt
> >>>
> >>> Dec 2007    Delivering request-URI and parameters to UAS via=20
> >>> proxy to WGLC =20
> >>> Feb 2008    Delivering request-URI and parameters to UAS via=20
> >>> proxy to IESG (PS)
> >>>
> >>> This work is currently stalled and the editor needs input.
> >>>
> >>> The document contains various example scenarios where a=20
> solution is=20
> >>> required, for which there appears to be no dispute that a=20
> >> solution is=20
> >>> needed.
> >>>
> >>> The document proposes one solution to resolve these example=20
> >> scenarios,=20
> >>> but this solution is not gaining consensus. At least one other=20
> >>> solution has been talked about, but there is no=20
> >> documentation on the=20
> >>> table.
> >>>
> >>> This mail is to identify a deadline for other solutions to=20
> >> the example=20
> >>> scenarios to be documented as internet drafts, showing how the=20
> >>> solution works for those scenarios. This deadline is:
> >>>
> >>> 	January 11th 2008
> >>>
> >>> It is appropriate fo these documents to identify any other=20
> >> scenarios=20
> >>> where such a solution is appropriate. Any other input is=20
> >> also welcome=20
> >>> in identifying other scenarios.
> >>>
> >>> If we have no such internet-drafts by this deadline, we=20
> >> will proceed=20
> >>> with completing the solution we have.
> >>>
> >>>
> >>> Regards
> >>>
> >>> Keith
> >>>
> >>>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip=20
> >> Use sipping@ietf.org for new developments on the application of sip
> >>
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
Date: Fri, 11 Jan 2008 12:58:04 -0600
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From: "Francois Audet" <audet@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
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	Keith \(Keith\)" <drage@alcatel-lucent.com>
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I'm also totally confused now about what is being proposed.

The Target seems identical to the To header field.

I tought that what you (Christer) was arguing for is using the =
P-Called-Header
instead of the new loose route proposal, i.e., not for the "initial =
target", but for
the "current target".

=20

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]=20
> Sent: Friday, January 11, 2008 09:01
> To: Paul Kyzivat; Christer Holmberg
> Cc: sip@ietf.org; DRAGE, Keith (Keith)
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy
>=20
> Christer,
>=20
> In addition to Paul's concerns, I am having difficulty=20
> understanding why the To header field is insufficient for the=20
> cases where the Target header field is proposed. The initial=20
> source of this confusion was the statement "the Target header=20
> field represents
>    the initial target identity that was used to initiate a session to
>    the target"
>=20
> So we now have 4 fields (ignoring Route):
> - To
> - Target
> - P-Called-ID
> - Request-URI
> I think it would benefit from a clearer exposition of the=20
> semantics and relationship between these.
>=20
> John
>=20
> =20
>=20
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > Sent: 11 January 2008 15:29
> > To: Christer Holmberg
> > Cc: sip@ietf.org; DRAGE, Keith (Keith)
> > Subject: Re: [Sip] RE: Delivering request-URI and parameters to UAS=20
> > via proxy
> >=20
> > Christer,
> >=20
> > I have some questions and comments:
> >=20
> > - I don't understand your examples in section 3. They are a bit=20
> > sketchy about the assumptions they are making, and in=20
> notation. I get=20
> > lost about which referenced component has which address,=20
> etc. I am far=20
> > from convinced that these are problems with appropriate use of the=20
> > mechanism.
> >=20
> > - It seems from your analysis of use cases that it is=20
> P-Called-Party=20
> > that solves many of them, not Target. So both headers seem=20
> to be part=20
> > of the solution. Its not entirely clear to me at the moment whether=20
> > the R-URI in the loose-route approach aligns with Target or=20
> > P-Called-Party.
> > Since they are different, it can't align with both. So=20
> there must be=20
> > some features it doesn't cover. I haven't fully grokked that yet.
> >=20
> > 	Paul
> >=20
> >=20
> > Christer Holmberg wrote:
> > > Hi,
> > >=20
> > > We have submitted a draft with an alternative proposal.
> > >=20
> > > It can also be found at:
> > >=20
> > >=20
> > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > et-uri-del
> > > ivery-00.txt
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > > =20
> > >=20
> > >> -----Original Message-----
> > >> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> > >> Sent: 9. tammikuuta 2008 18:32
> > >> To: sip@ietf.org
> > >> Subject: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via=20
> > >> proxy
> > >>
> > >> A reminder of the deadline on the 11th January for submitting=20
> > >> alternative proposals on the way forward.
> > >>
> > >> Regards
> > >>
> > >> Keith
> > >>
> > >>> -----Original Message-----
> > >>> From: DRAGE, Keith (Keith)
> > >>> Sent: Tuesday, December 04, 2007 3:27 PM
> > >>> To: sip@ietf.org
> > >>> Subject: Delivering request-URI and parameters to UAS via proxy
> > >>>
> > >>> (As WG chair)
> > >>>
> > >>> We have a couple of milestones that we generated as a result of=20
> > >>> discussion of
> > >>>
> > >>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> > >>> se-route-01.txt
> > >>>
> > >>> Dec 2007    Delivering request-URI and parameters to UAS via=20
> > >>> proxy to WGLC =20
> > >>> Feb 2008    Delivering request-URI and parameters to UAS via=20
> > >>> proxy to IESG (PS)
> > >>>
> > >>> This work is currently stalled and the editor needs input.
> > >>>
> > >>> The document contains various example scenarios where a
> > solution is
> > >>> required, for which there appears to be no dispute that a
> > >> solution is
> > >>> needed.
> > >>>
> > >>> The document proposes one solution to resolve these example
> > >> scenarios,
> > >>> but this solution is not gaining consensus. At least one other=20
> > >>> solution has been talked about, but there is no
> > >> documentation on the
> > >>> table.
> > >>>
> > >>> This mail is to identify a deadline for other solutions to
> > >> the example
> > >>> scenarios to be documented as internet drafts, showing how the=20
> > >>> solution works for those scenarios. This deadline is:
> > >>>
> > >>> 	January 11th 2008
> > >>>
> > >>> It is appropriate fo these documents to identify any other
> > >> scenarios
> > >>> where such a solution is appropriate. Any other input is
> > >> also welcome
> > >>> in identifying other scenarios.
> > >>>
> > >>> If we have no such internet-drafts by this deadline, we
> > >> will proceed
> > >>> with completing the solution we have.
> > >>>
> > >>>
> > >>> Regards
> > >>>
> > >>> Keith
> > >>>
> > >>>
> > >>
> > >> _______________________________________________
> > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >> This list is for NEW development of the core SIP Protocol Use=20
> > >> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > >> sipping@ietf.org for new developments on the application of sip
> > >>
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi Paul,=20

>I have some questions and comments:
>=20
>- I don't understand your examples in section 3. They are a=20
>bit sketchy about the assumptions they are making, and in=20
>notation. I get lost about which referenced component has=20
>which address, etc. I am far from convinced that these are=20
>problems with appropriate use of the mechanism.

As indicated in chapter 3, one of the main issues/limitations with the
J'draft solution is that the entity wanting to use it must have
knowledge whether the next hop also supports it. My understanding from
off-line discussions I had with Jonathan in Vancouver is also that
Jonathan agrees to that issue/limitation.

The purpose of the examples is just to show what can go wrong if the
next hop does not understand the mechanism. But, if we all agree to the
limitation with the J'draft solution I don't know whether we need to
spend too much time on the examples.

>- It seems from your analysis of use cases that it is=20
>P-Called-Party that solves many of them, not Target. So both=20
>headers seem to be part of the solution.

No, as far as the alternative to the J'draft solution is concerned, the
alternative is Target. We believe it can be used for all listed
use-cases.

Since I myself have been talking about P-CPI as an alternative, we
wanted to show in which cases it could be used as an alternative, and
why we think that P-CPI still would have to be used in IMS.=20

But, we have realized that P-CPI as defined and used today probably
would not solve all the use-cases, and that is the reason we came up
with the Target header.

>Its not entirely clear to me at the moment whether the R-URI in the=20
>loose-route approach aligns with Target or P-Called-Party.=20
>Since they are different, it can't align with both. So there=20
>must be some features it doesn't cover. I haven't fully=20
>grokked that yet.

I appologize if the P-CPI text causes confusion. Again, the alternative
to the J'draft solution we propose is Target.

Regards,

Christer








> Christer Holmberg wrote:
> > Hi,
> >=20
> > We have submitted a draft with an alternative proposal.
> >=20
> > It can also be found at:
> >=20
> >=20
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-target-uri-d
> > el
> > ivery-00.txt
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> > =20
> >=20
> >> -----Original Message-----
> >> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> >> Sent: 9. tammikuuta 2008 18:32
> >> To: sip@ietf.org
> >> Subject: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via=20
> >> proxy
> >>
> >> A reminder of the deadline on the 11th January for submitting=20
> >> alternative proposals on the way forward.
> >>
> >> Regards
> >>
> >> Keith
> >>
> >>> -----Original Message-----
> >>> From: DRAGE, Keith (Keith)
> >>> Sent: Tuesday, December 04, 2007 3:27 PM
> >>> To: sip@ietf.org
> >>> Subject: Delivering request-URI and parameters to UAS via proxy
> >>>
> >>> (As WG chair)
> >>>
> >>> We have a couple of milestones that we generated as a result of=20
> >>> discussion of
> >>>
> >>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> >>> se-route-01.txt
> >>>
> >>> Dec 2007    Delivering request-URI and parameters to UAS via=20
> >>> proxy to WGLC =20
> >>> Feb 2008    Delivering request-URI and parameters to UAS via=20
> >>> proxy to IESG (PS)
> >>>
> >>> This work is currently stalled and the editor needs input.
> >>>
> >>> The document contains various example scenarios where a=20
> solution is=20
> >>> required, for which there appears to be no dispute that a
> >> solution is
> >>> needed.
> >>>
> >>> The document proposes one solution to resolve these example
> >> scenarios,
> >>> but this solution is not gaining consensus. At least one other=20
> >>> solution has been talked about, but there is no
> >> documentation on the
> >>> table.
> >>>
> >>> This mail is to identify a deadline for other solutions to
> >> the example
> >>> scenarios to be documented as internet drafts, showing how the=20
> >>> solution works for those scenarios. This deadline is:
> >>>
> >>> 	January 11th 2008
> >>>
> >>> It is appropriate fo these documents to identify any other
> >> scenarios
> >>> where such a solution is appropriate. Any other input is
> >> also welcome
> >>> in identifying other scenarios.
> >>>
> >>> If we have no such internet-drafts by this deadline, we
> >> will proceed
> >>> with completing the solution we have.
> >>>
> >>>
> >>> Regards
> >>>
> >>> Keith
> >>>
> >>>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >>
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Francois Audet" <audet@nortel.com>
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Hi,

I answer both John's and Francois' questions in this reply.

>I'm also totally confused now about what is being proposed.
>=20
>The Target seems identical to the To header field.

I don't think they are identical. Jonathan's draft describes the issues
with using the To header.

>I tought that what you (Christer) was arguing for is using=20
>the P-Called-Header instead of the new loose route proposal,=20
>i.e., not for the "initial target", but for the "current target".

Yes, I have.

But, as the P-PCI header is currently defined and used (e.g. in 3GPP) we
are not sure it could be used for all use-cases.

Also, if you read the abstract it does say "current target". I guess the
same wording should be used elsewhere too.

>>In addition to Paul's concerns, I am having difficulty understanding=20
>>why the To header field is insufficient for the cases where the Target

>>header field is proposed. The initial source of this confusion was the

>>statement "the Target header field represents the initial target
identity that was used to initiate a=20
>>session to the target"
>>=20
>>So we now have 4 fields (ignoring Route):
>>- To
>>- Target
>>- P-Called-ID
>>- Request-URI
>>I think it would benefit from a clearer exposition of the semantics=20
>>and relationship between these.

That main purpose of the document was to define an alternative to the
solution in Jonathan's draft, and due to time issues that is what we
focused on.

But, I do agree that there are lots of things that could be clarified
and explained - no matter what solution we move forward with.

Regards,

Christer








> > > -----Original Message-----
> > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > > Sent: 11 January 2008 15:29
> > > To: Christer Holmberg
> > > Cc: sip@ietf.org; DRAGE, Keith (Keith)
> > > Subject: Re: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> > > via proxy
> > >=20
> > > Christer,
> > >=20
> > > I have some questions and comments:
> > >=20
> > > - I don't understand your examples in section 3. They are a bit=20
> > > sketchy about the assumptions they are making, and in
> > notation. I get
> > > lost about which referenced component has which address,
> > etc. I am far
> > > from convinced that these are problems with appropriate=20
> use of the=20
> > > mechanism.
> > >=20
> > > - It seems from your analysis of use cases that it is
> > P-Called-Party
> > > that solves many of them, not Target. So both headers seem
> > to be part
> > > of the solution. Its not entirely clear to me at the=20
> moment whether=20
> > > the R-URI in the loose-route approach aligns with Target or=20
> > > P-Called-Party.
> > > Since they are different, it can't align with both. So
> > there must be
> > > some features it doesn't cover. I haven't fully grokked that yet.
> > >=20
> > > 	Paul
> > >=20
> > >=20
> > > Christer Holmberg wrote:
> > > > Hi,
> > > >=20
> > > > We have submitted a draft with an alternative proposal.
> > > >=20
> > > > It can also be found at:
> > > >=20
> > > >=20
> > > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > et-uri-del
> > > > ivery-00.txt
> > > >=20
> > > > Regards,
> > > >=20
> > > > Christer
> > > >=20
> > > > =20
> > > >=20
> > > >> -----Original Message-----
> > > >> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> > > >> Sent: 9. tammikuuta 2008 18:32
> > > >> To: sip@ietf.org
> > > >> Subject: [Sip] RE: Delivering request-URI and parameters
> > to UAS via
> > > >> proxy
> > > >>
> > > >> A reminder of the deadline on the 11th January for submitting=20
> > > >> alternative proposals on the way forward.
> > > >>
> > > >> Regards
> > > >>
> > > >> Keith
> > > >>
> > > >>> -----Original Message-----
> > > >>> From: DRAGE, Keith (Keith)
> > > >>> Sent: Tuesday, December 04, 2007 3:27 PM
> > > >>> To: sip@ietf.org
> > > >>> Subject: Delivering request-URI and parameters to UAS=20
> via proxy
> > > >>>
> > > >>> (As WG chair)
> > > >>>
> > > >>> We have a couple of milestones that we generated as a=20
> result of=20
> > > >>> discussion of
> > > >>>
> > > >>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> > > >>> se-route-01.txt
> > > >>>
> > > >>> Dec 2007    Delivering request-URI and parameters to UAS via=20
> > > >>> proxy to WGLC =20
> > > >>> Feb 2008    Delivering request-URI and parameters to UAS via=20
> > > >>> proxy to IESG (PS)
> > > >>>
> > > >>> This work is currently stalled and the editor needs input.
> > > >>>
> > > >>> The document contains various example scenarios where a
> > > solution is
> > > >>> required, for which there appears to be no dispute that a
> > > >> solution is
> > > >>> needed.
> > > >>>
> > > >>> The document proposes one solution to resolve these example
> > > >> scenarios,
> > > >>> but this solution is not gaining consensus. At least=20
> one other=20
> > > >>> solution has been talked about, but there is no
> > > >> documentation on the
> > > >>> table.
> > > >>>
> > > >>> This mail is to identify a deadline for other solutions to
> > > >> the example
> > > >>> scenarios to be documented as internet drafts,=20
> showing how the=20
> > > >>> solution works for those scenarios. This deadline is:
> > > >>>
> > > >>> 	January 11th 2008
> > > >>>
> > > >>> It is appropriate fo these documents to identify any other
> > > >> scenarios
> > > >>> where such a solution is appropriate. Any other input is
> > > >> also welcome
> > > >>> in identifying other scenarios.
> > > >>>
> > > >>> If we have no such internet-drafts by this deadline, we
> > > >> will proceed
> > > >>> with completing the solution we have.
> > > >>>
> > > >>>
> > > >>> Regards
> > > >>>
> > > >>> Keith
> > > >>>
> > > >>>
> > > >>
> > > >> _______________________________________________
> > > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > >> This list is for NEW development of the core SIP Protocol Use=20
> > > >> sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > >> sipping@ietf.org for new developments on the application of sip
> > > >>
> > > >=20
> > > >=20
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > > >=20
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com><CA9998CD4A020D418654FCDEF4E707DF040266B1@esealmw113.eemea.ericsson.se>
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Christer Holmberg wrote:
> Hi Paul, 
> 
>> I have some questions and comments:
>>
>> - I don't understand your examples in section 3. They are a 
>> bit sketchy about the assumptions they are making, and in 
>> notation. I get lost about which referenced component has 
>> which address, etc. I am far from convinced that these are 
>> problems with appropriate use of the mechanism.
> 
> As indicated in chapter 3, one of the main issues/limitations with the
> J'draft solution is that the entity wanting to use it must have
> knowledge whether the next hop also supports it. My understanding from
> off-line discussions I had with Jonathan in Vancouver is also that
> Jonathan agrees to that issue/limitation.
> 
> The purpose of the examples is just to show what can go wrong if the
> next hop does not understand the mechanism. But, if we all agree to the
> limitation with the J'draft solution I don't know whether we need to
> spend too much time on the examples.

The limitation is the limitation. It means that you don't use the 
mechanism if you don't know the next hop supports it. So its useless to 
speculate what would go wrong if you use the mechanism with a next hop 
that doesn't support it.

>> - It seems from your analysis of use cases that it is 
>> P-Called-Party that solves many of them, not Target. So both 
>> headers seem to be part of the solution.
> 
> No, as far as the alternative to the J'draft solution is concerned, the
> alternative is Target. We believe it can be used for all listed
> use-cases.

I don't understand. Several of them called for using P-CPI.

> Since I myself have been talking about P-CPI as an alternative, we
> wanted to show in which cases it could be used as an alternative, and
> why we think that P-CPI still would have to be used in IMS. 
> 
> But, we have realized that P-CPI as defined and used today probably
> would not solve all the use-cases, and that is the reason we came up
> with the Target header.
> 
>> Its not entirely clear to me at the moment whether the R-URI in the 
>> loose-route approach aligns with Target or P-Called-Party. 
>> Since they are different, it can't align with both. So there 
>> must be some features it doesn't cover. I haven't fully 
>> grokked that yet.
> 
> I appologize if the P-CPI text causes confusion. Again, the alternative
> to the J'draft solution we propose is Target.
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> 
> 
> 
> 
>> Christer Holmberg wrote:
>>> Hi,
>>>
>>> We have submitted a draft with an alternative proposal.
>>>
>>> It can also be found at:
>>>
>>>
>> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-target-uri-d
>>> el
>>> ivery-00.txt
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>  
>>>
>>>> -----Original Message-----
>>>> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
>>>> Sent: 9. tammikuuta 2008 18:32
>>>> To: sip@ietf.org
>>>> Subject: [Sip] RE: Delivering request-URI and parameters 
>> to UAS via 
>>>> proxy
>>>>
>>>> A reminder of the deadline on the 11th January for submitting 
>>>> alternative proposals on the way forward.
>>>>
>>>> Regards
>>>>
>>>> Keith
>>>>
>>>>> -----Original Message-----
>>>>> From: DRAGE, Keith (Keith)
>>>>> Sent: Tuesday, December 04, 2007 3:27 PM
>>>>> To: sip@ietf.org
>>>>> Subject: Delivering request-URI and parameters to UAS via proxy
>>>>>
>>>>> (As WG chair)
>>>>>
>>>>> We have a couple of milestones that we generated as a result of 
>>>>> discussion of
>>>>>
>>>>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
>>>>> se-route-01.txt
>>>>>
>>>>> Dec 2007    Delivering request-URI and parameters to UAS via 
>>>>> proxy to WGLC  
>>>>> Feb 2008    Delivering request-URI and parameters to UAS via 
>>>>> proxy to IESG (PS)
>>>>>
>>>>> This work is currently stalled and the editor needs input.
>>>>>
>>>>> The document contains various example scenarios where a 
>> solution is 
>>>>> required, for which there appears to be no dispute that a
>>>> solution is
>>>>> needed.
>>>>>
>>>>> The document proposes one solution to resolve these example
>>>> scenarios,
>>>>> but this solution is not gaining consensus. At least one other 
>>>>> solution has been talked about, but there is no
>>>> documentation on the
>>>>> table.
>>>>>
>>>>> This mail is to identify a deadline for other solutions to
>>>> the example
>>>>> scenarios to be documented as internet drafts, showing how the 
>>>>> solution works for those scenarios. This deadline is:
>>>>>
>>>>> 	January 11th 2008
>>>>>
>>>>> It is appropriate fo these documents to identify any other
>>>> scenarios
>>>>> where such a solution is appropriate. Any other input is
>>>> also welcome
>>>>> in identifying other scenarios.
>>>>>
>>>>> If we have no such internet-drafts by this deadline, we
>>>> will proceed
>>>>> with completing the solution we have.
>>>>>
>>>>>
>>>>> Regards
>>>>>
>>>>> Keith
>>>>>
>>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP 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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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Friday, January 11, 2008 11:33
> To: Audet, Francois (SC100:3055)
> Cc: sip@ietf.org; Elwell, John; Paul Kyzivat; DRAGE,Keith (Keith)
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy
>
> >The Target seems identical to the To header field.
>=20
> I don't think they are identical. Jonathan's draft describes=20
> the issues with using the To header.

Well, your description of "Target" in your draft is indentical to
the description of the "To" header.

What Jonathan describes as limitations of "To" is that it doesn't
reflect any "retargeting".

I think what you meant by Target was more the "Current" target as =
opposed
to the Initiatl Target.

And if that's the case, then I don't see why it is different from =
P-Called-ID
(although I might be missing something with what the P-Called_ID is =
supposed
to be).


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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
Date: Fri, 11 Jan 2008 21:02:14 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi,=20

>>>I have some questions and comments:
>>>
>>>- I don't understand your examples in section 3. They are a bit=20
>>>sketchy about the assumptions they are making, and in notation. I get

>>>lost about which referenced component has which address, etc. I am=20
>>>far from convinced that these are problems with appropriate use of=20
>>>the mechanism.
>>=20
>>As indicated in chapter 3, one of the main issues/limitations with the

>>J'draft solution is that the entity wanting to use it must have=20
>>knowledge whether the next hop also supports it. My understanding from

>>off-line discussions I had with Jonathan in Vancouver is also that=20
>>Jonathan agrees to that issue/limitation.
>>=20
>>The purpose of the examples is just to show what can go wrong if the=20
>>next hop does not understand the mechanism. But, if we all agree to=20
>>the limitation with the J'draft solution I don't know whether we need=20
>>to spend too much time on the examples.
>=20
>The limitation is the limitation. It means that you don't use=20
>the mechanism if you don't know the next hop supports it. So=20
>its useless to speculate what would go wrong if you use the=20
>mechanism with a next hop that doesn't support it.

We thought it would be good to show some examples, in case people don't
understand WHY the limitation is there.

>>>- It seems from your analysis of use cases that it is P-Called-Party=20
>>>that solves many of them, not Target. So both headers seem to be part

>>>of the solution.
>>=20
>>No, as far as the alternative to the J'draft solution is concerned,=20
>>the alternative is Target. We believe it can be used for all listed=20
>>use-cases.
>=20
>I don't understand. Several of them called for using P-CPI.

No, we say that there are certain use-case we belive the P-CPI was
defined to be used for, but we also say that the Target header can be
used for those use-cases.

Again, the Target header is our proposal for an alternative to the
J'draft solution.

Regards,

Christer








> >> Christer Holmberg wrote:
> >>> Hi,
> >>>
> >>> We have submitted a draft with an alternative proposal.
> >>>
> >>> It can also be found at:
> >>>
> >>>
> >>=20
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-target-uri-
> >> d
> >>> el
> >>> ivery-00.txt
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>> =20
> >>>
> >>>> -----Original Message-----
> >>>> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> >>>> Sent: 9. tammikuuta 2008 18:32
> >>>> To: sip@ietf.org
> >>>> Subject: [Sip] RE: Delivering request-URI and parameters
> >> to UAS via
> >>>> proxy
> >>>>
> >>>> A reminder of the deadline on the 11th January for submitting=20
> >>>> alternative proposals on the way forward.
> >>>>
> >>>> Regards
> >>>>
> >>>> Keith
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: DRAGE, Keith (Keith)
> >>>>> Sent: Tuesday, December 04, 2007 3:27 PM
> >>>>> To: sip@ietf.org
> >>>>> Subject: Delivering request-URI and parameters to UAS via proxy
> >>>>>
> >>>>> (As WG chair)
> >>>>>
> >>>>> We have a couple of milestones that we generated as a result of=20
> >>>>> discussion of
> >>>>>
> >>>>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> >>>>> se-route-01.txt
> >>>>>
> >>>>> Dec 2007    Delivering request-URI and parameters to UAS via=20
> >>>>> proxy to WGLC =20
> >>>>> Feb 2008    Delivering request-URI and parameters to UAS via=20
> >>>>> proxy to IESG (PS)
> >>>>>
> >>>>> This work is currently stalled and the editor needs input.
> >>>>>
> >>>>> The document contains various example scenarios where a
> >> solution is
> >>>>> required, for which there appears to be no dispute that a
> >>>> solution is
> >>>>> needed.
> >>>>>
> >>>>> The document proposes one solution to resolve these example
> >>>> scenarios,
> >>>>> but this solution is not gaining consensus. At least one other=20
> >>>>> solution has been talked about, but there is no
> >>>> documentation on the
> >>>>> table.
> >>>>>
> >>>>> This mail is to identify a deadline for other solutions to
> >>>> the example
> >>>>> scenarios to be documented as internet drafts, showing how the=20
> >>>>> solution works for those scenarios. This deadline is:
> >>>>>
> >>>>> 	January 11th 2008
> >>>>>
> >>>>> It is appropriate fo these documents to identify any other
> >>>> scenarios
> >>>>> where such a solution is appropriate. Any other input is
> >>>> also welcome
> >>>>> in identifying other scenarios.
> >>>>>
> >>>>> If we have no such internet-drafts by this deadline, we
> >>>> will proceed
> >>>>> with completing the solution we have.
> >>>>>
> >>>>>
> >>>>> Regards
> >>>>>
> >>>>> Keith
> >>>>>
> >>>>>
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sipping@ietf.org for new developments on the application of sip
> >>>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
> >>>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >>
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,=20

>>>The Target seems identical to the To header field.
>>=20
>>I don't think they are identical. Jonathan's draft describes the=20
>>issues with using the To header.
>=20
>Well, your description of "Target" in your draft is=20
>indentical to the description of the "To" header.
>=20
>What Jonathan describes as limitations of "To" is that it=20
>doesn't reflect any "retargeting".
>=20
>I think what you meant by Target was more the "Current"=20
>target as opposed to the Initiatl Target.

Yes, and I think I did say that futher down in my previous reply :)

>And if that's the case, then I don't see why it is different=20
>from P-Called-ID (although I might be missing something with=20
>what the P-Called_ID is supposed to be).

I will get back to you on this one. I am writing this e-mail from my
home machine, so I don't have all the documents etc.

Regards,

Christer





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


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From: "Nadya Zanon" <nadya.zanon@gmail.com>
To: "Farnham, Bruce" <Bruce.Farnham@twtelecom.com>
Subject: Re: [Sip] I am also a beginner with SIP
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Hello Bruce,

I think there is no RFC for Q.931 to SIP mapping. For such implementations,
the Q-SIG is used as base.
Like I'm also a SIP beginner,  I can be wrong.... :)

Best regards,

Nadya Zanon

On 1/3/08, Farnham, Bruce <Bruce.Farnham@twtelecom.com> wrote:
>
>  Hello,
>
> I have been looking for some time for an RFC or specification that
> details the mapping of SIP to Q.931. Does anyone know if such a
> document exists?
>
> Thanks in advance,
>
> Bruce Farnham
> Time Warner Telecom
>   The content contained in this electronic message is not intended to
> constitute formation of a contract binding TWTC. TWTC will be
> contractually bound only upon execution, by an authorized officer, of
> a contract including agreed terms and conditions or by express
> application of its tariffs.
>
> This message is intended only for the use of the individual or entity
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> intended recipient, or the employee or agent responsible for
> delivering the message to the intended recipient, you are hereby
> notified that any dissemination, distribution or copying of this
> message is strictly prohibited. If you have received this
> communication in error, please notify us immediately by replying to
> the sender of this E-Mail or by telephone.
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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>

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<div>Hello Bruce,</div>
<div>&nbsp;</div>
<div>I think there is no RFC for Q.931 to SIP mapping. For such implementations, the Q-SIG is used as base. </div>
<div>Like I&#39;m also a SIP beginner,&nbsp; I can be wrong.... :)</div>
<div>&nbsp;</div>
<div>Best regards,</div>
<div>&nbsp;</div>
<div>Nadya Zanon<br>&nbsp;</div>
<div><span class="gmail_quote">On 1/3/08, <b class="gmail_sendername">Farnham, Bruce</b> &lt;<a href="mailto:Bruce.Farnham@twtelecom.com">Bruce.Farnham@twtelecom.com</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div>
<p><font face="Comic Sans MS" color="#000000">Hello,</font> </p>
<p><font face="Comic Sans MS" color="#000000">I have been looking for some time for an RFC or specification that </font><br><font face="Comic Sans MS" color="#000000">details the mapping of SIP to Q.931. Does anyone know if such a 
</font><br><font face="Comic Sans MS" color="#000000">document exists?</font> </p>
<p><font face="Comic Sans MS" color="#000000">Thanks in advance,</font> </p>
<p><font face="Comic Sans MS" color="#000000">Bruce Farnham</font> <br><font face="Comic Sans MS" color="#000000">Time Warner Telecom</font> </p></div>
<table>
<tbody>
<tr>
<td bgcolor="#ffffff"><font color="#000000">The content contained in this electronic message is not intended to<br>constitute formation of a contract binding TWTC. TWTC will be<br>contractually bound only upon execution, by an authorized officer, of
<br>a contract including agreed terms and conditions or by express<br>application of its tariffs.<br><br>This message is intended only for the use of the individual or entity<br>to which it is addressed. If the reader of this message is not the
<br>intended recipient, or the employee or agent responsible for<br>delivering the message to the intended recipient, you are hereby<br>notified that any dissemination, distribution or copying of this<br>message is strictly prohibited. If you have received this
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From: "Nadya Zanon" <nadya.zanon@gmail.com>
To: "Farnham, Bruce" <Bruce.Farnham@twtelecom.com>
Subject: Re: [Sip] I am also a beginner with SIP
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Would you like to work in that with me?
I'm really interested.

Nadya Zanon

On 1/12/08, Nadya Zanon <nadya.zanon@gmail.com> wrote:
>
> Hello Bruce,
>
> I think there is no RFC for Q.931 to SIP mapping. For such
> implementations, the Q-SIG is used as base.
> Like I'm also a SIP beginner,  I can be wrong.... :)
>
> Best regards,
>
> Nadya Zanon
>
>  On 1/3/08, Farnham, Bruce <Bruce.Farnham@twtelecom.com> wrote:
>
> >  Hello,
> >
> > I have been looking for some time for an RFC or specification that
> > details the mapping of SIP to Q.931. Does anyone know if such a
> > document exists?
> >
> > Thanks in advance,
> >
> > Bruce Farnham
> > Time Warner Telecom
> >   The content contained in this electronic message is not intended to
> > constitute formation of a contract binding TWTC. TWTC will be
> > contractually bound only upon execution, by an authorized officer, of
> > a contract including agreed terms and conditions or by express
> > application of its tariffs.
> >
> > This message is intended only for the use of the individual or entity
> > to which it is addressed. If the reader of this message is not the
> > intended recipient, or the employee or agent responsible for
> > delivering the message to the intended recipient, you are hereby
> > notified that any dissemination, distribution or copying of this
> > message is strictly prohibited. If you have received this
> > communication in error, please notify us immediately by replying to
> > the sender of this E-Mail or by telephone.
> >
> > _______________________________________________
> > Sip mailing list   https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
>

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<div>&nbsp;</div>
<div>Would you like to work in that with me?</div>
<div>I&#39;m really interested.<br>&nbsp;</div>
<div>Nadya Zanon<br>&nbsp;</div>
<div><span class="gmail_quote">On 1/12/08, <b class="gmail_sendername">Nadya Zanon</b> &lt;<a href="mailto:nadya.zanon@gmail.com">nadya.zanon@gmail.com</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div>Hello Bruce,</div>
<div>&nbsp;</div>
<div>I think there is no RFC for Q.931 to SIP mapping. For such implementations, the Q-SIG is used as base. </div>
<div>Like I&#39;m also a SIP beginner,&nbsp; I can be wrong.... :)</div>
<div>&nbsp;</div>
<div>Best regards,</div>
<div>&nbsp;</div>
<div>Nadya Zanon<br>&nbsp;</div>
<div>
<div><span class="e" id="q_1176e2ddd2b8f19a_1"><span class="gmail_quote">On 1/3/08, <b class="gmail_sendername">Farnham, Bruce</b> &lt;<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:Bruce.Farnham@twtelecom.com" target="_blank">
Bruce.Farnham@twtelecom.com</a>&gt; wrote:</span> </span></div>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div><span class="e" id="q_1176e2ddd2b8f19a_3">
<div>
<p><font face="Comic Sans MS" color="#000000">Hello,</font> </p>
<p><font face="Comic Sans MS" color="#000000">I have been looking for some time for an RFC or specification that </font><br><font face="Comic Sans MS" color="#000000">details the mapping of SIP to Q.931. Does anyone know if such a 
</font><br><font face="Comic Sans MS" color="#000000">document exists?</font> </p>
<p><font face="Comic Sans MS" color="#000000">Thanks in advance,</font> </p>
<p><font face="Comic Sans MS" color="#000000">Bruce Farnham</font> <br><font face="Comic Sans MS" color="#000000">Time Warner Telecom</font> </p></div>
<table>
<tbody>
<tr>
<td bgcolor="#ffffff"><font color="#000000">The content contained in this electronic message is not intended to<br>constitute formation of a contract binding TWTC. TWTC will be<br>contractually bound only upon execution, by an authorized officer, of 
<br>a contract including agreed terms and conditions or by express<br>application of its tariffs.<br><br>This message is intended only for the use of the individual or entity<br>to which it is addressed. If the reader of this message is not the 
<br>intended recipient, or the employee or agent responsible for<br>delivering the message to the intended recipient, you are hereby<br>notified that any dissemination, distribution or copying of this<br>message is strictly prohibited. If you have received this 
<br>communication in error, please notify us immediately by replying to<br>the sender of this E-Mail or by telephone.<br></font></td></tr></tbody></table><br></span></div>_______________________________________________<br>
Sip mailing list &nbsp; <a onclick="return top.js.OpenExtLink(window,event,this)" href="https://www1.ietf.org/mailman/listinfo/sip" target="_blank">https://www1.ietf.org/mailman/listinfo/sip</a><br>This list is for NEW development of the core SIP Protocol 
<br>Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip-implementors@cs.columbia.edu" target="_blank">sip-implementors@cs.columbia.edu</a> for questions on current sip<br>Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sipping@ietf.org" target="_blank">
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From: "Nadya Zanon" <nadya.zanon@gmail.com>
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Subject: Re: [Sip] I'm a SIP beginner
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Hello Aslam,

I took a look at RFC3666 and I can see only flows for call establishment.
I'd like to know if there is some other recommendation or draft about
mapping of ISDN features in context ISDN PBX - SIP.

Thanks for your attention. I really appreciated.


On 1/2/08, Aslam Khan <Aslam.Khan@nominum.com> wrote:
>
>  Hi,
>
>
>
> For SIP-PSTN interface you can look at: rfc3398/rfc3666.
>
>
>
> Thanks
>  ------------------------------
>
> *From:* Nadya Zanon [mailto:nadya.zanon@gmail.com]
> *Sent:* Wednesday, January 02, 2008 11:32 AM
> *To:* sip@ietf.org
> *Subject:* [Sip] I'm a SIP beginner
>
>
>
> Hello All,
>
>
>
> I'm a SIP beginner, probably my questions in the begin will be silly but I
> believe that you will understand me.
>
>
>
> I have two questions for now:
>
>
>
> 1) Are there some place where I can find a bunch of drafts for mapping of
> PSTN features on SIP?
>
> 2) Are ther some gateway and standard that maps SMS messages on SIP? I
> mean, are there some mapping from GSM for example to SIP?
>
>
>
>
>
> Thanks in advance,
>
>
>
> Nadya Zanon
>

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<div>Hello Aslam,</div>
<div>&nbsp;</div>
<div>I took a look at RFC3666 and I can see only flows for call establishment.</div>
<div>I&#39;d like to know if there is some other recommendation or draft about mapping of ISDN features in context ISDN PBX - SIP.</div>
<div>&nbsp;</div>
<div>Thanks for your attention. I really appreciated.<br><br>&nbsp;</div>
<div><span class="gmail_quote">On 1/2/08, <b class="gmail_sendername">Aslam Khan</b> &lt;<a href="mailto:Aslam.Khan@nominum.com">Aslam.Khan@nominum.com</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div lang="EN-US" vlink="purple" link="blue">
<div>
<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Hi,</span></font></p>
<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">&nbsp;</span></font></p>
<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">For SIP-PSTN interface you can look at: rfc3398/rfc3666.</span></font></p>
<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">&nbsp;</span></font></p>
<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Thanks</span></font></p>
<div>
<div style="TEXT-ALIGN: center" align="center"><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">
<hr align="center" width="100%" size="2">
</span></font></div>
<p><b><font face="Tahoma" size="2"><span style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</span></font></b><font face="Tahoma" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> Nadya Zanon [mailto:
<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:nadya.zanon@gmail.com" target="_blank">nadya.zanon@gmail.com</a>] <br><b><span style="FONT-WEIGHT: bold">Sent:</span></b> Wednesday, January 02, 2008 11:32 AM
<br><b><span style="FONT-WEIGHT: bold">To:</span></b> <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip@ietf.org" target="_blank">sip@ietf.org</a><br><b><span style="FONT-WEIGHT: bold">Subject:</span>
</b> [Sip] I&#39;m a SIP beginner</span></font></p></div>
<div><span class="e" id="q_1173cabdda9a47b7_1">
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">&nbsp;</span></font></p>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">Hello All,</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">&nbsp;</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">I&#39;m a SIP beginner, probably my questions in the begin will be silly but I believe that you will understand me.</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">&nbsp;</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">I have two questions for now:</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">&nbsp;</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">1) Are there some place where I can find a bunch of drafts for mapping of PSTN features on SIP?</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">2) Are ther some gateway and standard that maps SMS messages on SIP? I mean, are there some mapping from GSM for example to SIP?</span></font></p></div>

<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">&nbsp;</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">&nbsp;</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">Thanks in advance,</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">&nbsp;</span></font></p></div>
<div>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">Nadya Zanon</span></font></p></div></span></div></div></div></blockquote></div><br>

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inline...

Dean Willis wrote:
> 
> On Jan 9, 2008, at 3:58 AM, Elwell, John wrote:
> 

>> 4. The use of offer-answer to achieve hold-retrieve fails to give an
>> explicit indication of what is happening. An explicit indication of
>> things like hold, retrieve, microphone mute, speaker mute, camera
>> on/off, recorder on/off, etc. might be quite helpful. Of course, there
>> will normally be an offer-answer exchange at the same time, so inclusion
>> of the notification in a re-INVITE or UPDATE request might be more
>> useful than using INFO.
> 
> I suspect adding something to the reINVITE to convey explicit semantics 
> would be much better than an INFO here.

The problem with giving an explicit indication of what is going on is 
that it requires agreement on the set of things that *can* go on.

This means standardizing a set of features, which probably isn't going 
to happen, because it will have the effect of limiting feature development.

>> 5. Concerning 6.7 (soft-key labels), a problem here is that the
>> notifying UA might not have knowledge of soft key availability at the
>> receiving UA. In certain closed environments (e.g., involving a UA and
>> its local voicemail server) it may work, but I am not sure it would work
>> well in general. Having said that, where it does work, perhaps other
>> parameters at the device can be controlled in this way, e.g., microphone
>> and speaker volume (especially if this control is based on say a web
>> interface, so the user is really controlling his/her own settings from
>> the web page rather than directly on the device).
> 
> A soft-key INFO package would need to describe availability. This is a 
> fundamental problem from KPML, and to the extent it is solvable by KPML 
> in a SUBSCRIBE dialog, it should be solvable in an INVITE dialog.
> 
>>
>> 6. When a B2BUA performs 3PCC there is plenty of scope for the UAs to
>> lose track of what is happening. For example, when a B2BUA holds,
>> retrieves, transfers calls, joins calls, pre-empts calls, etc., there is
>> currently no explicit information - only perhaps a re-INVITE request.
>> Thus it can be difficult to know how to interpret things like identity.
>> Say a UA receives a re-INVITE request with a new PAI, it might deduce
>> that something like call transfer has occurred. However, if it receives
>> a re-INVITE request without a PAI, does it mean the previous PAI is
>> still valid, or does it mean, following transfer, there is no longer any
>> identity available? Of course, INFO might not always be the most
>> appropriate method to use for such notification if an offer-answer
>> exchange is needed at the same time.
> 
> I'm thinking this is totally NOT an INFO problem. If it needs to be 
> solved, it needs to happen in the 3PCC reINVITE sequence.

Same comment as for (4).

	Paul

>> 7. There might be some usages to do with interworking with TDM, e.g.,
>> advice of charge during a call, user-to-user information during a call.
>>
> 
> Clearly, all the old TDM interworking mid-call events work better with 
> the new INFO. Advice-of-charge would appear to be easily solved with 
> INFO, except that some use cases for it require delivering the final AOC 
> after the call has completed. There may some flexibility on this, of 
> course, but a dialog-correlated MESSAGE to the GRUU may be cleaner.
> 
> -- 
> Dean
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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From sip-bounces@ietf.org Mon Jan 14 06:56:30 2008
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi Francois,=20

>I think what you meant by Target was more the "Current"=20
>target as opposed to the Initiatl Target.

Yes.
=20
>And if that's the case, then I don't see why it is different=20
>from P-Called-ID (although I might be missing something with=20
>what the P-Called_ID is supposed to be).

In the draft we try to explain the difference. But, we are working on
the text to make it more clear.

The P-CPI is inserted when the R-URI is rewritten by the Contact address
of the UAS. RFC3455 calls that operation "retargeting", but we don't
think that is the definition for retarget used in the ua-loose-route
draft, which says:

"When a home proxy receives a request and accesses a location
service, the resulting contact(s) obtained from the location
service are considered the last hop in the route towards the
entity addressed by the Request-URI.  Since that target, almost
by definition, can claim the identity of the URI prior to
translation, the operation is one of routing and not retargeting."

So, if we follow the definitions in the ua-loose-route draft, P-CPI
would be inserted due to a reroute - not retarget.

But, no matter whether we call it retarget or reroute, the point is that
the P-CPI is inserted when the R-URI is rewritten with the Contact
address of the UAS. The scope of Target is wider than that, and can be
used in any retargeting situation.

Regards,

Christer



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From sip-bounces@ietf.org Mon Jan 14 07:15:37 2008
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Avshalom, sorry for slow response...

ext Avshalom Houri wrote:
>
> Wish to get a clarification regarding what will happen when we 
> subscribe to the
> resources below and the first entry does not exist while the other two 
> do exist.
> Assume that we get a pending notification, when and how will it be 
> possible to
> know which resource does exist and which one does not.
>
> What will be the content of the pending notification and how it will 
> be possible
> to know which one exist?
>
>
>    <?xml version="1.0" encoding="UTF-8"?>
>    <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists">
>     <list>
>      <entry uri="resource-lists/users/sip:joe@example.com/index"/>
>      <entry uri="rls-services/users/sip:joe@example.com/index/
>    ~~/*/service%5b@uri='sip:marketing@example.com'%5d"/>
>      <entry uri="pidf-manipulation/"/>
>     </list>
>    </resource-lists>
> --Avshalom
>

The initial full notification will list all existing resources, so given 
the resource-lists document (index) doesn't exist it will not be 
indicated within the notification. So the element will be indicated and 
possibly zero or multiple documents within the pidf-manipulataion 
collection. So there are no reason indications like error codes within 
the notifications, the resources simply doesn't exist within the 
notification body. So a client knows what it tries to retrieve and the 
full notification will list what resources are available, i.e. some 
pretty simple logic on a client side.

br, jari


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(As WG chair)

In fulfilment of our charter items of

Dec 2007    Delivering request-URI and parameters to UAS via proxy to
WGLC =20
Feb 2008    Delivering request-URI and parameters to UAS via proxy to
IESG (PS)

We now have a couple of proposals on the table for solving the problem.

The original draft from Jonathan and which led to the creation of the
charter items by the WG is unfortunately expired, but is at:

http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt

The alternative document from Christer, etc is at:

http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-uri-delive
ry-00.txt

We obviously need to make a decision between the two approaches so
please attempt to address the following specific points via the mailing
list:

1)	Problem cases: These are summarised in section 4.4 of
draft-rosenberg-sip-ua-loose-route-01 and from my read of the other
draft, I don't believe that this draft adds any others. If you believe
there are other cases that should be covered by the solution, then
please identify them. If there is support on any new problem cases, I
would encourage the authors of both drafts to add text concerning these
problem cases.

2)	Clarifications: If for any reason you don't understand either
draft, or believe that there are technical issues that are not
represented in the current draft, please post your questions / comments
to the list. I would encourage authors of both drafts to revise as
frequently as appropriate to reflect the current state of discussion.

3)	Support for either position. If you wish to indicate support for
either position please do so, but please accompany this is technical
reasoning as to why you have this position, as that will help other
members of the WG form a position.

I would encourage as much list discussion as possible before
Philadelphia. I suspect we will need to have a discussion at the
face-to-face meeting in Philadephia, but list discussion is essential
prior to that. If we can solve this positions on list, then well and
good, and even better (that is how we are meant to make decisions).

Regards


Keith


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From sip-bounces@ietf.org Mon Jan 14 20:08:36 2008
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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Hum. I guess then P-Called-ID would then be useful with Loose-route
as well (although now I'm thinking that History-Info covers it).

I think explaining all that in great and precise details, with a =
concrete=20
example would be very useful.

And then we could compare P-Target with Loose-route.=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Monday, January 14, 2008 03:56
> To: Audet, Francois (SC100:3055)
> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy
>=20
>=20
> Hi Francois,=20
>=20
> >I think what you meant by Target was more the "Current"=20
> >target as opposed to the Initiatl Target.
>=20
> Yes.
> =20
> >And if that's the case, then I don't see why it is different from=20
> >P-Called-ID (although I might be missing something with what the=20
> >P-Called_ID is supposed to be).
>=20
> In the draft we try to explain the difference. But, we are=20
> working on the text to make it more clear.
>=20
> The P-CPI is inserted when the R-URI is rewritten by the=20
> Contact address of the UAS. RFC3455 calls that operation=20
> "retargeting", but we don't think that is the definition for=20
> retarget used in the ua-loose-route draft, which says:
>=20
> "When a home proxy receives a request and accesses a location=20
> service, the resulting contact(s) obtained from the location=20
> service are considered the last hop in the route towards the=20
> entity addressed by the Request-URI.  Since that target,=20
> almost by definition, can claim the identity of the URI prior=20
> to translation, the operation is one of routing and not retargeting."
>=20
> So, if we follow the definitions in the ua-loose-route draft,=20
> P-CPI would be inserted due to a reroute - not retarget.
>=20
> But, no matter whether we call it retarget or reroute, the=20
> point is that the P-CPI is inserted when the R-URI is=20
> rewritten with the Contact address of the UAS. The scope of=20
> Target is wider than that, and can be used in any retargeting=20
> situation.
>=20
> Regards,
>=20
> Christer
>=20
>=20


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Hi,=20

P-CPI could probably be useful in some cases, in addition to Loose
Route/Target. And, as the draft says, P-CPI will still have to be used
in IMS, because there are procedures defined for it.

However, again, the purpose of the draft was to provide an alternative
to the Loose Route alternative, and that alternative is Target only.=20

I am working on an updated version of the draft to make that more clear.

Regards,

Christer





>Hum. I guess then P-Called-ID would then be useful with=20
>Loose-route as well (although now I'm thinking that=20
>History-Info covers it).
>=20
>I think explaining all that in great and precise details,=20
>with a concrete example would be very useful.
>=20
>And then we could compare P-Target with Loose-route.=20





>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: Monday, January 14, 2008 03:56
> > To: Audet, Francois (SC100:3055)
> > Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> > Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS=20
> > via proxy
> >=20
> >=20
> > Hi Francois,
> >=20
> > >I think what you meant by Target was more the "Current"=20
> > >target as opposed to the Initiatl Target.
> >=20
> > Yes.
> > =20
> > >And if that's the case, then I don't see why it is different from=20
> > >P-Called-ID (although I might be missing something with what the=20
> > >P-Called_ID is supposed to be).
> >=20
> > In the draft we try to explain the difference. But, we are=20
> working on=20
> > the text to make it more clear.
> >=20
> > The P-CPI is inserted when the R-URI is rewritten by the Contact=20
> > address of the UAS. RFC3455 calls that operation=20
> "retargeting", but we=20
> > don't think that is the definition for retarget used in the=20
> > ua-loose-route draft, which says:
> >=20
> > "When a home proxy receives a request and accesses a=20
> location service,=20
> > the resulting contact(s) obtained from the location service are=20
> > considered the last hop in the route towards the entity=20
> addressed by=20
> > the Request-URI.  Since that target, almost by definition,=20
> can claim=20
> > the identity of the URI prior to translation, the operation=20
> is one of=20
> > routing and not retargeting."
> >=20
> > So, if we follow the definitions in the ua-loose-route draft, P-CPI=20
> > would be inserted due to a reroute - not retarget.
> >=20
> > But, no matter whether we call it retarget or reroute, the point is=20
> > that the P-CPI is inserted when the R-URI is rewritten with the=20
> > Contact address of the UAS. The scope of Target is wider than that,=20
> > and can be used in any retargeting situation.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
>=20


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From sip-bounces@ietf.org Tue Jan 15 04:20:11 2008
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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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Christer,

I guess what still confuses me is, when both Target and P-CPI are used,
which comes first, i.e., which represents the earlier target? When I
read the draft, I thought Target was earlier. From various clarifying
emails I now get the impression that Target is later. Can you confirm?

Picking up on Francois' point about History-Info, with the introduction
of Target and P-CPI we do indeed have a lot of URIs, and of course
History-Info can already convey all these URIs and any others. The
difference is that History-Info does not give particular semantics to
each of the URIs it conveys - they are simply a succession of targets.
With Target and P-CPI we are aiming to define specific semantics. I am
concerned whether we will be able to define these semantics tightly
enough to ensure consistent implementations. The more URIs we try to
define, the harder it will be to assign each one a clearly
distinguishable meaning. I hope the next draft will help.

John

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 15 January 2008 08:48
> To: Francois Audet
> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy
>=20
>=20
> Hi,=20
>=20
> P-CPI could probably be useful in some cases, in addition to Loose
> Route/Target. And, as the draft says, P-CPI will still have to be used
> in IMS, because there are procedures defined for it.
>=20
> However, again, the purpose of the draft was to provide an alternative
> to the Loose Route alternative, and that alternative is Target only.=20
>=20
> I am working on an updated version of the draft to make that=20
> more clear.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
> >Hum. I guess then P-Called-ID would then be useful with=20
> >Loose-route as well (although now I'm thinking that=20
> >History-Info covers it).
> >=20
> >I think explaining all that in great and precise details,=20
> >with a concrete example would be very useful.
> >=20
> >And then we could compare P-Target with Loose-route.=20
>=20
>=20
>=20
>=20
>=20
> >=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: Monday, January 14, 2008 03:56
> > > To: Audet, Francois (SC100:3055)
> > > Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> > > Subject: RE: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> > > via proxy
> > >=20
> > >=20
> > > Hi Francois,
> > >=20
> > > >I think what you meant by Target was more the "Current"=20
> > > >target as opposed to the Initiatl Target.
> > >=20
> > > Yes.
> > > =20
> > > >And if that's the case, then I don't see why it is=20
> different from=20
> > > >P-Called-ID (although I might be missing something with what the=20
> > > >P-Called_ID is supposed to be).
> > >=20
> > > In the draft we try to explain the difference. But, we are=20
> > working on=20
> > > the text to make it more clear.
> > >=20
> > > The P-CPI is inserted when the R-URI is rewritten by the Contact=20
> > > address of the UAS. RFC3455 calls that operation=20
> > "retargeting", but we=20
> > > don't think that is the definition for retarget used in the=20
> > > ua-loose-route draft, which says:
> > >=20
> > > "When a home proxy receives a request and accesses a=20
> > location service,=20
> > > the resulting contact(s) obtained from the location service are=20
> > > considered the last hop in the route towards the entity=20
> > addressed by=20
> > > the Request-URI.  Since that target, almost by definition,=20
> > can claim=20
> > > the identity of the URI prior to translation, the operation=20
> > is one of=20
> > > routing and not retargeting."
> > >=20
> > > So, if we follow the definitions in the ua-loose-route=20
> draft, P-CPI=20
> > > would be inserted due to a reroute - not retarget.
> > >=20
> > > But, no matter whether we call it retarget or reroute,=20
> the point is=20
> > > that the P-CPI is inserted when the R-URI is rewritten with the=20
> > > Contact address of the UAS. The scope of Target is wider=20
> than that,=20
> > > and can be used in any retargeting situation.
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > >=20
> >=20
>=20


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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
Date: Tue, 15 Jan 2008 11:41:15 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	"Francois Audet" <audet@nortel.com>
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Hi John,

>I guess what still confuses me is, when both Target and P-CPI=20
>are used, which comes first, i.e., which represents the=20
>earlier target? When I read the draft, I thought Target was=20
>earlier. From various clarifying emails I now get the=20
>impression that Target is later. Can you confirm?

I am not sure what you mean by "earlier" target.=20

Target, if present, always represents the current target. A receiving
entity that receives a request without a Target header has to assume
that the Request-URI represents the current target. When no Target
header is available on the request then the Target is inserted when a
reroute is performed.

P-CPI is ONLY used to maintain the R-URI which is rewritten by the
contact of the UAS. I.e. by the home proxy which is usually at the end
of a chain of proxies.

When both Target and P-CPI are used: Target always represents the
current target. P-CPI always represents the AOR received by the home
proxy which may in some cases be the same as the current target but it
may also be the last route (not equal to the current target) taken to
deliver the request.=20

>Picking up on Francois' point about History-Info, with the=20
>introduction of Target and P-CPI we do indeed have a lot of=20
>URIs, and of course History-Info can already convey all these=20
>URIs and any others. The difference is that History-Info does=20
>not give particular semantics to each of the URIs it conveys=20
>- they are simply a succession of targets.

Yes, we agree with your analysis regarding History-Info. The
UA-loose-draft discusses further issues with using History-Info.

>With Target and P-CPI we are aiming to define specific=20
>semantics. I am concerned whether we will be able to define=20
>these semantics tightly enough to ensure consistent=20
>implementations.

Target always represents the current target. See above.

P-CPI does not represent the current target in all cases. See above.

>The more URIs we try to define, the harder it will be to assign each
one a clearly distinguishable meaning. I hope the next draft will help.

In the next version of the draft I will remove as much as possible about
P-CPI, in order not to cause confusion. I will only keep some text where
I describe the semantical difference between Target and P-CPI.

I also do agree with one of your previous comments, that we should
clarify the meaning of all the URIs we're using - and that should be
done no matter if we adopt Target or not.

Regards.

Christer














> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 15 January 2008 08:48
> > To: Francois Audet
> > Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> > Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS=20
> > via proxy
> >=20
> >=20
> > Hi,
> >=20
> > P-CPI could probably be useful in some cases, in addition to Loose=20
> > Route/Target. And, as the draft says, P-CPI will still have=20
> to be used=20
> > in IMS, because there are procedures defined for it.
> >=20
> > However, again, the purpose of the draft was to provide an=20
> alternative=20
> > to the Loose Route alternative, and that alternative is Target only.
> >=20
> > I am working on an updated version of the draft to make that more=20
> > clear.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> >=20
> > >Hum. I guess then P-Called-ID would then be useful with=20
> Loose-route=20
> > >as well (although now I'm thinking that History-Info covers it).
> > >=20
> > >I think explaining all that in great and precise details, with a=20
> > >concrete example would be very useful.
> > >=20
> > >And then we could compare P-Target with Loose-route.=20
> >=20
> >=20
> >=20
> >=20
> >=20
> > >=20
> > > > -----Original Message-----
> > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > > Sent: Monday, January 14, 2008 03:56
> > > > To: Audet, Francois (SC100:3055)
> > > > Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat;=20
> Elwell, John
> > > > Subject: RE: [Sip] RE: Delivering request-URI and
> > parameters to UAS
> > > > via proxy
> > > >=20
> > > >=20
> > > > Hi Francois,
> > > >=20
> > > > >I think what you meant by Target was more the "Current"=20
> > > > >target as opposed to the Initiatl Target.
> > > >=20
> > > > Yes.
> > > > =20
> > > > >And if that's the case, then I don't see why it is
> > different from
> > > > >P-Called-ID (although I might be missing something=20
> with what the=20
> > > > >P-Called_ID is supposed to be).
> > > >=20
> > > > In the draft we try to explain the difference. But, we are
> > > working on
> > > > the text to make it more clear.
> > > >=20
> > > > The P-CPI is inserted when the R-URI is rewritten by=20
> the Contact=20
> > > > address of the UAS. RFC3455 calls that operation
> > > "retargeting", but we
> > > > don't think that is the definition for retarget used in the=20
> > > > ua-loose-route draft, which says:
> > > >=20
> > > > "When a home proxy receives a request and accesses a
> > > location service,
> > > > the resulting contact(s) obtained from the location service are=20
> > > > considered the last hop in the route towards the entity
> > > addressed by
> > > > the Request-URI.  Since that target, almost by definition,
> > > can claim
> > > > the identity of the URI prior to translation, the operation
> > > is one of
> > > > routing and not retargeting."
> > > >=20
> > > > So, if we follow the definitions in the ua-loose-route
> > draft, P-CPI
> > > > would be inserted due to a reroute - not retarget.
> > > >=20
> > > > But, no matter whether we call it retarget or reroute,
> > the point is
> > > > that the P-CPI is inserted when the R-URI is rewritten with the=20
> > > > Contact address of the UAS. The scope of Target is wider
> > than that,
> > > > and can be used in any retargeting situation.
> > > >=20
> > > > Regards,
> > > >=20
> > > > Christer
> > > >=20
> > > >=20
> > >=20
> >=20
>=20


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Christer,=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 15 January 2008 10:41
> To: Elwell, John; Francois Audet
> Cc: sip@ietf.org; Paul Kyzivat; DRAGE, Keith (Keith)
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy
>=20
>=20
> Hi John,
>=20
> >I guess what still confuses me is, when both Target and P-CPI=20
> >are used, which comes first, i.e., which represents the=20
> >earlier target? When I read the draft, I thought Target was=20
> >earlier. From various clarifying emails I now get the=20
> >impression that Target is later. Can you confirm?
>=20
> I am not sure what you mean by "earlier" target.=20
[JRE]Assuming that the target changes as the request progresses
downstream, I meant the one that occurs first.

>=20
> Target, if present, always represents the current target. A receiving
> entity that receives a request without a Target header has to assume
> that the Request-URI represents the current target. When no Target
> header is available on the request then the Target is inserted when a
> reroute is performed.
>=20
> P-CPI is ONLY used to maintain the R-URI which is rewritten by the
> contact of the UAS. I.e. by the home proxy which is usually at the end
> of a chain of proxies.
>=20
> When both Target and P-CPI are used: Target always represents the
> current target. P-CPI always represents the AOR received by the home
> proxy which may in some cases be the same as the current target but it
> may also be the last route (not equal to the current target) taken to
> deliver the request.=20
[JRE] I think I am getting there. So the target could be some sort of
call centre group, for example, and P-CPI could be the AoR of the
selected member of that group. Correct?
I am a little concerned about using the term "AoR" in the definition of
these AoRs. What about a request targeted at a GRUU? I would imagine the
Target and P-CPI would both contain the GRUU, which is not an AoR.

John
=20


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Hi John,=20

>>Target, if present, always represents the current target. A receiving=20
>>entity that receives a request without a Target header has to assume=20
>>that the Request-URI represents the current target. When no Target=20
>>header is available on the request then the Target is inserted when a=20
>>reroute is performed.
>>=20
>>P-CPI is ONLY used to maintain the R-URI which is rewritten by the=20
>>contact of the UAS. I.e. by the home proxy which is usually=20
>>at the end of a chain of proxies.
>>=20
>>When both Target and P-CPI are used: Target always represents the=20
>>current target. P-CPI always represents the AOR received by the home=20
>>proxy which may in some cases be the same as the current target but it

>>may also be the last route (not equal to the current target) taken to=20
>>deliver the request.
>>
>[JRE] I think I am getting there. So the target could be some=20
>sort of call centre group, for example, and P-CPI could be=20
>the AoR of the selected member of that group. Correct?

Correct.

>I am a little concerned about using the term "AoR" in the=20
>definition of these AoRs. What about a request targeted at a=20
>GRUU? I would imagine the Target and P-CPI would both contain=20
>the GRUU, which is not an AoR.

Good point. I guess we could simply say URI. We'll fix that in the next
version of the document.

Regards,

Christer


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To: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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I am still confused by the distinction between Target and P-CPI.

I agree that you have described rules for updating Target that are 
different from those for updating P-CPI. But it seems to me that is a 
distinction without a significant difference. For the recipient of the 
request, the two are used the same way, toward the same end. If anything 
I would view the rules for Target to be just a bug fix on P-CPI.

	Paul

Christer Holmberg wrote:
> Hi John,
> 
>> I guess what still confuses me is, when both Target and P-CPI 
>> are used, which comes first, i.e., which represents the 
>> earlier target? When I read the draft, I thought Target was 
>> earlier. From various clarifying emails I now get the 
>> impression that Target is later. Can you confirm?
> 
> I am not sure what you mean by "earlier" target. 
> 
> Target, if present, always represents the current target. A receiving
> entity that receives a request without a Target header has to assume
> that the Request-URI represents the current target. When no Target
> header is available on the request then the Target is inserted when a
> reroute is performed.
> 
> P-CPI is ONLY used to maintain the R-URI which is rewritten by the
> contact of the UAS. I.e. by the home proxy which is usually at the end
> of a chain of proxies.
> 
> When both Target and P-CPI are used: Target always represents the
> current target. P-CPI always represents the AOR received by the home
> proxy which may in some cases be the same as the current target but it
> may also be the last route (not equal to the current target) taken to
> deliver the request. 
> 
>> Picking up on Francois' point about History-Info, with the 
>> introduction of Target and P-CPI we do indeed have a lot of 
>> URIs, and of course History-Info can already convey all these 
>> URIs and any others. The difference is that History-Info does 
>> not give particular semantics to each of the URIs it conveys 
>> - they are simply a succession of targets.
> 
> Yes, we agree with your analysis regarding History-Info. The
> UA-loose-draft discusses further issues with using History-Info.
> 
>> With Target and P-CPI we are aiming to define specific 
>> semantics. I am concerned whether we will be able to define 
>> these semantics tightly enough to ensure consistent 
>> implementations.
> 
> Target always represents the current target. See above.
> 
> P-CPI does not represent the current target in all cases. See above.
> 
>> The more URIs we try to define, the harder it will be to assign each
> one a clearly distinguishable meaning. I hope the next draft will help.
> 
> In the next version of the draft I will remove as much as possible about
> P-CPI, in order not to cause confusion. I will only keep some text where
> I describe the semantical difference between Target and P-CPI.
> 
> I also do agree with one of your previous comments, that we should
> clarify the meaning of all the URIs we're using - and that should be
> done no matter if we adopt Target or not.
> 
> Regards.
> 
> Christer
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>>> -----Original Message-----
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: 15 January 2008 08:48
>>> To: Francois Audet
>>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
>>> Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS 
>>> via proxy
>>>
>>>
>>> Hi,
>>>
>>> P-CPI could probably be useful in some cases, in addition to Loose 
>>> Route/Target. And, as the draft says, P-CPI will still have 
>> to be used 
>>> in IMS, because there are procedures defined for it.
>>>
>>> However, again, the purpose of the draft was to provide an 
>> alternative 
>>> to the Loose Route alternative, and that alternative is Target only.
>>>
>>> I am working on an updated version of the draft to make that more 
>>> clear.
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>>
>>>
>>>> Hum. I guess then P-Called-ID would then be useful with 
>> Loose-route 
>>>> as well (although now I'm thinking that History-Info covers it).
>>>>
>>>> I think explaining all that in great and precise details, with a 
>>>> concrete example would be very useful.
>>>>
>>>> And then we could compare P-Target with Loose-route. 
>>>
>>>
>>>
>>>
>>>>> -----Original Message-----
>>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>>> Sent: Monday, January 14, 2008 03:56
>>>>> To: Audet, Francois (SC100:3055)
>>>>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; 
>> Elwell, John
>>>>> Subject: RE: [Sip] RE: Delivering request-URI and
>>> parameters to UAS
>>>>> via proxy
>>>>>
>>>>>
>>>>> Hi Francois,
>>>>>
>>>>>> I think what you meant by Target was more the "Current" 
>>>>>> target as opposed to the Initiatl Target.
>>>>> Yes.
>>>>>  
>>>>>> And if that's the case, then I don't see why it is
>>> different from
>>>>>> P-Called-ID (although I might be missing something 
>> with what the 
>>>>>> P-Called_ID is supposed to be).
>>>>> In the draft we try to explain the difference. But, we are
>>>> working on
>>>>> the text to make it more clear.
>>>>>
>>>>> The P-CPI is inserted when the R-URI is rewritten by 
>> the Contact 
>>>>> address of the UAS. RFC3455 calls that operation
>>>> "retargeting", but we
>>>>> don't think that is the definition for retarget used in the 
>>>>> ua-loose-route draft, which says:
>>>>>
>>>>> "When a home proxy receives a request and accesses a
>>>> location service,
>>>>> the resulting contact(s) obtained from the location service are 
>>>>> considered the last hop in the route towards the entity
>>>> addressed by
>>>>> the Request-URI.  Since that target, almost by definition,
>>>> can claim
>>>>> the identity of the URI prior to translation, the operation
>>>> is one of
>>>>> routing and not retargeting."
>>>>>
>>>>> So, if we follow the definitions in the ua-loose-route
>>> draft, P-CPI
>>>>> would be inserted due to a reroute - not retarget.
>>>>>
>>>>> But, no matter whether we call it retarget or reroute,
>>> the point is
>>>>> that the P-CPI is inserted when the R-URI is rewritten with the 
>>>>> Contact address of the UAS. The scope of Target is wider
>>> than that,
>>>>> and can be used in any retargeting situation.
>>>>>
>>>>> Regards,
>>>>>
>>>>> Christer
>>>>>
>>>>>
> 


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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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From: "Francois Audet" <audet@nortel.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	"Christer Holmberg" <christer.holmberg@ericsson.com>
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I think we should charge customers per URI in the protocol.

We'd be rich.=20

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]=20
> Sent: Tuesday, January 15, 2008 01:20
> To: Christer Holmberg; Audet, Francois (SC100:3055)
> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy
>=20
> Christer,
>=20
> I guess what still confuses me is, when both Target and P-CPI=20
> are used, which comes first, i.e., which represents the=20
> earlier target? When I read the draft, I thought Target was=20
> earlier. From various clarifying emails I now get the=20
> impression that Target is later. Can you confirm?
>=20
> Picking up on Francois' point about History-Info, with the=20
> introduction of Target and P-CPI we do indeed have a lot of=20
> URIs, and of course History-Info can already convey all these=20
> URIs and any others. The difference is that History-Info does=20
> not give particular semantics to each of the URIs it conveys=20
> - they are simply a succession of targets.
> With Target and P-CPI we are aiming to define specific=20
> semantics. I am concerned whether we will be able to define=20
> these semantics tightly enough to ensure consistent=20
> implementations. The more URIs we try to define, the harder=20
> it will be to assign each one a clearly distinguishable=20
> meaning. I hope the next draft will help.
>=20
> John
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 15 January 2008 08:48
> > To: Francois Audet
> > Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> > Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS=20
> > via proxy
> >=20
> >=20
> > Hi,
> >=20
> > P-CPI could probably be useful in some cases, in addition to Loose=20
> > Route/Target. And, as the draft says, P-CPI will still have=20
> to be used=20
> > in IMS, because there are procedures defined for it.
> >=20
> > However, again, the purpose of the draft was to provide an=20
> alternative=20
> > to the Loose Route alternative, and that alternative is Target only.
> >=20
> > I am working on an updated version of the draft to make that more=20
> > clear.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> >=20
> > >Hum. I guess then P-Called-ID would then be useful with=20
> Loose-route=20
> > >as well (although now I'm thinking that History-Info covers it).
> > >=20
> > >I think explaining all that in great and precise details, with a=20
> > >concrete example would be very useful.
> > >=20
> > >And then we could compare P-Target with Loose-route.=20
> >=20
> >=20
> >=20
> >=20
> >=20
> > >=20
> > > > -----Original Message-----
> > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > > Sent: Monday, January 14, 2008 03:56
> > > > To: Audet, Francois (SC100:3055)
> > > > Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat;=20
> Elwell, John
> > > > Subject: RE: [Sip] RE: Delivering request-URI and
> > parameters to UAS
> > > > via proxy
> > > >=20
> > > >=20
> > > > Hi Francois,
> > > >=20
> > > > >I think what you meant by Target was more the "Current"=20
> > > > >target as opposed to the Initiatl Target.
> > > >=20
> > > > Yes.
> > > > =20
> > > > >And if that's the case, then I don't see why it is
> > different from
> > > > >P-Called-ID (although I might be missing something=20
> with what the=20
> > > > >P-Called_ID is supposed to be).
> > > >=20
> > > > In the draft we try to explain the difference. But, we are
> > > working on
> > > > the text to make it more clear.
> > > >=20
> > > > The P-CPI is inserted when the R-URI is rewritten by=20
> the Contact=20
> > > > address of the UAS. RFC3455 calls that operation
> > > "retargeting", but we
> > > > don't think that is the definition for retarget used in the=20
> > > > ua-loose-route draft, which says:
> > > >=20
> > > > "When a home proxy receives a request and accesses a
> > > location service,
> > > > the resulting contact(s) obtained from the location service are=20
> > > > considered the last hop in the route towards the entity
> > > addressed by
> > > > the Request-URI.  Since that target, almost by definition,
> > > can claim
> > > > the identity of the URI prior to translation, the operation
> > > is one of
> > > > routing and not retargeting."
> > > >=20
> > > > So, if we follow the definitions in the ua-loose-route
> > draft, P-CPI
> > > > would be inserted due to a reroute - not retarget.
> > > >=20
> > > > But, no matter whether we call it retarget or reroute,
> > the point is
> > > > that the P-CPI is inserted when the R-URI is rewritten with the=20
> > > > Contact address of the UAS. The scope of Target is wider
> > than that,
> > > > and can be used in any retargeting situation.
> > > >=20
> > > > Regards,
> > > >=20
> > > > Christer
> > > >=20
> > > >=20
> > >=20
> >=20
>=20


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From: Jeroen van Bemmel <jbemmel@zonnet.nl>
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To: Francois Audet <audet@nortel.com>
Subject: Re: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com>	<CA9998CD4A020D418654FCDEF4E707DF040266B1@esealmw113.eemea.ericsson.se>	<47878B1E.3010303@cisco.com>	<0D5F89FAC29E2C41B98A6A762007F5D0549A47@GBNTHT12009MSX.gb002.siemens.net>	<1ECE0EB50388174790F9694F77522CCF1428F69B@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF04051C9D@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1428F846@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF040960B7@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>	A	<CA9998CD4A020D418654FCDEF4E707DF040D69C7@esealmw113.eemea.ericsson.se>	<0D5F89FAC29E2C41B98A6A762007F5D0549D3F@GBNTHT12009MSX.gb002.siemens.net>
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	John" <john.elwell@siemens.com>,
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I think we'll also need a new response code: 4xx I'm confused, too many 
URIs in request

Seriously, a solution based on yet another header with an URI IMHO 
introduces more complexity/confusion than solving the issue is worth.

Better to limit the scope of the problem we're trying to solve and 
document the known limitations / use cases. If the problem is to deliver 
the request URI to a UAS which is registered with its home proxy, and 
there is a mechanism by which the proxy can detect support by the UAS, 
Jonathan's solution is fine. To other proxies being in the path not 
supporting this mechanism it would still appear as basic loose routing, 
so I don't see the issue there (if needed we could extend the Path 
mechanism with some flag as done for outbound)

Regards,
Jeroen

Francois Audet wrote:
> I think we should charge customers per URI in the protocol.
>
> We'd be rich. 
>
>   
>> -----Original Message-----
>> From: Elwell, John [mailto:john.elwell@siemens.com] 
>> Sent: Tuesday, January 15, 2008 01:20
>> To: Christer Holmberg; Audet, Francois (SC100:3055)
>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat
>> Subject: RE: [Sip] RE: Delivering request-URI and parameters 
>> to UAS via proxy
>>
>> Christer,
>>
>> I guess what still confuses me is, when both Target and P-CPI 
>> are used, which comes first, i.e., which represents the 
>> earlier target? When I read the draft, I thought Target was 
>> earlier. From various clarifying emails I now get the 
>> impression that Target is later. Can you confirm?
>>
>> Picking up on Francois' point about History-Info, with the 
>> introduction of Target and P-CPI we do indeed have a lot of 
>> URIs, and of course History-Info can already convey all these 
>> URIs and any others. The difference is that History-Info does 
>> not give particular semantics to each of the URIs it conveys 
>> - they are simply a succession of targets.
>> With Target and P-CPI we are aiming to define specific 
>> semantics. I am concerned whether we will be able to define 
>> these semantics tightly enough to ensure consistent 
>> implementations. The more URIs we try to define, the harder 
>> it will be to assign each one a clearly distinguishable 
>> meaning. I hope the next draft will help.
>>
>> John
>>
>>     
>>> -----Original Message-----
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: 15 January 2008 08:48
>>> To: Francois Audet
>>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
>>> Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS 
>>> via proxy
>>>
>>>
>>> Hi,
>>>
>>> P-CPI could probably be useful in some cases, in addition to Loose 
>>> Route/Target. And, as the draft says, P-CPI will still have 
>>>       
>> to be used 
>>     
>>> in IMS, because there are procedures defined for it.
>>>
>>> However, again, the purpose of the draft was to provide an 
>>>       
>> alternative 
>>     
>>> to the Loose Route alternative, and that alternative is Target only.
>>>
>>> I am working on an updated version of the draft to make that more 
>>> clear.
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>>
>>>
>>>       
>>>> Hum. I guess then P-Called-ID would then be useful with 
>>>>         
>> Loose-route 
>>     
>>>> as well (although now I'm thinking that History-Info covers it).
>>>>
>>>> I think explaining all that in great and precise details, with a 
>>>> concrete example would be very useful.
>>>>
>>>> And then we could compare P-Target with Loose-route. 
>>>>         
>>>
>>>
>>>
>>>       
>>>>> -----Original Message-----
>>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>>> Sent: Monday, January 14, 2008 03:56
>>>>> To: Audet, Francois (SC100:3055)
>>>>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; 
>>>>>           
>> Elwell, John
>>     
>>>>> Subject: RE: [Sip] RE: Delivering request-URI and
>>>>>           
>>> parameters to UAS
>>>       
>>>>> via proxy
>>>>>
>>>>>
>>>>> Hi Francois,
>>>>>
>>>>>           
>>>>>> I think what you meant by Target was more the "Current" 
>>>>>> target as opposed to the Initiatl Target.
>>>>>>             
>>>>> Yes.
>>>>>  
>>>>>           
>>>>>> And if that's the case, then I don't see why it is
>>>>>>             
>>> different from
>>>       
>>>>>> P-Called-ID (although I might be missing something 
>>>>>>             
>> with what the 
>>     
>>>>>> P-Called_ID is supposed to be).
>>>>>>             
>>>>> In the draft we try to explain the difference. But, we are
>>>>>           
>>>> working on
>>>>         
>>>>> the text to make it more clear.
>>>>>
>>>>> The P-CPI is inserted when the R-URI is rewritten by 
>>>>>           
>> the Contact 
>>     
>>>>> address of the UAS. RFC3455 calls that operation
>>>>>           
>>>> "retargeting", but we
>>>>         
>>>>> don't think that is the definition for retarget used in the 
>>>>> ua-loose-route draft, which says:
>>>>>
>>>>> "When a home proxy receives a request and accesses a
>>>>>           
>>>> location service,
>>>>         
>>>>> the resulting contact(s) obtained from the location service are 
>>>>> considered the last hop in the route towards the entity
>>>>>           
>>>> addressed by
>>>>         
>>>>> the Request-URI.  Since that target, almost by definition,
>>>>>           
>>>> can claim
>>>>         
>>>>> the identity of the URI prior to translation, the operation
>>>>>           
>>>> is one of
>>>>         
>>>>> routing and not retargeting."
>>>>>
>>>>> So, if we follow the definitions in the ua-loose-route
>>>>>           
>>> draft, P-CPI
>>>       
>>>>> would be inserted due to a reroute - not retarget.
>>>>>
>>>>> But, no matter whether we call it retarget or reroute,
>>>>>           
>>> the point is
>>>       
>>>>> that the P-CPI is inserted when the R-URI is rewritten with the 
>>>>> Contact address of the UAS. The scope of Target is wider
>>>>>           
>>> than that,
>>>       
>>>>> and can be used in any retargeting situation.
>>>>>
>>>>> Regards,
>>>>>
>>>>> Christer
>>>>>
>>>>>
>>>>>           
>
>
> _______________________________________________
> 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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>
>   


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From sip-bounces@ietf.org Wed Jan 16 03:09:37 2008
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi,=20

>I am still confused by the distinction between Target and P-CPI.
>=20
>I agree that you have described rules for updating Target=20
>that are different from those for updating P-CPI. But it=20
>seems to me that is a distinction without a significant=20
>difference. For the recipient of the request, the two are=20
>used the same way, toward the same end. If anything I would=20
>view the rules for Target to be just a bug fix on P-CPI.

You don't update P-CPI. It's inserted by the home proxy, and nothing
else.

The headers have differenct semantics. I don't see that as a bug.

Maybe we could try to change the semantics of P-CPI, and re-define the
header, but that was never my intention when I proposed to use P-CPI.

Regards,

Christer=20





> 	Paul
>=20
> Christer Holmberg wrote:
> > Hi John,
> >=20
> >> I guess what still confuses me is, when both Target and P-CPI are=20
> >> used, which comes first, i.e., which represents the=20
> earlier target?=20
> >> When I read the draft, I thought Target was earlier. From various=20
> >> clarifying emails I now get the impression that Target is=20
> later. Can=20
> >> you confirm?
> >=20
> > I am not sure what you mean by "earlier" target.=20
> >=20
> > Target, if present, always represents the current target. A=20
> receiving=20
> > entity that receives a request without a Target header has=20
> to assume=20
> > that the Request-URI represents the current target. When no Target=20
> > header is available on the request then the Target is=20
> inserted when a=20
> > reroute is performed.
> >=20
> > P-CPI is ONLY used to maintain the R-URI which is rewritten by the=20
> > contact of the UAS. I.e. by the home proxy which is usually=20
> at the end=20
> > of a chain of proxies.
> >=20
> > When both Target and P-CPI are used: Target always represents the=20
> > current target. P-CPI always represents the AOR received by=20
> the home=20
> > proxy which may in some cases be the same as the current=20
> target but it=20
> > may also be the last route (not equal to the current=20
> target) taken to=20
> > deliver the request.
> >=20
> >> Picking up on Francois' point about History-Info, with the=20
> >> introduction of Target and P-CPI we do indeed have a lot=20
> of URIs, and=20
> >> of course History-Info can already convey all these URIs and any=20
> >> others. The difference is that History-Info does not give=20
> particular=20
> >> semantics to each of the URIs it conveys
> >> - they are simply a succession of targets.
> >=20
> > Yes, we agree with your analysis regarding History-Info. The=20
> > UA-loose-draft discusses further issues with using History-Info.
> >=20
> >> With Target and P-CPI we are aiming to define specific=20
> semantics. I=20
> >> am concerned whether we will be able to define these semantics=20
> >> tightly enough to ensure consistent implementations.
> >=20
> > Target always represents the current target. See above.
> >=20
> > P-CPI does not represent the current target in all cases. See above.
> >=20
> >> The more URIs we try to define, the harder it will be to=20
> assign each
> > one a clearly distinguishable meaning. I hope the next=20
> draft will help.
> >=20
> > In the next version of the draft I will remove as much as possible=20
> > about P-CPI, in order not to cause confusion. I will only keep some=20
> > text where I describe the semantical difference between=20
> Target and P-CPI.
> >=20
> > I also do agree with one of your previous comments, that we should=20
> > clarify the meaning of all the URIs we're using - and that=20
> should be=20
> > done no matter if we adopt Target or not.
> >=20
> > Regards.
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >>> -----Original Message-----
> >>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>> Sent: 15 January 2008 08:48
> >>> To: Francois Audet
> >>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> >>> Subject: RE: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> >>> via proxy
> >>>
> >>>
> >>> Hi,
> >>>
> >>> P-CPI could probably be useful in some cases, in addition=20
> to Loose=20
> >>> Route/Target. And, as the draft says, P-CPI will still have
> >> to be used
> >>> in IMS, because there are procedures defined for it.
> >>>
> >>> However, again, the purpose of the draft was to provide an
> >> alternative
> >>> to the Loose Route alternative, and that alternative is=20
> Target only.
> >>>
> >>> I am working on an updated version of the draft to make that more=20
> >>> clear.
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>> Hum. I guess then P-Called-ID would then be useful with
> >> Loose-route
> >>>> as well (although now I'm thinking that History-Info covers it).
> >>>>
> >>>> I think explaining all that in great and precise details, with a=20
> >>>> concrete example would be very useful.
> >>>>
> >>>> And then we could compare P-Target with Loose-route.=20
> >>>
> >>>
> >>>
> >>>
> >>>>> -----Original Message-----
> >>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>>>> Sent: Monday, January 14, 2008 03:56
> >>>>> To: Audet, Francois (SC100:3055)
> >>>>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat;
> >> Elwell, John
> >>>>> Subject: RE: [Sip] RE: Delivering request-URI and
> >>> parameters to UAS
> >>>>> via proxy
> >>>>>
> >>>>>
> >>>>> Hi Francois,
> >>>>>
> >>>>>> I think what you meant by Target was more the "Current"=20
> >>>>>> target as opposed to the Initiatl Target.
> >>>>> Yes.
> >>>>> =20
> >>>>>> And if that's the case, then I don't see why it is
> >>> different from
> >>>>>> P-Called-ID (although I might be missing something
> >> with what the
> >>>>>> P-Called_ID is supposed to be).
> >>>>> In the draft we try to explain the difference. But, we are
> >>>> working on
> >>>>> the text to make it more clear.
> >>>>>
> >>>>> The P-CPI is inserted when the R-URI is rewritten by
> >> the Contact
> >>>>> address of the UAS. RFC3455 calls that operation
> >>>> "retargeting", but we
> >>>>> don't think that is the definition for retarget used in the=20
> >>>>> ua-loose-route draft, which says:
> >>>>>
> >>>>> "When a home proxy receives a request and accesses a
> >>>> location service,
> >>>>> the resulting contact(s) obtained from the location service are=20
> >>>>> considered the last hop in the route towards the entity
> >>>> addressed by
> >>>>> the Request-URI.  Since that target, almost by definition,
> >>>> can claim
> >>>>> the identity of the URI prior to translation, the operation
> >>>> is one of
> >>>>> routing and not retargeting."
> >>>>>
> >>>>> So, if we follow the definitions in the ua-loose-route
> >>> draft, P-CPI
> >>>>> would be inserted due to a reroute - not retarget.
> >>>>>
> >>>>> But, no matter whether we call it retarget or reroute,
> >>> the point is
> >>>>> that the P-CPI is inserted when the R-URI is rewritten with the=20
> >>>>> Contact address of the UAS. The scope of Target is wider
> >>> than that,
> >>>>> and can be used in any retargeting situation.
> >>>>>
> >>>>> Regards,
> >>>>>
> >>>>> Christer
> >>>>>
> >>>>>
> >=20
>=20


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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
Date: Wed, 16 Jan 2008 09:12:11 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
	"Francois Audet" <audet@nortel.com>
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Hi Jeroen,=20

>I think we'll also need a new response code: 4xx I'm confused, too many
URIs in request
>=20
>Seriously, a solution based on yet another header with an URI IMHO
introduces more complexity/confusion than solving the=20
>issue is worth.
>=20
>Better to limit the scope of the problem we're trying to=20
>solve and document the known limitations / use cases. If the=20
>problem is to deliver the request URI to a UAS which is=20
>registered with its home proxy, and there is a mechanism by=20
>which the proxy can detect support by the UAS, Jonathan's=20
>solution is fine.=20

I don't think that is the whole problem (if it was, P-Called-Party-ID
would probably work just fine). To my understanding it is not
necessarily the home proxy of the UAS which rewrites the Request-URI in
the use-cases presented in the drafts. It can be another entity in the
path - an entity which havs no clue about what the UAS supports or not.

Regards,

Christer






> >> -----Original Message-----
> >> From: Elwell, John [mailto:john.elwell@siemens.com]
> >> Sent: Tuesday, January 15, 2008 01:20
> >> To: Christer Holmberg; Audet, Francois (SC100:3055)
> >> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat
> >> Subject: RE: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> >> via proxy
> >>
> >> Christer,
> >>
> >> I guess what still confuses me is, when both Target and P-CPI are=20
> >> used, which comes first, i.e., which represents the=20
> earlier target?=20
> >> When I read the draft, I thought Target was earlier. From various=20
> >> clarifying emails I now get the impression that Target is=20
> later. Can=20
> >> you confirm?
> >>
> >> Picking up on Francois' point about History-Info, with the=20
> >> introduction of Target and P-CPI we do indeed have a lot=20
> of URIs, and=20
> >> of course History-Info can already convey all these URIs and any=20
> >> others. The difference is that History-Info does not give=20
> particular=20
> >> semantics to each of the URIs it conveys
> >> - they are simply a succession of targets.
> >> With Target and P-CPI we are aiming to define specific=20
> semantics. I=20
> >> am concerned whether we will be able to define these semantics=20
> >> tightly enough to ensure consistent implementations. The=20
> more URIs we=20
> >> try to define, the harder it will be to assign each one a clearly=20
> >> distinguishable meaning. I hope the next draft will help.
> >>
> >> John
> >>
> >>    =20
> >>> -----Original Message-----
> >>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>> Sent: 15 January 2008 08:48
> >>> To: Francois Audet
> >>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John
> >>> Subject: RE: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> >>> via proxy
> >>>
> >>>
> >>> Hi,
> >>>
> >>> P-CPI could probably be useful in some cases, in addition=20
> to Loose=20
> >>> Route/Target. And, as the draft says, P-CPI will still have
> >>>      =20
> >> to be used
> >>    =20
> >>> in IMS, because there are procedures defined for it.
> >>>
> >>> However, again, the purpose of the draft was to provide an
> >>>      =20
> >> alternative
> >>    =20
> >>> to the Loose Route alternative, and that alternative is=20
> Target only.
> >>>
> >>> I am working on an updated version of the draft to make that more=20
> >>> clear.
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>      =20
> >>>> Hum. I guess then P-Called-ID would then be useful with
> >>>>        =20
> >> Loose-route
> >>    =20
> >>>> as well (although now I'm thinking that History-Info covers it).
> >>>>
> >>>> I think explaining all that in great and precise details, with a=20
> >>>> concrete example would be very useful.
> >>>>
> >>>> And then we could compare P-Target with Loose-route.=20
> >>>>        =20
> >>>
> >>>
> >>>
> >>>      =20
> >>>>> -----Original Message-----
> >>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>>>> Sent: Monday, January 14, 2008 03:56
> >>>>> To: Audet, Francois (SC100:3055)
> >>>>> Cc: sip@ietf.org; DRAGE, Keith (Keith); Paul Kyzivat;
> >>>>>          =20
> >> Elwell, John
> >>    =20
> >>>>> Subject: RE: [Sip] RE: Delivering request-URI and
> >>>>>          =20
> >>> parameters to UAS
> >>>      =20
> >>>>> via proxy
> >>>>>
> >>>>>
> >>>>> Hi Francois,
> >>>>>
> >>>>>          =20
> >>>>>> I think what you meant by Target was more the "Current"=20
> >>>>>> target as opposed to the Initiatl Target.
> >>>>>>            =20
> >>>>> Yes.
> >>>>> =20
> >>>>>          =20
> >>>>>> And if that's the case, then I don't see why it is
> >>>>>>            =20
> >>> different from
> >>>      =20
> >>>>>> P-Called-ID (although I might be missing something
> >>>>>>            =20
> >> with what the
> >>    =20
> >>>>>> P-Called_ID is supposed to be).
> >>>>>>            =20
> >>>>> In the draft we try to explain the difference. But, we are
> >>>>>          =20
> >>>> working on
> >>>>        =20
> >>>>> the text to make it more clear.
> >>>>>
> >>>>> The P-CPI is inserted when the R-URI is rewritten by
> >>>>>          =20
> >> the Contact
> >>    =20
> >>>>> address of the UAS. RFC3455 calls that operation
> >>>>>          =20
> >>>> "retargeting", but we
> >>>>        =20
> >>>>> don't think that is the definition for retarget used in the=20
> >>>>> ua-loose-route draft, which says:
> >>>>>
> >>>>> "When a home proxy receives a request and accesses a
> >>>>>          =20
> >>>> location service,
> >>>>        =20
> >>>>> the resulting contact(s) obtained from the location service are=20
> >>>>> considered the last hop in the route towards the entity
> >>>>>          =20
> >>>> addressed by
> >>>>        =20
> >>>>> the Request-URI.  Since that target, almost by definition,
> >>>>>          =20
> >>>> can claim
> >>>>        =20
> >>>>> the identity of the URI prior to translation, the operation
> >>>>>          =20
> >>>> is one of
> >>>>        =20
> >>>>> routing and not retargeting."
> >>>>>
> >>>>> So, if we follow the definitions in the ua-loose-route
> >>>>>          =20
> >>> draft, P-CPI
> >>>      =20
> >>>>> would be inserted due to a reroute - not retarget.
> >>>>>
> >>>>> But, no matter whether we call it retarget or reroute,
> >>>>>          =20
> >>> the point is
> >>>      =20
> >>>>> that the P-CPI is inserted when the R-URI is rewritten with the=20
> >>>>> Contact address of the UAS. The scope of Target is wider
> >>>>>          =20
> >>> than that,
> >>>      =20
> >>>>> and can be used in any retargeting situation.
> >>>>>
> >>>>> Regards,
> >>>>>
> >>>>> Christer
> >>>>>
> >>>>>
> >>>>>          =20
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >
> >  =20
>=20


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,

We've uploaded a new version (-01) of the Target draft.

We've tried to make things more clear. I've also removed all text about
P-Called-Party-ID, except from one chapter where we try to explain the
semantical difference between Target and P-CPI.

You can also find the draft from:
http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-target-uri-del
ivery-01.txt

Regards,

Christer


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Subject: [Sip] RE: Delivering request-URI and parameters to UAS via proxy -
 new version of the draft-holmberg-sip-target-uri-delivery draft
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Christer,

Thanks for this revision, which makes things somewhat clearer. I do have
a couple of comments:

1. I am not sure I agree with the assertions in section 4 concerning
issues with the mechanisms in loose-route. Taking example 1, the Route
header field should contain enough entries to get you to the registered
contact, not just to an intermediate proxy. Therefore this situation
should not arise with a correctly implemented home proxy. It is not
clear to me how example 2 could arise either, for similar reasons. The
MGC case can be resolved by taking into account the option tag in the
REGISTER request, or if it is permanently registered, through
provisioning.

2. Comparing the mechanism proposed with the loose-route mechanism, my
understanding is:
a)When retargeting occurs, the loose-route mechanism places the new
target in the Request URI. Your proposal places the new target in both
the Request-URI and the Target header field.
b) When rerouting, the loose-route mechanism places the new route (i.e.,
the registered contact) in the Route header field. Your proposal places
the new route in the Request-URI (the latter as per RFC 3261).
So the two mechanisms solve exactly the same problems using a slightly
different mechanism. Correct?

3. How P-Called-Party-ID fits into this is not really relevant from an
IETF perspective - it seems there are some 3GPP-specific situations
where the contents of P-Called-Party-ID will not equal the contents of
Target. Correct?

4. If my suggestion in point 1 above that the loose-route mechanism does
not suffer from the problems suggested, then each mechanism will work
and each addresses the same problem. So it is just down to a beauty
contest between the two. Correct?

John


> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 16 January 2008 08:41
> To: sip@ietf.org
> Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John; Jeroen=20
> van Bemmel; Francois Audet
> Subject: Delivering request-URI and parameters to UAS via=20
> proxy - new version of the=20
> draft-holmberg-sip-target-uri-delivery draft
>=20
>=20
> Hi,
>=20
> We've uploaded a new version (-01) of the Target draft.
>=20
> We've tried to make things more clear. I've also removed all=20
> text about
> P-Called-Party-ID, except from one chapter where we try to explain the
> semantical difference between Target and P-CPI.
>=20
> You can also find the draft from:
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> et-uri-del
> ivery-01.txt
>=20
> Regards,
>=20
> Christer
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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	89FAC29E2C41B98A6A762007F5D0593C0E@GBNTHT12009MSX.gb002.siemens.net>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	<sip@ietf.org>
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Subject: [Sip] RE: Delivering request-URI and parameters to UAS via proxy -
	new version of the draft-holmberg-sip-target-uri-delivery draft
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Hi John,=20

>Thanks for this revision, which makes things somewhat=20
>clearer. I do have a couple of comments:
>=20
>1. I am not sure I agree with the assertions in section 4=20
>concerning issues with the mechanisms in loose-route. Taking=20
>example 1, the Route header field should contain enough=20
>entries to get you to the registered contact, not just to an=20
>intermediate proxy. Therefore this situation should not arise=20
>with a correctly implemented home proxy. It is not clear to=20
>me how example 2 could arise either, for similar reasons. The=20
>MGC case can be resolved by taking into account the option=20
>tag in the REGISTER request, or if it is permanently=20
>registered, through provisioning.

The examples are not meant to show bugs in the loose-route mechanism.
They are meant to help people understand the limitations with the
loose-route mechanism.
=20
>2. Comparing the mechanism proposed with the loose-route=20
>mechanism, my understanding is:
>a)When retargeting occurs, the loose-route mechanism places=20
>the new target in the Request URI. Your proposal places the=20
>new target in both the Request-URI and the Target header field.
>b) When rerouting, the loose-route mechanism places the new=20
>route (i.e., the registered contact) in the Route header=20
>field. Your proposal places the new route in the Request-URI=20
>(the latter as per RFC 3261).
>So the two mechanisms solve exactly the same problems using a=20
>slightly different mechanism. Correct?

Yes. The two solutions intend to solve the same problem.
=20
>3. How P-Called-Party-ID fits into this is not really=20
>relevant from an IETF perspective - it seems there are some=20
>3GPP-specific situations where the contents of=20
>P-Called-Party-ID will not equal the contents of Target. Correct?

Correct.

>4. If my suggestion in point 1 above that the loose-route=20
>mechanism does not suffer from the problems suggested, then=20
>each mechanism will work and each addresses the same problem.=20
>So it is just down to a beauty contest between the two. Correct?

See question 1.

We believe that our solution does not have the same limitations as the
loose-route solution. But, again, both solutions intend to solve the
same problem.

Regards,

Christer






> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 16 January 2008 08:41
> > To: sip@ietf.org
> > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John; Jeroen van=20
> > Bemmel; Francois Audet
> > Subject: Delivering request-URI and parameters to UAS via=20
> proxy - new=20
> > version of the draft-holmberg-sip-target-uri-delivery draft
> >=20
> >=20
> > Hi,
> >=20
> > We've uploaded a new version (-01) of the Target draft.
> >=20
> > We've tried to make things more clear. I've also removed all text=20
> > about P-Called-Party-ID, except from one chapter where we try to=20
> > explain the semantical difference between Target and P-CPI.
> >=20
> > You can also find the draft from:
> > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > et-uri-del
> > ivery-01.txt
> >=20
> > Regards,
> >=20
> > Christer
> >=20
>=20


_______________________________________________
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Date: Wed, 16 Jan 2008 11:19:25 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy -
	new version of the draft-holmberg-sip-target-uri-delivery draft
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Christer,

I hear what you are saying, but I don't really see that the points you
raised in section 4 are real limitations of the loose route mechanism.

John=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 16 January 2008 11:06
> To: Elwell, John; sip@ietf.org
> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> Subject: [Sip] RE: Delivering request-URI and parameters to=20
> UAS via proxy - new version of the=20
> draft-holmberg-sip-target-uri-delivery draft
>=20
>=20
> Hi John,=20
>=20
> >Thanks for this revision, which makes things somewhat=20
> >clearer. I do have a couple of comments:
> >=20
> >1. I am not sure I agree with the assertions in section 4=20
> >concerning issues with the mechanisms in loose-route. Taking=20
> >example 1, the Route header field should contain enough=20
> >entries to get you to the registered contact, not just to an=20
> >intermediate proxy. Therefore this situation should not arise=20
> >with a correctly implemented home proxy. It is not clear to=20
> >me how example 2 could arise either, for similar reasons. The=20
> >MGC case can be resolved by taking into account the option=20
> >tag in the REGISTER request, or if it is permanently=20
> >registered, through provisioning.
>=20
> The examples are not meant to show bugs in the loose-route mechanism.
> They are meant to help people understand the limitations with the
> loose-route mechanism.
> =20
> >2. Comparing the mechanism proposed with the loose-route=20
> >mechanism, my understanding is:
> >a)When retargeting occurs, the loose-route mechanism places=20
> >the new target in the Request URI. Your proposal places the=20
> >new target in both the Request-URI and the Target header field.
> >b) When rerouting, the loose-route mechanism places the new=20
> >route (i.e., the registered contact) in the Route header=20
> >field. Your proposal places the new route in the Request-URI=20
> >(the latter as per RFC 3261).
> >So the two mechanisms solve exactly the same problems using a=20
> >slightly different mechanism. Correct?
>=20
> Yes. The two solutions intend to solve the same problem.
> =20
> >3. How P-Called-Party-ID fits into this is not really=20
> >relevant from an IETF perspective - it seems there are some=20
> >3GPP-specific situations where the contents of=20
> >P-Called-Party-ID will not equal the contents of Target. Correct?
>=20
> Correct.
>=20
> >4. If my suggestion in point 1 above that the loose-route=20
> >mechanism does not suffer from the problems suggested, then=20
> >each mechanism will work and each addresses the same problem.=20
> >So it is just down to a beauty contest between the two. Correct?
>=20
> See question 1.
>=20
> We believe that our solution does not have the same limitations as the
> loose-route solution. But, again, both solutions intend to solve the
> same problem.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: 16 January 2008 08:41
> > > To: sip@ietf.org
> > > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John; Jeroen van=20
> > > Bemmel; Francois Audet
> > > Subject: Delivering request-URI and parameters to UAS via=20
> > proxy - new=20
> > > version of the draft-holmberg-sip-target-uri-delivery draft
> > >=20
> > >=20
> > > Hi,
> > >=20
> > > We've uploaded a new version (-01) of the Target draft.
> > >=20
> > > We've tried to make things more clear. I've also removed all text=20
> > > about P-Called-Party-ID, except from one chapter where we try to=20
> > > explain the semantical difference between Target and P-CPI.
> > >=20
> > > You can also find the draft from:
> > > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > et-uri-del
> > > ivery-01.txt
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: [Sip] Delivering request-URI and parameters to UAS via proxy
Date: Wed, 16 Jan 2008 12:35:12 +0100
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: <sip@ietf.org>
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(As WG chair)

I am seeing consensus on the second draft.......


.....that everyone is confused by the semantics applied to the various
headers, and the mail exchange does not seem to be clearing it up.

I urge Christer and Hans Erik to get a new version out as soon as
possible in the hope of removing some of the confusion.

Regards

Keith=20

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Sent: Monday, January 14, 2008 4:58 PM
> To: sip@ietf.org
> Subject: [Sip] Delivering request-URI and parameters to UAS via proxy
>=20
> (As WG chair)
>=20
> In fulfilment of our charter items of
>=20
> Dec 2007    Delivering request-URI and parameters to UAS via proxy to
> WGLC =20
> Feb 2008    Delivering request-URI and parameters to UAS via proxy to
> IESG (PS)
>=20
> We now have a couple of proposals on the table for solving=20
> the problem.
>=20
> The original draft from Jonathan and which led to the=20
> creation of the charter items by the WG is unfortunately=20
> expired, but is at:
>=20
> http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
>=20
> The alternative document from Christer, etc is at:
>=20
> http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
> uri-delive
> ry-00.txt
>=20
> We obviously need to make a decision between the two=20
> approaches so please attempt to address the following=20
> specific points via the mailing
> list:
>=20
> 1)	Problem cases: These are summarised in section 4.4 of
> draft-rosenberg-sip-ua-loose-route-01 and from my read of the=20
> other draft, I don't believe that this draft adds any others.=20
> If you believe there are other cases that should be covered=20
> by the solution, then please identify them. If there is=20
> support on any new problem cases, I would encourage the=20
> authors of both drafts to add text concerning these problem cases.
>=20
> 2)	Clarifications: If for any reason you don't understand either
> draft, or believe that there are technical issues that are=20
> not represented in the current draft, please post your=20
> questions / comments to the list. I would encourage authors=20
> of both drafts to revise as frequently as appropriate to=20
> reflect the current state of discussion.
>=20
> 3)	Support for either position. If you wish to indicate support for
> either position please do so, but please accompany this is=20
> technical reasoning as to why you have this position, as that=20
> will help other members of the WG form a position.
>=20
> I would encourage as much list discussion as possible before=20
> Philadelphia. I suspect we will need to have a discussion at=20
> the face-to-face meeting in Philadephia, but list discussion=20
> is essential prior to that. If we can solve this positions on=20
> list, then well and good, and even better (that is how we are=20
> meant to make decisions).
>=20
> Regards
>=20
>=20
> Keith
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



From sip-bounces@ietf.org Wed Jan 16 06:38:26 2008
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Subject: RE: [Sip] Delivering request-URI and parameters to UAS via proxy
Date: Wed, 16 Jan 2008 12:38:19 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>,
	<sip@ietf.org>
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Hi Keith,

I submitted a new version earlier today.

Regards,

Christer=20

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Sent: 16. tammikuuta 2008 13:35
> To: sip@ietf.org
> Subject: RE: [Sip] Delivering request-URI and parameters to=20
> UAS via proxy
>=20
> (As WG chair)
>=20
> I am seeing consensus on the second draft.......
>=20
>=20
> .....that everyone is confused by the semantics applied to=20
> the various headers, and the mail exchange does not seem to=20
> be clearing it up.
>=20
> I urge Christer and Hans Erik to get a new version out as=20
> soon as possible in the hope of removing some of the confusion.
>=20
> Regards
>=20
> Keith=20
>=20
> > -----Original Message-----
> > From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> > Sent: Monday, January 14, 2008 4:58 PM
> > To: sip@ietf.org
> > Subject: [Sip] Delivering request-URI and parameters to UAS=20
> via proxy
> >=20
> > (As WG chair)
> >=20
> > In fulfilment of our charter items of
> >=20
> > Dec 2007    Delivering request-URI and parameters to UAS=20
> via proxy to
> > WGLC =20
> > Feb 2008    Delivering request-URI and parameters to UAS=20
> via proxy to
> > IESG (PS)
> >=20
> > We now have a couple of proposals on the table for solving the=20
> > problem.
> >=20
> > The original draft from Jonathan and which led to the=20
> creation of the=20
> > charter items by the WG is unfortunately expired, but is at:
> >=20
> > http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
> >=20
> > The alternative document from Christer, etc is at:
> >=20
> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
> > uri-delive
> > ry-00.txt
> >=20
> > We obviously need to make a decision between the two approaches so=20
> > please attempt to address the following specific points via the=20
> > mailing
> > list:
> >=20
> > 1)	Problem cases: These are summarised in section 4.4 of
> > draft-rosenberg-sip-ua-loose-route-01 and from my read of the other=20
> > draft, I don't believe that this draft adds any others.
> > If you believe there are other cases that should be covered by the=20
> > solution, then please identify them. If there is support on any new=20
> > problem cases, I would encourage the authors of both drafts to add=20
> > text concerning these problem cases.
> >=20
> > 2)	Clarifications: If for any reason you don't understand either
> > draft, or believe that there are technical issues that are not=20
> > represented in the current draft, please post your questions /=20
> > comments to the list. I would encourage authors of both drafts to=20
> > revise as frequently as appropriate to reflect the current state of=20
> > discussion.
> >=20
> > 3)	Support for either position. If you wish to indicate support for
> > either position please do so, but please accompany this is=20
> technical=20
> > reasoning as to why you have this position, as that will help other=20
> > members of the WG form a position.
> >=20
> > I would encourage as much list discussion as possible before=20
> > Philadelphia. I suspect we will need to have a discussion at the=20
> > face-to-face meeting in Philadephia, but list discussion is=20
> essential=20
> > prior to that. If we can solve this positions on list, then=20
> well and=20
> > good, and even better (that is how we are meant to make decisions).
> >=20
> > Regards
> >=20
> >=20
> > Keith
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



From sip-bounces@ietf.org Wed Jan 16 06:51:18 2008
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
	-new version of the draft-holmberg-sip-target-uri-delivery draft
Date: Wed, 16 Jan 2008 12:50:54 +0100
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References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com><CA9998CD4A020D418654FCDEF4E707DF040266B1@esealmw113.eemea.ericsson.se><47878B1E.3010303@cisco.com><0D5F89FAC29E2C41B98A6A762007F5D0549A47@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1428F69B@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF04051C9D@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1428F846@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF040960B7@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	<sip@ietf.org>
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	Francois Audet <audet@nortel.com>
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Hi,=20

>I hear what you are saying, but I don't really see that the points you
raised in section 4 are real limitations of the loose route mechanism.

I think it is a limitation that an entity using the mechanism has to
know whether the the next hop supports it or not, and that the mechanism
can only be used if the subsequent hops support it.

Regards,

Christer




> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 16 January 2008 11:06
> > To: Elwell, John; sip@ietf.org
> > Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> > Subject: [Sip] RE: Delivering request-URI and parameters to UAS via=20
> > proxy - new version of the draft-holmberg-sip-target-uri-delivery=20
> > draft
> >=20
> >=20
> > Hi John,
> >=20
> > >Thanks for this revision, which makes things somewhat=20
> clearer. I do=20
> > >have a couple of comments:
> > >=20
> > >1. I am not sure I agree with the assertions in section 4=20
> concerning=20
> > >issues with the mechanisms in loose-route. Taking example 1, the=20
> > >Route header field should contain enough entries to get you to the=20
> > >registered contact, not just to an intermediate proxy.=20
> Therefore this=20
> > >situation should not arise with a correctly implemented=20
> home proxy.=20
> > >It is not clear to me how example 2 could arise either,=20
> for similar=20
> > >reasons. The MGC case can be resolved by taking into account the=20
> > >option tag in the REGISTER request, or if it is permanently=20
> > >registered, through provisioning.
> >=20
> > The examples are not meant to show bugs in the loose-route=20
> mechanism.
> > They are meant to help people understand the limitations with the=20
> > loose-route mechanism.
> > =20
> > >2. Comparing the mechanism proposed with the loose-route=20
> mechanism,=20
> > >my understanding is:
> > >a)When retargeting occurs, the loose-route mechanism=20
> places the new=20
> > >target in the Request URI. Your proposal places the new target in=20
> > >both the Request-URI and the Target header field.
> > >b) When rerouting, the loose-route mechanism places the new route=20
> > >(i.e., the registered contact) in the Route header field. Your=20
> > >proposal places the new route in the Request-URI (the=20
> latter as per=20
> > >RFC 3261).
> > >So the two mechanisms solve exactly the same problems using a=20
> > >slightly different mechanism. Correct?
> >=20
> > Yes. The two solutions intend to solve the same problem.
> > =20
> > >3. How P-Called-Party-ID fits into this is not really=20
> relevant from=20
> > >an IETF perspective - it seems there are some 3GPP-specific=20
> > >situations where the contents of P-Called-Party-ID will=20
> not equal the=20
> > >contents of Target. Correct?
> >=20
> > Correct.
> >=20
> > >4. If my suggestion in point 1 above that the loose-route=20
> mechanism=20
> > >does not suffer from the problems suggested, then each=20
> mechanism will=20
> > >work and each addresses the same problem.
> > >So it is just down to a beauty contest between the two. Correct?
> >=20
> > See question 1.
> >=20
> > We believe that our solution does not have the same=20
> limitations as the=20
> > loose-route solution. But, again, both solutions intend to=20
> solve the=20
> > same problem.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> > > > -----Original Message-----
> > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > > Sent: 16 January 2008 08:41
> > > > To: sip@ietf.org
> > > > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;=20
> Jeroen van=20
> > > > Bemmel; Francois Audet
> > > > Subject: Delivering request-URI and parameters to UAS via
> > > proxy - new
> > > > version of the draft-holmberg-sip-target-uri-delivery draft
> > > >=20
> > > >=20
> > > > Hi,
> > > >=20
> > > > We've uploaded a new version (-01) of the Target draft.
> > > >=20
> > > > We've tried to make things more clear. I've also=20
> removed all text=20
> > > > about P-Called-Party-ID, except from one chapter where=20
> we try to=20
> > > > explain the semantical difference between Target and P-CPI.
> > > >=20
> > > > You can also find the draft from:
> > > > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > > et-uri-del
> > > > ivery-01.txt
> > > >=20
> > > > Regards,
> > > >=20
> > > > Christer
> > > >=20
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



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To: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
	-new version of the draft-holmberg-sip-target-uri-delivery draft
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com><CA9998CD4A020D418654FCDEF4E707DF04051C9D@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1428F846@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF040960B7@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>
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Christer Holmberg wrote:
> Hi, 
> 
>> I hear what you are saying, but I don't really see that the points you
> raised in section 4 are real limitations of the loose route mechanism.
> 
> I think it is a limitation that an entity using the mechanism has to
> know whether the the next hop supports it or not, and that the mechanism
> can only be used if the subsequent hops support it.

In theory it is a problem. Its not at all clear that it is a problem in 
practice. That depends on whether this *will* be known in the places 
where the mechanism is used.

AFAIK the loose route draft requires that the behavior of the next hop 
*be* known before the mechanism is used. So raising issues based on 
problems if it is used inappropriately are not valid. What *would* be 
valid are places where the mechanism is not used because support it not 
known, resulting is less functionality. I remain to be convinced there 
are any such cases.

So I think these two mechanisms are getting close to be just being in a 
beauty contest.

I think the crucial point in both of these is deciding when you are 
retargeting vs rerouting.

	Paul

> Regards,
> 
> Christer
> 
> 
> 
> 
>>> -----Original Message-----
>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>> Sent: 16 January 2008 11:06
>>> To: Elwell, John; sip@ietf.org
>>> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
>>> Subject: [Sip] RE: Delivering request-URI and parameters to UAS via 
>>> proxy - new version of the draft-holmberg-sip-target-uri-delivery 
>>> draft
>>>
>>>
>>> Hi John,
>>>
>>>> Thanks for this revision, which makes things somewhat 
>> clearer. I do 
>>>> have a couple of comments:
>>>>
>>>> 1. I am not sure I agree with the assertions in section 4 
>> concerning 
>>>> issues with the mechanisms in loose-route. Taking example 1, the 
>>>> Route header field should contain enough entries to get you to the 
>>>> registered contact, not just to an intermediate proxy. 
>> Therefore this 
>>>> situation should not arise with a correctly implemented 
>> home proxy. 
>>>> It is not clear to me how example 2 could arise either, 
>> for similar 
>>>> reasons. The MGC case can be resolved by taking into account the 
>>>> option tag in the REGISTER request, or if it is permanently 
>>>> registered, through provisioning.
>>> The examples are not meant to show bugs in the loose-route 
>> mechanism.
>>> They are meant to help people understand the limitations with the 
>>> loose-route mechanism.
>>>  
>>>> 2. Comparing the mechanism proposed with the loose-route 
>> mechanism, 
>>>> my understanding is:
>>>> a)When retargeting occurs, the loose-route mechanism 
>> places the new 
>>>> target in the Request URI. Your proposal places the new target in 
>>>> both the Request-URI and the Target header field.
>>>> b) When rerouting, the loose-route mechanism places the new route 
>>>> (i.e., the registered contact) in the Route header field. Your 
>>>> proposal places the new route in the Request-URI (the 
>> latter as per 
>>>> RFC 3261).
>>>> So the two mechanisms solve exactly the same problems using a 
>>>> slightly different mechanism. Correct?
>>> Yes. The two solutions intend to solve the same problem.
>>>  
>>>> 3. How P-Called-Party-ID fits into this is not really 
>> relevant from 
>>>> an IETF perspective - it seems there are some 3GPP-specific 
>>>> situations where the contents of P-Called-Party-ID will 
>> not equal the 
>>>> contents of Target. Correct?
>>> Correct.
>>>
>>>> 4. If my suggestion in point 1 above that the loose-route 
>> mechanism 
>>>> does not suffer from the problems suggested, then each 
>> mechanism will 
>>>> work and each addresses the same problem.
>>>> So it is just down to a beauty contest between the two. Correct?
>>> See question 1.
>>>
>>> We believe that our solution does not have the same 
>> limitations as the 
>>> loose-route solution. But, again, both solutions intend to 
>> solve the 
>>> same problem.
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>>
>>>
>>>
>>>>> -----Original Message-----
>>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>>> Sent: 16 January 2008 08:41
>>>>> To: sip@ietf.org
>>>>> Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John; 
>> Jeroen van 
>>>>> Bemmel; Francois Audet
>>>>> Subject: Delivering request-URI and parameters to UAS via
>>>> proxy - new
>>>>> version of the draft-holmberg-sip-target-uri-delivery draft
>>>>>
>>>>>
>>>>> Hi,
>>>>>
>>>>> We've uploaded a new version (-01) of the Target draft.
>>>>>
>>>>> We've tried to make things more clear. I've also 
>> removed all text 
>>>>> about P-Called-Party-ID, except from one chapter where 
>> we try to 
>>>>> explain the semantical difference between Target and P-CPI.
>>>>>
>>>>> You can also find the draft from:
>>>>> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
>>>>> et-uri-del
>>>>> ivery-01.txt
>>>>>
>>>>> Regards,
>>>>>
>>>>> Christer
>>>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip 
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Hi there,
Can some one suggest me info on SIP Routers.
Are there any SIP Routers available in the market today?
=20
Regards
Kamal
=20

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<HTML dir=3Dltr><HEAD><TITLE>Re: [Sip] RE: Delivering request-URI and =
parameters to UAS via proxy-new version of the =
draft-holmberg-sip-target-uri-delivery draft</TITLE>=0A=
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dunicode">=0A=
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there,</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Can some one suggest me info =
on SIP Routers.</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Are there any SIP Routers =
available in the market today?</FONT></DIV>=0A=
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<DIV dir=3Dltr><FONT face=3DArial size=3D2>Regards</FONT></DIV>=0A=
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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
--===============1491679590==--





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	sip@ietf.org; Wed, 16 Jan 2008 21:30:24 +0800 (CST)
Date: Wed, 16 Jan 2008 19:00:22 +0530
From: Srinivas <srinivasj@huawei.com>
To: sip@ietf.org
Message-id: <00fb01c85843$f2be9ad0$7103120a@china.huawei.com>
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Subject: [Sip] regarding session modification
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Hi all,

Consider the following scenarios. 

 

1. 

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

 

Now the SIP session is established b/w UAC and UAS. speechrecog resource is active on the established sip session

 

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK 

Offer/Answer fail for the new resource, TTS (speechsynth).

 

 

2. 

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

 

Now the SIP session is established b/w UAC and UAS. speechrecog resource is active on the established sip session.

 

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK 

Offer/Answer fail for the for the existing ASR(speechrecog) resource also.

 

3.

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

 

Now the SIP session is established b/w UAC and UAS. speechrecog resource is active

 

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK

C->S: ACK

 

Now both speechrecog and speechsynth are active on the sip session

 

C->S: INVITE (re-INVITE for removing speechsynth resource)

S->C: 486 BUSY HERE

 

 

What should be the behaviour of UAC in the above scenarios?

 

 

Regds,

Srinivas

 

 


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@list l9
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	mso-level-tab-stop:255.1pt;
	mso-level-number-position:left;
	margin-left:255.1pt;
	text-indent:-85.0pt;}
ol
	{margin-bottom:0in;}
ul
	{margin-bottom:0in;}
-->
</style>
<!--[if gte mso 9]><xml>
 <o:shapedefaults v:ext=3D"edit" spidmax=3D"3074" />
</xml><![endif]--><!--[if gte mso 9]><xml>
 <o:shapelayout v:ext=3D"edit">
  <o:idmap v:ext=3D"edit" data=3D"2" />
 </o:shapelayout></xml><![endif]-->
</head>

<body lang=3DEN-US link=3Dblue vlink=3Dpurple =
style=3D'text-justify-trim:punctuation'>

<div class=3DSection1 style=3D'layout-grid:15.6pt'>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Hi =
all,<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Consider the following =
scenarios. <o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>1. =
<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: INVITE =
(for
speechrecog)<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>S-&gt;C: 200 =
OK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: =
ACK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Now the SIP session is =
established b/w
UAC and UAS. speechrecog resource is active on the established sip =
session<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: INVITE =
(re-INVITE
for adding speechsynth resource)<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>S-&gt;C: 200 OK =
<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Offer/Answer fail for the new =
resource,
TTS (speechsynth).<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>2. =
<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: INVITE =
(for
speechrecog)<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>S-&gt;C: 200 =
OK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: =
ACK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Now the SIP session is =
established b/w
UAC and UAS. speechrecog resource is active on the established sip =
session.<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: INVITE =
(re-INVITE
for adding speechsynth resource)<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>S-&gt;C: 200 OK =
<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Offer/Answer fail for the for =
the existing
ASR(speechrecog) resource also.<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>3.<o:p></o:p></s=
pan></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: INVITE =
(for
speechrecog)<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>S-&gt;C: 200 =
OK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: =
ACK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Now the SIP session is =
established b/w
UAC and UAS. speechrecog resource is active<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: INVITE =
(re-INVITE
for adding speechsynth resource)<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>S-&gt;C: 200 =
OK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: =
ACK<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Now both speechrecog and =
speechsynth
are active on the sip session<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>C-&gt;S: INVITE =
(re-INVITE
for removing speechsynth resource)<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><b><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333;font-weight:bold'>S-&gt;C: 486 =
BUSY HERE<o:p></o:p></span></font></b></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>What should be the behaviour of =
UAC in
the above scenarios?<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Regds,<o:p></o:p></span></font></=
p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:
150%;font-family:Verdana;color:#333333'>Srinivas<o:p></o:p></span></font>=
</p>

<p class=3DMsoNormal =
style=3D'margin-left:21.0pt;layout-grid-mode:char'><font
size=3D2 color=3D"#333333" face=3DVerdana><span =
style=3D'font-size:10.5pt;line-height:
150%;font-family:Verdana;color:#333333'><o:p>&nbsp;</o:p></span></font></=
p>

<p class=3DMsoNormal style=3D'margin-left:21.0pt'><font size=3D2 =
color=3D"#333333"
face=3DVerdana><span =
style=3D'font-size:10.0pt;line-height:150%;font-family:Verdana;
color:#333333'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

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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
--===============1827363563==--





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Subject: RE: [Sip] regarding session modification
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What is the header field that is changed in all re-INVITEs.
What is the value of this header field in each scenario?

________________________________

From: Srinivas [mailto:srinivasj@huawei.com]
Sent: Wed 1/16/2008 7:00 PM
To: sip@ietf.org
Subject: [Sip] regarding session modification



Hi all,

Consider the following scenarios.=20

=20

1.=20

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

=20

Now the SIP session is established b/w UAC and UAS. speechrecog resource =
is active on the established sip session

=20

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK=20

Offer/Answer fail for the new resource, TTS (speechsynth).

=20

=20

2.=20

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

=20

Now the SIP session is established b/w UAC and UAS. speechrecog resource =
is active on the established sip session.

=20

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK=20

Offer/Answer fail for the for the existing ASR(speechrecog) resource =
also.

=20

3.

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

=20

Now the SIP session is established b/w UAC and UAS. speechrecog resource =
is active

=20

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK

C->S: ACK

=20

Now both speechrecog and speechsynth are active on the sip session

=20

C->S: INVITE (re-INVITE for removing speechsynth resource)

S->C: 486 BUSY HERE

=20

=20

What should be the behaviour of UAC in the above scenarios?

=20

=20

Regds,

Srinivas

=20

=20


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<DIV id=3DidOWAReplyText45579 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>What is the =
header field that is changed in all re-INVITEs.</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>What is the value of this =
header field in each scenario?</FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>From:</B> Srinivas =
[mailto:srinivasj@huawei.com]<BR><B>Sent:</B> Wed 1/16/2008 7:00 =
PM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] regarding session =
modification<BR></FONT><BR></DIV>=0A=
<DIV>=0A=
<DIV class=3DSection1>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Hi all,</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Consider the following =
scenarios. </SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">1. =
</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
INVITE (for speechrecog)</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C: =
200 OK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
ACK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Now the SIP session is =
established b/w UAC and UAS. speechrecog resource is active on the =
established sip session</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
INVITE (re-INVITE for adding speechsynth resource)</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C: =
200 OK </SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Offer/Answer fail for the new =
resource, TTS (speechsynth).</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">2. =
</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
INVITE (for speechrecog)</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C: =
200 OK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
ACK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Now the SIP session is =
established b/w UAC and UAS. speechrecog resource is active on the =
established sip session.</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
INVITE (re-INVITE for adding speechsynth resource)</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C: =
200 OK </SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Offer/Answer fail for the for =
the existing ASR(speechrecog) resource also.</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: =
Verdana">3.</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
INVITE (for speechrecog)</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C: =
200 OK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
ACK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Now the SIP session is =
established b/w UAC and UAS. speechrecog resource is =
active</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
INVITE (re-INVITE for adding speechsynth resource)</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C: =
200 OK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
ACK</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Now both speechrecog and =
speechsynth are active on the sip session</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S: =
INVITE (re-INVITE for removing speechsynth =
resource)</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><B><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt; COLOR: #333333; LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C: =
486 BUSY HERE</SPAN></FONT></B></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">What should be the behaviour of =
UAC in the above scenarios?</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Regds,</SPAN></FONT></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">Srinivas</SPAN></FONT></P>=0A=
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color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10.5pt; COLOR: =
#333333; LINE-HEIGHT: 150%; FONT-FAMILY: =
Verdana"></SPAN></FONT>&nbsp;</P>=0A=
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333 size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: =
Verdana"></SPAN></FONT>&nbsp;</P></DIV></DIV></BODY></HTML>
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From sip-bounces@ietf.org Wed Jan 16 08:52:52 2008
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Date: Wed, 16 Jan 2008 13:52:32 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
	-new version of the draft-holmberg-sip-target-uri-delivery draft
In-reply-to: A
	<CA9998CD4A020D418654FCDEF4E707DF04173CB8@esealmw113.eemea.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>, sip@ietf.org
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Christer,

I don't think the entity using the mechanism needs to know about next
hop support for the loose route mechanism - only whether the UAS
supports it. If the UA registers, then the loose route draft provides an
automatic mechanism. If the UA does not register, support would need to
be indicated by provisioning. I am not sure this ranks as a significant
limitation, since a certain amount of provisioning needs to be done
anyway for UAs such as gateways that do not register.

John=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 16 January 2008 11:51
> To: Elwell, John; sip@ietf.org
> Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy -new version of the=20
> draft-holmberg-sip-target-uri-delivery draft
>=20
>=20
> Hi,=20
>=20
> >I hear what you are saying, but I don't really see that the=20
> points you
> raised in section 4 are real limitations of the loose route mechanism.
>=20
> I think it is a limitation that an entity using the mechanism has to
> know whether the the next hop supports it or not, and that=20
> the mechanism
> can only be used if the subsequent hops support it.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: 16 January 2008 11:06
> > > To: Elwell, John; sip@ietf.org
> > > Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> > > Subject: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via=20
> > > proxy - new version of the draft-holmberg-sip-target-uri-delivery=20
> > > draft
> > >=20
> > >=20
> > > Hi John,
> > >=20
> > > >Thanks for this revision, which makes things somewhat=20
> > clearer. I do=20
> > > >have a couple of comments:
> > > >=20
> > > >1. I am not sure I agree with the assertions in section 4=20
> > concerning=20
> > > >issues with the mechanisms in loose-route. Taking example 1, the=20
> > > >Route header field should contain enough entries to get=20
> you to the=20
> > > >registered contact, not just to an intermediate proxy.=20
> > Therefore this=20
> > > >situation should not arise with a correctly implemented=20
> > home proxy.=20
> > > >It is not clear to me how example 2 could arise either,=20
> > for similar=20
> > > >reasons. The MGC case can be resolved by taking into account the=20
> > > >option tag in the REGISTER request, or if it is permanently=20
> > > >registered, through provisioning.
> > >=20
> > > The examples are not meant to show bugs in the loose-route=20
> > mechanism.
> > > They are meant to help people understand the limitations with the=20
> > > loose-route mechanism.
> > > =20
> > > >2. Comparing the mechanism proposed with the loose-route=20
> > mechanism,=20
> > > >my understanding is:
> > > >a)When retargeting occurs, the loose-route mechanism=20
> > places the new=20
> > > >target in the Request URI. Your proposal places the new=20
> target in=20
> > > >both the Request-URI and the Target header field.
> > > >b) When rerouting, the loose-route mechanism places the=20
> new route=20
> > > >(i.e., the registered contact) in the Route header field. Your=20
> > > >proposal places the new route in the Request-URI (the=20
> > latter as per=20
> > > >RFC 3261).
> > > >So the two mechanisms solve exactly the same problems using a=20
> > > >slightly different mechanism. Correct?
> > >=20
> > > Yes. The two solutions intend to solve the same problem.
> > > =20
> > > >3. How P-Called-Party-ID fits into this is not really=20
> > relevant from=20
> > > >an IETF perspective - it seems there are some 3GPP-specific=20
> > > >situations where the contents of P-Called-Party-ID will=20
> > not equal the=20
> > > >contents of Target. Correct?
> > >=20
> > > Correct.
> > >=20
> > > >4. If my suggestion in point 1 above that the loose-route=20
> > mechanism=20
> > > >does not suffer from the problems suggested, then each=20
> > mechanism will=20
> > > >work and each addresses the same problem.
> > > >So it is just down to a beauty contest between the two. Correct?
> > >=20
> > > See question 1.
> > >=20
> > > We believe that our solution does not have the same=20
> > limitations as the=20
> > > loose-route solution. But, again, both solutions intend to=20
> > solve the=20
> > > same problem.
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > >=20
> > > > > -----Original Message-----
> > > > > From: Christer Holmberg=20
> [mailto:christer.holmberg@ericsson.com]
> > > > > Sent: 16 January 2008 08:41
> > > > > To: sip@ietf.org
> > > > > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;=20
> > Jeroen van=20
> > > > > Bemmel; Francois Audet
> > > > > Subject: Delivering request-URI and parameters to UAS via
> > > > proxy - new
> > > > > version of the draft-holmberg-sip-target-uri-delivery draft
> > > > >=20
> > > > >=20
> > > > > Hi,
> > > > >=20
> > > > > We've uploaded a new version (-01) of the Target draft.
> > > > >=20
> > > > > We've tried to make things more clear. I've also=20
> > removed all text=20
> > > > > about P-Called-Party-ID, except from one chapter where=20
> > we try to=20
> > > > > explain the semantical difference between Target and P-CPI.
> > > > >=20
> > > > > You can also find the draft from:
> > > > > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > > > et-uri-del
> > > > > ivery-01.txt
> > > > >=20
> > > > > Regards,
> > > > >=20
> > > > > Christer
> > > > >=20
> > > >=20
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip=20
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20


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From sip-bounces@ietf.org Wed Jan 16 09:32:54 2008
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
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	<0D5F89FAC29E2C41B98A6A762007F5D0593CFF@GBNTHT12009MSX.gb002.siemens.net>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	<sip@ietf.org>
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John,

According to the UA-loose-route draft, every proxy in the path need to
support the mechanism in order for the "current target" to be retained
in the R-URI. A good example is the "Service Invocation" use case in the
UA loose route draft.

Maybe the "loose route" wording is confusing (I think it is also
mentioned in the UA-loose-route draft that a better naming may be
needed)?

Also, maybe one can assume that a home proxy knows what entities not
registering (MGCs etc) support. But, again, we are not only talking
about home proxies.

Regards,

Christer=20





> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]=20
> Sent: 16. tammikuuta 2008 15:53
> To: Christer Holmberg; sip@ietf.org
> Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy -new version of the=20
> draft-holmberg-sip-target-uri-delivery draft
>=20
> Christer,
>=20
> I don't think the entity using the mechanism needs to know=20
> about next hop support for the loose route mechanism - only=20
> whether the UAS supports it. If the UA registers, then the=20
> loose route draft provides an automatic mechanism. If the UA=20
> does not register, support would need to be indicated by=20
> provisioning. I am not sure this ranks as a significant=20
> limitation, since a certain amount of provisioning needs to=20
> be done anyway for UAs such as gateways that do not register.
>=20
> John=20
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: 16 January 2008 11:51
> > To: Elwell, John; sip@ietf.org
> > Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS=20
> > via proxy -new version of the=20
> draft-holmberg-sip-target-uri-delivery=20
> > draft
> >=20
> >=20
> > Hi,
> >=20
> > >I hear what you are saying, but I don't really see that the
> > points you
> > raised in section 4 are real limitations of the loose route=20
> mechanism.
> >=20
> > I think it is a limitation that an entity using the=20
> mechanism has to=20
> > know whether the the next hop supports it or not, and that the=20
> > mechanism can only be used if the subsequent hops support it.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> > > > -----Original Message-----
> > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > > Sent: 16 January 2008 11:06
> > > > To: Elwell, John; sip@ietf.org
> > > > Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> > > > Subject: [Sip] RE: Delivering request-URI and parameters
> > to UAS via
> > > > proxy - new version of the=20
> draft-holmberg-sip-target-uri-delivery
> > > > draft
> > > >=20
> > > >=20
> > > > Hi John,
> > > >=20
> > > > >Thanks for this revision, which makes things somewhat
> > > clearer. I do
> > > > >have a couple of comments:
> > > > >=20
> > > > >1. I am not sure I agree with the assertions in section 4
> > > concerning
> > > > >issues with the mechanisms in loose-route. Taking=20
> example 1, the=20
> > > > >Route header field should contain enough entries to get
> > you to the
> > > > >registered contact, not just to an intermediate proxy.=20
> > > Therefore this
> > > > >situation should not arise with a correctly implemented
> > > home proxy.=20
> > > > >It is not clear to me how example 2 could arise either,
> > > for similar
> > > > >reasons. The MGC case can be resolved by taking into=20
> account the=20
> > > > >option tag in the REGISTER request, or if it is permanently=20
> > > > >registered, through provisioning.
> > > >=20
> > > > The examples are not meant to show bugs in the loose-route
> > > mechanism.
> > > > They are meant to help people understand the=20
> limitations with the=20
> > > > loose-route mechanism.
> > > > =20
> > > > >2. Comparing the mechanism proposed with the loose-route
> > > mechanism,
> > > > >my understanding is:
> > > > >a)When retargeting occurs, the loose-route mechanism
> > > places the new
> > > > >target in the Request URI. Your proposal places the new
> > target in
> > > > >both the Request-URI and the Target header field.
> > > > >b) When rerouting, the loose-route mechanism places the
> > new route
> > > > >(i.e., the registered contact) in the Route header field. Your=20
> > > > >proposal places the new route in the Request-URI (the
> > > latter as per
> > > > >RFC 3261).
> > > > >So the two mechanisms solve exactly the same problems using a=20
> > > > >slightly different mechanism. Correct?
> > > >=20
> > > > Yes. The two solutions intend to solve the same problem.
> > > > =20
> > > > >3. How P-Called-Party-ID fits into this is not really
> > > relevant from
> > > > >an IETF perspective - it seems there are some 3GPP-specific=20
> > > > >situations where the contents of P-Called-Party-ID will
> > > not equal the
> > > > >contents of Target. Correct?
> > > >=20
> > > > Correct.
> > > >=20
> > > > >4. If my suggestion in point 1 above that the loose-route
> > > mechanism
> > > > >does not suffer from the problems suggested, then each
> > > mechanism will
> > > > >work and each addresses the same problem.
> > > > >So it is just down to a beauty contest between the=20
> two. Correct?
> > > >=20
> > > > See question 1.
> > > >=20
> > > > We believe that our solution does not have the same
> > > limitations as the
> > > > loose-route solution. But, again, both solutions intend to
> > > solve the
> > > > same problem.
> > > >=20
> > > > Regards,
> > > >=20
> > > > Christer
> > > >=20
> > > >=20
> > > >=20
> > > >=20
> > > >=20
> > > >=20
> > > > > > -----Original Message-----
> > > > > > From: Christer Holmberg
> > [mailto:christer.holmberg@ericsson.com]
> > > > > > Sent: 16 January 2008 08:41
> > > > > > To: sip@ietf.org
> > > > > > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;
> > > Jeroen van
> > > > > > Bemmel; Francois Audet
> > > > > > Subject: Delivering request-URI and parameters to UAS via
> > > > > proxy - new
> > > > > > version of the draft-holmberg-sip-target-uri-delivery draft
> > > > > >=20
> > > > > >=20
> > > > > > Hi,
> > > > > >=20
> > > > > > We've uploaded a new version (-01) of the Target draft.
> > > > > >=20
> > > > > > We've tried to make things more clear. I've also
> > > removed all text
> > > > > > about P-Called-Party-ID, except from one chapter where
> > > we try to
> > > > > > explain the semantical difference between Target and P-CPI.
> > > > > >=20
> > > > > > You can also find the draft from:
> > > > > >=20
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > > > > et-uri-del
> > > > > > ivery-01.txt
> > > > > >=20
> > > > > > Regards,
> > > > > >=20
> > > > > > Christer
> > > > > >=20
> > > > >=20
> > > >=20
> > > >=20
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > > >=20
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
>=20


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



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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
	-new version of the draft-holmberg-sip-target-uri-delivery draft
Date: Wed, 16 Jan 2008 15:44:18 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi Paul,=20

>>>I hear what you are saying, but I don't really see that the points
you
>>>raised in section 4 are real limitations of the loose route
mechanism.
>>=20
>>I think it is a limitation that an entity using the mechanism has to=20
>>know whether the the next hop supports it or not, and that the=20
>>mechanism can only be used if the subsequent hops support it.
>=20
>In theory it is a problem. Its not at all clear that it is a=20
>problem in practice. That depends on whether this *will* be=20
>known in the places where the mechanism is used.

I don't think it's a theoretical problem.

You may have next hop entities in different domains, belonging to
different operators, assigned dynamically for each call (using DNS, load
sharing etc), being updated with more/less capabilities very frequencly
etc etc etc. So, I think there are lots of scenarios where it will
simply be impossible for an entity to know what the next physical hop
will support.

Yes, an UA can indicate to its home proxy what it supports, but again I
think it is important that we are not only talking about home proxies.

And, if we are going to start making protocol design based on the fact
that entities know what other entities support, why do we need
option-tags, why do we need Require, why do we need SDP? Why don't we
just assume everything is provisioned...

I thought the idea behind SIP, and the whole internet model, was to NOT
rely on provisioning.

>AFAIK the loose route draft requires that the behavior of the next hop
>*be* known before the mechanism is used.
>
>So raising issues based on problems if it is used inappropriately are
not valid.

Again, the examples are just there to explain the limitation.

Also, the issue about possible services that may break is not related to
inappropriately use of the mechanism. The entity that does trigger the
service relying on the mechanism may very well know that the next hop
supports it. But, the hop after the next hop, and/or the hop after the
hop of the next hop, may not support it. So, somewhere in the chain the
service relying on the mechanism will be "broken" - even if the hop that
doesn't support the mechanism has nothing to do with the service as
such.

We think that limits the usability of the mechanism (again it has
nothing to do with inappropriately use of the mechanism).

>What *would* be valid are places where the mechanism is not used
because support it not known, resulting is less=20
>functionality. I remain to be convinced there are any such cases.
>=20
>So I think these two mechanisms are getting close to be just being in a
beauty contest.
>=20
>I think the crucial point in both of these is deciding when you are
retargeting vs rerouting.

Yes, I agree. We should get a clear defintion, once and for all.

Regards,

Christer





> >=20
> >=20
> >=20
> >=20
> >>> -----Original Message-----
> >>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>> Sent: 16 January 2008 11:06
> >>> To: Elwell, John; sip@ietf.org
> >>> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> >>> Subject: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via=20
> >>> proxy - new version of the draft-holmberg-sip-target-uri-delivery
> >>> draft
> >>>
> >>>
> >>> Hi John,
> >>>
> >>>> Thanks for this revision, which makes things somewhat
> >> clearer. I do
> >>>> have a couple of comments:
> >>>>
> >>>> 1. I am not sure I agree with the assertions in section 4
> >> concerning
> >>>> issues with the mechanisms in loose-route. Taking example 1, the=20
> >>>> Route header field should contain enough entries to get=20
> you to the=20
> >>>> registered contact, not just to an intermediate proxy.
> >> Therefore this
> >>>> situation should not arise with a correctly implemented
> >> home proxy.=20
> >>>> It is not clear to me how example 2 could arise either,
> >> for similar
> >>>> reasons. The MGC case can be resolved by taking into account the=20
> >>>> option tag in the REGISTER request, or if it is permanently=20
> >>>> registered, through provisioning.
> >>> The examples are not meant to show bugs in the loose-route
> >> mechanism.
> >>> They are meant to help people understand the limitations with the=20
> >>> loose-route mechanism.
> >>> =20
> >>>> 2. Comparing the mechanism proposed with the loose-route
> >> mechanism,
> >>>> my understanding is:
> >>>> a)When retargeting occurs, the loose-route mechanism
> >> places the new
> >>>> target in the Request URI. Your proposal places the new=20
> target in=20
> >>>> both the Request-URI and the Target header field.
> >>>> b) When rerouting, the loose-route mechanism places the=20
> new route=20
> >>>> (i.e., the registered contact) in the Route header field. Your=20
> >>>> proposal places the new route in the Request-URI (the
> >> latter as per
> >>>> RFC 3261).
> >>>> So the two mechanisms solve exactly the same problems using a=20
> >>>> slightly different mechanism. Correct?
> >>> Yes. The two solutions intend to solve the same problem.
> >>> =20
> >>>> 3. How P-Called-Party-ID fits into this is not really
> >> relevant from
> >>>> an IETF perspective - it seems there are some 3GPP-specific=20
> >>>> situations where the contents of P-Called-Party-ID will
> >> not equal the
> >>>> contents of Target. Correct?
> >>> Correct.
> >>>
> >>>> 4. If my suggestion in point 1 above that the loose-route
> >> mechanism
> >>>> does not suffer from the problems suggested, then each
> >> mechanism will
> >>>> work and each addresses the same problem.
> >>>> So it is just down to a beauty contest between the two. Correct?
> >>> See question 1.
> >>>
> >>> We believe that our solution does not have the same
> >> limitations as the
> >>> loose-route solution. But, again, both solutions intend to
> >> solve the
> >>> same problem.
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>>> -----Original Message-----
> >>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>>>> Sent: 16 January 2008 08:41
> >>>>> To: sip@ietf.org
> >>>>> Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;
> >> Jeroen van
> >>>>> Bemmel; Francois Audet
> >>>>> Subject: Delivering request-URI and parameters to UAS via
> >>>> proxy - new
> >>>>> version of the draft-holmberg-sip-target-uri-delivery draft
> >>>>>
> >>>>>
> >>>>> Hi,
> >>>>>
> >>>>> We've uploaded a new version (-01) of the Target draft.
> >>>>>
> >>>>> We've tried to make things more clear. I've also
> >> removed all text
> >>>>> about P-Called-Party-ID, except from one chapter where
> >> we try to
> >>>>> explain the semantical difference between Target and P-CPI.
> >>>>>
> >>>>> You can also find the draft from:
> >>>>> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> >>>>> et-uri-del
> >>>>> ivery-01.txt
> >>>>>
> >>>>> Regards,
> >>>>>
> >>>>> Christer
> >>>>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
> >>>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >>
> >=20
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Wed Jan 16 09:45:50 2008
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Message-ID: <478E1897.8030905@joelhalpern.com>
Date: Wed, 16 Jan 2008 09:45:43 -0500
From: "Joel M. Halpern" <jmh@joelhalpern.com>
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To: IETF SIP List <sip@ietf.org>
Subject: Re: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
	-new version of the draft-holmberg-sip-target-uri-delivery draft
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com><CA9998CD4A020D418654FCDEF4E707DF04051C9D@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1428F846@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF040960B7@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>	A<CA9998CD4A020D418654FCDEF4E707DF040D69C7@esealmw113.eemea.ericsson.se><0D5F89FAC29E2C41B98A6A762007F5D0549D3F@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1438F1B0@zrc2hxm0.corp.nortel.com><478CEFB4.6070002@zonnet.nl><CA9998CD4A020D418654FCDEF4E707DF0413D587@esealmw113.eemea.ericsson.se><"0D5F89FAC29E2C41B98A6A762007F5D0593C0E"@GBNTHT12009MSX.gb002.siemens.net>A<CA9998CD	4A020D41
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Maybe I am missing something simple.
But, as I said at the meeting when Jonathan's draft was discussed, the 
main problem I have with the proposed solution(s) relates to the 
distinction between retargeting and routing.  This distinction seems to 
be central to both proposals.
The description of the distinction looks clear enough in the abstract.
But when I try to analyze how a proxy would actually know which 
operation it is performing, it is not clear.  Sometimes I am sure it 
will "just work."  But the description, and thinking about the wide 
range of actual things proxies do, leaves me very uncertain as to the 
clarity.

So I would at least like to ask if other folks find the distinction 
sufficiently clear to be reliable for meeting the need.  My own hope had 
been that this process would either clarify the distinction or provide a 
different distinction which was clearer.

Again, I may be off in left field, but after reviewing the drafts again 
I felt I had to ask.

Joel



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



From sip-bounces@ietf.org Wed Jan 16 12:41:59 2008
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Date: Wed, 16 Jan 2008 18:43:50 +0100
From: Enrico Marocco <enrico.marocco@telecomitalia.it>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Delivering request-URI and parameters to UAS via proxy
	- new	version of the draft-holmberg-sip-target-uri-delivery draft
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com>	<CA9998CD4A020D418654FCDEF4E707DF040266B1@esealmw113.eemea.ericsson.se>	<47878B1E.3010303@cisco.com>	<0D5F89FAC29E2C41B98A6A762007F5D0549A47@GBNTHT12009MSX.gb002.siemens.net>	<1ECE0EB50388174790F9694F77522CCF1428F69B@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF04051C9D@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1428F846@zrc2hxm0.corp.nortel.com>	<CA9998CD4A020D418654FCDEF4E707DF040960B7@esealmw113.eemea.ericsson.se>	<1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>	A	<CA9998CD4A020D418654FCDEF4E707DF040D69C7@esealmw113.eemea.ericsson.se>	<0D5F89FAC29E2C41B98A6A762007F5D0549D3F@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1438F1B0@zrc2hxm0.corp.nortel.com><478CEFB4.6070002@zonnet.nl>	<CA9998CD4A020D418654FCDEF4E707DF0413D587@esealmw113.eemea.ericsson.se>
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Christer Holmberg wrote:
> We've uploaded a new version (-01) of the Target draft.
> 
> We've tried to make things more clear. I've also removed all text about
> P-Called-Party-ID, except from one chapter where we try to explain the
> semantical difference between Target and P-CPI.

Thanks Christer, the new version is pretty clear about the difference
between Target and P-Called-Party-ID.  However, other than the semantic
difference, I'd also add that the Target header could be used in a
cross-domain registration case, when the registered contact for a
specific URI is an AOR itself targeting a different domain.  As far as I
know, such a case is uncommon in current SIP deployments, but it could
appear more frequently with P2PSIP overlays, e.g. when peers with
limited connectivity make themselves globally reachable using P2P URIs
(maybe temporary ones).

-- 
Ciao,
Enrico

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At 07:00 AM 1/16/2008, kamalakar.mergu@wipro.com wrote:
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>
>Hi there,
>Can some one suggest me info on SIP Routers.
>Are there any SIP Routers available in the market today?

Kamal

which OSI layer are you wanting these SIP Routers to be operating at 
(layer 3 or layer 7 or another layer)?

>
>Regards
>Kamal
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP 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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Thread-Topic: [Sip] regarding session modification
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From: "Zhang, Rocky Fan Wen \(Rocky\)" <fanwenz@alcatel-lucent.com>
To: "Srinivas" <srinivasj@huawei.com>, <sip@ietf.org>
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Hi,
=20
My concerns:
=20
1. SHOULD the called party reply with 200 OK if the negotiation fails?
2. Why the first two tries to add speechsynth resource failed but the
third success?
3. As my understanding, if the re-negotiation fails, it will not affect
the current media.
=20
=20
Best Regards,
=20
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Fanwen zhang
FS5000 CPE IOT, Alcatel-Lucent Qingdao, R&D
Tel: +86-532-88615493
Email:fanwenz@alcatel-lucent.com
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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

________________________________

From: Srinivas [mailto:srinivasj@huawei.com]=20
Sent: Wednesday, January 16, 2008 9:30 PM
To: sip@ietf.org
Subject: [Sip] regarding session modification



Hi all,

Consider the following scenarios.=20

=20

1.=20

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

=20

Now the SIP session is established b/w UAC and UAS. speechrecog resource
is active on the established sip session

=20

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK=20

Offer/Answer fail for the new resource, TTS (speechsynth).

=20

=20

2.=20

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

=20

Now the SIP session is established b/w UAC and UAS. speechrecog resource
is active on the established sip session.

=20

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK=20

Offer/Answer fail for the for the existing ASR(speechrecog) resource
also.

=20

3.

C->S: INVITE (for speechrecog)

S->C: 200 OK

C->S: ACK

=20

Now the SIP session is established b/w UAC and UAS. speechrecog resource
is active

=20

C->S: INVITE (re-INVITE for adding speechsynth resource)

S->C: 200 OK

C->S: ACK

=20

Now both speechrecog and speechsynth are active on the sip session

=20

C->S: INVITE (re-INVITE for removing speechsynth resource)

S->C: 486 BUSY HERE

=20

=20

What should be the behaviour of UAC in the above scenarios?

=20

=20

Regds,

Srinivas

=20

=20


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<BODY lang=3DEN-US style=3D"TEXT-JUSTIFY-TRIM: punctuation" =
vLink=3Dpurple link=3Dblue>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D515044500-17012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D515044500-17012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D515044500-17012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>My concerns:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D515044500-17012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D515044500-17012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>1. SHOULD the called party reply with 200 OK if =
the=20
negotiation fails?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D515044500-17012008><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>2. Why the first two tries to add speechsynth =
resource=20
failed but the third success?</FONT></SPAN></DIV>
<DIV><SPAN class=3D515044500-17012008><FONT face=3D&#23435;&#20307; =
color=3D#0000ff size=3D2>3. As my=20
understanding, if the re-negotiation fails, it will not affect the =
current=20
media.</FONT></SPAN></DIV>
<DIV align=3Dleft><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV align=3Dleft><FONT size=3D2>Best Regards,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT =
size=3D2>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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></D=
IV>
<DIV><FONT size=3D2>Fanwen zhang<BR>FS5000 CPE IOT, Alcatel-Lucent =
Qingdao,=20
R&amp;D<BR>Tel:=20
+86-532-88615493<BR>Email:fanwenz@alcatel-lucent.com</FONT></DIV>
<DIV><FONT =
size=3D2>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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></D=
IV>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Srinivas =
[mailto:srinivasj@huawei.com]=20
<BR><B>Sent:</B> Wednesday, January 16, 2008 9:30 PM<BR><B>To:</B>=20
sip@ietf.org<BR><B>Subject:</B> [Sip] regarding session=20
modification<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1 style=3D"LAYOUT-GRID:  15.6pt none">
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Hi=20
all,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Consider=20
the following scenarios. <o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">1.=20
<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
INVITE (for speechrecog)<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C:=20
200 OK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
ACK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Now=20
the SIP session is established b/w UAC and UAS. speechrecog resource is =
active=20
on the established sip session<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
INVITE (re-INVITE for adding speechsynth=20
resource)<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C:=20
200 OK <o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Offer/Answer=20
fail for the new resource, TTS =
(speechsynth).<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">2.=20
<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
INVITE (for speechrecog)<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C:=20
200 OK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
ACK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Now=20
the SIP session is established b/w UAC and UAS. speechrecog resource is =
active=20
on the established sip session.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
INVITE (re-INVITE for adding speechsynth=20
resource)<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C:=20
200 OK <o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Offer/Answer=20
fail for the for the existing ASR(speechrecog) resource=20
also.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: =
Verdana">3.<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
INVITE (for speechrecog)<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C:=20
200 OK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
ACK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Now=20
the SIP session is established b/w UAC and UAS. speechrecog resource is=20
active<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
INVITE (re-INVITE for adding speechsynth=20
resource)<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C:=20
200 OK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
ACK<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Now=20
both speechrecog and speechsynth are active on the sip=20
session<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">C-&gt;S:=20
INVITE (re-INVITE for removing speechsynth=20
resource)<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><B><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #333333; =
LINE-HEIGHT: 150%; FONT-FAMILY: Verdana">S-&gt;C:=20
486 BUSY HERE<o:p></o:p></SPAN></FONT></B></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">What=20
should be the behaviour of UAC in the above=20
scenarios?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Regds,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana">Srinivas<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt; LAYOUT-GRID-MODE: =
char"><FONT=20
face=3DVerdana color=3D#333333 size=3D2><SPAN=20
style=3D"FONT-SIZE: 10.5pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal style=3D"MARGIN-LEFT: 21pt"><FONT face=3DVerdana =
color=3D#333333=20
size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #333333; LINE-HEIGHT: 150%; =
FONT-FAMILY: =
Verdana"><o:p>&nbsp;</o:p></SPAN></FONT></P></DIV></BODY></HTML>

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   From: Srinivas <srinivasj@huawei.com>

   1. 

   C->S: INVITE (for speechrecog)

   S->C: 200 OK

   C->S: ACK

   Now the SIP session is established b/w UAC and UAS. speechrecog
   resource is active on the established sip session

   C->S: INVITE (re-INVITE for adding speechsynth resource)

   S->C: 200 OK 

   Offer/Answer fail for the new resource, TTS (speechsynth).

What?

OK, you don't specify whether the re-INVITE contains an SDP offer, but
it appears to do so as you've labeled it "for adding speechsynth
resource".  The S responds 200, but you then say "offer/answer fails
for the new resource".  If S determines that it cannot provide a
usable answer, then it must reject the re-INVITE with a failure
response.  The 200 response indicates offer/answer success.

Dale


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Date: Thu, 17 Jan 2008 07:58:54 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via
	proxy-new version of the draft-holmberg-sip-target-uri-delivery draft
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Christer,

We still seem to have a disconnect in our understanding. See below.

John=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 16 January 2008 14:32
> To: Elwell, John; sip@ietf.org
> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy-new version of the=20
> draft-holmberg-sip-target-uri-delivery draft
>=20
>=20
> John,
>=20
> According to the UA-loose-route draft, every proxy in the path need to
> support the mechanism in order for the "current target" to be retained
> in the R-URI. A good example is the "Service Invocation" use=20
> case in the
> UA loose route draft.
[JRE] The Service Invocation section in loose-route tells me nothing
about proxies on the path. It simply talks about invocation of a service
at the UAS.

>=20
> Maybe the "loose route" wording is confusing (I think it is also
> mentioned in the UA-loose-route draft that a better naming may be
> needed)?
>=20
> Also, maybe one can assume that a home proxy knows what entities not
> registering (MGCs etc) support. But, again, we are not only talking
> about home proxies.
[JRE] Surely it is only the home proxy that takes an AoR and determines
a contact URI to route to. Maybe my understanding of home proxy is
different.

>=20
> Regards,
>=20
> Christer=20
>=20
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com]=20
> > Sent: 16. tammikuuta 2008 15:53
> > To: Christer Holmberg; sip@ietf.org
> > Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> > to UAS via proxy -new version of the=20
> > draft-holmberg-sip-target-uri-delivery draft
> >=20
> > Christer,
> >=20
> > I don't think the entity using the mechanism needs to know=20
> > about next hop support for the loose route mechanism - only=20
> > whether the UAS supports it. If the UA registers, then the=20
> > loose route draft provides an automatic mechanism. If the UA=20
> > does not register, support would need to be indicated by=20
> > provisioning. I am not sure this ranks as a significant=20
> > limitation, since a certain amount of provisioning needs to=20
> > be done anyway for UAs such as gateways that do not register.
> >=20
> > John=20
> >=20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: 16 January 2008 11:51
> > > To: Elwell, John; sip@ietf.org
> > > Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > > Subject: RE: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> > > via proxy -new version of the=20
> > draft-holmberg-sip-target-uri-delivery=20
> > > draft
> > >=20
> > >=20
> > > Hi,
> > >=20
> > > >I hear what you are saying, but I don't really see that the
> > > points you
> > > raised in section 4 are real limitations of the loose route=20
> > mechanism.
> > >=20
> > > I think it is a limitation that an entity using the=20
> > mechanism has to=20
> > > know whether the the next hop supports it or not, and that the=20
> > > mechanism can only be used if the subsequent hops support it.
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > >=20
> > >=20
> > >=20
> > > > > -----Original Message-----
> > > > > From: Christer Holmberg=20
> [mailto:christer.holmberg@ericsson.com]
> > > > > Sent: 16 January 2008 11:06
> > > > > To: Elwell, John; sip@ietf.org
> > > > > Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> > > > > Subject: [Sip] RE: Delivering request-URI and parameters
> > > to UAS via
> > > > > proxy - new version of the=20
> > draft-holmberg-sip-target-uri-delivery
> > > > > draft
> > > > >=20
> > > > >=20
> > > > > Hi John,
> > > > >=20
> > > > > >Thanks for this revision, which makes things somewhat
> > > > clearer. I do
> > > > > >have a couple of comments:
> > > > > >=20
> > > > > >1. I am not sure I agree with the assertions in section 4
> > > > concerning
> > > > > >issues with the mechanisms in loose-route. Taking=20
> > example 1, the=20
> > > > > >Route header field should contain enough entries to get
> > > you to the
> > > > > >registered contact, not just to an intermediate proxy.=20
> > > > Therefore this
> > > > > >situation should not arise with a correctly implemented
> > > > home proxy.=20
> > > > > >It is not clear to me how example 2 could arise either,
> > > > for similar
> > > > > >reasons. The MGC case can be resolved by taking into=20
> > account the=20
> > > > > >option tag in the REGISTER request, or if it is permanently=20
> > > > > >registered, through provisioning.
> > > > >=20
> > > > > The examples are not meant to show bugs in the loose-route
> > > > mechanism.
> > > > > They are meant to help people understand the=20
> > limitations with the=20
> > > > > loose-route mechanism.
> > > > > =20
> > > > > >2. Comparing the mechanism proposed with the loose-route
> > > > mechanism,
> > > > > >my understanding is:
> > > > > >a)When retargeting occurs, the loose-route mechanism
> > > > places the new
> > > > > >target in the Request URI. Your proposal places the new
> > > target in
> > > > > >both the Request-URI and the Target header field.
> > > > > >b) When rerouting, the loose-route mechanism places the
> > > new route
> > > > > >(i.e., the registered contact) in the Route header=20
> field. Your=20
> > > > > >proposal places the new route in the Request-URI (the
> > > > latter as per
> > > > > >RFC 3261).
> > > > > >So the two mechanisms solve exactly the same=20
> problems using a=20
> > > > > >slightly different mechanism. Correct?
> > > > >=20
> > > > > Yes. The two solutions intend to solve the same problem.
> > > > > =20
> > > > > >3. How P-Called-Party-ID fits into this is not really
> > > > relevant from
> > > > > >an IETF perspective - it seems there are some 3GPP-specific=20
> > > > > >situations where the contents of P-Called-Party-ID will
> > > > not equal the
> > > > > >contents of Target. Correct?
> > > > >=20
> > > > > Correct.
> > > > >=20
> > > > > >4. If my suggestion in point 1 above that the loose-route
> > > > mechanism
> > > > > >does not suffer from the problems suggested, then each
> > > > mechanism will
> > > > > >work and each addresses the same problem.
> > > > > >So it is just down to a beauty contest between the=20
> > two. Correct?
> > > > >=20
> > > > > See question 1.
> > > > >=20
> > > > > We believe that our solution does not have the same
> > > > limitations as the
> > > > > loose-route solution. But, again, both solutions intend to
> > > > solve the
> > > > > same problem.
> > > > >=20
> > > > > Regards,
> > > > >=20
> > > > > Christer
> > > > >=20
> > > > >=20
> > > > >=20
> > > > >=20
> > > > >=20
> > > > >=20
> > > > > > > -----Original Message-----
> > > > > > > From: Christer Holmberg
> > > [mailto:christer.holmberg@ericsson.com]
> > > > > > > Sent: 16 January 2008 08:41
> > > > > > > To: sip@ietf.org
> > > > > > > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;
> > > > Jeroen van
> > > > > > > Bemmel; Francois Audet
> > > > > > > Subject: Delivering request-URI and parameters to UAS via
> > > > > > proxy - new
> > > > > > > version of the=20
> draft-holmberg-sip-target-uri-delivery draft
> > > > > > >=20
> > > > > > >=20
> > > > > > > Hi,
> > > > > > >=20
> > > > > > > We've uploaded a new version (-01) of the Target draft.
> > > > > > >=20
> > > > > > > We've tried to make things more clear. I've also
> > > > removed all text
> > > > > > > about P-Called-Party-ID, except from one chapter where
> > > > we try to
> > > > > > > explain the semantical difference between Target=20
> and P-CPI.
> > > > > > >=20
> > > > > > > You can also find the draft from:
> > > > > > >=20
> > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > > > > > et-uri-del
> > > > > > > ivery-01.txt
> > > > > > >=20
> > > > > > > Regards,
> > > > > > >=20
> > > > > > > Christer
> > > > > > >=20
> > > > > >=20
> > > > >=20
> > > > >=20
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > > sip-implementors@cs.columbia.edu for questions on=20
> > current sip Use=20
> > > > > sipping@ietf.org for new developments on the=20
> application of sip
> > > > >=20
> > > >=20
> > > >=20
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > > >=20
> > >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via
	proxy-new version of the draft-holmberg-sip-target-uri-delivery draft
Date: Thu, 17 Jan 2008 11:54:34 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	<sip@ietf.org>
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>>According to the UA-loose-route draft, every proxy in the path need to

>>support the mechanism in order for the "current target" to be retained

>>in the R-URI. A good example is the "Service Invocation" use case in=20
>>the UA loose route draft.
>[JRE] The Service Invocation section in loose-route tells me=20
>nothing about proxies on the path. It simply talks about=20
>invocation of a service at the UAS.

This is what the loose draft says:

"5. Proposed Solution

   The proposed solution is simple.  When handling a request, a proxy
   only rewrites the Request-URI when performing a retargeting
   operation.  If, instead, the proxy is trying to route the request via
   some entity (whether its a proxy or UA) to reach the target, the
   Request-URI is retained, and Route header fields are pushed into the
   request to reach the target.

etc."

This applies to ALL proxies on the path, otherwise one will not always
find the current target (service invocation uri) in the Request-URI.=20
=20
>>Maybe the "loose route" wording is confusing (I think it is also=20
>>mentioned in the UA-loose-route draft that a better naming may be=20
>>needed)?
>>=20
>>Also, maybe one can assume that a home proxy knows what entities not=20
>>registering (MGCs etc) support. But, again, we are not only talking=20
>>about home proxies.
>[JRE] Surely it is only the home proxy that takes an AoR and determines
a contact URI to route to. Maybe my understanding of home proxy is
different.

So, what is your understanding?

Regards,

Christer




> >=20
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > Sent: 16. tammikuuta 2008 15:53
> > > To: Christer Holmberg; sip@ietf.org
> > > Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > > Subject: RE: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> > > via proxy -new version of the=20
> draft-holmberg-sip-target-uri-delivery=20
> > > draft
> > >=20
> > > Christer,
> > >=20
> > > I don't think the entity using the mechanism needs to know about=20
> > > next hop support for the loose route mechanism - only whether the=20
> > > UAS supports it. If the UA registers, then the loose route draft=20
> > > provides an automatic mechanism. If the UA does not register,=20
> > > support would need to be indicated by provisioning. I am not sure=20
> > > this ranks as a significant limitation, since a certain amount of=20
> > > provisioning needs to be done anyway for UAs such as=20
> gateways that=20
> > > do not register.
> > >=20
> > > John
> > >=20
> > > > -----Original Message-----
> > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > > Sent: 16 January 2008 11:51
> > > > To: Elwell, John; sip@ietf.org
> > > > Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > > > Subject: RE: [Sip] RE: Delivering request-URI and
> > parameters to UAS
> > > > via proxy -new version of the
> > > draft-holmberg-sip-target-uri-delivery
> > > > draft
> > > >=20
> > > >=20
> > > > Hi,
> > > >=20
> > > > >I hear what you are saying, but I don't really see that the
> > > > points you
> > > > raised in section 4 are real limitations of the loose route
> > > mechanism.
> > > >=20
> > > > I think it is a limitation that an entity using the
> > > mechanism has to
> > > > know whether the the next hop supports it or not, and that the=20
> > > > mechanism can only be used if the subsequent hops support it.
> > > >=20
> > > > Regards,
> > > >=20
> > > > Christer
> > > >=20
> > > >=20
> > > >=20
> > > >=20
> > > > > > -----Original Message-----
> > > > > > From: Christer Holmberg
> > [mailto:christer.holmberg@ericsson.com]
> > > > > > Sent: 16 January 2008 11:06
> > > > > > To: Elwell, John; sip@ietf.org
> > > > > > Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> > > > > > Subject: [Sip] RE: Delivering request-URI and parameters
> > > > to UAS via
> > > > > > proxy - new version of the
> > > draft-holmberg-sip-target-uri-delivery
> > > > > > draft
> > > > > >=20
> > > > > >=20
> > > > > > Hi John,
> > > > > >=20
> > > > > > >Thanks for this revision, which makes things somewhat
> > > > > clearer. I do
> > > > > > >have a couple of comments:
> > > > > > >=20
> > > > > > >1. I am not sure I agree with the assertions in section 4
> > > > > concerning
> > > > > > >issues with the mechanisms in loose-route. Taking
> > > example 1, the
> > > > > > >Route header field should contain enough entries to get
> > > > you to the
> > > > > > >registered contact, not just to an intermediate proxy.=20
> > > > > Therefore this
> > > > > > >situation should not arise with a correctly implemented
> > > > > home proxy.=20
> > > > > > >It is not clear to me how example 2 could arise either,
> > > > > for similar
> > > > > > >reasons. The MGC case can be resolved by taking into
> > > account the
> > > > > > >option tag in the REGISTER request, or if it is=20
> permanently=20
> > > > > > >registered, through provisioning.
> > > > > >=20
> > > > > > The examples are not meant to show bugs in the loose-route
> > > > > mechanism.
> > > > > > They are meant to help people understand the
> > > limitations with the
> > > > > > loose-route mechanism.
> > > > > > =20
> > > > > > >2. Comparing the mechanism proposed with the loose-route
> > > > > mechanism,
> > > > > > >my understanding is:
> > > > > > >a)When retargeting occurs, the loose-route mechanism
> > > > > places the new
> > > > > > >target in the Request URI. Your proposal places the new
> > > > target in
> > > > > > >both the Request-URI and the Target header field.
> > > > > > >b) When rerouting, the loose-route mechanism places the
> > > > new route
> > > > > > >(i.e., the registered contact) in the Route header
> > field. Your
> > > > > > >proposal places the new route in the Request-URI (the
> > > > > latter as per
> > > > > > >RFC 3261).
> > > > > > >So the two mechanisms solve exactly the same
> > problems using a
> > > > > > >slightly different mechanism. Correct?
> > > > > >=20
> > > > > > Yes. The two solutions intend to solve the same problem.
> > > > > > =20
> > > > > > >3. How P-Called-Party-ID fits into this is not really
> > > > > relevant from
> > > > > > >an IETF perspective - it seems there are some=20
> 3GPP-specific=20
> > > > > > >situations where the contents of P-Called-Party-ID will
> > > > > not equal the
> > > > > > >contents of Target. Correct?
> > > > > >=20
> > > > > > Correct.
> > > > > >=20
> > > > > > >4. If my suggestion in point 1 above that the loose-route
> > > > > mechanism
> > > > > > >does not suffer from the problems suggested, then each
> > > > > mechanism will
> > > > > > >work and each addresses the same problem.
> > > > > > >So it is just down to a beauty contest between the
> > > two. Correct?
> > > > > >=20
> > > > > > See question 1.
> > > > > >=20
> > > > > > We believe that our solution does not have the same
> > > > > limitations as the
> > > > > > loose-route solution. But, again, both solutions intend to
> > > > > solve the
> > > > > > same problem.
> > > > > >=20
> > > > > > Regards,
> > > > > >=20
> > > > > > Christer
> > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > > > > > -----Original Message-----
> > > > > > > > From: Christer Holmberg
> > > > [mailto:christer.holmberg@ericsson.com]
> > > > > > > > Sent: 16 January 2008 08:41
> > > > > > > > To: sip@ietf.org
> > > > > > > > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;
> > > > > Jeroen van
> > > > > > > > Bemmel; Francois Audet
> > > > > > > > Subject: Delivering request-URI and parameters=20
> to UAS via
> > > > > > > proxy - new
> > > > > > > > version of the
> > draft-holmberg-sip-target-uri-delivery draft
> > > > > > > >=20
> > > > > > > >=20
> > > > > > > > Hi,
> > > > > > > >=20
> > > > > > > > We've uploaded a new version (-01) of the Target draft.
> > > > > > > >=20
> > > > > > > > We've tried to make things more clear. I've also
> > > > > removed all text
> > > > > > > > about P-Called-Party-ID, except from one chapter where
> > > > > we try to
> > > > > > > > explain the semantical difference between Target
> > and P-CPI.
> > > > > > > >=20
> > > > > > > > You can also find the draft from:
> > > > > > > >=20
> > > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > > > > > > et-uri-del
> > > > > > > > ivery-01.txt
> > > > > > > >=20
> > > > > > > > Regards,
> > > > > > > >=20
> > > > > > > > Christer
> > > > > > > >=20
> > > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > > > _______________________________________________
> > > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > > This list is for NEW development of the core SIP=20
> Protocol Use=20
> > > > > > sip-implementors@cs.columbia.edu for questions on
> > > current sip Use
> > > > > > sipping@ietf.org for new developments on the
> > application of sip
> > > > > >=20
> > > > >=20
> > > > >=20
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > > sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > > > sipping@ietf.org for new developments on the=20
> application of sip
> > > > >=20
> > > >=20
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20


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



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Date: Thu, 17 Jan 2008 11:11:12 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via
	proxy-new version of the draft-holmberg-sip-target-uri-delivery draft
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 17 January 2008 10:55
> To: Elwell, John; sip@ietf.org
> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> Subject: RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy-new version of the=20
> draft-holmberg-sip-target-uri-delivery draft
>=20
>=20
> >>According to the UA-loose-route draft, every proxy in the=20
> path need to
>=20
> >>support the mechanism in order for the "current target" to=20
> be retained
>=20
> >>in the R-URI. A good example is the "Service Invocation"=20
> use case in=20
> >>the UA loose route draft.
> >[JRE] The Service Invocation section in loose-route tells me=20
> >nothing about proxies on the path. It simply talks about=20
> >invocation of a service at the UAS.
>=20
> This is what the loose draft says:
>=20
> "5. Proposed Solution
>=20
>    The proposed solution is simple.  When handling a request, a proxy
>    only rewrites the Request-URI when performing a retargeting
>    operation.  If, instead, the proxy is trying to route the=20
> request via
>    some entity (whether its a proxy or UA) to reach the target, the
>    Request-URI is retained, and Route header fields are=20
> pushed into the
>    request to reach the target.
>=20
> etc."
>=20
> This applies to ALL proxies on the path, otherwise one will not always
> find the current target (service invocation uri) in the Request-URI.=20
[JRE] Yes, but it doesn suggest to me that the proxies you need to route
through are going to perform service invocation, so the fact that the
information needed for service invocation is in the Request-URI should
not be a concern for those proxies - they will do what any good proxy
will do - route on the contents of the Route header field. Service
information in the Request-URI is for the UAS.

John

> =20
> >>Maybe the "loose route" wording is confusing (I think it is also=20
> >>mentioned in the UA-loose-route draft that a better naming may be=20
> >>needed)?
> >>=20
> >>Also, maybe one can assume that a home proxy knows what=20
> entities not=20
> >>registering (MGCs etc) support. But, again, we are not only talking=20
> >>about home proxies.
> >[JRE] Surely it is only the home proxy that takes an AoR and=20
> determines
> a contact URI to route to. Maybe my understanding of home proxy is
> different.
>=20
> So, what is your understanding?
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
> > >=20
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > > Sent: 16. tammikuuta 2008 15:53
> > > > To: Christer Holmberg; sip@ietf.org
> > > > Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > > > Subject: RE: [Sip] RE: Delivering request-URI and=20
> > parameters to UAS=20
> > > > via proxy -new version of the=20
> > draft-holmberg-sip-target-uri-delivery=20
> > > > draft
> > > >=20
> > > > Christer,
> > > >=20
> > > > I don't think the entity using the mechanism needs to=20
> know about=20
> > > > next hop support for the loose route mechanism - only=20
> whether the=20
> > > > UAS supports it. If the UA registers, then the loose=20
> route draft=20
> > > > provides an automatic mechanism. If the UA does not register,=20
> > > > support would need to be indicated by provisioning. I=20
> am not sure=20
> > > > this ranks as a significant limitation, since a certain=20
> amount of=20
> > > > provisioning needs to be done anyway for UAs such as=20
> > gateways that=20
> > > > do not register.
> > > >=20
> > > > John
> > > >=20
> > > > > -----Original Message-----
> > > > > From: Christer Holmberg=20
> [mailto:christer.holmberg@ericsson.com]
> > > > > Sent: 16 January 2008 11:51
> > > > > To: Elwell, John; sip@ietf.org
> > > > > Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > > > > Subject: RE: [Sip] RE: Delivering request-URI and
> > > parameters to UAS
> > > > > via proxy -new version of the
> > > > draft-holmberg-sip-target-uri-delivery
> > > > > draft
> > > > >=20
> > > > >=20
> > > > > Hi,
> > > > >=20
> > > > > >I hear what you are saying, but I don't really see that the
> > > > > points you
> > > > > raised in section 4 are real limitations of the loose route
> > > > mechanism.
> > > > >=20
> > > > > I think it is a limitation that an entity using the
> > > > mechanism has to
> > > > > know whether the the next hop supports it or not, and=20
> that the=20
> > > > > mechanism can only be used if the subsequent hops support it.
> > > > >=20
> > > > > Regards,
> > > > >=20
> > > > > Christer
> > > > >=20
> > > > >=20
> > > > >=20
> > > > >=20
> > > > > > > -----Original Message-----
> > > > > > > From: Christer Holmberg
> > > [mailto:christer.holmberg@ericsson.com]
> > > > > > > Sent: 16 January 2008 11:06
> > > > > > > To: Elwell, John; sip@ietf.org
> > > > > > > Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> > > > > > > Subject: [Sip] RE: Delivering request-URI and parameters
> > > > > to UAS via
> > > > > > > proxy - new version of the
> > > > draft-holmberg-sip-target-uri-delivery
> > > > > > > draft
> > > > > > >=20
> > > > > > >=20
> > > > > > > Hi John,
> > > > > > >=20
> > > > > > > >Thanks for this revision, which makes things somewhat
> > > > > > clearer. I do
> > > > > > > >have a couple of comments:
> > > > > > > >=20
> > > > > > > >1. I am not sure I agree with the assertions in section 4
> > > > > > concerning
> > > > > > > >issues with the mechanisms in loose-route. Taking
> > > > example 1, the
> > > > > > > >Route header field should contain enough entries to get
> > > > > you to the
> > > > > > > >registered contact, not just to an intermediate proxy.=20
> > > > > > Therefore this
> > > > > > > >situation should not arise with a correctly implemented
> > > > > > home proxy.=20
> > > > > > > >It is not clear to me how example 2 could arise either,
> > > > > > for similar
> > > > > > > >reasons. The MGC case can be resolved by taking into
> > > > account the
> > > > > > > >option tag in the REGISTER request, or if it is=20
> > permanently=20
> > > > > > > >registered, through provisioning.
> > > > > > >=20
> > > > > > > The examples are not meant to show bugs in the loose-route
> > > > > > mechanism.
> > > > > > > They are meant to help people understand the
> > > > limitations with the
> > > > > > > loose-route mechanism.
> > > > > > > =20
> > > > > > > >2. Comparing the mechanism proposed with the loose-route
> > > > > > mechanism,
> > > > > > > >my understanding is:
> > > > > > > >a)When retargeting occurs, the loose-route mechanism
> > > > > > places the new
> > > > > > > >target in the Request URI. Your proposal places the new
> > > > > target in
> > > > > > > >both the Request-URI and the Target header field.
> > > > > > > >b) When rerouting, the loose-route mechanism places the
> > > > > new route
> > > > > > > >(i.e., the registered contact) in the Route header
> > > field. Your
> > > > > > > >proposal places the new route in the Request-URI (the
> > > > > > latter as per
> > > > > > > >RFC 3261).
> > > > > > > >So the two mechanisms solve exactly the same
> > > problems using a
> > > > > > > >slightly different mechanism. Correct?
> > > > > > >=20
> > > > > > > Yes. The two solutions intend to solve the same problem.
> > > > > > > =20
> > > > > > > >3. How P-Called-Party-ID fits into this is not really
> > > > > > relevant from
> > > > > > > >an IETF perspective - it seems there are some=20
> > 3GPP-specific=20
> > > > > > > >situations where the contents of P-Called-Party-ID will
> > > > > > not equal the
> > > > > > > >contents of Target. Correct?
> > > > > > >=20
> > > > > > > Correct.
> > > > > > >=20
> > > > > > > >4. If my suggestion in point 1 above that the loose-route
> > > > > > mechanism
> > > > > > > >does not suffer from the problems suggested, then each
> > > > > > mechanism will
> > > > > > > >work and each addresses the same problem.
> > > > > > > >So it is just down to a beauty contest between the
> > > > two. Correct?
> > > > > > >=20
> > > > > > > See question 1.
> > > > > > >=20
> > > > > > > We believe that our solution does not have the same
> > > > > > limitations as the
> > > > > > > loose-route solution. But, again, both solutions intend to
> > > > > > solve the
> > > > > > > same problem.
> > > > > > >=20
> > > > > > > Regards,
> > > > > > >=20
> > > > > > > Christer
> > > > > > >=20
> > > > > > >=20
> > > > > > >=20
> > > > > > >=20
> > > > > > >=20
> > > > > > >=20
> > > > > > > > > -----Original Message-----
> > > > > > > > > From: Christer Holmberg
> > > > > [mailto:christer.holmberg@ericsson.com]
> > > > > > > > > Sent: 16 January 2008 08:41
> > > > > > > > > To: sip@ietf.org
> > > > > > > > > Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;
> > > > > > Jeroen van
> > > > > > > > > Bemmel; Francois Audet
> > > > > > > > > Subject: Delivering request-URI and parameters=20
> > to UAS via
> > > > > > > > proxy - new
> > > > > > > > > version of the
> > > draft-holmberg-sip-target-uri-delivery draft
> > > > > > > > >=20
> > > > > > > > >=20
> > > > > > > > > Hi,
> > > > > > > > >=20
> > > > > > > > > We've uploaded a new version (-01) of the=20
> Target draft.
> > > > > > > > >=20
> > > > > > > > > We've tried to make things more clear. I've also
> > > > > > removed all text
> > > > > > > > > about P-Called-Party-ID, except from one chapter where
> > > > > > we try to
> > > > > > > > > explain the semantical difference between Target
> > > and P-CPI.
> > > > > > > > >=20
> > > > > > > > > You can also find the draft from:
> > > > > > > > >=20
> > > > http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > > > > > > > > et-uri-del
> > > > > > > > > ivery-01.txt
> > > > > > > > >=20
> > > > > > > > > Regards,
> > > > > > > > >=20
> > > > > > > > > Christer
> > > > > > > > >=20
> > > > > > > >=20
> > > > > > >=20
> > > > > > >=20
> > > > > > > _______________________________________________
> > > > > > > Sip mailing list =20
> https://www1.ietf.org/mailman/listinfo/sip
> > > > > > > This list is for NEW development of the core SIP=20
> > Protocol Use=20
> > > > > > > sip-implementors@cs.columbia.edu for questions on
> > > > current sip Use
> > > > > > > sipping@ietf.org for new developments on the
> > > application of sip
> > > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > > > _______________________________________________
> > > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > > This list is for NEW development of the core SIP=20
> Protocol Use=20
> > > > > > sip-implementors@cs.columbia.edu for questions on
> > > current sip Use
> > > > > > sipping@ietf.org for new developments on the=20
> > application of sip
> > > > > >=20
> > > > >=20
> > > >=20
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
>=20


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



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Subject: [Sip] 
	Vocabulary and problem statement for Request-URI, retargeting,
	and SIP routing (long, but read it!)
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The ongoing discussion around parameter passing mechanisms is getting  
wrapped up in some concepts that we really haven't yet described well  
enough to be communicating clearly. This message is an attempt to  
develop a better understanding of the basic problem.

Even given something as simple as a "SIP proxy", we have three  
fundamentally different modes of operation, and one of those modes  
has two sub-cases that require different thinking.

In the first fundamental mode, the destination of the request is not  
changed by the proxy. The proxy gets the request, looks at it, and  
makes a best guess as to where the request should be sent so as to  
get the request closer to the destination encoded in that request.  
This is roughly analogous to IP routing, where the router does not  
change any of the addresses encoded in the packet. I call this  
fundamental mode "request routing".

In the second fundamental mode, the proxy alters the target of the  
request. Under RFC 3261 as-is, this means that the request-URI is  
changed by the action of the proxy. In general, I call this  
fundamental mode "request retargeting", except that our vocabulary  
isn't completely consistent here. Incidentally, while the first  
fundamental mode is roughly analogous to IP level routing, this  
second fundamental mode is more similar to IP-level network address  
translation. Yes, we've re-invented NAT. For the purpose of future  
clarity, I propose we start calling this fundamental mode "request  
translation".

There are two sub-cases of the second fundamental mode -- that is,  
two kinds of request translation.

In the first sub-case, the proxy transforms the request-URI in a way  
that preserves the identity of the target. That is, the request is  
still going to the same distinct entity, it's just using a different  
request URI to get there. This is what happens when a "home proxy"  
does what I have in the past called "contact routing" and the  
original request URI is transformed from an AOR into a contact URI  
registered against that AOR. The important thing here is that it is  
reasonable for the message sender to expect the message recipient to  
be able to authenticate using the identities expressed in the To:  
header field, which is in general the same as the identity expressed  
in the original Request URI -- that is, as the AOR. So for example:  
sip:Bob@example.com registers from sip:Bob@192.168.1.1. Alice calls  
sip:Bob@example.com. Bob's proxy transforms this to  
sip:Bob@192.168.1.1. Bob gets the request, extracts the To" header  
field, and could reasonably insert an Identity header into the  
response claiming to be "sip:Bob@example.com".

As I said, I've previously called this first sub-case "contact  
routing". We probably shouldn't, since properly it is a translation  
(the request URI changes), not routing. I suggest that we call this  
"translation with preservation of target identity" until somebody  
comes up with a catchier name. The most common example of translation  
with preservation of identity is probably the "home proxy" applying a  
Contact:, which we might call "contact translation". Perhaps we could  
use "rerouting" as a short name, and use "contact rerouting" for the  
thing that a home proxy does.

In the second sub-case of the second fundamental mode, the proxy  
transforms the request URI in a way that does NOT preserve the  
identity of the target. That is, the request is going to a distinctly  
different entity from that indicated by the original request URI and  
the To: header field. Such a case arises if Bob forwards his calls to  
Charlie. Alice sends a request to "sip:Bob@example.com" and gets back  
a response can only be authenticated as being from  
"sip:Charlie@example.com". It is this second case that gives rise to  
the unanticipated response problem and has lead people to propose  
that all retargeting be replaced with redirection. We might call this  
sub-case "translation without preservation of target identity". We've  
often called this sub-case "retargeting" in the past.

We also have a third fundamental mode, "redirection". In this mode,  
the proxy sends a response that encodes a new target for subsequent  
requests. The identity of this new target is independent from the  
identity of the original target -- it may or may not be the same, and  
the act of redirection says nothing about this relationship. As an  
example, Alice calls sip:Bob@example.com. The proxy at example.com  
sends Alice a response redirecting her to "sip:carol@example.com".  
Alice then calls "sip:carol@example.com".


So to recap, we have three basic proxy operations in RFC 3261:

1) Request routing, which preserves the request URI.

2) Request rerouting, which changes the request URI but preserves the  
identity of the request's target.

3) Request retargeting, which changes the request URI and changes the  
identity of the request's target.

We also have a related operation:

4) Request redirection, which informs an upstream node about a new  
identity to target instead of the original.


As a further layer of complexity. note that these proxy operations  
can be stacked. That is, we might, in the process of dealing with one  
request, see multiple applications of the first three, followed by an  
application of the fourth.


Leaving the unanticipated response problem aside, we have two  
fundamental problems with both rerouting and retargeting. The various  
Called-Party-ID, Target, UA-Loose-Route,and other such mechanisms are  
attempting to solve these problems. History-Info provides a related  
set of solution that also encompasses request redirection.

Problem 1: Expressing the identity that was originally the target of  
the request.

Problem 2: Delivering any end-to-end parameters that might have been  
encoded in the original request URI. Note that this is complicated by  
the possibility of end-to-middle parameters that need to be stripped  
by a mid-path node.

We also have a related problem that, AFAIK, neither CPID, nor Target  
nor UA-Loose-Route attempt to solve, but that History-Info does  
attempt, which is:

Problem 3: Expressing what operations have occurred, and WHY they  
have occurred, in such a way that the target, mid-path boxes, and/or  
the requester can do something useful with this knowledge.

I've always held that this last problem is infinitely complex and  
that we should not try to solve it in the general case, which  
explains my reluctance to attack the History-Info problem (or indeed  
the old Diversion header, which also tickled part of this problem  
space). Building applications that require Ultimate Understanding  
seems to me to be a Bad Idea. Much as the invention of time travel  
will have made the concept of future-perfect tense untenable (the  
future will have been found not to have been perfect after all), so  
too is the concept of Ultimate Understanding untenable -- if you know  
everything, you can't possibly understand it since understanding  
requires knowledge outside the subject being considered, and if you  
understand everything you know, you'll understand you don't know  
everything. And that sums up the way I currently feel about the whole  
US-Loose-Route/CPID/Target discussion.

Conclusion:

New proposed vocabulary for proxy operations: request routing,  
rerouting, retargeting, and redirection.

Call for clarity on exactly which problems we're solving with UA- 
Loose-Routing, target, CPID, etc. Proposed: target identity, end-to- 
end parameters.


--
Dean Willis


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Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: [Sip] Vocabulary and problem statement for Request-URI,
	retargeting, and SIP routing (long, but read it!)
Date: Thu, 17 Jan 2008 20:35:23 +0100
Message-ID: <5D1A7985295922448D5550C94DE2918001B490C0@DEEXC1U01.de.lucent.com>
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References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com><CA9998CD4A020D418654FCDEF4E707DF040266B1@esealmw113.eemea.ericsson.se><47878B1E.3010303@cisco.com><0D5F89FAC29E2C41B98A6A762007F5D0549A47@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1428F69B@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF04051C9D@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1428F846@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF040960B7@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>A<CA9998CD4A020D418654FCDEF4E707DF040D69C7@esealmw113.eemea.ericsson.se><0D5F89FAC29E2C41B98A6A762007F5D0549D3F@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1438F1B0@zrc2hxm0.corp.nortel.com><478CEFB4.6070002@zonnet.nl><CA9998CD4A020D418654FCDEF4E707DF0413D587@esealmw113.eemea.ericsson.se><"0D5F89FAC29E2C41B98A6A762007F5D0593C68"@GBNTHT12009MSX.gb002.siemen!s
	.net> A <CA99! 98CD4A020 D418654FCDEF
	4E707DF04173CB8@esealmw113.eemea.ericsson.se><0D5F89FAC29E2C41B98A6A762007F5D0593CFF@GBNTHT12009MSX.gb002.siemens.net>A<CA9998CD4A020D418654FCDEF4E707DF041743D6@esealmw113.eemea.ericsson.se><0D5F89FAC29E2C41B98A6A762007F5D0593E13@GBNTHT12009MSX.gb002.siemens.net><CA9998CD4A020D418654FCDEF4E707DF041C939B@esealmw113.eemea.ericsson.se>
	<C8D63C78-437F-430E-950C-2E63C69E3CEF@softarmor.com>
From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Dean Willis" <dean.willis@softarmor.com>, "IETF SIP List" <sip@ietf.org>
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So out of interest which group would you slot the subdomain to domain
translation in, e.g.

drage@swindon.alcatel-lucent.com to drage@alcatel-lucent.com=20

And vice versa.

And I agree, we do need some vocabulary in this area.

There was a definition of some terms at a higher level in=20

http://tools.ietf.org/id/draft-rosenberg-rai-phone-names-numbers-00.txt

Which we may want to reuse as well.

Regards

Keith

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: Thursday, January 17, 2008 7:13 PM
> To: IETF SIP List
> Subject: [Sip] Vocabulary and problem statement for=20
> Request-URI, retargeting,and SIP routing (long, but read it!)
>=20
>=20
> The ongoing discussion around parameter passing mechanisms is=20
> getting wrapped up in some concepts that we really haven't=20
> yet described well enough to be communicating clearly. This=20
> message is an attempt to develop a better understanding of=20
> the basic problem.
>=20
> Even given something as simple as a "SIP proxy", we have=20
> three fundamentally different modes of operation, and one of=20
> those modes has two sub-cases that require different thinking.
>=20
> In the first fundamental mode, the destination of the request=20
> is not changed by the proxy. The proxy gets the request,=20
> looks at it, and makes a best guess as to where the request=20
> should be sent so as to get the request closer to the=20
> destination encoded in that request. =20
> This is roughly analogous to IP routing, where the router=20
> does not change any of the addresses encoded in the packet. I=20
> call this fundamental mode "request routing".
>=20
> In the second fundamental mode, the proxy alters the target=20
> of the request. Under RFC 3261 as-is, this means that the=20
> request-URI is changed by the action of the proxy. In=20
> general, I call this fundamental mode "request retargeting",=20
> except that our vocabulary isn't completely consistent here.=20
> Incidentally, while the first fundamental mode is roughly=20
> analogous to IP level routing, this second fundamental mode=20
> is more similar to IP-level network address translation. Yes,=20
> we've re-invented NAT. For the purpose of future clarity, I=20
> propose we start calling this fundamental mode "request translation".
>=20
> There are two sub-cases of the second fundamental mode --=20
> that is, two kinds of request translation.
>=20
> In the first sub-case, the proxy transforms the request-URI=20
> in a way that preserves the identity of the target. That is,=20
> the request is still going to the same distinct entity, it's=20
> just using a different request URI to get there. This is what=20
> happens when a "home proxy" =20
> does what I have in the past called "contact routing" and the=20
> original request URI is transformed from an AOR into a=20
> contact URI registered against that AOR. The important thing=20
> here is that it is reasonable for the message sender to=20
> expect the message recipient to be able to authenticate using=20
> the identities expressed in the To: =20
> header field, which is in general the same as the identity=20
> expressed in the original Request URI -- that is, as the AOR.=20
> So for example: =20
> sip:Bob@example.com registers from sip:Bob@192.168.1.1. Alice=20
> calls sip:Bob@example.com. Bob's proxy transforms this to=20
> sip:Bob@192.168.1.1. Bob gets the request, extracts the To"=20
> header field, and could reasonably insert an Identity header=20
> into the response claiming to be "sip:Bob@example.com".
>=20
> As I said, I've previously called this first sub-case=20
> "contact routing". We probably shouldn't, since properly it=20
> is a translation (the request URI changes), not routing. I=20
> suggest that we call this "translation with preservation of=20
> target identity" until somebody comes up with a catchier=20
> name. The most common example of translation with=20
> preservation of identity is probably the "home proxy"=20
> applying a Contact:, which we might call "contact=20
> translation". Perhaps we could use "rerouting" as a short=20
> name, and use "contact rerouting" for the thing that a home=20
> proxy does.
>=20
> In the second sub-case of the second fundamental mode, the=20
> proxy transforms the request URI in a way that does NOT=20
> preserve the identity of the target. That is, the request is=20
> going to a distinctly different entity from that indicated by=20
> the original request URI and the To: header field. Such a=20
> case arises if Bob forwards his calls to Charlie. Alice sends=20
> a request to "sip:Bob@example.com" and gets back a response=20
> can only be authenticated as being from=20
> "sip:Charlie@example.com". It is this second case that gives=20
> rise to the unanticipated response problem and has lead=20
> people to propose that all retargeting be replaced with=20
> redirection. We might call this sub-case "translation without=20
> preservation of target identity". We've often called this=20
> sub-case "retargeting" in the past.
>=20
> We also have a third fundamental mode, "redirection". In this=20
> mode, the proxy sends a response that encodes a new target=20
> for subsequent requests. The identity of this new target is=20
> independent from the identity of the original target -- it=20
> may or may not be the same, and the act of redirection says=20
> nothing about this relationship. As an example, Alice calls=20
> sip:Bob@example.com. The proxy at example.com sends Alice a=20
> response redirecting her to "sip:carol@example.com". =20
> Alice then calls "sip:carol@example.com".
>=20
>=20
> So to recap, we have three basic proxy operations in RFC 3261:
>=20
> 1) Request routing, which preserves the request URI.
>=20
> 2) Request rerouting, which changes the request URI but=20
> preserves the identity of the request's target.
>=20
> 3) Request retargeting, which changes the request URI and=20
> changes the identity of the request's target.
>=20
> We also have a related operation:
>=20
> 4) Request redirection, which informs an upstream node about=20
> a new identity to target instead of the original.
>=20
>=20
> As a further layer of complexity. note that these proxy=20
> operations can be stacked. That is, we might, in the process=20
> of dealing with one request, see multiple applications of the=20
> first three, followed by an application of the fourth.
>=20
>=20
> Leaving the unanticipated response problem aside, we have two=20
> fundamental problems with both rerouting and retargeting. The=20
> various Called-Party-ID, Target, UA-Loose-Route,and other=20
> such mechanisms are attempting to solve these problems.=20
> History-Info provides a related set of solution that also=20
> encompasses request redirection.
>=20
> Problem 1: Expressing the identity that was originally the=20
> target of the request.
>=20
> Problem 2: Delivering any end-to-end parameters that might=20
> have been encoded in the original request URI. Note that this=20
> is complicated by the possibility of end-to-middle parameters=20
> that need to be stripped by a mid-path node.
>=20
> We also have a related problem that, AFAIK, neither CPID, nor=20
> Target nor UA-Loose-Route attempt to solve, but that=20
> History-Info does attempt, which is:
>=20
> Problem 3: Expressing what operations have occurred, and WHY=20
> they have occurred, in such a way that the target, mid-path=20
> boxes, and/or the requester can do something useful with this=20
> knowledge.
>=20
> I've always held that this last problem is infinitely complex=20
> and that we should not try to solve it in the general case,=20
> which explains my reluctance to attack the History-Info=20
> problem (or indeed the old Diversion header, which also=20
> tickled part of this problem space). Building applications=20
> that require Ultimate Understanding seems to me to be a Bad=20
> Idea. Much as the invention of time travel will have made the=20
> concept of future-perfect tense untenable (the future will=20
> have been found not to have been perfect after all), so too=20
> is the concept of Ultimate Understanding untenable -- if you=20
> know everything, you can't possibly understand it since=20
> understanding requires knowledge outside the subject being=20
> considered, and if you understand everything you know, you'll=20
> understand you don't know everything. And that sums up the=20
> way I currently feel about the whole=20
> US-Loose-Route/CPID/Target discussion.
>=20
> Conclusion:
>=20
> New proposed vocabulary for proxy operations: request=20
> routing, rerouting, retargeting, and redirection.
>=20
> Call for clarity on exactly which problems we're solving with=20
> UA- Loose-Routing, target, CPID, etc. Proposed: target=20
> identity, end-to- end parameters.
>=20
>=20
> --
> Dean Willis
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Thu Jan 17 15:24:04 2008
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To: sip@ietf.org
From: Dale.Worley@comcast.net
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Subject: [Sip] Interesting problem with PRACK and resent INVITEs
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I was looking at a problem were were having with a phone and
discovered the following interesting situation with PRACK and INVITEs
that have to be resent with authentication.

Consider an INVITE that gets serially forked to two destinations, the
first one doesn't answer and the second one demands authentication:

    UAC             Proxy           UAS 1           UAS 2

     |                |               |               |
F1   | INVITE CSeq 1  |               |               |
     |--------------->|               |               |
F2   |                | INVITE CSeq 1 |               |
     |                |-------------->|               |
F3   |                | 180 CSeq 1    |               |
     |                | to-tag abc    |               |
     |                |<--------------|               |
F4   | 180 CSeq 1     |               |               |
     | to-tag abc     |               |               |
     |<---------------|               |               |
F5   | PRACK CSeq 2   |               |               |
     | to-tag abc     |               |               |
     |------------------------------->|               |
F6   |                | 200 CSeq 2    |               |
     |                | to-tag abc    |               |
     |<-------------------------------|               |
     |                |               |               |
    ---------------------- delay ------------------------
     |                |               |               |
F7   |                | CANCEL CSq 1  |               |
     |                |-------------->|               |
F8   |                | 200 CANCEL    |               |
     |                |<--------------|               |
F9   |                | 487 CSeq 1    |               |
     |                | to-tag abc    |               |
     |                |<--------------|               |
F10  |                | INVITE CSeq 1 |               |
     |                |------------------------------>|
F11  |                |               | 407 CSeq 1    |
     |                |               | to-tag def    |
     |                |<------------------------------|
F12  | 407 CSeq 1     |               |               |
     | to-tag def     |               |               |
     |<---------------|               |               |
F13  | INVITE CSeq 2  |               |               |
     |--------------->|               |               |
F14  |                | INVITE CSeq 2 |               |
     |                |-------------->|               |
F15  |                | 500 CSeq 2    |               |
     |                | to-tag pqr    |               |
     |                |<--------------|               |
F16  | 500 CSeq 2     |               |               |
     | to-tag pqr     |               |               |
     |<---------------|               |               |
     |                |               |               |

Since the PRACK F5 is sent within a forked dialog, we don't think of
it as using up CSeq number space in the space of out-of-dialog
requests.  So when it comes time to re-send the INVITE F13, it's easy
to think that you can just use a CSeq one higher than the original
INVITE F1.  But transaction processing at UAS 1 is likely to notice
that it's already seen CSeq 2 with that Call-Id.  It looks like one
must make sure that the re-sent INVITE has a CSeq higher than has been
used in any derived dialog of the original INVITE.

Dale


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Subject: RE: [Sip] Interesting problem with PRACK and resent INVITEs
Date: Thu, 17 Jan 2008 16:06:22 -0500
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Thread-Topic: [Sip] Interesting problem with PRACK and resent INVITEs
Thread-Index: AchZRw8Z4IHlyKMKSDKKZzkyuAWvCAABK/Og
From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: <Dale.Worley@comcast.net>, <sip@ietf.org>
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=20
Couple of questions.

Since dialog is a combination of Call-Id + tags, and since F14 does not
have a to-tag, how does UAS correlate this request to the earlier
dialog.

After getting 500 from UAS1, why is not sending the request to UAS2 and
instead sends 500 upstream.

Sanjay


>-----Original Message-----
>From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]=20
>Sent: Thursday, January 17, 2008 3:23 PM
>To: sip@ietf.org
>Subject: [Sip] Interesting problem with PRACK and resent INVITEs
>
>I was looking at a problem were were having with a phone and=20
>discovered the following interesting situation with PRACK and=20
>INVITEs that have to be resent with authentication.
>
>Consider an INVITE that gets serially forked to two=20
>destinations, the first one doesn't answer and the second one=20
>demands authentication:
>
>    UAC             Proxy           UAS 1           UAS 2
>
>     |                |               |               |
>F1   | INVITE CSeq 1  |               |               |
>     |--------------->|               |               |
>F2   |                | INVITE CSeq 1 |               |
>     |                |-------------->|               |
>F3   |                | 180 CSeq 1    |               |
>     |                | to-tag abc    |               |
>     |                |<--------------|               |
>F4   | 180 CSeq 1     |               |               |
>     | to-tag abc     |               |               |
>     |<---------------|               |               |
>F5   | PRACK CSeq 2   |               |               |
>     | to-tag abc     |               |               |
>     |------------------------------->|               |
>F6   |                | 200 CSeq 2    |               |
>     |                | to-tag abc    |               |
>     |<-------------------------------|               |
>     |                |               |               |
>    ---------------------- delay ------------------------
>     |                |               |               |
>F7   |                | CANCEL CSq 1  |               |
>     |                |-------------->|               |
>F8   |                | 200 CANCEL    |               |
>     |                |<--------------|               |
>F9   |                | 487 CSeq 1    |               |
>     |                | to-tag abc    |               |
>     |                |<--------------|               |
>F10  |                | INVITE CSeq 1 |               |
>     |                |------------------------------>|
>F11  |                |               | 407 CSeq 1    |
>     |                |               | to-tag def    |
>     |                |<------------------------------|
>F12  | 407 CSeq 1     |               |               |
>     | to-tag def     |               |               |
>     |<---------------|               |               |
>F13  | INVITE CSeq 2  |               |               |
>     |--------------->|               |               |
>F14  |                | INVITE CSeq 2 |               |
>     |                |-------------->|               |
>F15  |                | 500 CSeq 2    |               |
>     |                | to-tag pqr    |               |
>     |                |<--------------|               |
>F16  | 500 CSeq 2     |               |               |
>     | to-tag pqr     |               |               |
>     |<---------------|               |               |
>     |                |               |               |
>
>Since the PRACK F5 is sent within a forked dialog, we don't=20
>think of it as using up CSeq number space in the space of=20
>out-of-dialog requests.  So when it comes time to re-send the=20
>INVITE F13, it's easy to think that you can just use a CSeq=20
>one higher than the original INVITE F1.  But transaction=20
>processing at UAS 1 is likely to notice that it's already seen=20
>CSeq 2 with that Call-Id.  It looks like one must make sure=20
>that the re-sent INVITE has a CSeq higher than has been used=20
>in any derived dialog of the original INVITE.
>
>Dale
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use=20
>sip-implementors@cs.columbia.edu for questions on current sip=20
>Use sipping@ietf.org for new developments on the application of sip
>


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To: Dean Willis <dean.willis@softarmor.com>, IETF SIP List <sip@ietf.org>
Date: Thu, 17 Jan 2008 16:08:48 -0500
Subject: RE: [Sip] 	Vocabulary and problem statement for Request-URI,
	retargeting,	and SIP routing (long, but read it!)
Thread-Topic: [Sip] 	Vocabulary and problem statement for Request-URI,
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Thursday, January 17, 2008 2:13 PM
> To: IETF SIP List
> Subject: [Sip] Vocabulary and problem statement for Request-URI,
> retargeting, and SIP routing (long, but read it!)
>
> So to recap, we have three basic proxy operations in RFC 3261:
>
> 1) Request routing, which preserves the request URI.
>
> 2) Request rerouting, which changes the request URI but preserves the
> identity of the request's target.
>
> 3) Request retargeting, which changes the request URI and changes the
> identity of the request's target.
>
> We also have a related operation:
>
> 4) Request redirection, which informs an upstream node about a new
> identity to target instead of the original.

This is described from the aspect of the downstream node sending the 3xx, r=
ight?  Vs. what it is for the upstream node recursing on the 3xx? (which ha=
s no idea if it's performing (2) or (3) when this occurs, but I guess could=
 be added to the 3xx contact as a param such as described in JR's draft)

So if a proxy is doing "redirection" it is sending a 3xx, whereas the upstr=
eam proxy or UAC getting that 3xx is not doing redirection, but in fact "re=
cursing".


> Conclusion:
>
> New proposed vocabulary for proxy operations: request routing,
> rerouting, retargeting, and redirection.

Sure.  Except I'm with Joel in that I don't know how a system knows when it=
's doing rerouting vs. retargeting in all cases, even if we stick to the 44=
74-type litmus test.  And unfortunately a lot of systems in the path have u=
ser-configured rules for what they do with the req-uri.

-hadriel



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To: Dean Willis <dean.willis@softarmor.com>, IETF SIP List <sip@ietf.org>
Date: Thu, 17 Jan 2008 16:10:05 -0500
Subject: RE: [Sip] 	Vocabulary and problem statement for Request-URI,
	retargeting,	and SIP routing (long, but read it!)
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Thursday, January 17, 2008 2:13 PM
>
> Call for clarity on exactly which problems we're solving with UA-
> Loose-Routing, target, CPID, etc. Proposed: target identity, end-to-
> end parameters.

Not exactly.  I believe the idea in the UALR draft is to get the target ide=
ntity/params created by the nearest re-targeting event relative to the UAS,=
 where those created by the UAC is the default such event (and thus the log=
ical "farthest").  In other words, what you say is true, except if the requ=
est got retargeted, the identity and params will be those of the new target=
, and thus not end-to-end and not always what the UAC sent.  Right?

For example, given the following, where "RTRG" means Re-Targeting, "RRT" me=
ans Re-Routing, "RT" means Routing, and the single letters represent connec=
tions:

                    RRT
                   +---+
                   |R1 |
                B /+---+\ C
            RT   /       \  RT
           +---+/         \+---+
           |P1 |           |P1 |
        A /+---+           +---+\ D
         /                       \
   +---+/                         \+---+
   |UAC|                           |UAS|
   +---+                           +---+

UA-Loose-routing wants the req-uri seen on connection "A" I think.
To header gives you A.
PCPID gives you B.
Hist-Info gives you A,B,C.


But in the re-targeting scenario such as:
                    RTRG                    RRT
                   +---+                   +---+
                   |R1 |                   |R2 |
                B /+---+\ C             E /+---+\ F
            RT   /       \  RT      RT   /       \  RT
           +---+/         \+---+ D +---+/         \+---+
           |P1 |           |P2 +---+P3 |           |P4 |
        A /+---+           +---+   +---+           +---+\ G
         /                                               \
   +---+/                                                 \+---+
   |UAC|                                                   |UAS|
   +---+                                                   +---+

UA-Loose-routing wants the req-uri seen on connection "C" I think.
To header gives you A.
PCPID gives you E.
Hist-Info gives you A,B,C,D,E,F.

-hadriel



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From: "Theo Zourzouvillys" <theo@crazygreek.co.uk>
To: Dale.Worley@comcast.net
Subject: Re: [Sip] Interesting problem with PRACK and resent INVITEs
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On Jan 17, 2008 8:23 PM,  <Dale.Worley@comcast.net> wrote:
> Since the PRACK F5 is sent within a forked dialog, we don't think of
> it as using up CSeq number space in the space of out-of-dialog
> requests.  So when it comes time to re-send the INVITE F13, it's easy
> to think that you can just use a CSeq one higher than the original
> INVITE F1.  But transaction processing at UAS 1 is likely to notice
> that it's already seen CSeq 2 with that Call-Id.  It looks like one
> must make sure that the re-sent INVITE has a CSeq higher than has been

Dale,

Surely there is no dialog for the INVITE in F14 to be interpreted in
between UAC and UAS1 when it arrives as UAS1, as the previous 487 (F9)
would have terminated the dialog?

 ~ Theo


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From: Dean Willis <dean.willis@softarmor.com>
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [Sip] 	Vocabulary and problem statement for Request-URI,
	retargeting, and SIP routing (long, but read it!)
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Hadriel Kaplan wrote:
> 
>> We also have a related operation:
>> 
>> 4) Request redirection, which informs an upstream node about a new 
>> identity to target instead of the original.
> 
> 
> This is described from the aspect of the downstream node sending the
> 3xx, right?  Vs. what it is for the upstream node recursing on the
> 3xx? (which has no idea if it's performing (2) or (3) when this
> occurs, but I guess could be added to the 3xx contact as a param such
> as described in JR's draft)

right.

> So if a proxy is doing "redirection" it is sending a 3xx,

Right.

> whereas the
> upstream proxy or UAC getting that 3xx is not doing redirection, but
> in fact "recursing".
> 

The upstream node is just generating a new request. It really
doesn't know where that request is going, except that the new request
URI was obtained during an attempt to reach the old one. We'd actually
have more information from a REFER request.

We've already seen all sorts of hideous problems related to an upstream
proxy recursing on a 3xx. My take is that we should never do this in a
proxy.

--
Dean


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From sip-bounces@ietf.org Thu Jan 17 18:37:53 2008
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Subject: Re: [Sip] 	Vocabulary and problem statement for Request-URI,
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Hadriel Kaplan wrote:
> 
>>-----Original Message-----
>>From: Dean Willis [mailto:dean.willis@softarmor.com]
>>Sent: Thursday, January 17, 2008 2:13 PM
>>
>>Call for clarity on exactly which problems we're solving with UA-
>>Loose-Routing, target, CPID, etc. Proposed: target identity, end-to-
>>end parameters.
> 
> 
> Not exactly.  I believe the idea in the UALR draft is to get the
> target identity/params created by the nearest re-targeting event
> relative to the UAS, where those created by the UAC is the default
> such event (and thus the logical "farthest").  In other words, what
> you say is true, except if the request got retargeted, the identity
> and params will be those of the new target, and thus not end-to-end
> and not always what the UAC sent.  Right?

AFAIK, the original goal of UALR was to get parameters to applications. 
The last time I thought I understood the draft, these were preserved 
back somewhere on teh route stack so that the application could get them.

The other thing UALR does, and this is kind of neat, is allow different 
sets of parameters to be stuffed in at different places, and for the UAS 
to be able to differentiate where those parameters were inserted (at 
least a little bit -- "original", "middle", and "last hop" being easy to 
parse.).

> 
> For example, given the following, where "RTRG" means Re-Targeting,
"RRT" means Re-Routing, "RT" means Routing, and the single letters
represent connections:
> 
>                     RRT
>                    +---+
>                    |R1 |
>                 B /+---+\ C
>             RT   /       \  RT
>            +---+/         \+---+
>            |P1 |           |P1 |
>         A /+---+           +---+\ D
>          /                       \
>    +---+/                         \+---+
>    |UAC|                           |UAS|
>    +---+                           +---+
> 
> UA-Loose-routing wants the req-uri seen on connection "A" I think.

I think so too.

> To header gives you A.
> PCPID gives you B.
> Hist-Info gives you A,B,C.

and D is directly visible to UAS, should RT have inserted any.


> 
> But in the re-targeting scenario such as:
>                     RTRG                    RRT
>                    +---+                   +---+
>                    |R1 |                   |R2 |
>                 B /+---+\ C             E /+---+\ F
>             RT   /       \  RT      RT   /       \  RT
>            +---+/         \+---+ D +---+/         \+---+
>            |P1 |           |P2 +---+P3 |           |P4 |
>         A /+---+           +---+   +---+           +---+\ G
>          /                                               \
>    +---+/                                                 \+---+
>    |UAC|                                                   |UAS|
>    +---+                                                   +---+
> 
> UA-Loose-routing wants the req-uri seen on connection "C" I think.
> To header gives you A.
> PCPID gives you E.
> Hist-Info gives you A,B,C,D,E,F.

As I thought I understood it, UALR gives us the stack of A and C.

But I may well not understand.


However, this does illustrate that there's another axis of proxy 
operations that can be happening.

In addition to affecting the route, the target, and the identity of the 
target, proxies may also affect parameters.

But "repararametering" is such an ugly word. And it's an ugly concept, 
when you think about the consequences. I'm tempted to think of all 
reparametering as retargeting.

--
Dean


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Subject: RE: [Sip] 	Vocabulary and problem statement for Request-URI,
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Inline ...

>-----Original Message-----
>From: Dean Willis [mailto:dean.willis@softarmor.com]=20
>Sent: Thursday, January 17, 2008 6:34 PM
>To: Hadriel Kaplan
>Cc: IETF SIP List
>Subject: Re: [Sip] Vocabulary and problem statement for=20
>Request-URI,retargeting, and SIP routing (long, but read it!)
>
>Hadriel Kaplan wrote:
>>=20
>>> We also have a related operation:
>>>=20
>>> 4) Request redirection, which informs an upstream node about a new=20
>>> identity to target instead of the original.
>>=20
>>=20
>> This is described from the aspect of the downstream node sending the=20
>> 3xx, right?  Vs. what it is for the upstream node recursing on the=20
>> 3xx? (which has no idea if it's performing (2) or (3) when this=20
>> occurs, but I guess could be added to the 3xx contact as a=20
>param such=20
>> as described in JR's draft)
>
>right.
>
>> So if a proxy is doing "redirection" it is sending a 3xx,
>
>Right.
>
>> whereas the
>> upstream proxy or UAC getting that 3xx is not doing redirection, but=20
>> in fact "recursing".
>>=20
>
>The upstream node is just generating a new request. It really=20
>doesn't know where that request is going, except that the new=20
>request URI was obtained during an attempt to reach the old=20
>one.=20
Not sure what you mean here? The next request-uri is taken from Contact
of the redirect response. So it knows where the request is going.

> We'd actually have more information from a REFER request.
How is the Refer-To header different that Contacts in redirect response?

Sanjay

>
>We've already seen all sorts of hideous problems related to an=20
>upstream proxy recursing on a 3xx. My take is that we should=20
>never do this in a proxy.
>
>--
>Dean
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use=20
>sip-implementors@cs.columbia.edu for questions on current sip=20
>Use sipping@ietf.org for new developments on the application of sip
>


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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Dale.Worley@comcast.net
Subject: Re: [Sip] Interesting problem with PRACK and resent INVITEs
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Interesting. But I'm not certain I buy it. CSeqs only make sense within 
a dialog. F13 doesn't contain the To-tag abc, so it ought not be 
expected to use a valid CSeq for that dialog.

A different issue here is that that F13 ought not go to UAS1 at all. But 
we treat it as a new transaction, and don't expect the proxy to maintain 
any history across transactions, so we don't have a way to prevent that.

There ought to be something in F11 that causes the request to go only to 
  (in this case) UA2. (In general to whichever node generated the 
challenge, which might have been a proxy rather than a UA.)

	Paul

Dale.Worley@comcast.net wrote:
> I was looking at a problem were were having with a phone and
> discovered the following interesting situation with PRACK and INVITEs
> that have to be resent with authentication.
> 
> Consider an INVITE that gets serially forked to two destinations, the
> first one doesn't answer and the second one demands authentication:
> 
>     UAC             Proxy           UAS 1           UAS 2
> 
>      |                |               |               |
> F1   | INVITE CSeq 1  |               |               |
>      |--------------->|               |               |
> F2   |                | INVITE CSeq 1 |               |
>      |                |-------------->|               |
> F3   |                | 180 CSeq 1    |               |
>      |                | to-tag abc    |               |
>      |                |<--------------|               |
> F4   | 180 CSeq 1     |               |               |
>      | to-tag abc     |               |               |
>      |<---------------|               |               |
> F5   | PRACK CSeq 2   |               |               |
>      | to-tag abc     |               |               |
>      |------------------------------->|               |
> F6   |                | 200 CSeq 2    |               |
>      |                | to-tag abc    |               |
>      |<-------------------------------|               |
>      |                |               |               |
>     ---------------------- delay ------------------------
>      |                |               |               |
> F7   |                | CANCEL CSq 1  |               |
>      |                |-------------->|               |
> F8   |                | 200 CANCEL    |               |
>      |                |<--------------|               |
> F9   |                | 487 CSeq 1    |               |
>      |                | to-tag abc    |               |
>      |                |<--------------|               |
> F10  |                | INVITE CSeq 1 |               |
>      |                |------------------------------>|
> F11  |                |               | 407 CSeq 1    |
>      |                |               | to-tag def    |
>      |                |<------------------------------|
> F12  | 407 CSeq 1     |               |               |
>      | to-tag def     |               |               |
>      |<---------------|               |               |
> F13  | INVITE CSeq 2  |               |               |
>      |--------------->|               |               |
> F14  |                | INVITE CSeq 2 |               |
>      |                |-------------->|               |
> F15  |                | 500 CSeq 2    |               |
>      |                | to-tag pqr    |               |
>      |                |<--------------|               |
> F16  | 500 CSeq 2     |               |               |
>      | to-tag pqr     |               |               |
>      |<---------------|               |               |
>      |                |               |               |
> 
> Since the PRACK F5 is sent within a forked dialog, we don't think of
> it as using up CSeq number space in the space of out-of-dialog
> requests.  So when it comes time to re-send the INVITE F13, it's easy
> to think that you can just use a CSeq one higher than the original
> INVITE F1.  But transaction processing at UAS 1 is likely to notice
> that it's already seen CSeq 2 with that Call-Id.  It looks like one
> must make sure that the re-sent INVITE has a CSeq higher than has been
> used in any derived dialog of the original INVITE.
> 
> Dale
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Thu, 17 Jan 2008 23:30:44 -0500
Subject: RE: [Sip] 	Vocabulary and problem statement for Request-URI,
	retargeting, and SIP routing (long, but read it!)
Thread-Topic: [Sip] 	Vocabulary and problem statement for Request-URI,
	retargeting, and SIP routing (long, but read it!)
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Thursday, January 17, 2008 6:45 PM
>
> AFAIK, the original goal of UALR was to get parameters to applications.
> The last time I thought I understood the draft, these were preserved
> back somewhere on teh route stack so that the application could get them.

That may have been the intent, but as I read draft-rosenberg-sip-ua-loose-r=
oute-01.txt, it doesn't do that if a retarget occurs.  In a retargeting cas=
e the req-uri is replaced with the new target.  In the rerouting case, the =
req-uri is left alone, and the newly resolved destination URI (for lack of =
a better term) would be pushed as a Route header.

It does, though, say the previous req-uri would be pushed into the History-=
Info header.  It would have been before this draft too, if people implement=
ed it, except with this new draft ONLY retarget cases would push a History-=
Info.  So I guess in that sense you're right, the pre-retargeting data coul=
d be there in that.


> >
> > But in the re-targeting scenario such as:
> >                     RTRG                    RRT
> >                    +---+                   +---+
> >                    |R1 |                   |R2 |
> >                 B /+---+\ C             E /+---+\ F
> >             RT   /       \  RT      RT   /       \  RT
> >            +---+/         \+---+ D +---+/         \+---+
> >            |P1 |           |P2 +---+P3 |           |P4 |
> >         A /+---+           +---+   +---+           +---+\ G
> >          /                                               \
> >    +---+/                                                 \+---+
> >    |UAC|                                                   |UAS|
> >    +---+                                                   +---+
> >
> > UA-Loose-routing wants the req-uri seen on connection "C" I think.
> > To header gives you A.
> > PCPID gives you E.
> > Hist-Info gives you A,B,C,D,E,F.
>
> As I thought I understood it, UALR gives us the stack of A and C.

If R1 and R2 did UALR, the UAS would get the req-uri from C, a History-Info=
 with B, and there'd be a Route header with UAS' contact.

-hadriel



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From sip-bounces@ietf.org Fri Jan 18 00:04:23 2008
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>
Date: Fri, 18 Jan 2008 00:01:23 -0500
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Subject: [Sip] Questions on draft-rosenberg-sip-ua-loose-route-01.txt
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If I read the draft correctly, the proposed solution is that rerouting acti=
on leaves the req-uri alone, and pushes a Route for the resolved target's U=
RI (contact/whatever).  Whereas a retargeting action replaces the req-uri w=
ith the new target, and possibly pushes the previous req-uri into History-I=
nfo.  The draft proposes retargeting be defined as when the proxy determine=
s the new target could not claim the previous req-uri as its Identity.  In =
other words, if the new target were to originate a request using the previo=
us req-uri as its From addr-spec, would its domain sign it using rfc4474 - =
if so then it's a reroute, if not then it's a retarget.  Correct?

The draft also describes several cases, some of which don't seem to follow =
the above rules.

For example, Section 5 example 3 says:
   3.  When a proxy receiving a request identifies a next hop server
       that is needed to process the request, that next hop server is a
       route.  A next hop server is not a UA and would never be able to
       claim its identity.  Its URI is pushed into a Route header field
       and the Request-URI is retained.  An important use case for this
       are proxies that select PSTN gateways for call egress to the
       PSTN.  Such selection would place the SIP URI of the gateway into
       the topmost Route header field value and retain the Request-URI.

So this says a PSTN gateway could not claim the identity, yet the behavior =
is still rerouting?

Another example of this is in Section 2.6 for Emergency Numbers.  It states=
 the SOS URN needs to stay in place in the req-uri all the way to the call-=
taker.  I'm not sure if such a call-taker could claim Identity to an SOS UR=
N, and furthermore the odds are the call-taker is in another domain, which =
per this draft would constitute a retargeting and change the req-uri.  No?

This draft should also describe what happens for service invocation, which =
is a stated problem space in section 2.5.  Again, a voicemail system or IVR=
 or App Server could not claim Identity for the req-uri, and thus it would =
be a retarget per the draft's definition.

-hadriel


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Subject: RE: [Sip] Interesting problem with PRACK and resent INVITEs
Date: Fri, 18 Jan 2008 06:17:39 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	<Dale.Worley@comcast.net>
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Hi,=20

>Interesting. But I'm not certain I buy it. CSeqs only make=20
>sense within a dialog. F13 doesn't contain the To-tag abc, so=20
>it ought not be expected to use a valid CSeq for that dialog.

Correct.

And, the INVITEs shall contain different From-tag values (and Call-ID
values), so UAS 2 will be able to differentiate the dialogs based on
those parameters.

Regards,

Christer




>=20
>A different issue here is that that F13 ought not go to UAS1=20
>at all. But we treat it as a new transaction, and don't=20
>expect the proxy to maintain any history across transactions,=20
>so we don't have a way to prevent that.
>=20
>There ought to be something in F11 that causes the request to=20
>go only to (in this case) UA2. (In general to whichever node generated=20
>the challenge, which might have been a proxy rather than a UA.)
>=20
> 	Paul
>=20
> Dale.Worley@comcast.net wrote:
> > I was looking at a problem were were having with a phone and=20
> > discovered the following interesting situation with PRACK=20
> and INVITEs=20
> > that have to be resent with authentication.
> >=20
> > Consider an INVITE that gets serially forked to two=20
> destinations, the=20
> > first one doesn't answer and the second one demands authentication:
> >=20
> >     UAC             Proxy           UAS 1           UAS 2
> >=20
> >      |                |               |               |
> > F1   | INVITE CSeq 1  |               |               |
> >      |--------------->|               |               |
> > F2   |                | INVITE CSeq 1 |               |
> >      |                |-------------->|               |
> > F3   |                | 180 CSeq 1    |               |
> >      |                | to-tag abc    |               |
> >      |                |<--------------|               |
> > F4   | 180 CSeq 1     |               |               |
> >      | to-tag abc     |               |               |
> >      |<---------------|               |               |
> > F5   | PRACK CSeq 2   |               |               |
> >      | to-tag abc     |               |               |
> >      |------------------------------->|               |
> > F6   |                | 200 CSeq 2    |               |
> >      |                | to-tag abc    |               |
> >      |<-------------------------------|               |
> >      |                |               |               |
> >     ---------------------- delay ------------------------
> >      |                |               |               |
> > F7   |                | CANCEL CSq 1  |               |
> >      |                |-------------->|               |
> > F8   |                | 200 CANCEL    |               |
> >      |                |<--------------|               |
> > F9   |                | 487 CSeq 1    |               |
> >      |                | to-tag abc    |               |
> >      |                |<--------------|               |
> > F10  |                | INVITE CSeq 1 |               |
> >      |                |------------------------------>|
> > F11  |                |               | 407 CSeq 1    |
> >      |                |               | to-tag def    |
> >      |                |<------------------------------|
> > F12  | 407 CSeq 1     |               |               |
> >      | to-tag def     |               |               |
> >      |<---------------|               |               |
> > F13  | INVITE CSeq 2  |               |               |
> >      |--------------->|               |               |
> > F14  |                | INVITE CSeq 2 |               |
> >      |                |-------------->|               |
> > F15  |                | 500 CSeq 2    |               |
> >      |                | to-tag pqr    |               |
> >      |                |<--------------|               |
> > F16  | 500 CSeq 2     |               |               |
> >      | to-tag pqr     |               |               |
> >      |<---------------|               |               |
> >      |                |               |               |
> >=20
> > Since the PRACK F5 is sent within a forked dialog, we don't=20
> think of=20
> > it as using up CSeq number space in the space of out-of-dialog=20
> > requests.  So when it comes time to re-send the INVITE F13,=20
> it's easy=20
> > to think that you can just use a CSeq one higher than the original=20
> > INVITE F1.  But transaction processing at UAS 1 is likely to notice=20
> > that it's already seen CSeq 2 with that Call-Id.  It looks like one=20
> > must make sure that the re-sent INVITE has a CSeq higher=20
> than has been=20
> > used in any derived dialog of the original INVITE.
> >=20
> > Dale
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Fri Jan 18 00:34:40 2008
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	retargeting, and SIP routing (long, but read it!)
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Sanjay Sinha (sanjsinh) wrote:
> Inline ...
>
>   
>>> whereas the
>>> upstream proxy or UAC getting that 3xx is not doing redirection, but 
>>> in fact "recursing".
>>>
>>>       
>> The upstream node is just generating a new request. It really 
>> doesn't know where that request is going, except that the new 
>> request URI was obtained during an attempt to reach the old 
>> one. 
>>     
> Not sure what you mean here? The next request-uri is taken from Contact
> of the redirect response. So it knows where the request is going.
>   

It knows the URI that the request is going to. It doesn't know "who" 
that URI will reach, or what their relationship to the original URI is 
(unless perhaps "History-Info" explains it). However, this is at least 
enough knowledge to be able to prevent the unanticipated response 
problem sometimes, because we can at least check that the identity 
expressed in the eventual response matches that received in the 302. 
Assuming, of course, that the new URI doesn't go through some sort of 
retargeting operation that does not preserve the request identity.

>   
>> We'd actually have more information from a REFER request.
>>     
> How is the Refer-To header different that Contacts in redirect response?
>
>   
We're not really sure where the 3xx response came from. All we know is 
that it was generated somewhere along the path towards the original 
request-URI. REFER, however, is a request instead of a response, so we 
can use the various request-authentication techniques to figure out who 
sent it.


--
Dean



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From sip-bounces@ietf.org Fri Jan 18 02:26:11 2008
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Interesting problem with PRACK and resent INVITEs
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	<CA9998CD4A020D418654FCDEF4E707DF041F5175@esealmw113.eemea.ericsson.se>
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Christer,

No, see RFC3261 8.1.1.4:

Note that when requests are retried after certain failure responses that solicit an amendment to a request (for
   example, a challenge for authentication), these retried requests are
   not considered new requests, and therefore do not need new Call-ID
   header fields; see Section 8.1.3.5.

Mid-dialog challenges should use the same from-tag + Call-ID, so although it seems allowed to start anew, IMHO it is better to keep them the same. 
The text could be more clear though (and should have some normative words).

Regarding from tags there is no explicit text, but 22.2 says:

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.

This hints at treating the challenge response as a mid-call request (although it may have no to-tag in case of an initial INVITE)

Regarding Dale's case below, this can only occur if UAS1 has a broken SIP implementation (F1, F5 and F13 are 3 independent transactions)

Regards,
Jeroen

Christer Holmberg wrote:
> Hi, 
>
>> Interesting. But I'm not certain I buy it. CSeqs only make 
>> sense within a dialog. F13 doesn't contain the To-tag abc, so 
>> it ought not be expected to use a valid CSeq for that dialog.
>
> Correct.
>
> And, the INVITEs shall contain different From-tag values (and Call-ID
> values), so UAS 2 will be able to differentiate the dialogs based on
> those parameters.
>
> Regards,
>
> Christer
>
>
>
>
>> A different issue here is that that F13 ought not go to UAS1 
>> at all. But we treat it as a new transaction, and don't 
>> expect the proxy to maintain any history across transactions, 
>> so we don't have a way to prevent that.
>>
>> There ought to be something in F11 that causes the request to 
>> go only to (in this case) UA2. (In general to whichever node generated 
>> the challenge, which might have been a proxy rather than a UA.)
>>
>> 	Paul
>>
>> Dale.Worley@comcast.net wrote:
>>> I was looking at a problem were were having with a phone and 
>>> discovered the following interesting situation with PRACK 
>> and INVITEs 
>>> that have to be resent with authentication.
>>>
>>> Consider an INVITE that gets serially forked to two 
>> destinations, the 
>>> first one doesn't answer and the second one demands authentication:
>>>
>>>     UAC             Proxy           UAS 1           UAS 2
>>>
>>>      |                |               |               |
>>> F1   | INVITE CSeq 1  |               |               |
>>>      |--------------->|               |               |
>>> F2   |                | INVITE CSeq 1 |               |
>>>      |                |-------------->|               |
>>> F3   |                | 180 CSeq 1    |               |
>>>      |                | to-tag abc    |               |
>>>      |                |<--------------|               |
>>> F4   | 180 CSeq 1     |               |               |
>>>      | to-tag abc     |               |               |
>>>      |<---------------|               |               |
>>> F5   | PRACK CSeq 2   |               |               |
>>>      | to-tag abc     |               |               |
>>>      |------------------------------->|               |
>>> F6   |                | 200 CSeq 2    |               |
>>>      |                | to-tag abc    |               |
>>>      |<-------------------------------|               |
>>>      |                |               |               |
>>>     ---------------------- delay ------------------------
>>>      |                |               |               |
>>> F7   |                | CANCEL CSq 1  |               |
>>>      |                |-------------->|               |
>>> F8   |                | 200 CANCEL    |               |
>>>      |                |<--------------|               |
>>> F9   |                | 487 CSeq 1    |               |
>>>      |                | to-tag abc    |               |
>>>      |                |<--------------|               |
>>> F10  |                | INVITE CSeq 1 |               |
>>>      |                |------------------------------>|
>>> F11  |                |               | 407 CSeq 1    |
>>>      |                |               | to-tag def    |
>>>      |                |<------------------------------|
>>> F12  | 407 CSeq 1     |               |               |
>>>      | to-tag def     |               |               |
>>>      |<---------------|               |               |
>>> F13  | INVITE CSeq 2  |               |               |
>>>      |--------------->|               |               |
>>> F14  |                | INVITE CSeq 2 |               |
>>>      |                |-------------->|               |
>>> F15  |                | 500 CSeq 2    |               |
>>>      |                | to-tag pqr    |               |
>>>      |                |<--------------|               |
>>> F16  | 500 CSeq 2     |               |               |
>>>      | to-tag pqr     |               |               |
>>>      |<---------------|               |               |
>>>      |                |               |               |
>>>
>>> Since the PRACK F5 is sent within a forked dialog, we don't 
>> think of 
>>> it as using up CSeq number space in the space of out-of-dialog 
>>> requests.  So when it comes time to re-send the INVITE F13, 
>> it's easy 
>>> to think that you can just use a CSeq one higher than the original 
>>> INVITE F1.  But transaction processing at UAS 1 is likely to notice 
>>> that it's already seen CSeq 2 with that Call-Id.  It looks like one 
>>> must make sure that the re-sent INVITE has a CSeq higher 
>> than has been 
>>> used in any derived dialog of the original INVITE.
>>>
>>> Dale
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip 
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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Subject: RE: [Sip] Interesting problem with PRACK and resent INVITEs
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Hi,=20

>No, see RFC3261 8.1.1.4:
>=20
>Note that when requests are retried after certain failure=20
>responses that solicit an amendment to a request (for
>example, a challenge for authentication), these retried=20
>requests are not considered new requests, and therefore do not need new
Call-ID
>header fields; see Section 8.1.3.5.

Correct.

>Mid-dialog challenges should use the same from-tag + Call-ID,=20
>so although it seems allowed to start anew, IMHO it is better=20
>to keep them the same.=20
>The text could be more clear though (and should have some=20
>normative words).
>
>=20
>Regarding from tags there is no explicit text, but 22.2 says:
>=20
>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.

So, I guess that means that the second INVITE in Dale's example should
have Cseq =3D> 3.
=20
>This hints at treating the challenge response as a mid-call=20
>request (although it may have no to-tag in case of an initial INVITE)

Personally I think that EVERY new initial INVITE should be update the
Call-ID and/or From-tag, no matter what triggered it.

But, I guess there is a good reason behind the current text...

Regards,

Christer




> Christer Holmberg wrote:
> > Hi,
> >
> >> Interesting. But I'm not certain I buy it. CSeqs only make sense=20
> >> within a dialog. F13 doesn't contain the To-tag abc, so it=20
> ought not=20
> >> be expected to use a valid CSeq for that dialog.
> >
> > Correct.
> >
> > And, the INVITEs shall contain different From-tag values=20
> (and Call-ID=20
> > values), so UAS 2 will be able to differentiate the dialogs=20
> based on=20
> > those parameters.
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> >
> >> A different issue here is that that F13 ought not go to=20
> UAS1 at all.=20
> >> But we treat it as a new transaction, and don't expect the=20
> proxy to=20
> >> maintain any history across transactions, so we don't have=20
> a way to=20
> >> prevent that.
> >>
> >> There ought to be something in F11 that causes the request=20
> to go only=20
> >> to (in this case) UA2. (In general to whichever node generated the=20
> >> challenge, which might have been a proxy rather than a UA.)
> >>
> >> 	Paul
> >>
> >> Dale.Worley@comcast.net wrote:
> >>> I was looking at a problem were were having with a phone and=20
> >>> discovered the following interesting situation with PRACK
> >> and INVITEs
> >>> that have to be resent with authentication.
> >>>
> >>> Consider an INVITE that gets serially forked to two
> >> destinations, the
> >>> first one doesn't answer and the second one demands=20
> authentication:
> >>>
> >>>     UAC             Proxy           UAS 1           UAS 2
> >>>
> >>>      |                |               |               |
> >>> F1   | INVITE CSeq 1  |               |               |
> >>>      |--------------->|               |               |
> >>> F2   |                | INVITE CSeq 1 |               |
> >>>      |                |-------------->|               |
> >>> F3   |                | 180 CSeq 1    |               |
> >>>      |                | to-tag abc    |               |
> >>>      |                |<--------------|               |
> >>> F4   | 180 CSeq 1     |               |               |
> >>>      | to-tag abc     |               |               |
> >>>      |<---------------|               |               |
> >>> F5   | PRACK CSeq 2   |               |               |
> >>>      | to-tag abc     |               |               |
> >>>      |------------------------------->|               |
> >>> F6   |                | 200 CSeq 2    |               |
> >>>      |                | to-tag abc    |               |
> >>>      |<-------------------------------|               |
> >>>      |                |               |               |
> >>>     ---------------------- delay ------------------------
> >>>      |                |               |               |
> >>> F7   |                | CANCEL CSq 1  |               |
> >>>      |                |-------------->|               |
> >>> F8   |                | 200 CANCEL    |               |
> >>>      |                |<--------------|               |
> >>> F9   |                | 487 CSeq 1    |               |
> >>>      |                | to-tag abc    |               |
> >>>      |                |<--------------|               |
> >>> F10  |                | INVITE CSeq 1 |               |
> >>>      |                |------------------------------>|
> >>> F11  |                |               | 407 CSeq 1    |
> >>>      |                |               | to-tag def    |
> >>>      |                |<------------------------------|
> >>> F12  | 407 CSeq 1     |               |               |
> >>>      | to-tag def     |               |               |
> >>>      |<---------------|               |               |
> >>> F13  | INVITE CSeq 2  |               |               |
> >>>      |--------------->|               |               |
> >>> F14  |                | INVITE CSeq 2 |               |
> >>>      |                |-------------->|               |
> >>> F15  |                | 500 CSeq 2    |               |
> >>>      |                | to-tag pqr    |               |
> >>>      |                |<--------------|               |
> >>> F16  | 500 CSeq 2     |               |               |
> >>>      | to-tag pqr     |               |               |
> >>>      |<---------------|               |               |
> >>>      |                |               |               |
> >>>
> >>> Since the PRACK F5 is sent within a forked dialog, we don't
> >> think of
> >>> it as using up CSeq number space in the space of out-of-dialog=20
> >>> requests.  So when it comes time to re-send the INVITE F13,
> >> it's easy
> >>> to think that you can just use a CSeq one higher than the=20
> original=20
> >>> INVITE F1.  But transaction processing at UAS 1 is likely=20
> to notice=20
> >>> that it's already seen CSeq 2 with that Call-Id.  It=20
> looks like one=20
> >>> must make sure that the re-sent INVITE has a CSeq higher
> >> than has been
> >>> used in any derived dialog of the original INVITE.
> >>>
> >>> Dale
> >>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
> >>>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >>
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >
>=20


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	retargeting, and SIP routing (long, but read it!)
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Dean,

Regarding
>  
>
> We also have a related operation:
>
> 4) Request redirection, which informs an upstream node about a new 
> identity to target instead of the original.
>
Redirection does not always imply a new identity, it could be an 
alternative Contact for the same identity. The thing is that we cannot 
easily distinguish between these, since both use the same response code 
(302). So it can be either rerouting or retargeting. For the problem at 
hand this makes no difference though, in any case the request URI is 
replaced.

BTW what happened to the 'opaque' parameter solution that was part of a 
GRUU draft some versions ago? A special URI parameter that survives 
rerouting/retargeting would avoid introducing yet another identity 
header, and also detection-of-support issues.

Regards,
Jeroen



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Date: Fri, 18 Jan 2008 07:48:18 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] 	Vocabulary and problem statement for
	Request-URI,retargeting,	and SIP routing (long, but read it!)
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Hadriel,

Comment at the end.

John=20

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: 17 January 2008 21:10
> To: Dean Willis; IETF SIP List
> Subject: RE: [Sip] Vocabulary and problem statement for=20
> Request-URI,retargeting, and SIP routing (long, but read it!)
>=20
>=20
>=20
> > -----Original Message-----
> > From: Dean Willis [mailto:dean.willis@softarmor.com]
> > Sent: Thursday, January 17, 2008 2:13 PM
> >
> > Call for clarity on exactly which problems we're solving with UA-
> > Loose-Routing, target, CPID, etc. Proposed: target identity, end-to-
> > end parameters.
>=20
> Not exactly.  I believe the idea in the UALR draft is to get=20
> the target identity/params created by the nearest=20
> re-targeting event relative to the UAS, where those created=20
> by the UAC is the default such event (and thus the logical=20
> "farthest").  In other words, what you say is true, except if=20
> the request got retargeted, the identity and params will be=20
> those of the new target, and thus not end-to-end and not=20
> always what the UAC sent.  Right?
>=20
> For example, given the following, where "RTRG" means=20
> Re-Targeting, "RRT" means Re-Routing, "RT" means Routing, and=20
> the single letters represent connections:
>=20
>                     RRT
>                    +---+
>                    |R1 |
>                 B /+---+\ C
>             RT   /       \  RT
>            +---+/         \+---+
>            |P1 |           |P1 |
>         A /+---+           +---+\ D
>          /                       \
>    +---+/                         \+---+
>    |UAC|                           |UAS|
>    +---+                           +---+
>=20
> UA-Loose-routing wants the req-uri seen on connection "A" I think.
> To header gives you A.
> PCPID gives you B.
> Hist-Info gives you A,B,C.
>=20
>=20
> But in the re-targeting scenario such as:
>                     RTRG                    RRT
>                    +---+                   +---+
>                    |R1 |                   |R2 |
>                 B /+---+\ C             E /+---+\ F
>             RT   /       \  RT      RT   /       \  RT
>            +---+/         \+---+ D +---+/         \+---+
>            |P1 |           |P2 +---+P3 |           |P4 |
>         A /+---+           +---+   +---+           +---+\ G
>          /                                               \
>    +---+/                                                 \+---+
>    |UAC|                                                   |UAS|
>    +---+                                                   +---+
>=20
> UA-Loose-routing wants the req-uri seen on connection "C" I think.
> To header gives you A.
> PCPID gives you E.
> Hist-Info gives you A,B,C,D,E,F.
[JRE] According to Dean's definition of RT, it does not change the
Request-URI (only the Route header field presumably, or maybe not even
that).
So C, D and E are the same. Also A and B are the same, and F and G are
the same.
So I think:
- UA-Loose-routing gives you C/D/E
- To gives you A/B
- PCPID gives you C/D/E
- Target gives you C/D/E
- Hist-info gives you A/B, C/D/E and F/G.

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


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Christer,
>> .
>>
>>
>> Regarding from tags there is no explicit text, but 22.2 says:
>>
>> 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.
>>     
>
> So, I guess that means that the second INVITE in Dale's example should
> have Cseq => 3.

I interpreted this as "MUST increment (relative to the INVITE being 
challenged)". I guess for initial INVITEs the minimal requirement is 
that the CSeq be different, else merged request detection (8.2.2.2) 
could kick in and prevent the call from succeeding. So both 2 or 3 are 
valid, the text is ambiguous

Regards,
Jeroen


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Subject: RE: [Sip] Question on RFC3840/3841
Date: Fri, 18 Jan 2008 10:27:59 +0100
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	<sip@ietf.org>
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Hi Paul,
=20
>RFC3840 by itself, without 2533, is *not* sufficient to=20
>understand how to do the matching.
>=20
>I put an appendix in RFC 4596 that gives an algorithm without=20
>reference to 2533. I *think* it is right, but of course it is=20
>not normative.
>=20
>Note that IMO none of this is intuitive or straightforward.=20
>Every time I get a detailed question on it I have to do a=20
>bunch of review before I can answer it. (But maybe that's=20
>just because I don't have enough neurons left.)

And I thought SIP was supposed to be easy...

One question:

In chapter 6 (I guess that is what you call the annex) of 4596 you say:

"The simplification is primarily because a particular feature tag may
only
appear once in each Contact, Accept-Contact, or Reject-Contact header."

However, as seen before, a particular feature tag may appear more than
once, but in different A-C headers (or ac-values within a single
header).

For example:

Accept-Contact: *;tagQ
Accept-Contact: *;tagQ



And, from a matching perspective, does it really matter if the feature
tags were taken from one or more A-C headers?


For example:

Isn't:

Accept-Contact: *;tagY,tagW

...identical to:

Accept-Contact: *;tagY
Accept-Contact: *;tagW

...or:

Accept-Contact: *;tagY, *;tagW


They will all cause:

(& (tagY) (tagW))

Or?

Regards,

Christer








>=20
> 	Paul
>=20
> Christer Holmberg wrote:
> > Paul,
> >=20
> > Regarding the matching, are you saying that reading RFC3840=20
> is enough=20
> > to understand how the matching is done in a "SIP=20
> environment"? Or, do=20
> > I also need to read RFC2533?
> >=20
> > I have some questions on the matching, but I will come back to them=20
> > later. I still have some reading to do.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> > =20
> >=20
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: 10. tammikuuta 2008 16:21
> >> To: Mahesh Anjanappa
> >> Cc: Christer Holmberg; sip@ietf.org
> >> Subject: Re: [Sip] Question on RFC3840/3841
> >>
> >>
> >>
> >> Mahesh Anjanappa wrote:
> >>> Yes the ';' is a mistake. But i'm not sure if  each=20
> Contact should=20
> >>> contain "alice@atlanta.com"
> >> Well, it has to have a URI to be syntactically correct.
> >>
> >> Your problem is that you are trying to express something=20
> that cannot=20
> >> be expressed with the callee caps syntax. It only maps to=20
> a subset of=20
> >> the syntax supported by RFC 2533.
> >>
> >>> because the predicate that i'm trying to represent is as follows=20
> >>> meanig to say alice supports a communication which whose
> >> capability is
> >>> Duplex Audio + One Way Video.
> >> You just can't say that. Sorry.
> >>
> >> A little history may help you to understand, though it won't solve=20
> >> your
> >> problem:
> >>
> >> The initial draft for callerprefs came out a very long=20
> time ago. (The=20
> >> earliest draft I have is -03 which came out in november of=20
> 2000.) It=20
> >> set the basic syntax and the initial set of capabilities.
> >>
> >> It was however very weak on the semantics of matching. I=20
> eventually=20
> >> got involved because I was interested in it and identified many of=20
> >> the semantic issues. Jonathan and I worked on it for a long time.
> >> Eventually, because of some review comments I think,=20
> Jonathan tried=20
> >> to define the semantics using RFC 2533 because it was a=20
> standard that=20
> >> seemed to be at least partially suitable to the task. (In=20
> retrospect=20
> >> this may not have been such a good idea, since it adds an=20
> additional=20
> >> layer of complexity.)
> >>
> >> RFC 2533 syntax is much richer. But it was not a goal to=20
> provide that=20
> >> richness. (Mapping that all onto SIP header syntax would=20
> be awful.)=20
> >> It was only used as an intermediate form to define the=20
> semantics of=20
> >> the SIP headers.
> >>
> >> Its understood that there are lots of things one might want to=20
> >> express that you can't express.
> >>
> >> If you aren't already aware of it, you might want to look=20
> at RFC 4596=20
> >> (Guidelines for Usage of the Session Initiation Protocol=20
> (SIP) Caller=20
> >> Preferences Extension).
> >>
> >> 	Paul
> >>
> >>> If each of the COntact contains alice@atlanta.com wouldnt it be=20
> >>> interpreted as " alice supports duplex audio and oneway video=20
> >>> separatley and not both at the same time".
> >>> Actually i'm not quite sure how to represesnt the=20
> predicate in the=20
> >>> Contact header correctly, hence my Queries. It all stems from the=20
> >>> statement which you too noted " There MUST only be one
> >> instance of any
> >>> feature tag in feature-param." in RFC 3840
> >>>
> >>>
> >>> Predicate:
> >>> ------------
> >>> & ( & (sip.audio) (sip.duplex=3Dfull) )
> >>>    (  & (sip.video) (sip.duplex=3Dreceive-only) )
> >>>
> >>> regards
> >>> Mahesh
> >>> ----- Original Message ----- From: "Christer Holmberg"=20
> >>> <christer.holmberg@ericsson.com>
> >>> To: "Mahesh Anjanappa" <mahesha@samsung.com>; "Paul Kyzivat"=20
> >>> <pkyzivat@cisco.com>
> >>> Cc: <sip@ietf.org>
> >>> Sent: Thursday, January 10, 2008 11:30 AM
> >>> Subject: RE: [Sip] Question on RFC3840/3841
> >>>
> >>>
> >>>
> >>> Shouldn't each Contact also contain "alice@atlanta.com"?
> >>>
> >>> Also, why do you have ";" at the end of each Contact?
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Mahesh Anjanappa [mailto:mahesha@samsung.com]
> >>>> Sent: 10. tammikuuta 2008 6:16
> >>>> To: Paul Kyzivat
> >>>> Cc: sip@ietf.org; Christer Holmberg
> >>>> Subject: Re: [Sip] Question on RFC3840/3841
> >>>>
> >>>> hi Paul,
> >>>> Sorry there is a mistake in Contact header i=20
> described.Following is=20
> >>>> what i wanted to:
> >>>> Contact:sip:alice@atlanta.com;
> >>>> Contact:audio;duplex=3Dfull;
> >>>> Contact:video;duplex=3Drecieve-only;
> >>>>
> >>>> Thanks
> >>>> Mahesh
> >>>> ----- Original Message -----
> >>>> From: "Mahesh Anjanappa" <mahesha@samsung.com>
> >>>> To: "Paul Kyzivat" <pkyzivat@cisco.com>
> >>>> Cc: <sip@ietf.org>; "Christer Holmberg"
> >>>> <christer.holmberg@ericsson.com>
> >>>> Sent: Wednesday, January 09, 2008 10:48 PM
> >>>> Subject: Re: [Sip] Question on RFC3840/3841
> >>>>
> >>>>
> >>>>> Thanks Paul.
> >>>>> So would it be right if i have a Contact Headers describing
> >>>> Duplex Audio &
> >>>>> one-way Video Capabilities as follows?
> >>>>>
> >>>>>    Contact:sip:alice@atlanta.com;
> >>>>>    Contact:sip:audio;duplex=3Dfull;
> >>>>>    Contact:sip:video;duplex=3Drecieve-only;
> >>>>>
> >>>>> NOTE: The Predicate(for above) for which i'm attempting=20
> is & ( &=20
> >>>>> (sip.audio) (sip.duplex=3Dfull) )
> >>>>>    (  & (sip.video) (sip.duplex=3Dreceive-only) )
> >>>>>
> >>>>> regards
> >>>>> Mahesh
> >>>>>
> >>>>> ----- Original Message ----- > From: "Paul Kyzivat"=20
> >>>> <pkyzivat@cisco.com>
> >>>>> To: "Mahesh Anjanappa" <mahesha@samsung.com>
> >>>>> Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>;
> >>>> <sip@ietf.org>
> >>>>> Sent: Wednesday, January 09, 2008 7:55 PM
> >>>>> Subject: Re: [Sip] Question on RFC3840/3841
> >>>>>
> >>>>>
> >>>>>>
> >>>>>> Mahesh Anjanappa wrote:
> >>>>>>> hi
> >>>>>>> That is my understanding as well. But if that is the case
> >>>> then it should
> >>>>>>> apply to Contact header as well
> >>>>>>> when describing Capabilities using Feature parameters in
> >>>> Contact Header.
> >>>>>>> But i'm not sure if it applies the same way. I had posted
> >>>> this question
> >>>>>>> long ago but didnt get any comments on it.
> >>>>>> The rule forbidding multiple instances of a parameter
> >>>> applies to all
> >>>>>> headers, so it applies to Contact.  E.g. the following
> >> is illegal:
> >>>>>> =
Contact:sip:alice@atlanta.com;methods=3D"INVITE";methods=3D"CANCEL"
> >>>>>>
> >>>>>> Paul
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>>> sipping@ietf.org for new developments on the
> >> application of sip
> >>>>
> >>>
> >=20
>=20


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	<sip@ietf.org>
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=20
Hi,

One question is I have is related to the matching of list values.

First, when a user registers the SIP methods is supports, it can be a
list of methods (e.g. methods=3D"INVITE, BYE,....)

Then, when a request comes, it is automatically assigend an implicit
method preference, e.g. (methods=3D"INVITE").


But, what if the request also contains a list of values.



Assume the user registers the following feature-tag, with a list of
values:

myTag=3Da,b

Which will generate (| (myTag=3Da) (myTag=3Db))

Now, I can receive the following in the Contact:


1.	myTag=3Da,b			I assume that is a match
2.	myTag=3Da			Is that a match?
3.	myTag=3Db			Is that a match?
4.	myTag=3Db,a			Is that a match?
5.	myTag=3Da,b,c			Is that a match?



Now, IF it is enough that either a or b in my request is needed for a
match, but I want both a AND b to match, I guess I would have to do the
following:

Accept-Contact:*, myTag=3Da
Accept-Contact:*, myTag=3Db

(or, Accept-Contact: *;myTag=3Da, *;myTag=3Db

...which would generate:

(& (myTag=3Da) (myTag=3Db))


Regards,

Christer








> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: 10. tammikuuta 2008 17:04
> To: Christer Holmberg
> Cc: Mahesh Anjanappa; sip@ietf.org
> Subject: Re: [Sip] Question on RFC3840/3841
>=20
> Christer,
>=20
> RFC3840 by itself, without 2533, is *not* sufficient to=20
> understand how to do the matching.
>=20
> I put an appendix in RFC 4596 that gives an algorithm without=20
> reference to 2533. I *think* it is right, but of course it is=20
> not normative.
>=20
> Note that IMO none of this is intuitive or straightforward.=20
> Every time I get a detailed question on it I have to do a=20
> bunch of review before I can answer it. (But maybe that's=20
> just because I don't have enough neurons left.)
>=20
> 	Paul
>=20
> Christer Holmberg wrote:
> > Paul,
> >=20
> > Regarding the matching, are you saying that reading RFC3840=20
> is enough=20
> > to understand how the matching is done in a "SIP=20
> environment"? Or, do=20
> > I also need to read RFC2533?
> >=20
> > I have some questions on the matching, but I will come back to them=20
> > later. I still have some reading to do.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> > =20
> >=20
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: 10. tammikuuta 2008 16:21
> >> To: Mahesh Anjanappa
> >> Cc: Christer Holmberg; sip@ietf.org
> >> Subject: Re: [Sip] Question on RFC3840/3841
> >>
> >>
> >>
> >> Mahesh Anjanappa wrote:
> >>> Yes the ';' is a mistake. But i'm not sure if  each=20
> Contact should=20
> >>> contain "alice@atlanta.com"
> >> Well, it has to have a URI to be syntactically correct.
> >>
> >> Your problem is that you are trying to express something=20
> that cannot=20
> >> be expressed with the callee caps syntax. It only maps to=20
> a subset of=20
> >> the syntax supported by RFC 2533.
> >>
> >>> because the predicate that i'm trying to represent is as follows=20
> >>> meanig to say alice supports a communication which whose
> >> capability is
> >>> Duplex Audio + One Way Video.
> >> You just can't say that. Sorry.
> >>
> >> A little history may help you to understand, though it won't solve=20
> >> your
> >> problem:
> >>
> >> The initial draft for callerprefs came out a very long=20
> time ago. (The=20
> >> earliest draft I have is -03 which came out in november of=20
> 2000.) It=20
> >> set the basic syntax and the initial set of capabilities.
> >>
> >> It was however very weak on the semantics of matching. I=20
> eventually=20
> >> got involved because I was interested in it and identified many of=20
> >> the semantic issues. Jonathan and I worked on it for a long time.
> >> Eventually, because of some review comments I think,=20
> Jonathan tried=20
> >> to define the semantics using RFC 2533 because it was a=20
> standard that=20
> >> seemed to be at least partially suitable to the task. (In=20
> retrospect=20
> >> this may not have been such a good idea, since it adds an=20
> additional=20
> >> layer of complexity.)
> >>
> >> RFC 2533 syntax is much richer. But it was not a goal to=20
> provide that=20
> >> richness. (Mapping that all onto SIP header syntax would=20
> be awful.)=20
> >> It was only used as an intermediate form to define the=20
> semantics of=20
> >> the SIP headers.
> >>
> >> Its understood that there are lots of things one might want to=20
> >> express that you can't express.
> >>
> >> If you aren't already aware of it, you might want to look=20
> at RFC 4596=20
> >> (Guidelines for Usage of the Session Initiation Protocol=20
> (SIP) Caller=20
> >> Preferences Extension).
> >>
> >> 	Paul
> >>
> >>> If each of the COntact contains alice@atlanta.com wouldnt it be=20
> >>> interpreted as " alice supports duplex audio and oneway video=20
> >>> separatley and not both at the same time".
> >>> Actually i'm not quite sure how to represesnt the=20
> predicate in the=20
> >>> Contact header correctly, hence my Queries. It all stems from the=20
> >>> statement which you too noted " There MUST only be one
> >> instance of any
> >>> feature tag in feature-param." in RFC 3840
> >>>
> >>>
> >>> Predicate:
> >>> ------------
> >>> & ( & (sip.audio) (sip.duplex=3Dfull) )
> >>>    (  & (sip.video) (sip.duplex=3Dreceive-only) )
> >>>
> >>> regards
> >>> Mahesh
> >>> ----- Original Message ----- From: "Christer Holmberg"=20
> >>> <christer.holmberg@ericsson.com>
> >>> To: "Mahesh Anjanappa" <mahesha@samsung.com>; "Paul Kyzivat"=20
> >>> <pkyzivat@cisco.com>
> >>> Cc: <sip@ietf.org>
> >>> Sent: Thursday, January 10, 2008 11:30 AM
> >>> Subject: RE: [Sip] Question on RFC3840/3841
> >>>
> >>>
> >>>
> >>> Shouldn't each Contact also contain "alice@atlanta.com"?
> >>>
> >>> Also, why do you have ";" at the end of each Contact?
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Mahesh Anjanappa [mailto:mahesha@samsung.com]
> >>>> Sent: 10. tammikuuta 2008 6:16
> >>>> To: Paul Kyzivat
> >>>> Cc: sip@ietf.org; Christer Holmberg
> >>>> Subject: Re: [Sip] Question on RFC3840/3841
> >>>>
> >>>> hi Paul,
> >>>> Sorry there is a mistake in Contact header i=20
> described.Following is=20
> >>>> what i wanted to:
> >>>> Contact:sip:alice@atlanta.com;
> >>>> Contact:audio;duplex=3Dfull;
> >>>> Contact:video;duplex=3Drecieve-only;
> >>>>
> >>>> Thanks
> >>>> Mahesh
> >>>> ----- Original Message -----
> >>>> From: "Mahesh Anjanappa" <mahesha@samsung.com>
> >>>> To: "Paul Kyzivat" <pkyzivat@cisco.com>
> >>>> Cc: <sip@ietf.org>; "Christer Holmberg"
> >>>> <christer.holmberg@ericsson.com>
> >>>> Sent: Wednesday, January 09, 2008 10:48 PM
> >>>> Subject: Re: [Sip] Question on RFC3840/3841
> >>>>
> >>>>
> >>>>> Thanks Paul.
> >>>>> So would it be right if i have a Contact Headers describing
> >>>> Duplex Audio &
> >>>>> one-way Video Capabilities as follows?
> >>>>>
> >>>>>    Contact:sip:alice@atlanta.com;
> >>>>>    Contact:sip:audio;duplex=3Dfull;
> >>>>>    Contact:sip:video;duplex=3Drecieve-only;
> >>>>>
> >>>>> NOTE: The Predicate(for above) for which i'm attempting=20
> is & ( &=20
> >>>>> (sip.audio) (sip.duplex=3Dfull) )
> >>>>>    (  & (sip.video) (sip.duplex=3Dreceive-only) )
> >>>>>
> >>>>> regards
> >>>>> Mahesh
> >>>>>
> >>>>> ----- Original Message ----- > From: "Paul Kyzivat"=20
> >>>> <pkyzivat@cisco.com>
> >>>>> To: "Mahesh Anjanappa" <mahesha@samsung.com>
> >>>>> Cc: "Christer Holmberg" <christer.holmberg@ericsson.com>;
> >>>> <sip@ietf.org>
> >>>>> Sent: Wednesday, January 09, 2008 7:55 PM
> >>>>> Subject: Re: [Sip] Question on RFC3840/3841
> >>>>>
> >>>>>
> >>>>>>
> >>>>>> Mahesh Anjanappa wrote:
> >>>>>>> hi
> >>>>>>> That is my understanding as well. But if that is the case
> >>>> then it should
> >>>>>>> apply to Contact header as well
> >>>>>>> when describing Capabilities using Feature parameters in
> >>>> Contact Header.
> >>>>>>> But i'm not sure if it applies the same way. I had posted
> >>>> this question
> >>>>>>> long ago but didnt get any comments on it.
> >>>>>> The rule forbidding multiple instances of a parameter
> >>>> applies to all
> >>>>>> headers, so it applies to Contact.  E.g. the following
> >> is illegal:
> >>>>>> =
Contact:sip:alice@atlanta.com;methods=3D"INVITE";methods=3D"CANCEL"
> >>>>>>
> >>>>>> Paul
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>>> sipping@ietf.org for new developments on the
> >> application of sip
> >>>>
> >>>
> >=20
>=20


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From sip-bounces@ietf.org Fri Jan 18 08:33:58 2008
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Date: Fri, 18 Jan 2008 08:33:46 -0500
Subject: RE: [Sip] Interesting problem with PRACK and resent INVITEs
Thread-Topic: [Sip] Interesting problem with PRACK and resent INVITEs
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I believe that a CSeq of 2 is perfectly valid for the second INVITE (resubm=
itted with credentials). The RFC 3261 is clear that it should have the same=
 Call-ID and from-tag.  The CSeq number space is per dialog. This second IN=
VITE is not part of any dialog, therefore its CSeq should not be compared w=
ith any previous dialog.

If a UA does keep CSeq state for a Call-ID it should be independent of the =
dialog CSeq state, and not updated by in-dialog requests.

cheers,
(-:bob

-----Original Message-----
From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
Sent: Friday, January 18, 2008 3:02 AM
To: Christer Holmberg
Cc: sip@ietf.org; Paul Kyzivat
Subject: Re: [Sip] Interesting problem with PRACK and resent INVITEs

Christer,
>> .
>>
>>
>> Regarding from tags there is no explicit text, but 22.2 says:
>>
>> 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.
>>
>
> So, I guess that means that the second INVITE in Dale's example should
> have Cseq =3D> 3.

I interpreted this as "MUST increment (relative to the INVITE being
challenged)". I guess for initial INVITEs the minimal requirement is
that the CSeq be different, else merged request detection (8.2.2.2)
could kick in and prevent the call from succeeding. So both 2 or 3 are
valid, the text is ambiguous

Regards,
Jeroen


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


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From sip-bounces@ietf.org Fri Jan 18 08:39:31 2008
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Thread-Topic: Important deadlines for Philadelphia
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(As SIP WG chair)

Important dates for Philadelphia are at

http://www.ietf.org/meetings/71-cutoff_dates.html

The ones we are interested in for progressing the work of the SIP WG
are:

February 18, Monday - Internet Draft Cut-off for initial document (-00)
submission by 17:00 ET (22:00 UTC/GMT)
February 25, Monday - Internet Draft final submission cut-off by 17:00
ET (22:00 UTC/GMT)

If you think you have a WG draft at -00 level that you are submitting,
then it would be a good idea to talk to Dean and myself about it before
you submit it 3 minutes before the deadline.

We recommend you use the automatic submission tool if at all possible,
as this minimises the delay in getting your draft visible to the rest of
us out here who are all eagerly waiting to read it. It is here:

https://datatracker.ietf.org/idst/upload.cgi

It helps if you run ID-nits on the document before submission, and
correct the problems (that way you have only yourselves to blame if
boilerplate errors delay the submission):

http://tools.ietf.org/tools/idnits/

Please start thinking about what should be having agenda time. Dean will
be issuing his request for slots in due course.

Regards

Keith


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I've been busy so I have been only loosely following this discussion.

Hadriel Kaplan wrote:
> If I read the draft correctly, the proposed solution is that rerouting action leaves the req-uri alone, and pushes a Route for the resolved target's URI (contact/whatever).  Whereas a retargeting action replaces the req-uri with the new target, and possibly pushes the previous req-uri into History-Info.  The draft proposes retargeting be defined as when the proxy determines the new target could not claim the previous req-uri as its Identity.  In other words, if the new target were to originate a request using the previous req-uri as its From addr-spec, would its domain sign it using rfc4474 - if so then it's a reroute, if not then it's a retarget.  Correct?

Given dean's terminology the above statement seems a little imprecise. 
It puts things in only two buckets, not the three or more that Dean 
proposes. I think that impacts the following.

> The draft also describes several cases, some of which don't seem to follow the above rules.
> 
> For example, Section 5 example 3 says:
>    3.  When a proxy receiving a request identifies a next hop server
>        that is needed to process the request, that next hop server is a
>        route.  A next hop server is not a UA and would never be able to
>        claim its identity.  Its URI is pushed into a Route header field
>        and the Request-URI is retained.  An important use case for this
>        are proxies that select PSTN gateways for call egress to the
>        PSTN.  Such selection would place the SIP URI of the gateway into
>        the topmost Route header field value and retain the Request-URI.

The above is also less than precise.

> So this says a PSTN gateway could not claim the identity, yet the behavior is still rerouting?

There is a problem here one way or another. A PSTN GW *is* a UA. And if 
the address is a phone number, then it *can* claim identity to it.

> Another example of this is in Section 2.6 for Emergency Numbers.  It states the SOS URN needs to stay in place in the req-uri all the way to the call-taker.  I'm not sure if such a call-taker could claim Identity to an SOS URN, and furthermore the odds are the call-taker is in another domain, which per this draft would constitute a retargeting and change the req-uri.  No?

I see URNs as much like phone numbers, in that they aren't associated 
with domains. There semantics are intended to be known publicly, and 
there may be many different UAs that can claim to be responsible for 
them. In normal emergency processing, the receiving UA *should* be able 
to claim identity for that URN.

Of course there are technical problems for both phone numbers and URNs 
in that we don't know how to apply sip identity to them.

> This draft should also describe what happens for service invocation, which is a stated problem space in section 2.5.  Again, a voicemail system or IVR or App Server could not claim Identity for the req-uri, and thus it would be a retarget per the draft's definition.

I think it is debatable whether a VM or IVR system can claim identity 
for the r-uri. It depends on the deployment and the policies of the 
owner of that URI.

(If I have a phone in my home, listed as belonging to me, and I attach 
an answering machine to it, that answering machine gets to assert my 
identity as much as I do. If I contract with a SP to provide my 
answering services, it can still claim my identity if I want to allow it 
to do so.

Its the same with a human secretary. I can give a secretary an extension 
on my own AOR, in which case its possible to claim my identity, or I can 
program my proxy to retarget to the secretary, in which case the 
identity is different.)

	Paul

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


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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Jeroen van Bemmel <jbemmel@zonnet.nl>
Subject: Re: [Sip] 	Vocabulary and problem statement for Request-URI,
	retargeting, and SIP routing (long, but read it!)
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com>	<1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>	A	<CA9998CD4A020D418654FCDEF4E707DF040D69C7@esealmw113.eemea.ericsson.se>	<0D5F89FAC29E2C41B98A6A762007F5D0549D3F@GBNTHT12009MSX.gb002.siemens.net>	<1ECE0EB50388174790F9694F77522CCF1438F1B0@zrc2hxm0.corp.nortel.com>	<478CEFB4.6070002@zonnet.nl>	<CA9998CD4A020D418654FCDEF4E707DF0413D587@esealmw113.eemea.ericsson.se>	<"0D5F89FA	C29E2C41B98A6A762007F5D0593C68"@GBNTHT12009MSX.gb002.siemen!	s	.net>
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Jeroen van Bemmel wrote:

> BTW what happened to the 'opaque' parameter solution that was part of a 
> GRUU draft some versions ago? A special URI parameter that survives 
> rerouting/retargeting would avoid introducing yet another identity 
> header, and also detection-of-support issues.

It was judged to be a hack and rejected. That was one of the motivations 
for the current proposal, which (at the time) was thought to be cleaner. 
In hindsight maybe it wasn't so much cleaner. :-)

	Paul


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Elwell, John" <john.elwell@siemens.com>, Dean Willis
	<dean.willis@softarmor.com>, IETF SIP List <sip@ietf.org>
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Subject: RE: [Sip] 	Vocabulary and problem statement for
	Request-URI,retargeting,	and SIP routing (long, but read it!)
Thread-Topic: [Sip] 	Vocabulary and problem statement for
	Request-URI,retargeting,	and SIP routing (long, but read it!)
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Hey John,
Inline...

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Friday, January 18, 2008 2:48 AM
> >
> > But in the re-targeting scenario such as:
> >                     RTRG                    RRT
> >                    +---+                   +---+
> >                    |R1 |                   |R2 |
> >                 B /+---+\ C             E /+---+\ F
> >             RT   /       \  RT      RT   /       \  RT
> >            +---+/         \+---+ D +---+/         \+---+
> >            |P1 |           |P2 +---+P3 |           |P4 |
> >         A /+---+           +---+   +---+           +---+\ G
> >          /                                               \
> >    +---+/                                                 \+---+
> >    |UAC|                                                   |UAS|
> >    +---+                                                   +---+
> >
> > UA-Loose-routing wants the req-uri seen on connection "C" I think.
> > To header gives you A.
> > PCPID gives you E.
> > Hist-Info gives you A,B,C,D,E,F.
> [JRE] According to Dean's definition of RT, it does not change the
> Request-URI (only the Route header field presumably, or maybe not even
> that).
> So C, D and E are the same. Also A and B are the same, and F and G are
> the same.
> So I think:
> - UA-Loose-routing gives you C/D/E
> - To gives you A/B
> - PCPID gives you C/D/E
> - Target gives you C/D/E
> - Hist-info gives you A/B, C/D/E and F/G.

Yes, I agree that is the *theory*.  :)
I drew it that way though so we could argue about what the UAS/UALR-draft _=
wants_ to happen vs. what _will_ happen if P2 or P3 are not purely RT's and=
 didn't support a new draft. (Since it seemed the conversation was going th=
at way previously on this list, for example when Christer pointed out the d=
ifference between Target and PCPID)

For example, I think there is more than just a syntax difference between Ch=
rister's sip-target-uri-delivery draft (STUD?) and Jonathan's UALR approach=
.  Though I have no idea which one is better.

-hadriel



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Date: Fri, 18 Jan 2008 15:28:12 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] 	Vocabulary and problem statement
	forRequest-URI,retargeting,	and SIP routing (long, but read it!)
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Hadriel,

Well, I have been having a side thread with Christer and Hans Erik, and
the only difference other than syntax that they could convince me of was
support for UAs that do not register, in that with loose-route you would
need additional provisioning in the domain proxy to say that the UA
(gateway or whatever) supports loose-route. Given that you would need
provisioning in the proxy anyway for such UAs, I didn't see this as a
big deal, but it is a difference.

John=20

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: 18 January 2008 15:15
> To: Elwell, John; Dean Willis; IETF SIP List
> Subject: RE: [Sip] Vocabulary and problem statement=20
> forRequest-URI,retargeting, and SIP routing (long, but read it!)
>=20
> Hey John,
> Inline...
>=20
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com]
> > Sent: Friday, January 18, 2008 2:48 AM
> > >
> > > But in the re-targeting scenario such as:
> > >                     RTRG                    RRT
> > >                    +---+                   +---+
> > >                    |R1 |                   |R2 |
> > >                 B /+---+\ C             E /+---+\ F
> > >             RT   /       \  RT      RT   /       \  RT
> > >            +---+/         \+---+ D +---+/         \+---+
> > >            |P1 |           |P2 +---+P3 |           |P4 |
> > >         A /+---+           +---+   +---+           +---+\ G
> > >          /                                               \
> > >    +---+/                                                 \+---+
> > >    |UAC|                                                   |UAS|
> > >    +---+                                                   +---+
> > >
> > > UA-Loose-routing wants the req-uri seen on connection "C" I think.
> > > To header gives you A.
> > > PCPID gives you E.
> > > Hist-Info gives you A,B,C,D,E,F.
> > [JRE] According to Dean's definition of RT, it does not change the
> > Request-URI (only the Route header field presumably, or=20
> maybe not even
> > that).
> > So C, D and E are the same. Also A and B are the same, and=20
> F and G are
> > the same.
> > So I think:
> > - UA-Loose-routing gives you C/D/E
> > - To gives you A/B
> > - PCPID gives you C/D/E
> > - Target gives you C/D/E
> > - Hist-info gives you A/B, C/D/E and F/G.
>=20
> Yes, I agree that is the *theory*.  :)
> I drew it that way though so we could argue about what the=20
> UAS/UALR-draft _wants_ to happen vs. what _will_ happen if P2=20
> or P3 are not purely RT's and didn't support a new draft.=20
> (Since it seemed the conversation was going that way=20
> previously on this list, for example when Christer pointed=20
> out the difference between Target and PCPID)
>=20
> For example, I think there is more than just a syntax=20
> difference between Christer's sip-target-uri-delivery draft=20
> (STUD?) and Jonathan's UALR approach.  Though I have no idea=20
> which one is better.
>=20
> -hadriel
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Elwell, John" <john.elwell@siemens.com>, Dean Willis
	<dean.willis@softarmor.com>, IETF SIP List <sip@ietf.org>
Date: Fri, 18 Jan 2008 10:52:02 -0500
Subject: RE: [Sip] 	Vocabulary and problem statement
	forRequest-URI,retargeting,	and SIP routing (long, but read it!)
Thread-Topic: [Sip] 	Vocabulary and problem statement
	forRequest-URI,retargeting,	and SIP routing (long, but read it!)
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Yeah I like STUD because it's explicit.
But the big difference I see is what happens if R1 supports STUD but P2, or=
 more likely a domain between P2 and P3, Retarget but don't support STUD.  =
Then R2 gets a Target which is wrong - it's the URI of C vs. D (or somethin=
g between D).  In other words it could be the target uri of a completely di=
fferent user/domain.

With UALR, since retargeting proxies already replace the req-uri today, you=
'd get a req-uri which isn't wrong.  Even if P2 did Rerouting and not UALR =
(ie, still replaced the req-uri but was not really retargeting), the req-ur=
i received by R2 wouldn't be totally wrong.  It would lose the parameters a=
nd info that could have been useful, but it's less likely to cause harm. (m=
aybe?)

The counter-argument is what happens if P3 doesn't do UALR, but I would thi=
nk it's far more likely that an operator of a domain has control over the e=
quipment in their domain.

Though I'm probably thinking in circles - this whole topic makes my head hu=
rt.  :)

-hadriel

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Friday, January 18, 2008 10:28 AM
> To: Hadriel Kaplan; Dean Willis; IETF SIP List
> Subject: RE: [Sip] Vocabulary and problem statement forRequest-
> URI,retargeting, and SIP routing (long, but read it!)
>
> Hadriel,
>
> Well, I have been having a side thread with Christer and Hans Erik, and
> the only difference other than syntax that they could convince me of was
> support for UAs that do not register, in that with loose-route you would
> need additional provisioning in the domain proxy to say that the UA
> (gateway or whatever) supports loose-route. Given that you would need
> provisioning in the proxy anyway for such UAs, I didn't see this as a
> big deal, but it is a difference.
>
> John
>
> > -----Original Message-----
> > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > Sent: 18 January 2008 15:15
> > To: Elwell, John; Dean Willis; IETF SIP List
> > Subject: RE: [Sip] Vocabulary and problem statement
> > forRequest-URI,retargeting, and SIP routing (long, but read it!)
> >
> > Hey John,
> > Inline...
> >
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > Sent: Friday, January 18, 2008 2:48 AM
> > > >
> > > > But in the re-targeting scenario such as:
> > > >                     RTRG                    RRT
> > > >                    +---+                   +---+
> > > >                    |R1 |                   |R2 |
> > > >                 B /+---+\ C             E /+---+\ F
> > > >             RT   /       \  RT      RT   /       \  RT
> > > >            +---+/         \+---+ D +---+/         \+---+
> > > >            |P1 |           |P2 +---+P3 |           |P4 |
> > > >         A /+---+           +---+   +---+           +---+\ G
> > > >          /                                               \
> > > >    +---+/                                                 \+---+
> > > >    |UAC|                                                   |UAS|
> > > >    +---+                                                   +---+
> > > >
> > > > UA-Loose-routing wants the req-uri seen on connection "C" I think.
> > > > To header gives you A.
> > > > PCPID gives you E.
> > > > Hist-Info gives you A,B,C,D,E,F.
> > > [JRE] According to Dean's definition of RT, it does not change the
> > > Request-URI (only the Route header field presumably, or
> > maybe not even
> > > that).
> > > So C, D and E are the same. Also A and B are the same, and
> > F and G are
> > > the same.
> > > So I think:
> > > - UA-Loose-routing gives you C/D/E
> > > - To gives you A/B
> > > - PCPID gives you C/D/E
> > > - Target gives you C/D/E
> > > - Hist-info gives you A/B, C/D/E and F/G.
> >
> > Yes, I agree that is the *theory*.  :)
> > I drew it that way though so we could argue about what the
> > UAS/UALR-draft _wants_ to happen vs. what _will_ happen if P2
> > or P3 are not purely RT's and didn't support a new draft.
> > (Since it seemed the conversation was going that way
> > previously on this list, for example when Christer pointed
> > out the difference between Target and PCPID)
> >
> > For example, I think there is more than just a syntax
> > difference between Christer's sip-target-uri-delivery draft
> > (STUD?) and Jonathan's UALR approach.  Though I have no idea
> > which one is better.
> >
> > -hadriel
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Elwell, John wrote:
> Hadriel,
> 
> Well, I have been having a side thread with Christer and Hans Erik, and
> the only difference other than syntax that they could convince me of was
> support for UAs that do not register, in that with loose-route you would
> need additional provisioning in the domain proxy to say that the UA
> (gateway or whatever) supports loose-route. Given that you would need
> provisioning in the proxy anyway for such UAs, I didn't see this as a
> big deal, but it is a difference.

In the case of registration, it is indicated by the ;lr param on the 
contact. In the case of provisioning, you need to provision a contact, 
so its just a matter of provisioning it with ;lr. That is *still* just 
provisioning a uri, so its not really a difference.

	Paul

> John 
> 
>> -----Original Message-----
>> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
>> Sent: 18 January 2008 15:15
>> To: Elwell, John; Dean Willis; IETF SIP List
>> Subject: RE: [Sip] Vocabulary and problem statement 
>> forRequest-URI,retargeting, and SIP routing (long, but read it!)
>>
>> Hey John,
>> Inline...
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens.com]
>>> Sent: Friday, January 18, 2008 2:48 AM
>>>> But in the re-targeting scenario such as:
>>>>                     RTRG                    RRT
>>>>                    +---+                   +---+
>>>>                    |R1 |                   |R2 |
>>>>                 B /+---+\ C             E /+---+\ F
>>>>             RT   /       \  RT      RT   /       \  RT
>>>>            +---+/         \+---+ D +---+/         \+---+
>>>>            |P1 |           |P2 +---+P3 |           |P4 |
>>>>         A /+---+           +---+   +---+           +---+\ G
>>>>          /                                               \
>>>>    +---+/                                                 \+---+
>>>>    |UAC|                                                   |UAS|
>>>>    +---+                                                   +---+
>>>>
>>>> UA-Loose-routing wants the req-uri seen on connection "C" I think.
>>>> To header gives you A.
>>>> PCPID gives you E.
>>>> Hist-Info gives you A,B,C,D,E,F.
>>> [JRE] According to Dean's definition of RT, it does not change the
>>> Request-URI (only the Route header field presumably, or 
>> maybe not even
>>> that).
>>> So C, D and E are the same. Also A and B are the same, and 
>> F and G are
>>> the same.
>>> So I think:
>>> - UA-Loose-routing gives you C/D/E
>>> - To gives you A/B
>>> - PCPID gives you C/D/E
>>> - Target gives you C/D/E
>>> - Hist-info gives you A/B, C/D/E and F/G.
>> Yes, I agree that is the *theory*.  :)
>> I drew it that way though so we could argue about what the 
>> UAS/UALR-draft _wants_ to happen vs. what _will_ happen if P2 
>> or P3 are not purely RT's and didn't support a new draft. 
>> (Since it seemed the conversation was going that way 
>> previously on this list, for example when Christer pointed 
>> out the difference between Target and PCPID)
>>
>> For example, I think there is more than just a syntax 
>> difference between Christer's sip-target-uri-delivery draft 
>> (STUD?) and Jonathan's UALR approach.  Though I have no idea 
>> which one is better.
>>
>> -hadriel
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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Date: Fri, 18 Jan 2008 16:06:12 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] 	Vocabulary and problem statement
	forRequest-URI,retargeting,	and SIP routing (long, but read it!)
In-reply-to: <E6C2E8958BA59A4FB960963D475F7AC306E4ED4A3A@mail.acmepacket.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>,
	Dean Willis <dean.willis@softarmor.com>, IETF SIP List <sip@ietf.org>
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=20

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: 18 January 2008 15:52
> To: Elwell, John; Dean Willis; IETF SIP List
> Subject: RE: [Sip] Vocabulary and problem statement=20
> forRequest-URI,retargeting, and SIP routing (long, but read it!)
>=20
> Yeah I like STUD because it's explicit.
> But the big difference I see is what happens if R1 supports=20
> STUD but P2, or more likely a domain between P2 and P3,=20
> Retarget but don't support STUD.  Then R2 gets a Target which=20
> is wrong - it's the URI of C vs. D (or something between D). =20
> In other words it could be the target uri of a completely=20
> different user/domain.
>=20
> With UALR, since retargeting proxies already replace the=20
> req-uri today, you'd get a req-uri which isn't wrong.  Even=20
> if P2 did Rerouting and not UALR (ie, still replaced the=20
> req-uri but was not really retargeting), the req-uri received=20
> by R2 wouldn't be totally wrong.  It would lose the=20
> parameters and info that could have been useful, but it's=20
> less likely to cause harm. (maybe?)
>=20
> The counter-argument is what happens if P3 doesn't do UALR,=20
> but I would think it's far more likely that an operator of a=20
> domain has control over the equipment in their domain.
[JRE] And also, P2 should be just routing on the Route header field,
because there is still something in the stack. So it would have to be a
misbehaving proxy to get it wrong.
But in any case, I thought UALR was meant to be applied only on the last
hop (or the last set of hops) from the domain proxy to the UAS, so once
you invoke it there should not be any more retargeting or rerouting
proxies on the path. Maybe the domain proxy is never able to be certain
of this.

John

>=20
> Though I'm probably thinking in circles - this whole topic=20
> makes my head hurt.  :)
>=20
> -hadriel
>=20
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com]
> > Sent: Friday, January 18, 2008 10:28 AM
> > To: Hadriel Kaplan; Dean Willis; IETF SIP List
> > Subject: RE: [Sip] Vocabulary and problem statement forRequest-
> > URI,retargeting, and SIP routing (long, but read it!)
> >
> > Hadriel,
> >
> > Well, I have been having a side thread with Christer and=20
> Hans Erik, and
> > the only difference other than syntax that they could=20
> convince me of was
> > support for UAs that do not register, in that with=20
> loose-route you would
> > need additional provisioning in the domain proxy to say that the UA
> > (gateway or whatever) supports loose-route. Given that you=20
> would need
> > provisioning in the proxy anyway for such UAs, I didn't see=20
> this as a
> > big deal, but it is a difference.
> >
> > John
> >
> > > -----Original Message-----
> > > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > > Sent: 18 January 2008 15:15
> > > To: Elwell, John; Dean Willis; IETF SIP List
> > > Subject: RE: [Sip] Vocabulary and problem statement
> > > forRequest-URI,retargeting, and SIP routing (long, but read it!)
> > >
> > > Hey John,
> > > Inline...
> > >
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > > Sent: Friday, January 18, 2008 2:48 AM
> > > > >
> > > > > But in the re-targeting scenario such as:
> > > > >                     RTRG                    RRT
> > > > >                    +---+                   +---+
> > > > >                    |R1 |                   |R2 |
> > > > >                 B /+---+\ C             E /+---+\ F
> > > > >             RT   /       \  RT      RT   /       \  RT
> > > > >            +---+/         \+---+ D +---+/         \+---+
> > > > >            |P1 |           |P2 +---+P3 |           |P4 |
> > > > >         A /+---+           +---+   +---+           +---+\ G
> > > > >          /                                               \
> > > > >    +---+/                                            =20
>     \+---+
> > > > >    |UAC|                                             =20
>      |UAS|
> > > > >    +---+                                             =20
>      +---+
> > > > >
> > > > > UA-Loose-routing wants the req-uri seen on connection=20
> "C" I think.
> > > > > To header gives you A.
> > > > > PCPID gives you E.
> > > > > Hist-Info gives you A,B,C,D,E,F.
> > > > [JRE] According to Dean's definition of RT, it does not=20
> change the
> > > > Request-URI (only the Route header field presumably, or
> > > maybe not even
> > > > that).
> > > > So C, D and E are the same. Also A and B are the same, and
> > > F and G are
> > > > the same.
> > > > So I think:
> > > > - UA-Loose-routing gives you C/D/E
> > > > - To gives you A/B
> > > > - PCPID gives you C/D/E
> > > > - Target gives you C/D/E
> > > > - Hist-info gives you A/B, C/D/E and F/G.
> > >
> > > Yes, I agree that is the *theory*.  :)
> > > I drew it that way though so we could argue about what the
> > > UAS/UALR-draft _wants_ to happen vs. what _will_ happen if P2
> > > or P3 are not purely RT's and didn't support a new draft.
> > > (Since it seemed the conversation was going that way
> > > previously on this list, for example when Christer pointed
> > > out the difference between Target and PCPID)
> > >
> > > For example, I think there is more than just a syntax
> > > difference between Christer's sip-target-uri-delivery draft
> > > (STUD?) and Jonathan's UALR approach.  Though I have no idea
> > > which one is better.
> > >
> > > -hadriel
> > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the=20
> application of sip
> > >
>=20


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



From sip-bounces@ietf.org Fri Jan 18 11:07:05 2008
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Date: Fri, 18 Jan 2008 16:06:56 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] 	Vocabulary and problem statementforRequest-URI, retargeting,
	and SIP routing (long, but read it!)
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Errors-To: sip-bounces@ietf.org

Paul,

Yes, that was my feeling.

John=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: 18 January 2008 16:00
> To: Elwell, John
> Cc: Hadriel Kaplan; Dean Willis; IETF SIP List
> Subject: Re: [Sip] Vocabulary and problem=20
> statementforRequest-URI,retargeting, and SIP routing (long,=20
> but read it!)
>=20
>=20
>=20
> Elwell, John wrote:
> > Hadriel,
> >=20
> > Well, I have been having a side thread with Christer and=20
> Hans Erik, and
> > the only difference other than syntax that they could=20
> convince me of was
> > support for UAs that do not register, in that with=20
> loose-route you would
> > need additional provisioning in the domain proxy to say that the UA
> > (gateway or whatever) supports loose-route. Given that you=20
> would need
> > provisioning in the proxy anyway for such UAs, I didn't see=20
> this as a
> > big deal, but it is a difference.
>=20
> In the case of registration, it is indicated by the ;lr param on the=20
> contact. In the case of provisioning, you need to provision a=20
> contact,=20
> so its just a matter of provisioning it with ;lr. That is=20
> *still* just=20
> provisioning a uri, so its not really a difference.
>=20
> 	Paul
>=20
> > John=20
> >=20
> >> -----Original Message-----
> >> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> >> Sent: 18 January 2008 15:15
> >> To: Elwell, John; Dean Willis; IETF SIP List
> >> Subject: RE: [Sip] Vocabulary and problem statement=20
> >> forRequest-URI,retargeting, and SIP routing (long, but read it!)
> >>
> >> Hey John,
> >> Inline...
> >>
> >>> -----Original Message-----
> >>> From: Elwell, John [mailto:john.elwell@siemens.com]
> >>> Sent: Friday, January 18, 2008 2:48 AM
> >>>> But in the re-targeting scenario such as:
> >>>>                     RTRG                    RRT
> >>>>                    +---+                   +---+
> >>>>                    |R1 |                   |R2 |
> >>>>                 B /+---+\ C             E /+---+\ F
> >>>>             RT   /       \  RT      RT   /       \  RT
> >>>>            +---+/         \+---+ D +---+/         \+---+
> >>>>            |P1 |           |P2 +---+P3 |           |P4 |
> >>>>         A /+---+           +---+   +---+           +---+\ G
> >>>>          /                                               \
> >>>>    +---+/                                                 \+---+
> >>>>    |UAC|                                                   |UAS|
> >>>>    +---+                                                   +---+
> >>>>
> >>>> UA-Loose-routing wants the req-uri seen on connection=20
> "C" I think.
> >>>> To header gives you A.
> >>>> PCPID gives you E.
> >>>> Hist-Info gives you A,B,C,D,E,F.
> >>> [JRE] According to Dean's definition of RT, it does not change the
> >>> Request-URI (only the Route header field presumably, or=20
> >> maybe not even
> >>> that).
> >>> So C, D and E are the same. Also A and B are the same, and=20
> >> F and G are
> >>> the same.
> >>> So I think:
> >>> - UA-Loose-routing gives you C/D/E
> >>> - To gives you A/B
> >>> - PCPID gives you C/D/E
> >>> - Target gives you C/D/E
> >>> - Hist-info gives you A/B, C/D/E and F/G.
> >> Yes, I agree that is the *theory*.  :)
> >> I drew it that way though so we could argue about what the=20
> >> UAS/UALR-draft _wants_ to happen vs. what _will_ happen if P2=20
> >> or P3 are not purely RT's and didn't support a new draft.=20
> >> (Since it seemed the conversation was going that way=20
> >> previously on this list, for example when Christer pointed=20
> >> out the difference between Target and PCPID)
> >>
> >> For example, I think there is more than just a syntax=20
> >> difference between Christer's sip-target-uri-delivery draft=20
> >> (STUD?) and Jonathan's UALR approach.  Though I have no idea=20
> >> which one is better.
> >>
> >> -hadriel
> >>
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >>
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sip] 	Vocabulary and problem statement	forRequest-URI,
	retargeting, and SIP routing (long, but read it!)
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com>	<"CA9998CD4A0	20D418654FCDEF4E707DF04173CB8"@esealmw113.eemea.ericsson.se>	<"
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Elwell, John wrote:

> [JRE] And also, P2 should be just routing on the Route header field,
> because there is still something in the stack. So it would have to be a
> misbehaving proxy to get it wrong.

I had always assumed that as long as there was something in the Route 
stack the R-URI would be left alone.

But if you read 3261 carefully it nowhere says that. The rules for doing 
translations on the R-URI (if you are responsible for it) are 
independent of the routing rules. So it seems valid (maybe mandatory, 
though that isn't clear to me) that you might translate the R-URI even 
though there is a Route header, and *then* route based on the Route header.

	Paul

> But in any case, I thought UALR was meant to be applied only on the last
> hop (or the last set of hops) from the domain proxy to the UAS, so once
> you invoke it there should not be any more retargeting or rerouting
> proxies on the path. Maybe the domain proxy is never able to be certain
> of this.
> 
> John
> 
>> Though I'm probably thinking in circles - this whole topic 
>> makes my head hurt.  :)
>>
>> -hadriel
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens.com]
>>> Sent: Friday, January 18, 2008 10:28 AM
>>> To: Hadriel Kaplan; Dean Willis; IETF SIP List
>>> Subject: RE: [Sip] Vocabulary and problem statement forRequest-
>>> URI,retargeting, and SIP routing (long, but read it!)
>>>
>>> Hadriel,
>>>
>>> Well, I have been having a side thread with Christer and 
>> Hans Erik, and
>>> the only difference other than syntax that they could 
>> convince me of was
>>> support for UAs that do not register, in that with 
>> loose-route you would
>>> need additional provisioning in the domain proxy to say that the UA
>>> (gateway or whatever) supports loose-route. Given that you 
>> would need
>>> provisioning in the proxy anyway for such UAs, I didn't see 
>> this as a
>>> big deal, but it is a difference.
>>>
>>> John
>>>
>>>> -----Original Message-----
>>>> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
>>>> Sent: 18 January 2008 15:15
>>>> To: Elwell, John; Dean Willis; IETF SIP List
>>>> Subject: RE: [Sip] Vocabulary and problem statement
>>>> forRequest-URI,retargeting, and SIP routing (long, but read it!)
>>>>
>>>> Hey John,
>>>> Inline...
>>>>
>>>>> -----Original Message-----
>>>>> From: Elwell, John [mailto:john.elwell@siemens.com]
>>>>> Sent: Friday, January 18, 2008 2:48 AM
>>>>>> But in the re-targeting scenario such as:
>>>>>>                     RTRG                    RRT
>>>>>>                    +---+                   +---+
>>>>>>                    |R1 |                   |R2 |
>>>>>>                 B /+---+\ C             E /+---+\ F
>>>>>>             RT   /       \  RT      RT   /       \  RT
>>>>>>            +---+/         \+---+ D +---+/         \+---+
>>>>>>            |P1 |           |P2 +---+P3 |           |P4 |
>>>>>>         A /+---+           +---+   +---+           +---+\ G
>>>>>>          /                                               \
>>>>>>    +---+/                                             
>>     \+---+
>>>>>>    |UAC|                                              
>>      |UAS|
>>>>>>    +---+                                              
>>      +---+
>>>>>> UA-Loose-routing wants the req-uri seen on connection 
>> "C" I think.
>>>>>> To header gives you A.
>>>>>> PCPID gives you E.
>>>>>> Hist-Info gives you A,B,C,D,E,F.
>>>>> [JRE] According to Dean's definition of RT, it does not 
>> change the
>>>>> Request-URI (only the Route header field presumably, or
>>>> maybe not even
>>>>> that).
>>>>> So C, D and E are the same. Also A and B are the same, and
>>>> F and G are
>>>>> the same.
>>>>> So I think:
>>>>> - UA-Loose-routing gives you C/D/E
>>>>> - To gives you A/B
>>>>> - PCPID gives you C/D/E
>>>>> - Target gives you C/D/E
>>>>> - Hist-info gives you A/B, C/D/E and F/G.
>>>> Yes, I agree that is the *theory*.  :)
>>>> I drew it that way though so we could argue about what the
>>>> UAS/UALR-draft _wants_ to happen vs. what _will_ happen if P2
>>>> or P3 are not purely RT's and didn't support a new draft.
>>>> (Since it seemed the conversation was going that way
>>>> previously on this list, for example when Christer pointed
>>>> out the difference between Target and PCPID)
>>>>
>>>> For example, I think there is more than just a syntax
>>>> difference between Christer's sip-target-uri-delivery draft
>>>> (STUD?) and Jonathan's UALR approach.  Though I have no idea
>>>> which one is better.
>>>>
>>>> -hadriel
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the 
>> application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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 18 12:14:34 2008
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From: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>
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Subject: [Sip] Translation address mechanism from megaco to sip and vice
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Hi,

right now, i'm studying about interworking between megaco and sip. I would
like to know about translation address mechanism from megaco to sip and vice
versa. I've been searching on internet, but i haven't found the right
reference. Anyone can help me?

Thx before in advance

Regards,

Jati

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Hi,<br><br>right now, i&#39;m studying about interworking between megaco and sip. I would like to know about translation address mechanism from megaco to sip and vice versa. I&#39;ve been searching on internet, but i haven&#39;t found the right reference. Anyone can help me?
<br><br>Thx before in advance<br><br>Regards,<br><br>Jati<br>

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Subject: Re: [Sip] Re: SIP Location Conveyance revved (-09)
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Karl

Good catch.  I also noticed an incorrect word in what you quoted 
(entry instead of entity).  I added text to the bottom of this 
paragraph indicating the "inserted-by" parameter is necessary for any 
location inserting entity.  Hopefully I will finish -10 soon (there 
were a lot of changes because of what was said in Vancouver) and 
submit it for review...  ;-)

James

At 08:39 AM 1/7/2008, Karl Heinz Wolf wrote:
>by James M. Polk  Nov 21, 2007; 09:56pm
>< - made "inserted-by=" a mandatory header parameter.  This is because
>< there is a comparable header for errors that needs this parameter's
>< value. I'll mention that in a bit.
>
>
>Hi James,
>
>I just had a look at the draft because of the "inserted-by" parameter
>and noticed the following inconsistency:
>
>Section 3 says:
>
>    The Geolocation header can have parameters that are associated
>    with a URI in the header field.  The "inserted-by" parameter has
>    values of "endpoint" or "server", indicating which entry added
>    location to the message. This header parameter MAY be added every
>    time a new location is added into a message.
>
>But below it is a "must" and the parameter contains the host name.
>
>-karl heinz
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP 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] 	Vocabulary and problem statementforRequest-URI, retargeting,
	and SIP routing (long, but read it!)
Date: Mon, 21 Jan 2008 15:03:02 +0100
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Thread-Topic: [Sip] 	Vocabulary and problem statementforRequest-URI,
	retargeting, and SIP routing (long, but read it!)
Thread-Index: AchZPU+/dzx0k6eGS1WWuhtw80L2kAADrUvgABZyvrAADtRW8AABPONgAJQFFoA=
References: <5D1A7985295922448D5550C94DE2918001AC02E9@DEEXC1U01.de.lucent.com><CA9998CD4A020D418654FCDEF4E707DF040266B1@esealmw113.eemea.ericsson.se><47878B1E.3010303@cisco.com><0D5F89FAC29E2C41B98A6A762007F5D0549A47@GBNTHT12009MSX.gb002.siemens.net><1ECE0EB50388174790F9694F77522CCF1428F69B@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF04051C9D@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1428F846@zrc2hxm0.corp.nortel.com><CA9998CD4A020D418654FCDEF4E707DF040960B7@esealmw113.eemea.ericsson.se><1ECE0EB50388174790F9694F77522CCF1434B83B@zrc2hxm0.corp.nortel.com>
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	"Hadriel Kaplan" <HKaplan@acmepacket.com>,
	"Dean Willis" <dean.willis@softarmor.com>, "IETF SIP List" <sip@ietf.org>
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Hi,

>Well, I have been having a side thread with Christer and Hans=20
>Erik, and the only difference other than syntax that they=20
>could convince me of was support for UAs that do not=20
>register, in that with loose-route you would need additional=20
>provisioning in the domain proxy to say that the UA (gateway=20
>or whatever) supports loose-route. Given that you would need=20
>provisioning in the proxy anyway for such UAs, I didn't see=20
>this as a big deal, but it is a difference.

It's strange that we have convinced you about that, since it not the
use-case we have been talking about...  :)

Regards,

Christer


> > -----Original Message-----
> > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > Sent: 18 January 2008 15:15
> > To: Elwell, John; Dean Willis; IETF SIP List
> > Subject: RE: [Sip] Vocabulary and problem statement=20
> > forRequest-URI,retargeting, and SIP routing (long, but read it!)
> >=20
> > Hey John,
> > Inline...
> >=20
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > Sent: Friday, January 18, 2008 2:48 AM
> > > >
> > > > But in the re-targeting scenario such as:
> > > >                     RTRG                    RRT
> > > >                    +---+                   +---+
> > > >                    |R1 |                   |R2 |
> > > >                 B /+---+\ C             E /+---+\ F
> > > >             RT   /       \  RT      RT   /       \  RT
> > > >            +---+/         \+---+ D +---+/         \+---+
> > > >            |P1 |           |P2 +---+P3 |           |P4 |
> > > >         A /+---+           +---+   +---+           +---+\ G
> > > >          /                                               \
> > > >    +---+/                                                 \+---+
> > > >    |UAC|                                                   |UAS|
> > > >    +---+                                                   +---+
> > > >
> > > > UA-Loose-routing wants the req-uri seen on connection=20
> "C" I think.
> > > > To header gives you A.
> > > > PCPID gives you E.
> > > > Hist-Info gives you A,B,C,D,E,F.
> > > [JRE] According to Dean's definition of RT, it does not=20
> change the=20
> > > Request-URI (only the Route header field presumably, or
> > maybe not even
> > > that).
> > > So C, D and E are the same. Also A and B are the same, and
> > F and G are
> > > the same.
> > > So I think:
> > > - UA-Loose-routing gives you C/D/E
> > > - To gives you A/B
> > > - PCPID gives you C/D/E
> > > - Target gives you C/D/E
> > > - Hist-info gives you A/B, C/D/E and F/G.
> >=20
> > Yes, I agree that is the *theory*.  :) I drew it that way=20
> though so we=20
> > could argue about what the UAS/UALR-draft _wants_ to happen=20
> vs. what=20
> > _will_ happen if P2 or P3 are not purely RT's and didn't=20
> support a new=20
> > draft.
> > (Since it seemed the conversation was going that way previously on=20
> > this list, for example when Christer pointed out the difference=20
> > between Target and PCPID)
> >=20
> > For example, I think there is more than just a syntax difference=20
> > between Christer's sip-target-uri-delivery draft
> > (STUD?) and Jonathan's UALR approach.  Though I have no=20
> idea which one=20
> > is better.
> >=20
> > -hadriel
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Mon Jan 21 15:24:38 2008
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Thread-Topic: Question on 3841: extracting implicit preferences
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Hi,
=20
The text in chapter 7.2.2 of 3841 says:
=20
"If, and only if, the proxy did not find any explicit preferences in
the request (because there was no Accept-Contact or Reject-Contact
header field), the proxy extracts implicit preferences."
=20
Is this really true? My understanding is that e.g. the method preference =
is always implcitly set, even if there are A-C/R-C header field(s) in =
the request.
=20
Shouldn't the text say that if a specific preference wasn't explicitly =
set, it will be implicity set?
=20
Regards,
=20
Christer


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Subject: [Sip] updates to media-security-requirements
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At long last, what everyone has been waiting for:  -02 of
draft-ietf-sip-media-security-requirements has been submitted.  Changes from
-01 include:

  * changing requirement numbers to mnemonics (see Appendix E
    for details)
  * added discussion discussing re-use of cryptographic context to aid
    performance; this is now a 'protocol MUST support' and 'SHOULD
    implement' requirement.
  * removed requirement to upgrade from RTP to SRTP, as the
    requirement was never sufficiently clear.
  * added more text around several of the requirements.
  * fixed some nits.

Differences from -01 and -02 are at this link (rfcdiff):
  http://tinyurl.com/26zajs

Please reply if you have any comments on these changes.  I will ask the chairs
to start a WGLC soon.

-d



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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           : Requirements and Analysis of Media Security Management Protocols
	Author(s)       : D. Wing, et al.
	Filename        : draft-ietf-sip-media-security-requirements-02.txt
	Pages           : 46
	Date            : 2008-01-22

This documents describes requirements for a protocol to negotiate a
security context for SIP-signaled SRTP media.  In addition to the
natural security requirements, this negotiation protocol must
interoperate well with SIP in certain ways.  A number of proposals
have been published and a summary of these proposals is in the
appendix of this document.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-media-security-requirements-02.txt

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Below is the data which will enable a MIME compliant mail reader
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Internet-Draft.

--NextPart
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--OtherAccess
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Content-Type: text/plain
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ENCODING mime
FILE /internet-drafts/draft-ietf-sip-media-security-requirements-02.txt

--OtherAccess
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	site="ftp.ietf.org"; access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID: <2008-01-22191222.I-D\@ietf.org>


--OtherAccess--

--NextPart
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MIME-Version: 1.0
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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
--NextPart--





From sip-bounces@ietf.org Wed Jan 23 04:34:16 2008
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Subject: [Sip] End-to-end security for DTLS-SRTP (FW: I-D
	Action:draft-fischer-sip-e2e-sec-media-00.txt)
Date: Wed, 23 Jan 2008 10:33:57 +0100
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Thread-Topic: [Sip] End-to-end security for DTLS-SRTP (FW: I-D
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Thread-Index: AchdoYOBj7k5YqdfRt26nwywkqFi4gAAD1sg
From: "Fischer, Kai" <kai.fischer@siemens.com>
To: <sip@ietf.org>
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Hi,
I have submitted a draft proposing a solution to secure a DTLS-SRTP
handshake and hence SRTP end-to-end (in terms of end-domain to
end-domain). As discussed during the last IETF meetings and analyzed by
Dan's Identity-Media draft, current solutions like SIP Identity do not
protect the authenticity of the fingerprint end-to-end in certain
inter-domain scenarios. For example, a modification of SDP m-/c-lines or
the From header field by intermediaries breaks the SIP-Identity or
Identity-Media signature and causes a re-signing by a domain different
to the originating one. The draft proposes a solution for such scenarios
without the need to re-sign during domain traversal and which preserves
the original identity information.

I appreciate your comments and opinions to the draft and the proposed
solution.

Kai


> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20
> Sent: Mittwoch, 23. Januar 2008 10:20
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-fischer-sip-e2e-sec-media-00.txt=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>=20
> 	Title           : End-to-End Security for DTLS-SRTP
> 	Author(s)       : K. Fischer
> 	Filename        : draft-fischer-sip-e2e-sec-media-00.txt
> 	Pages           : 14
> 	Date            : 2008-01-23
>=20
> The end-to-end security properties of DTLS-SRTP depend on the
> authenticity of the certificate fingerprint exchanged in the
> signalling channel.  In current approaches the authenticity is
> protected by SIP-Identity or SIP-Identity-Media.  These types of
> signatures are broken if intermediaries like Session Border
> Controllers in other domains change specific information of the SIP
> header or the SIP body.  The end-to-end security property between the
> originating and terminating domain is lost if these intermediaries
> re-sign the SIP message and create a new identity signature using
> their own domain credentials.
>=20
> This document defines a new signature type 'Fingerprint-Identity'
> which is exchanged in the signalling channel.  Fingerprint-Identity
> covers only those elements of a SIP message necessary to authenticate
> the certificate fingerprint and to secure media end-to-end.  It is
> independent from SIP-Identity and SIP-Identity-Media and can be
> applied in parallel to them.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-fischer-sip-e2e-sec-
> media-00.txt
>=20
> To remove yourself from the I-D Announcement list, send a message to
> i-d-announce-request@ietf.org with the word unsubscribe in=20
> the body of=20
> the message.
> You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce
> to change your subscription settings.
>=20
> Internet-Drafts are also available by anonymous FTP. Login with the=20
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>=20
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>=20
> NOTE:   The mail server at ietf.org can return the document in
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> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
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> 	exhibit different behavior, especially when dealing with
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> 	how to manipulate these messages.
>=20
> Below is the data which will enable a MIME compliant mail reader
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> Internet-Draft.
>=20

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Content-Type: application/octet-stream;
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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
------_=_NextPart_001_01C85DA3.15D79A1D--





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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           : Message Body Handling in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo
	Filename        : draft-ietf-sip-body-handling-01.txt
	Pages           : 17
	Date            : 2008-01-23

This document specifies how message bodies are handled in SIP.
Additionally, it specifies SIP user agent support for MIME
(Multipurpose Internet Mail Extensions) in message bodies.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-01.txt

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--NextPart
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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           : Message Body Handling in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo
	Filename        : draft-ietf-sip-body-handling-01.txt
	Pages           : 17
	Date            : 2008-01-23

This document specifies how message bodies are handled in SIP.
Additionally, it specifies SIP user agent support for MIME
(Multipurpose Internet Mail Extensions) in message bodies.

A URL for this Internet-Draft is:
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Send a message to:
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NOTE:   The mail server at ietf.org can return the document in
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	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
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--NextPart
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	access-type="anon-ftp"; directory="internet-drafts"

Content-Type: text/plain
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--OtherAccess--

--NextPart
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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
--NextPart--





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

I have put together a new version of the body handling draft:

http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-01.txt

This version addresses all the comments we got in its APPS review, which 
included modifying the structure of the document. With respect to 
references to body parts, per the guidelines we got from the APPS 
people, the document now only allows forward references.

Comments are welcome.

Cheers,

Gonzalo



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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From sip-bounces@ietf.org Wed Jan 23 14:30:18 2008
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Folks,

I have put together a new version of the body handling draft:

http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-01.txt

This version addresses all the comments we got in its APPS review, which 
included modifying the structure of the document. With respect to 
references to body parts, per the guidelines we got from the APPS 
people, the document now only allows forward references.

Comments are welcome.

Cheers,

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



his list is for NEW development of the core SIP 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: question on refresh request to trigger full state notification
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From: "Ramamoorthy Subramanian (ramsubra)" <ramsubra@cisco.com>
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Subject: [Sip] question on refresh request to trigger full state notification
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We're implementing RFC4662 and wondering how to trigger full state
notification. The spec says as follows.
=20
"Immediate notifications triggered as a result of subsequent SUBSCRIBE
messages SHOULD include an RLMI document in which the full state is
indicated.  The RLS SHOULD also include state information for all
resources in the list for which the RLS has state, subject to policy
restrictions.  This allows the subscriber to refresh their state, and
to recover from lost notifications."
=20
This is not clear to me. Does this mean the subscriber needs to send a
refresh request by including RLMI doc?  I guess, not. According to
RFC3265, I think, you cant do this. If this means that the subscribe
should send another subscribe request (resubscribe) to get the
full-state notification? What am I missing here? Is there a simple way
to request a full-state notification by subscriber?
=20
Thanks,
=20
-rama
=20



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

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3243" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D953022400-24012008>We're =
implementing=20
RFC4662 and wondering how to trigger full state notification. The spec =
says as=20
follows.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D953022400-24012008></SPAN></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D953022400-24012008><FONT face=3DArial =
size=3D2>"Immediate=20
notifications triggered as a result of subsequent =
SUBSCRIBE&nbsp;messages SHOULD=20
include an RLMI document in which the full state =
is&nbsp;&nbsp;indicated.&nbsp;=20
The RLS SHOULD also include state information for =
all&nbsp;&nbsp;resources in=20
the list for which the RLS has state, subject to policy&nbsp;=20
restrictions.&nbsp; This allows the subscriber to refresh their state,=20
and&nbsp;&nbsp;&nbsp; to recover from lost =
notifications."</FONT></SPAN></DIV>
<DIV><SPAN class=3D953022400-24012008><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953022400-24012008></SPAN><SPAN=20
class=3D953022400-24012008><FONT><FONT face=3DArial size=3D2>This is not =
clear to me.=20
Does this mean the subscriber needs to send a refresh request by =
including RLMI=20
doc?&nbsp; I guess, not.&nbsp;According to RFC3265, I think, you cant do =
this.=20
If this means that the subscribe should send another subscribe request=20
(resubscribe) to get the full-state notification? What am I missing =
here? Is=20
there a simple way to request a full-state notification by=20
subscriber?</FONT></FONT></SPAN></DIV>
<DIV><SPAN class=3D953022400-24012008><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953022400-24012008><FONT face=3DArial=20
size=3D2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=3D953022400-24012008><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D953022400-24012008><FONT face=3DArial=20
size=3D2>-rama</FONT></SPAN></DIV>
<DIV><SPAN class=3D953022400-24012008><FONT><FONT face=3DArial=20
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><BR><FONT=20
size=3D2></FONT></FONT></DIV></FONT></SPAN></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
--===============0781331383==--





From sip-bounces@ietf.org Wed Jan 23 21:11:53 2008
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   From: "Ramamoorthy Subramanian (ramsubra)" <ramsubra@cisco.com>

   We're implementing RFC4662 and wondering how to trigger full state
   notification. The spec says as follows.

   "Immediate notifications triggered as a result of subsequent SUBSCRIBE
   messages SHOULD include an RLMI document in which the full state is
   indicated.  The RLS SHOULD also include state information for all
   resources in the list for which the RLS has state, subject to policy
   restrictions.  This allows the subscriber to refresh their state, and
   to recover from lost notifications."

   [...] [Does] this means that the subscribe[r] should send another
   subscribe request (resubscribe) to get the full-state notification?

Yes.

Dale


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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
Subject: Re: [Sip] New version of draft-ietf-sip-body-handling-01.txt
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Gonzalo,

This is *much* improved! (Not that it was bad before.) It is very 
straightforward and clear. I like it a lot.

I was only thinking about one use case that may, or may not, warrant 
some special explanation - the use of sipfrag.

A sipfrag is allowed to contain any subset of initial lines, headers, 
and body from a sip message. This could for instance lead to unsatisfied 
references to body parts, or to body parts that were intended to be 
referenced, but whose references have been elided. As a result it may 
not be possible to properly process the body parts of a sipfrag.

I think this this is pretty much intended behavior and so it should be 
ok to send such ill-constructed things. But it might be good, here or 
somewhere else, to clarify the construction and usage of sipfrags - what 
makes sense and what does not.

	Thanks,
	Paul

Gonzalo Camarillo wrote:
> Folks,
> 
> I have put together a new version of the body handling draft:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-01.txt
> 
> This version addresses all the comments we got in its APPS review, which 
> included modifying the structure of the document. With respect to 
> references to body parts, per the guidelines we got from the APPS 
> people, the document now only allows forward references.
> 
> Comments are welcome.
> 
> Cheers,
> 
> 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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Date: Thu, 24 Jan 2008 12:01:46 -0600
From: "David Roan" <j.d.roan@gmail.com>
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Subject: [Sip] Target Refresh Request and Contact Header
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--===============0591239539==
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According to RFC 3261, the following is stated in section 12.2.1.1:
A UAC SHOULD include a Contact header field in any target refresh requests
within a dialog, and unless there is a need to change it, the URI SHOULD be
the same as used in previous requests within the dialog.  If the "secure"
flag is true, that URI MUST be a SIPS URI.As discussed in Section 12.2.2, a
Contact header field in a target refresh request updates the remote target
URI.  This allows a UA to provide a new contact address, should its address
change during the duration of the dialog.

This seems to indicate that including the Contact header in a target refresh
request (ie, Re-INVITE or UPDATE) is only a strong recommendation (ie,
"SHOULD" vs "MUST").

However, table 3 in section 20 of RFC3261 seems to contradict this, as it
lists the Contact header as "m":
Header field              where       proxy ACK BYE CAN INV OPT REG
___________________________________________________________________
Contact                      R                      o       -      -
m     o      o

Also, table 1 in section 8 of RFC3311 lists the Contact header as "m" for
the UPDATE request.

Is there any consensus as to which is considered the correct statement? MUST
the Contact header be included in target refresh requests(as indicated by
the header tables in RFC3261 and RFC3311)? Or, SHOULD the Contact header be
included in the target refresh request(as indicated by section 12.2.1.1 of
RFC3261)?

Thanks in advance for any input, insight, or clarification.
John D. Roan

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<div>According to RFC 3261, the following is stated in section <a href="http://12.2.1.1">12.2.1.1</a>:</div>
<div>A UAC SHOULD include a Contact header field in any target refresh requests within a dialog, and unless there is a need to change it, the URI SHOULD be the same as used in previous requests within the dialog.&nbsp; If the &quot;secure&quot; flag is true, that URI MUST be a SIPS 
URI.As discussed in Section 12.2.2, a Contact header field in a target refresh request updates the remote target URI.&nbsp; This allows a UA to provide a new contact address, should its address change during the duration of the dialog.
</div>
<div>&nbsp;</div>
<div>This seems to indicate that including the Contact header in a target refresh request (ie, Re-INVITE or UPDATE) is only a strong recommendation (ie, &quot;SHOULD&quot; vs &quot;MUST&quot;). </div>
<div>&nbsp;</div>
<div>However, table 3 in section 20 of RFC3261 seems to contradict this, as it lists the Contact header as &quot;m&quot;:</div>
<div>Header field&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; where&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; proxy ACK BYE CAN INV OPT REG<br>___________________________________________________________________</div>
<div>Contact&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; R&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; o&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -&nbsp;&nbsp;&nbsp;&nbsp; m&nbsp;&nbsp;&nbsp;&nbsp; o&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;o</div>
<div>&nbsp;</div>
<div>Also, table 1 in section 8 of RFC3311&nbsp;lists the Contact header&nbsp;as &quot;m&quot; for the UPDATE request.</div>
<div>&nbsp;</div>
<div>Is there any consensus as to which&nbsp;is considered the correct statement? MUST the Contact header be included in target refresh requests(as indicated by the header tables in RFC3261 and RFC3311)? Or, SHOULD the Contact header be included in the target refresh request(as indicated by section 
<a href="http://12.2.1.1">12.2.1.1</a> of RFC3261)?<br><br>Thanks in advance&nbsp;for any input, insight, or clarification.</div>
<div>John D. Roan</div>

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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
--===============0591239539==--






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From: "Dan Wing" <dwing@cisco.com>
To: "'Fischer, Kai'" <kai.fischer@siemens.com>, <sip@ietf.org>
References: <198A10EC585EC74687BCA414E2A5971801FD49B7@MCHP7RDA.ww002.siemens.net>
Subject: RE: [Sip] End-to-end security for DTLS-SRTP (FW:
	I-DAction:draft-fischer-sip-e2e-sec-media-00.txt)
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I like this draft, and it solves a very real problem:  providing identity for
SIP requests that traverse B2BUAs and traverse SBCs.

-d

> -----Original Message-----
> From: Fischer, Kai [mailto:kai.fischer@siemens.com] 
> Sent: Wednesday, January 23, 2008 1:34 AM
> To: sip@ietf.org
> Subject: [Sip] End-to-end security for DTLS-SRTP (FW: 
> I-DAction:draft-fischer-sip-e2e-sec-media-00.txt)
> 
> Hi,
> I have submitted a draft proposing a solution to secure a DTLS-SRTP
> handshake and hence SRTP end-to-end (in terms of end-domain to
> end-domain). As discussed during the last IETF meetings and 
> analyzed by
> Dan's Identity-Media draft, current solutions like SIP Identity do not
> protect the authenticity of the fingerprint end-to-end in certain
> inter-domain scenarios. For example, a modification of SDP 
> m-/c-lines or
> the From header field by intermediaries breaks the SIP-Identity or
> Identity-Media signature and causes a re-signing by a domain different
> to the originating one. The draft proposes a solution for 
> such scenarios
> without the need to re-sign during domain traversal and which 
> preserves
> the original identity information.
> 
> I appreciate your comments and opinions to the draft and the proposed
> solution.
> 
> Kai
> 
> 
> > -----Original Message-----
> > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org] 
> > Sent: Mittwoch, 23. Januar 2008 10:20
> > To: i-d-announce@ietf.org
> > Subject: I-D Action:draft-fischer-sip-e2e-sec-media-00.txt 
> > 
> > A New Internet-Draft is available from the on-line 
> > Internet-Drafts directories.
> > 
> > 	Title           : End-to-End Security for DTLS-SRTP
> > 	Author(s)       : K. Fischer
> > 	Filename        : draft-fischer-sip-e2e-sec-media-00.txt
> > 	Pages           : 14
> > 	Date            : 2008-01-23
> > 
> > The end-to-end security properties of DTLS-SRTP depend on the
> > authenticity of the certificate fingerprint exchanged in the
> > signalling channel.  In current approaches the authenticity is
> > protected by SIP-Identity or SIP-Identity-Media.  These types of
> > signatures are broken if intermediaries like Session Border
> > Controllers in other domains change specific information of the SIP
> > header or the SIP body.  The end-to-end security property 
> between the
> > originating and terminating domain is lost if these intermediaries
> > re-sign the SIP message and create a new identity signature using
> > their own domain credentials.
> > 
> > This document defines a new signature type 'Fingerprint-Identity'
> > which is exchanged in the signalling channel.  Fingerprint-Identity
> > covers only those elements of a SIP message necessary to 
> authenticate
> > the certificate fingerprint and to secure media end-to-end.  It is
> > independent from SIP-Identity and SIP-Identity-Media and can be
> > applied in parallel to them.
> > 
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-fischer-sip-e2e-sec-
> > media-00.txt
> > 
> > To remove yourself from the I-D Announcement list, send a message to
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> > Internet-Drafts can also be obtained by e-mail.
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> 



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Subject: Re: [Sip] Target Refresh Request and Contact Header
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IMO it should be MUST. I suspect is is weaker for 2543 compatibility.

	Paul

David Roan wrote:
> According to RFC 3261, the following is stated in section 12.2.1.1 
> <http://12.2.1.1>:
> A UAC SHOULD include a Contact header field in any target refresh 
> requests within a dialog, and unless there is a need to change it, the 
> URI SHOULD be the same as used in previous requests within the dialog.  
> If the "secure" flag is true, that URI MUST be a SIPS URI.As discussed 
> in Section 12.2.2, a Contact header field in a target refresh request 
> updates the remote target URI.  This allows a UA to provide a new 
> contact address, should its address change during the duration of the 
> dialog.
>  
> This seems to indicate that including the Contact header in a target 
> refresh request (ie, Re-INVITE or UPDATE) is only a strong 
> recommendation (ie, "SHOULD" vs "MUST").
>  
> However, table 3 in section 20 of RFC3261 seems to contradict this, as 
> it lists the Contact header as "m":
> Header field              where       proxy ACK BYE CAN INV OPT REG
> ___________________________________________________________________
> Contact                      R                      o       -      -     
> m     o      o
>  
> Also, table 1 in section 8 of RFC3311 lists the Contact header as "m" 
> for the UPDATE request.
>  
> Is there any consensus as to which is considered the correct statement? 
> MUST the Contact header be included in target refresh requests(as 
> indicated by the header tables in RFC3261 and RFC3311)? Or, SHOULD the 
> Contact header be included in the target refresh request(as indicated by 
> section 12.2.1.1 <http://12.2.1.1> of RFC3261)?
> 
> Thanks in advance for any input, insight, or clarification.
> John D. Roan
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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] Target Refresh Request and Contact Header
Date: Thu, 24 Jan 2008 21:15:24 +0100
Message-ID: <CA9998CD4A020D418654FCDEF4E707DF043E1888@esealmw113.eemea.ericsson.se>
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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	"David Roan" <j.d.roan@gmail.com>
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Hi,

Is there something that could go wrong if it is not included?

Regards,

Christer=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: 24. tammikuuta 2008 20:41
> To: David Roan
> Cc: sip@ietf.org
> Subject: Re: [Sip] Target Refresh Request and Contact Header
>=20
> IMO it should be MUST. I suspect is is weaker for 2543 compatibility.
>=20
> 	Paul
>=20
> David Roan wrote:
> > According to RFC 3261, the following is stated in section 12.2.1.1
> > <http://12.2.1.1>:
> > A UAC SHOULD include a Contact header field in any target refresh=20
> > requests within a dialog, and unless there is a need to=20
> change it, the=20
> > URI SHOULD be the same as used in previous requests within=20
> the dialog.
> > If the "secure" flag is true, that URI MUST be a SIPS=20
> URI.As discussed=20
> > in Section 12.2.2, a Contact header field in a target=20
> refresh request=20
> > updates the remote target URI.  This allows a UA to provide a new=20
> > contact address, should its address change during the=20
> duration of the=20
> > dialog.
> > =20
> > This seems to indicate that including the Contact header in=20
> a target=20
> > refresh request (ie, Re-INVITE or UPDATE) is only a strong=20
> > recommendation (ie, "SHOULD" vs "MUST").
> > =20
> > However, table 3 in section 20 of RFC3261 seems to=20
> contradict this, as=20
> > it lists the Contact header as "m":
> > Header field              where       proxy ACK BYE CAN INV OPT REG
> > ___________________________________________________________________
> > Contact                      R                      o      =20
> -      -    =20
> > m     o      o
> > =20
> > Also, table 1 in section 8 of RFC3311 lists the Contact=20
> header as "m"=20
> > for the UPDATE request.
> > =20
> > Is there any consensus as to which is considered the=20
> correct statement?=20
> > MUST the Contact header be included in target refresh requests(as=20
> > indicated by the header tables in RFC3261 and RFC3311)? Or,=20
> SHOULD the=20
> > Contact header be included in the target refresh request(as=20
> indicated=20
> > by section 12.2.1.1 <http://12.2.1.1> of RFC3261)?
> >=20
> > Thanks in advance for any input, insight, or clarification.
> > John D. Roan
> >=20
> >=20
> >=20
> ----------------------------------------------------------------------
> > --
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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] Target Refresh Request and Contact Header
Date: Thu, 24 Jan 2008 15:22:20 -0500
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Thread-Topic: [Sip] Target Refresh Request and Contact Header
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>,
	"Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>,
	"David Roan" <j.d.roan@gmail.com>
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I guess most 3261 compliant implementations will send a 400 error
response. 2543 compliant spec, most likely, will accept it.

Sanjay=20

>-----Original Message-----
>From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
>Sent: Thursday, January 24, 2008 3:15 PM
>To: Paul Kyzivat (pkyzivat); David Roan
>Cc: sip@ietf.org
>Subject: RE: [Sip] Target Refresh Request and Contact Header
>
>
>Hi,
>
>Is there something that could go wrong if it is not included?
>
>Regards,
>
>Christer=20
>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: 24. tammikuuta 2008 20:41
>> To: David Roan
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Target Refresh Request and Contact Header
>>=20
>> IMO it should be MUST. I suspect is is weaker for 2543 compatibility.
>>=20
>> 	Paul
>>=20
>> David Roan wrote:
>> > According to RFC 3261, the following is stated in section 12.2.1.1
>> > <http://12.2.1.1>:
>> > A UAC SHOULD include a Contact header field in any target refresh=20
>> > requests within a dialog, and unless there is a need to
>> change it, the
>> > URI SHOULD be the same as used in previous requests within
>> the dialog.
>> > If the "secure" flag is true, that URI MUST be a SIPS
>> URI.As discussed
>> > in Section 12.2.2, a Contact header field in a target
>> refresh request
>> > updates the remote target URI.  This allows a UA to provide a new=20
>> > contact address, should its address change during the
>> duration of the
>> > dialog.
>> > =20
>> > This seems to indicate that including the Contact header in
>> a target
>> > refresh request (ie, Re-INVITE or UPDATE) is only a strong=20
>> > recommendation (ie, "SHOULD" vs "MUST").
>> > =20
>> > However, table 3 in section 20 of RFC3261 seems to
>> contradict this, as
>> > it lists the Contact header as "m":
>> > Header field              where       proxy ACK BYE CAN INV OPT REG
>> > ___________________________________________________________________
>> > Contact                      R                      o      =20
>> -      -    =20
>> > m     o      o
>> > =20
>> > Also, table 1 in section 8 of RFC3311 lists the Contact
>> header as "m"=20
>> > for the UPDATE request.
>> > =20
>> > Is there any consensus as to which is considered the
>> correct statement?=20
>> > MUST the Contact header be included in target refresh requests(as=20
>> > indicated by the header tables in RFC3261 and RFC3311)? Or,
>> SHOULD the
>> > Contact header be included in the target refresh request(as
>> indicated
>> > by section 12.2.1.1 <http://12.2.1.1> of RFC3261)?
>> >=20
>> > Thanks in advance for any input, insight, or clarification.
>> > John D. Roan
>> >=20
>> >=20
>> >=20
>>=20
>----------------------------------------------------------------------
>> > --
>> >=20
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol Use=20
>> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
>> > sipping@ietf.org for new developments on the application of sip
>>=20
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use=20
>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>> sipping@ietf.org for new developments on the application of sip
>>=20
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use=20
>sip-implementors@cs.columbia.edu for questions on current sip=20
>Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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	-new version of the draft-holmberg-sip-target-uri-delivery draft
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Sorry for jumping in late here. Trying to clarify some points here and 
there.

The UA-loose mechanism needs to know that the entity which would have 
previously been reached by the DNS/IP forwarding based on the R-URI, is 
now prepared for a request which reaches it with that target in the 
Route header field.

In the case of a UA registering through a chain of Path-inserting 
proxies, it means that only the *UA* has to support UA-loose route. 
Thats because of the assumption that the intermediate path-inserting 
proxies weren't being targeted based on the R-URI previously; they were 
(and still will) forward the request based on Route.

The 'next hop' use case that Christer is referring to are cases where we 
would have done a R-URI translation without a registration. For example, 
lets say a proxy receives an INVITE with sip:12345@proxy.com, and its 
using some number-prefix based routing rules, which have been CONFIGURED 
into the proxy, to determine a next-hop gateway. The UA loose route 
rules would say that, since this is a routing operation and not a 
retargeting, the request looks like:

INVITE sip:12345@proxy.com
Route: sip:ip-of-gateway

whereas current rules would require:

INVITE sip:12345@ip-of-gateway

in this use case, the entity which was previously being reached by the 
R-URI *is* the next hop.

And so, the main limitation that I see of the UA loose route mechanism 
is, in cases where the next-hop is being reached via a configured 
routing rule, or by some other non-registration means, and said 
operation is a re-route and not retarget, the UA_loose-route requires 
that this configured routing rule include information about whether the 
next hop uses UA loose routing, and if so, what URI to use in the Route 
header.

This is a limitation, I agree. Whether its significant or not is debatable.

I will note that, it is possible to separate these concerns. We could 
define UA loose route to strictly be used for the registration cases, 
since that is the specific problem at hand.

I agree CHrister's mechanism does not have this configuration 
limitation, and that this aspect of it is better. My main beef, as 
others have commented, is that it leaves us with nearly half-dozen 
header fields which are all awfully similar (To, P-C-ID, Route, 
History-Info, and now Target). I was hoping to reduce the set and 
include only the minimum. In that way, I think UA loose route is 
architecturally better since it gives some clarity to this.

-Jonathan R.


Elwell, John wrote:
> Christer,
> 
> I don't think the entity using the mechanism needs to know about next
> hop support for the loose route mechanism - only whether the UAS
> supports it. If the UA registers, then the loose route draft provides an
> automatic mechanism. If the UA does not register, support would need to
> be indicated by provisioning. I am not sure this ranks as a significant
> limitation, since a certain amount of provisioning needs to be done
> anyway for UAs such as gateways that do not register.
> 
> John 
> 
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] 
>> Sent: 16 January 2008 11:51
>> To: Elwell, John; sip@ietf.org
>> Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
>> Subject: RE: [Sip] RE: Delivering request-URI and parameters 
>> to UAS via proxy -new version of the 
>> draft-holmberg-sip-target-uri-delivery draft
>>
>>
>> Hi, 
>>
>>> I hear what you are saying, but I don't really see that the 
>> points you
>> raised in section 4 are real limitations of the loose route mechanism.
>>
>> I think it is a limitation that an entity using the mechanism has to
>> know whether the the next hop supports it or not, and that 
>> the mechanism
>> can only be used if the subsequent hops support it.
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>>>> -----Original Message-----
>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>> Sent: 16 January 2008 11:06
>>>> To: Elwell, John; sip@ietf.org
>>>> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
>>>> Subject: [Sip] RE: Delivering request-URI and parameters 
>> to UAS via 
>>>> proxy - new version of the draft-holmberg-sip-target-uri-delivery 
>>>> draft
>>>>
>>>>
>>>> Hi John,
>>>>
>>>>> Thanks for this revision, which makes things somewhat 
>>> clearer. I do 
>>>>> have a couple of comments:
>>>>>
>>>>> 1. I am not sure I agree with the assertions in section 4 
>>> concerning 
>>>>> issues with the mechanisms in loose-route. Taking example 1, the 
>>>>> Route header field should contain enough entries to get 
>> you to the 
>>>>> registered contact, not just to an intermediate proxy. 
>>> Therefore this 
>>>>> situation should not arise with a correctly implemented 
>>> home proxy. 
>>>>> It is not clear to me how example 2 could arise either, 
>>> for similar 
>>>>> reasons. The MGC case can be resolved by taking into account the 
>>>>> option tag in the REGISTER request, or if it is permanently 
>>>>> registered, through provisioning.
>>>> The examples are not meant to show bugs in the loose-route 
>>> mechanism.
>>>> They are meant to help people understand the limitations with the 
>>>> loose-route mechanism.
>>>>  
>>>>> 2. Comparing the mechanism proposed with the loose-route 
>>> mechanism, 
>>>>> my understanding is:
>>>>> a)When retargeting occurs, the loose-route mechanism 
>>> places the new 
>>>>> target in the Request URI. Your proposal places the new 
>> target in 
>>>>> both the Request-URI and the Target header field.
>>>>> b) When rerouting, the loose-route mechanism places the 
>> new route 
>>>>> (i.e., the registered contact) in the Route header field. Your 
>>>>> proposal places the new route in the Request-URI (the 
>>> latter as per 
>>>>> RFC 3261).
>>>>> So the two mechanisms solve exactly the same problems using a 
>>>>> slightly different mechanism. Correct?
>>>> Yes. The two solutions intend to solve the same problem.
>>>>  
>>>>> 3. How P-Called-Party-ID fits into this is not really 
>>> relevant from 
>>>>> an IETF perspective - it seems there are some 3GPP-specific 
>>>>> situations where the contents of P-Called-Party-ID will 
>>> not equal the 
>>>>> contents of Target. Correct?
>>>> Correct.
>>>>
>>>>> 4. If my suggestion in point 1 above that the loose-route 
>>> mechanism 
>>>>> does not suffer from the problems suggested, then each 
>>> mechanism will 
>>>>> work and each addresses the same problem.
>>>>> So it is just down to a beauty contest between the two. Correct?
>>>> See question 1.
>>>>
>>>> We believe that our solution does not have the same 
>>> limitations as the 
>>>> loose-route solution. But, again, both solutions intend to 
>>> solve the 
>>>> same problem.
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>> -----Original Message-----
>>>>>> From: Christer Holmberg 
>> [mailto:christer.holmberg@ericsson.com]
>>>>>> Sent: 16 January 2008 08:41
>>>>>> To: sip@ietf.org
>>>>>> Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John; 
>>> Jeroen van 
>>>>>> Bemmel; Francois Audet
>>>>>> Subject: Delivering request-URI and parameters to UAS via
>>>>> proxy - new
>>>>>> version of the draft-holmberg-sip-target-uri-delivery draft
>>>>>>
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> We've uploaded a new version (-01) of the Target draft.
>>>>>>
>>>>>> We've tried to make things more clear. I've also 
>>> removed all text 
>>>>>> about P-Called-Party-ID, except from one chapter where 
>>> we try to 
>>>>>> explain the semantical difference between Target and P-CPI.
>>>>>>
>>>>>> You can also find the draft from:
>>>>>> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
>>>>>> et-uri-del
>>>>>> ivery-01.txt
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> Christer
>>>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use 
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>> sipping@ietf.org for new developments on the application of sip
>>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip 
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
Cisco Fellow                                   Edison, NJ 08837
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com


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http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-route-02.txt

I didn't change anything since -01; but -01 had expired. Since we are 
discussing it it made sense to have a version in the archives.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
Cisco Fellow                                   Edison, NJ 08837
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com


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=20
There are other cases similar to what is described in section  2.1 =
Unknown Aliases of J. Rosenberg draft that should be taken into account. =


Those cases are where the customer side is composed of several SIP =
terminals that access the network through an intermediate entity which =
registers their associated aliases with its contact address. Thus, such =
client side is seen from the network as a single client having several =
aliases (aggregated endpoints). Moreover, such client side may be a mini =
private network composed of several entities. In these configurations, =
the intermediate entity is in charge of routeing incoming calls inside =
the client domain and may need to behave as a SIP proxy for incoming SIP =
messages. =20

As examples of such configuration:
- Corporate networks: in that case the PBX register all the served =
individual user identities with its contact address and routes incoming =
calls toward the individual called users.
- Home networks: the Home Gateway may need to register on behalf of all =
the served terminals.  The Home Gateway is in charge of routeing =
incoming calls toward the individual called users.=20

J. Rosenberg draft seems give a good solution to take into account these =
configurations.
  =20
Best regards,

Youssef


> -----Message d'origine-----
> De : DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Envoy=E9 : lundi 14 janvier 2008 17:58
> =C0 : sip@ietf.org
> Objet : [Sip] Delivering request-URI and parameters to UAS via proxy
>=20
> (As WG chair)
>=20
> In fulfilment of our charter items of
>=20
> Dec 2007    Delivering request-URI and parameters to UAS via proxy to
> WGLC =20
> Feb 2008    Delivering request-URI and parameters to UAS via proxy to
> IESG (PS)
>=20
> We now have a couple of proposals on the table for solving=20
> the problem.
>=20
> The original draft from Jonathan and which led to the=20
> creation of the charter items by the WG is unfortunately=20
> expired, but is at:
>=20
> http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
>=20
> The alternative document from Christer, etc is at:
>=20
> http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
> uri-delive
> ry-00.txt
>=20
> We obviously need to make a decision between the two=20
> approaches so please attempt to address the following=20
> specific points via the mailing
> list:
>=20
> 1)	Problem cases: These are summarised in section 4.4 of
> draft-rosenberg-sip-ua-loose-route-01 and from my read of the=20
> other draft, I don't believe that this draft adds any others.=20
> If you believe there are other cases that should be covered=20
> by the solution, then please identify them. If there is=20
> support on any new problem cases, I would encourage the=20
> authors of both drafts to add text concerning these problem cases.
>=20
> 2)	Clarifications: If for any reason you don't understand either
> draft, or believe that there are technical issues that are=20
> not represented in the current draft, please post your=20
> questions / comments to the list. I would encourage authors=20
> of both drafts to revise as frequently as appropriate to=20
> reflect the current state of discussion.
>=20
> 3)	Support for either position. If you wish to indicate support for
> either position please do so, but please accompany this is=20
> technical reasoning as to why you have this position, as that=20
> will help other members of the WG form a position.
>=20
> I would encourage as much list discussion as possible before=20
> Philadelphia. I suspect we will need to have a discussion at=20
> the face-to-face meeting in Philadephia, but list discussion=20
> is essential prior to that. If we can solve this positions on=20
> list, then well and good, and even better (that is how we are=20
> meant to make decisions).
>=20
> Regards
>=20
>=20
> Keith
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: Re: [Sip] Delivering request-URI and parameters to UAS via proxy
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These cases can all be handled by having intermediate entity use a 
unique user part, together with its own domain name, to construct a 
contact address for each terminal UA that it is registering. Then when 
it receives an incoming request to one of these uris it can simply 
translate it, using the user part as a key to its own local mappings.

We don't need anything new to solve these cases.

	Paul

youssef.chadli@orange-ftgroup.com wrote:
>  
> There are other cases similar to what is described in section  2.1 Unknown Aliases of J. Rosenberg draft that should be taken into account. 
> 
> Those cases are where the customer side is composed of several SIP terminals that access the network through an intermediate entity which registers their associated aliases with its contact address. Thus, such client side is seen from the network as a single client having several aliases (aggregated endpoints). Moreover, such client side may be a mini private network composed of several entities. In these configurations, the intermediate entity is in charge of routeing incoming calls inside the client domain and may need to behave as a SIP proxy for incoming SIP messages.  
> 
> As examples of such configuration:
> - Corporate networks: in that case the PBX register all the served individual user identities with its contact address and routes incoming calls toward the individual called users.
> - Home networks: the Home Gateway may need to register on behalf of all the served terminals.  The Home Gateway is in charge of routeing incoming calls toward the individual called users. 
> 
> J. Rosenberg draft seems give a good solution to take into account these configurations.
>    
> Best regards,
> 
> Youssef
> 
> 
>> -----Message d'origine-----
>> De : DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com] 
>> Envoyé : lundi 14 janvier 2008 17:58
>> À : sip@ietf.org
>> Objet : [Sip] Delivering request-URI and parameters to UAS via proxy
>>
>> (As WG chair)
>>
>> In fulfilment of our charter items of
>>
>> Dec 2007    Delivering request-URI and parameters to UAS via proxy to
>> WGLC  
>> Feb 2008    Delivering request-URI and parameters to UAS via proxy to
>> IESG (PS)
>>
>> We now have a couple of proposals on the table for solving 
>> the problem.
>>
>> The original draft from Jonathan and which led to the 
>> creation of the charter items by the WG is unfortunately 
>> expired, but is at:
>>
>> http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
>>
>> The alternative document from Christer, etc is at:
>>
>> http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
>> uri-delive
>> ry-00.txt
>>
>> We obviously need to make a decision between the two 
>> approaches so please attempt to address the following 
>> specific points via the mailing
>> list:
>>
>> 1)	Problem cases: These are summarised in section 4.4 of
>> draft-rosenberg-sip-ua-loose-route-01 and from my read of the 
>> other draft, I don't believe that this draft adds any others. 
>> If you believe there are other cases that should be covered 
>> by the solution, then please identify them. If there is 
>> support on any new problem cases, I would encourage the 
>> authors of both drafts to add text concerning these problem cases.
>>
>> 2)	Clarifications: If for any reason you don't understand either
>> draft, or believe that there are technical issues that are 
>> not represented in the current draft, please post your 
>> questions / comments to the list. I would encourage authors 
>> of both drafts to revise as frequently as appropriate to 
>> reflect the current state of discussion.
>>
>> 3)	Support for either position. If you wish to indicate support for
>> either position please do so, but please accompany this is 
>> technical reasoning as to why you have this position, as that 
>> will help other members of the WG form a position.
>>
>> I would encourage as much list discussion as possible before 
>> Philadelphia. I suspect we will need to have a discussion at 
>> the face-to-face meeting in Philadephia, but list discussion 
>> is essential prior to that. If we can solve this positions on 
>> list, then well and good, and even better (that is how we are 
>> meant to make decisions).
>>
>> Regards
>>
>>
>> Keith
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP 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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Dears
   
  this is hossam from jordan
   
  when the SAU server (A) send an invite message to CommuniGate PBX (B) , the call occured with nothing heared from the PBX IVR system, after checking the Sip call flow on the trace we found that the last Ack after the 200 ok message sent in the "TO" field instead of "CONTACT" field as appeared which is not the same as that in RFC 3261, how can we solve this issue and force the CommuniGate to accept the ACK message . 

hope to hear from you
   
  many thanx

       
---------------------------------
Never miss a thing.   Make Yahoo your homepage.
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<div>Dears</div>  <div>&nbsp;</div>  <div>this is hossam from jordan</div>  <div>&nbsp;</div>  <div>when the SAU server (A) send an invite message to CommuniGate PBX (B) , the call occured with nothing heared from the PBX IVR system, after checking the Sip call flow on the trace we found that the last Ack after the 200 ok message sent&nbsp;in the "TO" field instead of "CONTACT" field as appeared which is not the same as that in RFC 3261, how can we solve this issue and force the CommuniGate to accept the ACK message . <BR><BR>hope to hear from you</div>  <div>&nbsp;</div>  <div>many thanx</div><p>&#32;



      <hr size=1>Never miss a thing.  <a href="http://us.rd.yahoo.com/evt=51438/*http://www.yahoo.com/r/hs"> Make Yahoo your homepage.</a>


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Subject: [Sip] Contradictory Semantics header againest Join header in SIP
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Hi,
I have a query about the contradictory semantics header againest Join
header. In RFC 3911 it's saying the following

"The Join header has specific call control semantics.  If both a Join
  header field and another header field with contradictory semantics
  (for example a Replaces [8] header field) are present in a request,
  the request MUST be rejected with a 400 "Bad Request" response.
"
here it specifies that Replace is the one Contradictory header for
join but I seek to know that who else header has the contradictory
semantics againest Join header for an INVITE



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From: "suryakanta mandal" <suryakanta.mandal@gmail.com>
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Subject: [Sip] 180 for PRACK
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Hi ,

If you look at the RFC 3262, it says in Section 4: UAC behaviour

Header field          where   PRACK
___________________________________
 Alert-Info             180            -
Contact                1xx            -
 Contact                3xx           o
Error-Info           300-699         o
Record-Route      2xx,18x       o

 and If you look at the functionality of these column from RFC 3261
"The "where" column describes the request and response types in which
   the header field can be used.  Values in this column are:

      R: header field may only appear in requests;

      r: header field may only appear in responses;

      2xx, 4xx, etc.: A numerical value or range indicates response
           codes with which the header field can be used;

      c: header field is copied from the request to the response.

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

   The "proxy" column describes the operations a proxy may perform on a
   header field:

      a: A proxy can add or concatenate the header field if not present.

      m: A proxy can modify an existing header field value.

      d: A proxy can delete a header field value.

      r: A proxy must be able to read the header field, and thus this
           header field cannot be encrypted.

   The next six columns relate to the presence of a header field in a
   method:

      c: Conditional; requirements on the header field depend on the
           context of the message.

      m: The header field is mandatory.

      m*: The header field SHOULD be sent, but clients/servers need to
           be prepared to receive messages without that header field.

      o: The header field is optional.

      t: The header field SHOULD be sent, but clients/servers need to be
           prepared to receive messages without that header field.

           If a stream-based protocol (such as TCP) is used as a
           transport, then the header field MUST be sent.

      *: The header field is required if the message body is not empty.
           See Sections 20.14, 20.15 and 7.4 for details.

      -: The header field is not applicable.
"


I got struck on these above points.
Can you let me know the scenario when the UAS will generate 180 for PRACK ?
 and when UAS will generate 3xx response for PRACK because their it
mentioned optional means UAS may or may not generate 3xx response.

Surya
Aricent


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   From: HOSSAM ALDEBEI <haldebei@yahoo.com>

   when the SAU server (A) send an invite message to CommuniGate PBX
   (B) , the call occured with nothing heared from the PBX IVR system,
   after checking the Sip call flow on the trace we found that the
   last Ack after the 200 ok message sent in the "TO" field instead of
   "CONTACT" field as appeared which is not the same as that in RFC
   3261, how can we solve this issue and force the CommuniGate to
   accept the ACK message . 

>From your description, it appears that the ACK (which is being
generated by the SAU server) has an incorrect request-URI.  So the SAU
server needs to be corrected.

Dale


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Subject: RE: [Sip] Delivering request-URI and parameters to UAS via proxy
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I think there is a misunderstanding on these use cases:
 - The intermediate entity in the customer side does perform only one =
registration to the network. The AoRs associated to the served terminals =
inside the client domain are registred automatically by the home =
network, using implicit registration for example. In fact, it's not =
conceivable to register individually all the users of a corporate =
network for example.

- The "intermediate entity" should be able to route incoming requests =
inside its domain based on standard RFC3261 behaviour, that is, based on =
the Request-URI.

Best regards

Youssef


 =20
=20

> -----Message d'origine-----
> De : Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Envoy=E9 : lundi 28 janvier 2008 21:46
> =C0 : CHADLI Youssef RD-CORE-ISS
> Cc : drage@alcatel-lucent.com; sip@ietf.org
> Objet : Re: [Sip] Delivering request-URI and parameters to=20
> UAS via proxy
>=20
> These cases can all be handled by having intermediate entity=20
> use a unique user part, together with its own domain name, to=20
> construct a contact address for each terminal UA that it is=20
> registering. Then when it receives an incoming request to one=20
> of these uris it can simply translate it, using the user part=20
> as a key to its own local mappings.
>=20
> We don't need anything new to solve these cases.
>=20
> 	Paul
>=20
> youssef.chadli@orange-ftgroup.com wrote:
> > =20
> > There are other cases similar to what is described in=20
> section  2.1 Unknown Aliases of J. Rosenberg draft that=20
> should be taken into account.=20
> >=20
> > Those cases are where the customer side is composed of=20
> several SIP terminals that access the network through an=20
> intermediate entity which registers their associated aliases=20
> with its contact address. Thus, such client side is seen from=20
> the network as a single client having several aliases=20
> (aggregated endpoints). Moreover, such client side may be a=20
> mini private network composed of several entities. In these=20
> configurations, the intermediate entity is in charge of=20
> routeing incoming calls inside the client domain and may need=20
> to behave as a SIP proxy for incoming SIP messages. =20
> >=20
> > As examples of such configuration:
> > - Corporate networks: in that case the PBX register all the=20
> served individual user identities with its contact address=20
> and routes incoming calls toward the individual called users.
> > - Home networks: the Home Gateway may need to register on=20
> behalf of all the served terminals.  The Home Gateway is in=20
> charge of routeing incoming calls toward the individual called users.=20
> >=20
> > J. Rosenberg draft seems give a good solution to take into=20
> account these configurations.
> >   =20
> > Best regards,
> >=20
> > Youssef
> >=20
> >=20
> >> -----Message d'origine-----
> >> De : DRAGE, Keith (Keith)=20
> [mailto:drage@alcatel-lucent.com] Envoy=E9 :=20
> >> lundi 14 janvier 2008 17:58 =C0 : sip@ietf.org Objet : [Sip]=20
> Delivering=20
> >> request-URI and parameters to UAS via proxy
> >>
> >> (As WG chair)
> >>
> >> In fulfilment of our charter items of
> >>
> >> Dec 2007    Delivering request-URI and parameters to UAS=20
> via proxy to
> >> WGLC =20
> >> Feb 2008    Delivering request-URI and parameters to UAS=20
> via proxy to
> >> IESG (PS)
> >>
> >> We now have a couple of proposals on the table for solving the=20
> >> problem.
> >>
> >> The original draft from Jonathan and which led to the=20
> creation of the=20
> >> charter items by the WG is unfortunately expired, but is at:
> >>
> >> http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
> >>
> >> The alternative document from Christer, etc is at:
> >>
> >> http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
> >> uri-delive
> >> ry-00.txt
> >>
> >> We obviously need to make a decision between the two approaches so=20
> >> please attempt to address the following specific points via the=20
> >> mailing
> >> list:
> >>
> >> 1)	Problem cases: These are summarised in section 4.4 of
> >> draft-rosenberg-sip-ua-loose-route-01 and from my read of=20
> the other=20
> >> draft, I don't believe that this draft adds any others.
> >> If you believe there are other cases that should be covered by the=20
> >> solution, then please identify them. If there is support=20
> on any new=20
> >> problem cases, I would encourage the authors of both drafts to add=20
> >> text concerning these problem cases.
> >>
> >> 2)	Clarifications: If for any reason you don't understand either
> >> draft, or believe that there are technical issues that are not=20
> >> represented in the current draft, please post your questions /=20
> >> comments to the list. I would encourage authors of both drafts to=20
> >> revise as frequently as appropriate to reflect the current=20
> state of=20
> >> discussion.
> >>
> >> 3)	Support for either position. If you wish to indicate support for
> >> either position please do so, but please accompany this is=20
> technical=20
> >> reasoning as to why you have this position, as that will=20
> help other=20
> >> members of the WG form a position.
> >>
> >> I would encourage as much list discussion as possible before=20
> >> Philadelphia. I suspect we will need to have a discussion at the=20
> >> face-to-face meeting in Philadephia, but list discussion=20
> is essential=20
> >> prior to that. If we can solve this positions on list,=20
> then well and=20
> >> good, and even better (that is how we are meant to make decisions).
> >>
> >> Regards
> >>
> >>
> >> Keith
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >>
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20


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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via proxy
	-new version of the draft-holmberg-sip-target-uri-delivery draft
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>,
	"Elwell, John" <john.elwell@siemens.com>
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Hi Jonathan,

>Sorry for jumping in late here. Trying to clarify some points=20
>here and there.
>=20
>The UA-loose mechanism needs to know that the entity which=20
>would have previously been reached by the DNS/IP forwarding=20
>based on the R-URI, is now prepared for a request which=20
>reaches it with that target in the Route header field.
>=20
>In the case of a UA registering through a chain of=20
>Path-inserting proxies, it means that only the *UA* has to=20
>support UA-loose route.=20
>Thats because of the assumption that the intermediate=20
>path-inserting proxies weren't being targeted based on the=20
>R-URI previously; they were (and still will) forward the=20
>request based on Route.
>
>The 'next hop' use case that Christer is referring to are=20
>cases where we would have done a R-URI translation without a=20
>registration. For example, lets say a proxy receives an=20
>INVITE with sip:12345@proxy.com, and its using some=20
>number-prefix based routing rules, which have been CONFIGURED=20
>into the proxy, to determine a next-hop gateway. The UA loose=20
>route rules would say that, since this is a routing operation=20
>and not a retargeting, the request looks like:
>=20
>INVITE sip:12345@proxy.com
>Route: sip:ip-of-gateway
>=20
>whereas current rules would require:
>=20
>INVITE sip:12345@ip-of-gateway
>=20
>in this use case, the entity which was previously being=20
>reached by the R-URI *is* the next hop.
>=20
>And so, the main limitation that I see of the UA loose route=20
>mechanism is, in cases where the next-hop is being reached=20
>via a configured routing rule, or by some other=20
>non-registration means, and said operation is a re-route and=20
>not retarget, the UA_loose-route requires that this=20
>configured routing rule include information about whether the=20
>next hop uses UA loose routing, and if so, what URI to use in=20
>the Route header.
>=20
>This is a limitation, I agree. Whether its significant or not=20
>is debatable.
>=20
>I will note that, it is possible to separate these concerns.=20
>We could define UA loose route to strictly be used for the=20
>registration cases, since that is the specific problem at hand.
>=20
>I agree CHrister's mechanism does not have this configuration=20
>limitation, and that this aspect of it is better. My main=20
>beef, as others have commented, is that it leaves us with=20
>nearly half-dozen header fields which are all awfully similar=20
>(To, P-C-ID, Route, History-Info, and now Target). I was=20
>hoping to reduce the set and include only the minimum. In=20
>that way, I think UA loose route is architecturally better=20
>since it gives some clarity to this.

I don't think a solution with the configuration limitation is
architecturally better. =20

However, there is also a possibility to merge both solution proposals:
UA-loose-route would be used from the home proxy towards the UE (giving
Request-URI the same semantics as P-Called-Party-ID has today), and
Target to convey the current target. Yes, we would have another header.
But, the new header would have clear semantics, and we would get rid of
the configuration limitation. (Another possibility that does not require
a new header is to fix the semantics of the To header and allow it to be
modified, but that may have unwanted backward compatibility issues.)

The merged solution would probably also solve the issues raised by
Youssef - eventhough the "intermediate entity" (being a B2BUA) he is
talking about could as well map the Target header into a R-URI. However
we think it would be good to additionally cover the use case that is
behind this in the UALR work to justify such a combined solution.

Regards,

Christer









>=20
> Elwell, John wrote:
> > Christer,
> >=20
> > I don't think the entity using the mechanism needs to know=20
> about next=20
> > hop support for the loose route mechanism - only whether the UAS=20
> > supports it. If the UA registers, then the loose route=20
> draft provides=20
> > an automatic mechanism. If the UA does not register, support would=20
> > need to be indicated by provisioning. I am not sure this ranks as a=20
> > significant limitation, since a certain amount of=20
> provisioning needs=20
> > to be done anyway for UAs such as gateways that do not register.
> >=20
> > John
> >=20
> >> -----Original Message-----
> >> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >> Sent: 16 January 2008 11:51
> >> To: Elwell, John; sip@ietf.org
> >> Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> >> Subject: RE: [Sip] RE: Delivering request-URI and=20
> parameters to UAS=20
> >> via proxy -new version of the=20
> draft-holmberg-sip-target-uri-delivery=20
> >> draft
> >>
> >>
> >> Hi,
> >>
> >>> I hear what you are saying, but I don't really see that the
> >> points you
> >> raised in section 4 are real limitations of the loose=20
> route mechanism.
> >>
> >> I think it is a limitation that an entity using the=20
> mechanism has to=20
> >> know whether the the next hop supports it or not, and that the=20
> >> mechanism can only be used if the subsequent hops support it.
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >>
> >>
> >>>> -----Original Message-----
> >>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>>> Sent: 16 January 2008 11:06
> >>>> To: Elwell, John; sip@ietf.org
> >>>> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> >>>> Subject: [Sip] RE: Delivering request-URI and parameters
> >> to UAS via
> >>>> proxy - new version of the draft-holmberg-sip-target-uri-delivery
> >>>> draft
> >>>>
> >>>>
> >>>> Hi John,
> >>>>
> >>>>> Thanks for this revision, which makes things somewhat
> >>> clearer. I do
> >>>>> have a couple of comments:
> >>>>>
> >>>>> 1. I am not sure I agree with the assertions in section 4
> >>> concerning
> >>>>> issues with the mechanisms in loose-route. Taking=20
> example 1, the=20
> >>>>> Route header field should contain enough entries to get
> >> you to the
> >>>>> registered contact, not just to an intermediate proxy.=20
> >>> Therefore this
> >>>>> situation should not arise with a correctly implemented
> >>> home proxy.=20
> >>>>> It is not clear to me how example 2 could arise either,
> >>> for similar
> >>>>> reasons. The MGC case can be resolved by taking into=20
> account the=20
> >>>>> option tag in the REGISTER request, or if it is permanently=20
> >>>>> registered, through provisioning.
> >>>> The examples are not meant to show bugs in the loose-route
> >>> mechanism.
> >>>> They are meant to help people understand the limitations=20
> with the=20
> >>>> loose-route mechanism.
> >>>> =20
> >>>>> 2. Comparing the mechanism proposed with the loose-route
> >>> mechanism,
> >>>>> my understanding is:
> >>>>> a)When retargeting occurs, the loose-route mechanism
> >>> places the new
> >>>>> target in the Request URI. Your proposal places the new
> >> target in
> >>>>> both the Request-URI and the Target header field.
> >>>>> b) When rerouting, the loose-route mechanism places the
> >> new route
> >>>>> (i.e., the registered contact) in the Route header field. Your=20
> >>>>> proposal places the new route in the Request-URI (the
> >>> latter as per
> >>>>> RFC 3261).
> >>>>> So the two mechanisms solve exactly the same problems using a=20
> >>>>> slightly different mechanism. Correct?
> >>>> Yes. The two solutions intend to solve the same problem.
> >>>> =20
> >>>>> 3. How P-Called-Party-ID fits into this is not really
> >>> relevant from
> >>>>> an IETF perspective - it seems there are some 3GPP-specific=20
> >>>>> situations where the contents of P-Called-Party-ID will
> >>> not equal the
> >>>>> contents of Target. Correct?
> >>>> Correct.
> >>>>
> >>>>> 4. If my suggestion in point 1 above that the loose-route
> >>> mechanism
> >>>>> does not suffer from the problems suggested, then each
> >>> mechanism will
> >>>>> work and each addresses the same problem.
> >>>>> So it is just down to a beauty contest between the two. Correct?
> >>>> See question 1.
> >>>>
> >>>> We believe that our solution does not have the same
> >>> limitations as the
> >>>> loose-route solution. But, again, both solutions intend to
> >>> solve the
> >>>> same problem.
> >>>>
> >>>> Regards,
> >>>>
> >>>> Christer
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>>> -----Original Message-----
> >>>>>> From: Christer Holmberg
> >> [mailto:christer.holmberg@ericsson.com]
> >>>>>> Sent: 16 January 2008 08:41
> >>>>>> To: sip@ietf.org
> >>>>>> Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;
> >>> Jeroen van
> >>>>>> Bemmel; Francois Audet
> >>>>>> Subject: Delivering request-URI and parameters to UAS via
> >>>>> proxy - new
> >>>>>> version of the draft-holmberg-sip-target-uri-delivery draft
> >>>>>>
> >>>>>>
> >>>>>> Hi,
> >>>>>>
> >>>>>> We've uploaded a new version (-01) of the Target draft.
> >>>>>>
> >>>>>> We've tried to make things more clear. I've also
> >>> removed all text
> >>>>>> about P-Called-Party-ID, except from one chapter where
> >>> we try to
> >>>>>> explain the semantical difference between Target and P-CPI.
> >>>>>>
> >>>>>> You can also find the draft from:
> >>>>>> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> >>>>>> et-uri-del
> >>>>>> ivery-01.txt
> >>>>>>
> >>>>>> Regards,
> >>>>>>
> >>>>>> Christer
> >>>>>>
> >>>>
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sipping@ietf.org for new developments on the application of sip
> >>>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
> >>>
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
> Cisco Fellow                                   Edison, NJ 08837
> Cisco, Voice Technology Group
> jdrosen@cisco.com
> http://www.jdrosen.net                         PHONE: (408) 902-3084
> http://www.cisco.com
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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youssef.chadli@orange-ftgroup.com wrote:
> I think there is a misunderstanding on these use cases:

Quite possible.

>  - The intermediate entity in the customer side does perform only one registration to the network.
>    The AoRs associated to the served terminals inside the client domain are registred automatically
>    by the home network, using implicit registration for example. In fact, it's not conceivable to
>    register individually all the users of a corporate network for example.

You seem to be describing one of the models covered by the SIPForum in 
their SIPConnect specification. Is that what you mean?

 From the point of view of the registrar this seems similar to IMS 
implicit registration, except that the scale is much larger - there may 
be *very many* implicit registrations generated by this one explicit 
registration.

> - The "intermediate entity" should be able to route incoming requests inside its domain based on
>   standard RFC3261 behaviour, that is, based on the Request-URI.

I find the case much less compelling than the IMS case.

IMO this is an abuse of REGISTER to a purpose for which it is not well 
suited.

Treating it as registration means that there is an *expectation* that 
the R-URI will be translated. That isn't really appropriate in this 
case. The registration doesn't specify the mapping anyway - it is 
configuration in the network of all the addresses that belong to the 
"pbx" that does that. This is more appropriately viewed as just a 
routing operation, and so pushing the address of the enterprise server 
as a Route header is by far the best choice. Its only because this has 
been done using REGISTER that brings up how to preserve the old address 
when it is translated.

	Thanks,
	Paul

> Best regards
> 
> Youssef
> 
> 
>   
>  
> 
>> -----Message d'origine-----
>> De : Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Envoyé : lundi 28 janvier 2008 21:46
>> À : CHADLI Youssef RD-CORE-ISS
>> Cc : drage@alcatel-lucent.com; sip@ietf.org
>> Objet : Re: [Sip] Delivering request-URI and parameters to 
>> UAS via proxy
>>
>> These cases can all be handled by having intermediate entity 
>> use a unique user part, together with its own domain name, to 
>> construct a contact address for each terminal UA that it is 
>> registering. Then when it receives an incoming request to one 
>> of these uris it can simply translate it, using the user part 
>> as a key to its own local mappings.
>>
>> We don't need anything new to solve these cases.
>>
>> 	Paul
>>
>> youssef.chadli@orange-ftgroup.com wrote:
>>>  
>>> There are other cases similar to what is described in 
>> section  2.1 Unknown Aliases of J. Rosenberg draft that 
>> should be taken into account. 
>>> Those cases are where the customer side is composed of 
>> several SIP terminals that access the network through an 
>> intermediate entity which registers their associated aliases 
>> with its contact address. Thus, such client side is seen from 
>> the network as a single client having several aliases 
>> (aggregated endpoints). Moreover, such client side may be a 
>> mini private network composed of several entities. In these 
>> configurations, the intermediate entity is in charge of 
>> routeing incoming calls inside the client domain and may need 
>> to behave as a SIP proxy for incoming SIP messages.  
>>> As examples of such configuration:
>>> - Corporate networks: in that case the PBX register all the 
>> served individual user identities with its contact address 
>> and routes incoming calls toward the individual called users.
>>> - Home networks: the Home Gateway may need to register on 
>> behalf of all the served terminals.  The Home Gateway is in 
>> charge of routeing incoming calls toward the individual called users. 
>>> J. Rosenberg draft seems give a good solution to take into 
>> account these configurations.
>>>    
>>> Best regards,
>>>
>>> Youssef
>>>
>>>
>>>> -----Message d'origine-----
>>>> De : DRAGE, Keith (Keith) 
>> [mailto:drage@alcatel-lucent.com] Envoyé : 
>>>> lundi 14 janvier 2008 17:58 À : sip@ietf.org Objet : [Sip] 
>> Delivering 
>>>> request-URI and parameters to UAS via proxy
>>>>
>>>> (As WG chair)
>>>>
>>>> In fulfilment of our charter items of
>>>>
>>>> Dec 2007    Delivering request-URI and parameters to UAS 
>> via proxy to
>>>> WGLC  
>>>> Feb 2008    Delivering request-URI and parameters to UAS 
>> via proxy to
>>>> IESG (PS)
>>>>
>>>> We now have a couple of proposals on the table for solving the 
>>>> problem.
>>>>
>>>> The original draft from Jonathan and which led to the 
>> creation of the 
>>>> charter items by the WG is unfortunately expired, but is at:
>>>>
>>>> http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
>>>>
>>>> The alternative document from Christer, etc is at:
>>>>
>>>> http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
>>>> uri-delive
>>>> ry-00.txt
>>>>
>>>> We obviously need to make a decision between the two approaches so 
>>>> please attempt to address the following specific points via the 
>>>> mailing
>>>> list:
>>>>
>>>> 1)	Problem cases: These are summarised in section 4.4 of
>>>> draft-rosenberg-sip-ua-loose-route-01 and from my read of 
>> the other 
>>>> draft, I don't believe that this draft adds any others.
>>>> If you believe there are other cases that should be covered by the 
>>>> solution, then please identify them. If there is support 
>> on any new 
>>>> problem cases, I would encourage the authors of both drafts to add 
>>>> text concerning these problem cases.
>>>>
>>>> 2)	Clarifications: If for any reason you don't understand either
>>>> draft, or believe that there are technical issues that are not 
>>>> represented in the current draft, please post your questions / 
>>>> comments to the list. I would encourage authors of both drafts to 
>>>> revise as frequently as appropriate to reflect the current 
>> state of 
>>>> discussion.
>>>>
>>>> 3)	Support for either position. If you wish to indicate support for
>>>> either position please do so, but please accompany this is 
>> technical 
>>>> reasoning as to why you have this position, as that will 
>> help other 
>>>> members of the WG form a position.
>>>>
>>>> I would encourage as much list discussion as possible before 
>>>> Philadelphia. I suspect we will need to have a discussion at the 
>>>> face-to-face meeting in Philadephia, but list discussion 
>> is essential 
>>>> prior to that. If we can solve this positions on list, 
>> then well and 
>>>> good, and even better (that is how we are meant to make decisions).
>>>>
>>>> Regards
>>>>
>>>>
>>>> Keith
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP 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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SIPit 22 will be hosted by the University of New Hampshire  
Interoperability Laboratory
April 14 through 18, 2008.

Registration is $350 (USD) per person. Registration will close April 4.

Space is limited, so register early to ensure a seat.

Additional registration details are available at http://www.sipit.net








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Subject: RE: [Sip] WGLC for draft-ietf-sip-fork-loop-fix-06
Date: Wed, 30 Jan 2008 20:18:33 +0100
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>, <sip@ietf.org>
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(As WG chair)

There were no comments in response to this WGLC, so I propose to submit
a publish request for it shortly. If it is still on your pending list to
look at, then immediately would be a good time to get round to it.

Regards

Keith=20

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Sent: Friday, December 07, 2007 7:33 PM
> To: sip@ietf.org
> Cc: alan.ietf@polyphase.ca; slawrence@bluesocket.com
> Subject: [Sip] WGLC for draft-ietf-sip-fork-loop-fix-06
>=20
> (As WG chair)
>=20
> This document has a somewhat complicated history, in that we=20
> completed it once in the WG and sent it to the IESG, but it=20
> came back from them because of concerns over security issues.
>=20
> The editor's have now followed the instructions from the=20
> previous meeting, i.e. to:
>=20
>       Integrated Max-Breadth based on working group discussion of the
>       secdir review
>=20
>       Added a paragraph pointing out that removing or modifying other
>       node's branch parameters defeats their ability to loop detect
>=20
>       Moved the total number of messages from O(2^70) to O(2^71) based
>       on an observation by Jan Kolomaznik.  To see this, note that the
>       total number of requests is the sum from i=3D0 to Max-Forwards =
of
>       2^i which is 2^(Max-Forwards+1) - 1.  The point of the text
>       doesn't change - (the point being that the number is _big_).
>=20
>       Made the new 4xx concrete (choosing 440)
>=20
>       Added a sentence reinforcing that if you forward to only one
>       branch, you still potentially have a constant multiplier of
>       messages in the network as Max-Forwards runs out (based on
>       feedback from Thomas Cross.)
>=20
> These changes are significant enough to justify a revised=20
> WGLC, and I would therefore like the working group to review=20
> the document and provide comments to the SIP list and authors by=20
>=20
> 	Friday 21st December 2007.=20
>=20
> It would be appropriate to concentrate on the text with the=20
> changes identified about.
>=20
> The document may be found at:
>=20
> http://www.ietf.org/internet-drafts/draft-ietf-sip-fork-loop-f
> ix-06.txt
>=20
> And the wdiff document may be found at:
>=20
> http://tools.ietf.org/rfcdiff?difftype=3D--hwdiff&url2=3Dhttp://to
> ols.ietf.o
> rg/id/draft-ietf-sip-fork-loop-fix-06.txt
>=20
> [Note that unfortunately this identifies a significant=20
> portion of text as new text, when it fact it merely moved.]
>=20
> Regards
>=20
> Keith
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Wed Jan 30 19:22:22 2008
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Subject: RE: [Sip] Delivering request-URI and parameters to UAS via proxy
Date: Thu, 31 Jan 2008 01:21:59 +0100
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=20
Inline

Youssef

> -----Message d'origine-----
> De : Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Envoy=E9 : mercredi 30 janvier 2008 16:12
> =C0 : CHADLI Youssef RD-CORE-ISS
> Cc : drage@alcatel-lucent.com; sip@ietf.org
> Objet : Re: [Sip] Delivering request-URI and parameters to=20
> UAS via proxy
>=20
>=20
>=20
> youssef.chadli@orange-ftgroup.com wrote:
> > I think there is a misunderstanding on these use cases:
>=20
> Quite possible.
>=20
> >  - The intermediate entity in the customer side does=20
> perform only one registration to the network.
> >    The AoRs associated to the served terminals inside the=20
> client domain are registred automatically
> >    by the home network, using implicit registration for=20
> example. In fact, it's not conceivable to
> >    register individually all the users of a corporate=20
> network for example.
>=20
> You seem to be describing one of the models covered by the=20
> SIPForum in their SIPConnect specification. Is that what you mean?

The model described in Sipconnect is an example of such configuration. =
As I mentionned before there other cases such as home networks and other =
configurations of connecting corporate networks.
=20
>  From the point of view of the registrar this seems similar=20
> to IMS implicit registration, except that the scale is much=20
> larger - there may be *very many* implicit registrations=20
> generated by this one explicit registration.

Agreed

>=20
> > - The "intermediate entity" should be able to route=20
> incoming requests inside its domain based on
> >   standard RFC3261 behaviour, that is, based on the Request-URI.
>=20
> I find the case much less compelling than the IMS case.
>=20
> IMO this is an abuse of REGISTER to a purpose for which it is=20
> not well suited.
>=20
> Treating it as registration means that there is an=20
> *expectation* that the R-URI will be translated. That isn't=20
> really appropriate in this case. The registration doesn't=20
> specify the mapping anyway - it is configuration in the=20
> network of all the addresses that belong to the "pbx" that=20
> does that. This is more appropriately viewed as just a=20
> routing operation, and so pushing the address of the=20
> enterprise server as a Route header is by far the best=20
> choice. Its only because this has been done using REGISTER=20
> that brings up how to preserve the old address when it is translated.

Agreed
>=20
> 	Thanks,
> 	Paul
>=20
> > Best regards
> >=20
> > Youssef
> >=20
> >=20
> >  =20
> > =20
> >=20
> >> -----Message d'origine-----
> >> De : Paul Kyzivat [mailto:pkyzivat@cisco.com] Envoy=E9 : lundi 28=20
> >> janvier 2008 21:46 =C0 : CHADLI Youssef RD-CORE-ISS Cc :=20
> >> drage@alcatel-lucent.com; sip@ietf.org Objet : Re: [Sip]=20
> Delivering=20
> >> request-URI and parameters to UAS via proxy
> >>
> >> These cases can all be handled by having intermediate entity use a=20
> >> unique user part, together with its own domain name, to=20
> construct a=20
> >> contact address for each terminal UA that it is registering. Then=20
> >> when it receives an incoming request to one of these uris it can=20
> >> simply translate it, using the user part as a key to its own local=20
> >> mappings.
> >>
> >> We don't need anything new to solve these cases.
> >>
> >> 	Paul
> >>
> >> youssef.chadli@orange-ftgroup.com wrote:
> >>> =20
> >>> There are other cases similar to what is described in
> >> section  2.1 Unknown Aliases of J. Rosenberg draft that should be=20
> >> taken into account.
> >>> Those cases are where the customer side is composed of
> >> several SIP terminals that access the network through an=20
> intermediate=20
> >> entity which registers their associated aliases with its contact=20
> >> address. Thus, such client side is seen from the network=20
> as a single=20
> >> client having several aliases (aggregated endpoints).=20
> Moreover, such=20
> >> client side may be a mini private network composed of several=20
> >> entities. In these configurations, the intermediate entity is in=20
> >> charge of routeing incoming calls inside the client domain and may=20
> >> need to behave as a SIP proxy for incoming SIP messages.
> >>> As examples of such configuration:
> >>> - Corporate networks: in that case the PBX register all the
> >> served individual user identities with its contact address=20
> and routes=20
> >> incoming calls toward the individual called users.
> >>> - Home networks: the Home Gateway may need to register on
> >> behalf of all the served terminals.  The Home Gateway is=20
> in charge of=20
> >> routeing incoming calls toward the individual called users.
> >>> J. Rosenberg draft seems give a good solution to take into
> >> account these configurations.
> >>>   =20
> >>> Best regards,
> >>>
> >>> Youssef
> >>>
> >>>
> >>>> -----Message d'origine-----
> >>>> De : DRAGE, Keith (Keith)
> >> [mailto:drage@alcatel-lucent.com] Envoy=E9 :=20
> >>>> lundi 14 janvier 2008 17:58 =C0 : sip@ietf.org Objet : [Sip]
> >> Delivering
> >>>> request-URI and parameters to UAS via proxy
> >>>>
> >>>> (As WG chair)
> >>>>
> >>>> In fulfilment of our charter items of
> >>>>
> >>>> Dec 2007    Delivering request-URI and parameters to UAS=20
> >> via proxy to
> >>>> WGLC =20
> >>>> Feb 2008    Delivering request-URI and parameters to UAS=20
> >> via proxy to
> >>>> IESG (PS)
> >>>>
> >>>> We now have a couple of proposals on the table for solving the=20
> >>>> problem.
> >>>>
> >>>> The original draft from Jonathan and which led to the
> >> creation of the
> >>>> charter items by the WG is unfortunately expired, but is at:
> >>>>
> >>>>=20
> http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
> >>>>
> >>>> The alternative document from Christer, etc is at:
> >>>>
> >>>> http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
> >>>> uri-delive
> >>>> ry-00.txt
> >>>>
> >>>> We obviously need to make a decision between the two=20
> approaches so=20
> >>>> please attempt to address the following specific points via the=20
> >>>> mailing
> >>>> list:
> >>>>
> >>>> 1)	Problem cases: These are summarised in section 4.4 of
> >>>> draft-rosenberg-sip-ua-loose-route-01 and from my read of
> >> the other
> >>>> draft, I don't believe that this draft adds any others.
> >>>> If you believe there are other cases that should be=20
> covered by the=20
> >>>> solution, then please identify them. If there is support
> >> on any new
> >>>> problem cases, I would encourage the authors of both=20
> drafts to add=20
> >>>> text concerning these problem cases.
> >>>>
> >>>> 2)	Clarifications: If for any reason you don't=20
> understand either
> >>>> draft, or believe that there are technical issues that are not=20
> >>>> represented in the current draft, please post your questions /=20
> >>>> comments to the list. I would encourage authors of both=20
> drafts to=20
> >>>> revise as frequently as appropriate to reflect the current
> >> state of
> >>>> discussion.
> >>>>
> >>>> 3)	Support for either position. If you wish to=20
> indicate support for
> >>>> either position please do so, but please accompany this is
> >> technical
> >>>> reasoning as to why you have this position, as that will
> >> help other
> >>>> members of the WG form a position.
> >>>>
> >>>> I would encourage as much list discussion as possible before=20
> >>>> Philadelphia. I suspect we will need to have a discussion at the=20
> >>>> face-to-face meeting in Philadephia, but list discussion
> >> is essential
> >>>> prior to that. If we can solve this positions on list,
> >> then well and
> >>>> good, and even better (that is how we are meant to make=20
> decisions).
> >>>>
> >>>> Regards
> >>>>
> >>>>
> >>>> Keith
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sipping@ietf.org for new developments on the application of sip
> >>>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
> >>>
> >=20
>=20


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From sip-bounces@ietf.org Wed Jan 30 20:58:45 2008
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Subject: RE: [Sip] RE: Delivering request-URI and parameters to UAS via
	proxy-new version of the draft-holmberg-sip-target-uri-delivery draft
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Hi,=20

The merged solution proposed by Christer would have the following =
advantages:=20

 - The "UA loose route"  mechanism would solve the "registration cases", =
particularly the problem of routing incoming sessions to the right =
terminal where the client side is composed of several endpoints =
accessing the network through an intermediate entity which registers =
their associated aliases with its contact address (see the thread =
discussing these use cases).

 - The Target header would solve the backward compatibility limitation =
for the other cases.

Regards,=20

Youssef =20


> -----Message d'origine-----
> De : Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Envoy=E9 : mardi 29 janvier 2008 22:58
> =C0 : Jonathan Rosenberg; Elwell, John
> Cc : sip@ietf.org; Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> Objet : RE: [Sip] RE: Delivering request-URI and parameters=20
> to UAS via proxy-new version of the=20
> draft-holmberg-sip-target-uri-delivery draft
>=20
>=20
> Hi Jonathan,
>=20
> >Sorry for jumping in late here. Trying to clarify some=20
> points here and=20
> >there.
> >=20
> >The UA-loose mechanism needs to know that the entity which=20
> would have=20
> >previously been reached by the DNS/IP forwarding based on=20
> the R-URI, is=20
> >now prepared for a request which reaches it with that target in the=20
> >Route header field.
> >=20
> >In the case of a UA registering through a chain of Path-inserting=20
> >proxies, it means that only the *UA* has to support UA-loose route.
> >Thats because of the assumption that the intermediate path-inserting=20
> >proxies weren't being targeted based on the R-URI=20
> previously; they were=20
> >(and still will) forward the request based on Route.
> >
> >The 'next hop' use case that Christer is referring to are=20
> cases where=20
> >we would have done a R-URI translation without a registration. For=20
> >example, lets say a proxy receives an INVITE with=20
> sip:12345@proxy.com,=20
> >and its using some number-prefix based routing rules, which=20
> have been=20
> >CONFIGURED into the proxy, to determine a next-hop gateway. The UA=20
> >loose route rules would say that, since this is a routing=20
> operation and=20
> >not a retargeting, the request looks like:
> >=20
> >INVITE sip:12345@proxy.com
> >Route: sip:ip-of-gateway
> >=20
> >whereas current rules would require:
> >=20
> >INVITE sip:12345@ip-of-gateway
> >=20
> >in this use case, the entity which was previously being=20
> reached by the=20
> >R-URI *is* the next hop.
> >=20
> >And so, the main limitation that I see of the UA loose route=20
> mechanism=20
> >is, in cases where the next-hop is being reached via a configured=20
> >routing rule, or by some other non-registration means, and said=20
> >operation is a re-route and not retarget, the UA_loose-route=20
> requires=20
> >that this configured routing rule include information about=20
> whether the=20
> >next hop uses UA loose routing, and if so, what URI to use=20
> in the Route=20
> >header.
> >=20
> >This is a limitation, I agree. Whether its significant or not is=20
> >debatable.
> >=20
> >I will note that, it is possible to separate these concerns.=20
> >We could define UA loose route to strictly be used for the=20
> registration=20
> >cases, since that is the specific problem at hand.
> >=20
> >I agree CHrister's mechanism does not have this configuration=20
> >limitation, and that this aspect of it is better. My main beef, as=20
> >others have commented, is that it leaves us with nearly half-dozen=20
> >header fields which are all awfully similar (To, P-C-ID, Route,=20
> >History-Info, and now Target). I was hoping to reduce the set and=20
> >include only the minimum. In that way, I think UA loose route is=20
> >architecturally better since it gives some clarity to this.
>=20
> I don't think a solution with the configuration limitation is=20
> architecturally better. =20
>=20
> However, there is also a possibility to merge both solution proposals:
> UA-loose-route would be used from the home proxy towards the=20
> UE (giving Request-URI the same semantics as=20
> P-Called-Party-ID has today), and Target to convey the=20
> current target. Yes, we would have another header.
> But, the new header would have clear semantics, and we would=20
> get rid of the configuration limitation. (Another possibility=20
> that does not require a new header is to fix the semantics of=20
> the To header and allow it to be modified, but that may have=20
> unwanted backward compatibility issues.)
>=20
> The merged solution would probably also solve the issues=20
> raised by Youssef - eventhough the "intermediate entity"=20
> (being a B2BUA) he is talking about could as well map the=20
> Target header into a R-URI. However we think it would be good=20
> to additionally cover the use case that is behind this in the=20
> UALR work to justify such a combined solution.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> >=20
> > Elwell, John wrote:
> > > Christer,
> > >=20
> > > I don't think the entity using the mechanism needs to know
> > about next
> > > hop support for the loose route mechanism - only whether the UAS=20
> > > supports it. If the UA registers, then the loose route
> > draft provides
> > > an automatic mechanism. If the UA does not register,=20
> support would=20
> > > need to be indicated by provisioning. I am not sure this=20
> ranks as a=20
> > > significant limitation, since a certain amount of
> > provisioning needs
> > > to be done anyway for UAs such as gateways that do not register.
> > >=20
> > > John
> > >=20
> > >> -----Original Message-----
> > >> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > >> Sent: 16 January 2008 11:51
> > >> To: Elwell, John; sip@ietf.org
> > >> Cc: Paul Kyzivat; DRAGE,Keith (Keith); Francois Audet
> > >> Subject: RE: [Sip] RE: Delivering request-URI and
> > parameters to UAS
> > >> via proxy -new version of the
> > draft-holmberg-sip-target-uri-delivery
> > >> draft
> > >>
> > >>
> > >> Hi,
> > >>
> > >>> I hear what you are saying, but I don't really see that the
> > >> points you
> > >> raised in section 4 are real limitations of the loose
> > route mechanism.
> > >>
> > >> I think it is a limitation that an entity using the
> > mechanism has to
> > >> know whether the the next hop supports it or not, and that the=20
> > >> mechanism can only be used if the subsequent hops support it.
> > >>
> > >> Regards,
> > >>
> > >> Christer
> > >>
> > >>
> > >>
> > >>
> > >>>> -----Original Message-----
> > >>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > >>>> Sent: 16 January 2008 11:06
> > >>>> To: Elwell, John; sip@ietf.org
> > >>>> Cc: Paul Kyzivat; DRAGE, Keith (Keith); Francois Audet
> > >>>> Subject: [Sip] RE: Delivering request-URI and parameters
> > >> to UAS via
> > >>>> proxy - new version of the=20
> draft-holmberg-sip-target-uri-delivery
> > >>>> draft
> > >>>>
> > >>>>
> > >>>> Hi John,
> > >>>>
> > >>>>> Thanks for this revision, which makes things somewhat
> > >>> clearer. I do
> > >>>>> have a couple of comments:
> > >>>>>
> > >>>>> 1. I am not sure I agree with the assertions in section 4
> > >>> concerning
> > >>>>> issues with the mechanisms in loose-route. Taking
> > example 1, the
> > >>>>> Route header field should contain enough entries to get
> > >> you to the
> > >>>>> registered contact, not just to an intermediate proxy.=20
> > >>> Therefore this
> > >>>>> situation should not arise with a correctly implemented
> > >>> home proxy.=20
> > >>>>> It is not clear to me how example 2 could arise either,
> > >>> for similar
> > >>>>> reasons. The MGC case can be resolved by taking into
> > account the
> > >>>>> option tag in the REGISTER request, or if it is permanently=20
> > >>>>> registered, through provisioning.
> > >>>> The examples are not meant to show bugs in the loose-route
> > >>> mechanism.
> > >>>> They are meant to help people understand the limitations
> > with the
> > >>>> loose-route mechanism.
> > >>>> =20
> > >>>>> 2. Comparing the mechanism proposed with the loose-route
> > >>> mechanism,
> > >>>>> my understanding is:
> > >>>>> a)When retargeting occurs, the loose-route mechanism
> > >>> places the new
> > >>>>> target in the Request URI. Your proposal places the new
> > >> target in
> > >>>>> both the Request-URI and the Target header field.
> > >>>>> b) When rerouting, the loose-route mechanism places the
> > >> new route
> > >>>>> (i.e., the registered contact) in the Route header=20
> field. Your=20
> > >>>>> proposal places the new route in the Request-URI (the
> > >>> latter as per
> > >>>>> RFC 3261).
> > >>>>> So the two mechanisms solve exactly the same problems using a=20
> > >>>>> slightly different mechanism. Correct?
> > >>>> Yes. The two solutions intend to solve the same problem.
> > >>>> =20
> > >>>>> 3. How P-Called-Party-ID fits into this is not really
> > >>> relevant from
> > >>>>> an IETF perspective - it seems there are some 3GPP-specific=20
> > >>>>> situations where the contents of P-Called-Party-ID will
> > >>> not equal the
> > >>>>> contents of Target. Correct?
> > >>>> Correct.
> > >>>>
> > >>>>> 4. If my suggestion in point 1 above that the loose-route
> > >>> mechanism
> > >>>>> does not suffer from the problems suggested, then each
> > >>> mechanism will
> > >>>>> work and each addresses the same problem.
> > >>>>> So it is just down to a beauty contest between the=20
> two. Correct?
> > >>>> See question 1.
> > >>>>
> > >>>> We believe that our solution does not have the same
> > >>> limitations as the
> > >>>> loose-route solution. But, again, both solutions intend to
> > >>> solve the
> > >>>> same problem.
> > >>>>
> > >>>> Regards,
> > >>>>
> > >>>> Christer
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>>> -----Original Message-----
> > >>>>>> From: Christer Holmberg
> > >> [mailto:christer.holmberg@ericsson.com]
> > >>>>>> Sent: 16 January 2008 08:41
> > >>>>>> To: sip@ietf.org
> > >>>>>> Cc: DRAGE, Keith (Keith); Paul Kyzivat; Elwell, John;
> > >>> Jeroen van
> > >>>>>> Bemmel; Francois Audet
> > >>>>>> Subject: Delivering request-URI and parameters to UAS via
> > >>>>> proxy - new
> > >>>>>> version of the draft-holmberg-sip-target-uri-delivery draft
> > >>>>>>
> > >>>>>>
> > >>>>>> Hi,
> > >>>>>>
> > >>>>>> We've uploaded a new version (-01) of the Target draft.
> > >>>>>>
> > >>>>>> We've tried to make things more clear. I've also
> > >>> removed all text
> > >>>>>> about P-Called-Party-ID, except from one chapter where
> > >>> we try to
> > >>>>>> explain the semantical difference between Target and P-CPI.
> > >>>>>>
> > >>>>>> You can also find the draft from:
> > >>>>>>=20
> http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-targ
> > >>>>>> et-uri-del
> > >>>>>> ivery-01.txt
> > >>>>>>
> > >>>>>> Regards,
> > >>>>>>
> > >>>>>> Christer
> > >>>>>>
> > >>>>
> > >>>> _______________________________________________
> > >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >>>> This list is for NEW development of the core SIP Protocol Use=20
> > >>>> sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > >>>> sipping@ietf.org for new developments on the application of sip
> > >>>>
> > >>>
> > >>> _______________________________________________
> > >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >>> This list is for NEW development of the core SIP Protocol Use=20
> > >>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > >>> sipping@ietf.org for new developments on the application of sip
> > >>>
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
> > --=20
> > Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
> > Cisco Fellow                                   Edison, NJ 08837
> > Cisco, Voice Technology Group
> > jdrosen@cisco.com
> > http://www.jdrosen.net                         PHONE: (408) 902-3084
> > http://www.cisco.com
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20
>=20


_______________________________________________
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From sip-bounces@ietf.org Wed Jan 30 23:32:25 2008
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Thread-Topic: Extension mobility in SIP
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Hi ,

 Is there a separate RFC for the implementation of extension mobility in
SIP Uas ? If not, is it possible to achieve that ?

 Please provide your valuable suggestions and feedbacks.

Thanks & Regards,
Anisha

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<P><FONT COLOR=3D"#000080" FACE=3D"Garamond">Hi ,</FONT>
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<P><FONT COLOR=3D"#000080" FACE=3D"Garamond">&nbsp;Is there a separate =
RFC for the implementation of extension mobility in SIP Uas ? If not, is =
it possible to achieve that ?</FONT></P>

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valuable suggestions and feedbacks.</FONT>
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<P><FONT COLOR=3D"#000080" FACE=3D"Garamond">Thanks &amp; =
Regards,</FONT>

<BR><FONT COLOR=3D"#000080" FACE=3D"Garamond">Anisha</FONT>
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_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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--===============0410490703==--





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Subject: Re: [Sip] Extension mobility in SIP
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On Jan 30, 2008, at 10:30 PM, Anisha Kumar wrote:

> Hi ,
>
>  Is there a separate RFC for the implementation of extension  
> mobility in SIP Uas ? If not, is it possible to achieve that ?
>
What do you mean "extension mobility"? Are you thinking of the use  
case where a SIP user agent is moving around and occasionally changes  
its IP address and point of attachment to the Internet? Do you expect  
call continuity in such a case, quick recovery of the call afterward,  
or call drop?

--
Dean


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At 11:28 PM 1/30/2008, Dean Willis wrote:

>On Jan 30, 2008, at 10:30 PM, Anisha Kumar wrote:
>
>>Hi ,
>>
>>  Is there a separate RFC for the implementation of extension
>>mobility in SIP Uas ? If not, is it possible to achieve that ?
>What do you mean "extension mobility"? Are you thinking of the use
>case where a SIP user agent is moving around and occasionally changes
>its IP address and point of attachment to the Internet? Do you expect
>call continuity in such a case, quick recovery of the call afterward,
>or call drop?

A lot of times, 'extension mobility' means the number to call a phone 
(UA) changes based on who logged (or registered) onto it -- which is 
very different that what is above.


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



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



From sip-bounces@ietf.org Thu Jan 31 04:04:37 2008
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Subject: RE: [Sip] Extension mobility in SIP
Date: Thu, 31 Jan 2008 14:33:40 +0530
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From: "Anisha Kumar" <anisha.kumar@nestgroup.net>
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Hi,=20

The Extension Mobility feature dynamically configures a phone according
to the current user. Once a user logs in, the phone adopts
the user individual user default device profile information, including
line numbers, speed dials, services links, and other user-specific
properties of a phone.

For example, when user A occupies a desk and logs in to the phone, her
directory number(s), services, speed dials, and other properties appear
on that phone; but when user B uses the same desk at a different time,
his information appears.

Hope this is clear. How is this achieved in SIP ?

Thanks & Regards,
Anisha

-----Original Message-----
From: James M. Polk [mailto:jmpolk@cisco.com]=20
Sent: Thursday, January 31, 2008 11:58 AM
To: Dean Willis; Anisha Kumar
Cc: sip@ietf.org
Subject: Re: [Sip] Extension mobility in SIP

At 11:28 PM 1/30/2008, Dean Willis wrote:

>On Jan 30, 2008, at 10:30 PM, Anisha Kumar wrote:
>
>>Hi ,
>>
>>  Is there a separate RFC for the implementation of extension mobility

>>in SIP Uas ? If not, is it possible to achieve that ?
>What do you mean "extension mobility"? Are you thinking of the use case

>where a SIP user agent is moving around and occasionally changes its IP

>address and point of attachment to the Internet? Do you expect call=20
>continuity in such a case, quick recovery of the call afterward, or=20
>call drop?

A lot of times, 'extension mobility' means the number to call a phone
(UA) changes based on who logged (or registered) onto it -- which is
very different that what is above.


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



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



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Date: Thu, 31 Jan 2008 07:57:37 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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This is largely not a sip question. The UA can register with any AOR 
(phone number) it wants after some local conversation with the user 
determine what that AOR should be.

	Paul

Anisha Kumar wrote:
> Hi, 
> 
> The Extension Mobility feature dynamically configures a phone according
> to the current user. Once a user logs in, the phone adopts
> the user individual user default device profile information, including
> line numbers, speed dials, services links, and other user-specific
> properties of a phone.
> 
> For example, when user A occupies a desk and logs in to the phone, her
> directory number(s), services, speed dials, and other properties appear
> on that phone; but when user B uses the same desk at a different time,
> his information appears.
> 
> Hope this is clear. How is this achieved in SIP ?
> 
> Thanks & Regards,
> Anisha
> 
> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com] 
> Sent: Thursday, January 31, 2008 11:58 AM
> To: Dean Willis; Anisha Kumar
> Cc: sip@ietf.org
> Subject: Re: [Sip] Extension mobility in SIP
> 
> At 11:28 PM 1/30/2008, Dean Willis wrote:
> 
>> On Jan 30, 2008, at 10:30 PM, Anisha Kumar wrote:
>>
>>> Hi ,
>>>
>>>  Is there a separate RFC for the implementation of extension mobility
> 
>>> in SIP Uas ? If not, is it possible to achieve that ?
>> What do you mean "extension mobility"? Are you thinking of the use case
> 
>> where a SIP user agent is moving around and occasionally changes its IP
> 
>> address and point of attachment to the Internet? Do you expect call 
>> continuity in such a case, quick recovery of the call afterward, or 
>> call drop?
> 
> A lot of times, 'extension mobility' means the number to call a phone
> (UA) changes based on who logged (or registered) onto it -- which is
> very different that what is above.
> 
> 
>> --
>> Dean
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>> sipping@ietf.org for new developments on the application of sip
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP 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 31 08:06:03 2008
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Subject: RE: [Sip] Extension mobility in SIP
Date: Thu, 31 Jan 2008 05:05:46 -0800
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	<47A1C5C1.3090401@cisco.com>
From: "Steve Langstaff" <steve.langstaff@citel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	"Anisha Kumar" <anisha.kumar@nestgroup.net>
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Or maybe it's a SIPPING config framework question, after some local
conversation?

http://tools.ietf.org/html/draft-ietf-sipping-config-framework-14

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: 31 January 2008 12:58
> To: Anisha Kumar
> Cc: sip@ietf.org
> Subject: Re: [Sip] Extension mobility in SIP
>=20
> This is largely not a sip question. The UA can register with=20
> any AOR (phone number) it wants after some local conversation=20
> with the user determine what that AOR should be.
>=20
> 	Paul
>=20
> Anisha Kumar wrote:
> > Hi,
> >=20
> > The Extension Mobility feature dynamically configures a phone=20
> > according to the current user. Once a user logs in, the=20
> phone adopts=20
> > the user individual user default device profile=20
> information, including=20
> > line numbers, speed dials, services links, and other user-specific=20
> > properties of a phone.
> >=20
> > For example, when user A occupies a desk and logs in to the=20
> phone, her=20
> > directory number(s), services, speed dials, and other properties=20
> > appear on that phone; but when user B uses the same desk at a=20
> > different time, his information appears.
> >=20
> > Hope this is clear. How is this achieved in SIP ?
> >=20
> > Thanks & Regards,
> > Anisha
> >=20
> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: Thursday, January 31, 2008 11:58 AM
> > To: Dean Willis; Anisha Kumar
> > Cc: sip@ietf.org
> > Subject: Re: [Sip] Extension mobility in SIP
> >=20
> > At 11:28 PM 1/30/2008, Dean Willis wrote:
> >=20
> >> On Jan 30, 2008, at 10:30 PM, Anisha Kumar wrote:
> >>
> >>> Hi ,
> >>>
> >>>  Is there a separate RFC for the implementation of extension=20
> >>> mobility
> >=20
> >>> in SIP Uas ? If not, is it possible to achieve that ?
> >> What do you mean "extension mobility"? Are you thinking of the use=20
> >> case
> >=20
> >> where a SIP user agent is moving around and occasionally=20
> changes its=20
> >> IP
> >=20
> >> address and point of attachment to the Internet? Do you=20
> expect call=20
> >> continuity in such a case, quick recovery of the call=20
> afterward, or=20
> >> call drop?
> >=20
> > A lot of times, 'extension mobility' means the number to=20
> call a phone
> > (UA) changes based on who logged (or registered) onto it --=20
> which is=20
> > very different that what is above.
> >=20
> >=20
> >> --
> >> Dean
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



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Or,

it could be a matter of using the config framework first, discovering 
that a local conversation is needed to determine identity, then a local 
conversation, then more config framework, then registration.

But AFAIK there is no guidance on what might be in a configuration that 
would kick that off.

	Paul

Steve Langstaff wrote:
> Or maybe it's a SIPPING config framework question, after some local
> conversation?
> 
> http://tools.ietf.org/html/draft-ietf-sipping-config-framework-14
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: 31 January 2008 12:58
>> To: Anisha Kumar
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Extension mobility in SIP
>>
>> This is largely not a sip question. The UA can register with 
>> any AOR (phone number) it wants after some local conversation 
>> with the user determine what that AOR should be.
>>
>> 	Paul
>>
>> Anisha Kumar wrote:
>>> Hi,
>>>
>>> The Extension Mobility feature dynamically configures a phone 
>>> according to the current user. Once a user logs in, the 
>> phone adopts 
>>> the user individual user default device profile 
>> information, including 
>>> line numbers, speed dials, services links, and other user-specific 
>>> properties of a phone.
>>>
>>> For example, when user A occupies a desk and logs in to the 
>> phone, her 
>>> directory number(s), services, speed dials, and other properties 
>>> appear on that phone; but when user B uses the same desk at a 
>>> different time, his information appears.
>>>
>>> Hope this is clear. How is this achieved in SIP ?
>>>
>>> Thanks & Regards,
>>> Anisha
>>>
>>> -----Original Message-----
>>> From: James M. Polk [mailto:jmpolk@cisco.com]
>>> Sent: Thursday, January 31, 2008 11:58 AM
>>> To: Dean Willis; Anisha Kumar
>>> Cc: sip@ietf.org
>>> Subject: Re: [Sip] Extension mobility in SIP
>>>
>>> At 11:28 PM 1/30/2008, Dean Willis wrote:
>>>
>>>> On Jan 30, 2008, at 10:30 PM, Anisha Kumar wrote:
>>>>
>>>>> Hi ,
>>>>>
>>>>>  Is there a separate RFC for the implementation of extension 
>>>>> mobility
>>>>> in SIP Uas ? If not, is it possible to achieve that ?
>>>> What do you mean "extension mobility"? Are you thinking of the use 
>>>> case
>>>> where a SIP user agent is moving around and occasionally 
>> changes its 
>>>> IP
>>>> address and point of attachment to the Internet? Do you 
>> expect call 
>>>> continuity in such a case, quick recovery of the call 
>> afterward, or 
>>>> call drop?
>>> A lot of times, 'extension mobility' means the number to 
>> call a phone
>>> (UA) changes based on who logged (or registered) onto it -- 
>> which is 
>>> very different that what is above.
>>>
>>>
>>>> --
>>>> Dean
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use 
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>> sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP 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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--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           : A Framework for Consent-based Communications in the Session Initiation Protocol (SIP)
	Author(s)       : J. Rosenberg, et al.
	Filename        : draft-ietf-sip-consent-framework-04.txt
	Pages           : 32
	Date            : 2008-01-31

The Session Initiation Protocol (SIP) supports communications for
several services, including, real-time audio, video, text, instant
messaging, and presence.  In its current form, it allows session
invitations, instant messages, and other requests to be delivered
from one party to another without requiring explicit consent of the
recipient.  Without such consent, it is possible for SIP to be used
for malicious purposes, including amplification, and DoS (Denial of
Service) attacks.  This document identifies a framework for consent-
based communications in SIP.

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Those of you submitting drafts that define syntax using ABNF should  
probably now be referencing RFC 5234 instead of whatever it was last  
week.

--
Dean

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And a free beer for anyone who can figure out what of relevance  
changed from 4234 to this :-)

On Jan 31, 2008, at 9:59 PM, Dean Willis wrote:

>
> Those of you submitting drafts that define syntax using ABNF should
> probably now be referencing RFC 5234 instead of whatever it was last
> week.
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  http://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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	HELO_EQ_IP_ADDR=1.119, NORMAL_HTTP_TO_IP=0.001,
	RAZOR2_CF_RANGE_51_100=0.5, RAZOR2_CF_RANGE_E8_51_100=1.5,
	RAZOR2_CHECK=0.5, RCVD_IN_BL_SPAMCOP_NET=1.96, RCVD_IN_PBL=0.905,
	RCVD_IN_XBL=3.033, RCVD_NUMERIC_HELO=2.067, RDNS_NONE=0.1,
	STOX_REPLY_TYPE=0.001, TVD_FINGER_02=2.134, TVD_RCVD_IP=1.931]
X-Spam-Report:
 *  3.5 BAYES_99 BODY: Bayesian spam probability is 99 to 100%
 *      [score: 1.0000]
 *  1.1 HELO_EQ_DYNAMIC HELO_EQ_DYNAMIC
 *  3.5 HELO_DYNAMIC_SPLIT_IP Relay HELO'd using suspicious hostname (Split
 *      IP)
 *  1.1 HELO_EQ_IP_ADDR HELO using IP Address (not private)
 *  1.5 FH_RELAY_NODNS We could not determine your Reverse DNS
 *  4.4 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr
 *       2)
 *  0.0 STOX_REPLY_TYPE STOX_REPLY_TYPE
 *  2.1 TVD_FINGER_02 TVD_FINGER_02
 *  1.9 TVD_RCVD_IP TVD_RCVD_IP
 *  2.1 RCVD_NUMERIC_HELO Received: contains an IP address used for HELO
 *  0.0 NORMAL_HTTP_TO_IP URI: Uses a dotted-decimal IP address in URL
 *  1.5 RAZOR2_CF_RANGE_E8_51_100 Razor2 gives engine 8 confidence level
 *      above 50%
 *      [cf:  96]
 *  0.5 RAZOR2_CHECK Listed in Razor2 (http://razor.sf.net/)
 *  0.5 RAZOR2_CF_RANGE_51_100 Razor2 gives confidence level above 50%
 *      [cf:  96]
 *  2.0 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net
 *      [Blocked - see <http://www.spamcop.net/bl.shtml?121.247.0.93>]
 *  0.9 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL
 *      [121.247.0.93 listed in zen.spamhaus.org]
 *  3.0 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL
 *  0.1 RDNS_NONE Delivered to trusted network by a host with no rDNS
Received: from core3.amsl.com ([127.0.0.1])
	by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024)
	with ESMTP id axVzBpMtqHaZ for <ietfarch-sip-archive@core3.amsl.com>;
	Thu, 31 Jan 2008 22:57:31 -0800 (PST)
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Message-ID: <000e01c8649f$efd328f0$e9846ad2@arkj>
From: <bwebb@unimpeks.com>
To: <sip-archive@ietf.org>
Subject: ***SPAM*** 29.734 (5) I Love You Because
Date: Fri, 1 Feb 2008 12:29:06 +0530
MIME-Version: 1.0
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Hugging My Pillow http://89.37.39.244/

From to11@volition-inc.com  Thu Jan 31 22:58:38 2008
Return-Path: <to11@volition-inc.com>
X-Original-To: ietfarch-sip-archive@core3.amsl.com
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Received: from localhost (localhost [127.0.0.1])
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X-Virus-Scanned: amavisd-new at amsl.com
X-Spam-Flag: YES
X-Spam-Score: 37.599
X-Spam-Level: *************************************
X-Spam-Status: Yes, score=37.599 tagged_above=-999 required=5
	tests=[BAYES_99=3.5, FH_RELAY_NODNS=1.451, HELO_DYNAMIC_IPADDR2=4.395,
	HELO_DYNAMIC_SPLIT_IP=3.493, HELO_EQ_DYNAMIC=1.144,
	HELO_EQ_IP_ADDR=1.119, NORMAL_HTTP_TO_IP=0.001,
	RAZOR2_CF_RANGE_51_100=0.5, RAZOR2_CF_RANGE_E8_51_100=1.5,
	RAZOR2_CHECK=0.5, RCVD_IN_BL_SPAMCOP_NET=1.96, RCVD_IN_PBL=0.905,
	RCVD_IN_XBL=3.033, RCVD_NUMERIC_HELO=2.067, RDNS_NONE=0.1,
	TVD_RCVD_IP=1.931, URIBL_JP_SURBL=10]
X-Spam-Report:
 *  3.5 BAYES_99 BODY: Bayesian spam probability is 99 to 100%
 *      [score: 1.0000]
 *  1.1 HELO_EQ_DYNAMIC HELO_EQ_DYNAMIC
 *  3.5 HELO_DYNAMIC_SPLIT_IP Relay HELO'd using suspicious hostname (Split
 *      IP)
 *  1.1 HELO_EQ_IP_ADDR HELO using IP Address (not private)
 *  1.5 FH_RELAY_NODNS We could not determine your Reverse DNS
 *  4.4 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr
 *       2)
 *  1.9 TVD_RCVD_IP TVD_RCVD_IP
 *  2.1 RCVD_NUMERIC_HELO Received: contains an IP address used for HELO
 *  0.0 NORMAL_HTTP_TO_IP URI: Uses a dotted-decimal IP address in URL
 *  1.5 RAZOR2_CF_RANGE_E8_51_100 Razor2 gives engine 8 confidence level
 *      above 50%
 *      [cf: 100]
 *  0.5 RAZOR2_CHECK Listed in Razor2 (http://razor.sf.net/)
 *  0.5 RAZOR2_CF_RANGE_51_100 Razor2 gives confidence level above 50%
 *      [cf: 100]
 *   10 URIBL_JP_SURBL Contains an URL listed in the JP SURBL blocklist
 *      [URIs: 24.8.215.93]
 *  2.0 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net
 *      [Blocked - see <http://www.spamcop.net/bl.shtml?121.247.0.93>]
 *  3.0 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL
 *      [121.247.0.93 listed in zen.spamhaus.org]
 *  0.9 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL
 *  0.1 RDNS_NONE Delivered to trusted network by a host with no rDNS
Received: from core3.amsl.com ([127.0.0.1])
	by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024)
	with ESMTP id 7osqChCbcWy2 for <ietfarch-sip-archive@core3.amsl.com>;
	Thu, 31 Jan 2008 22:58:38 -0800 (PST)
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Message-ID: <47A2C37F.1060007@volition-inc.com>
Date: Fri, 1 Feb 2008 12:30:15 +0530
From: <to11@volition-inc.com>
User-Agent: Thunderbird 2.0.0.6 (Windows/20070728)
MIME-Version: 1.0
To: sip-archive@lists.ietf.org
Subject: ***SPAM*** 37.599 (5) You're In My Thoughts
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Eternity of Your Love http://24.8.215.93/

